From 3ba98e6ef8be6a41697c4ff86c77415b73742233 Mon Sep 17 00:00:00 2001 From: Connor McLaughlin Date: Fri, 10 Jan 2020 14:59:53 +1000 Subject: [PATCH] dep: Add cubeb --- dep/cubeb/AUTHORS | 16 + dep/cubeb/CMakeLists.txt | 332 ++ dep/cubeb/Config.cmake.in | 4 + dep/cubeb/INSTALL.md | 46 + dep/cubeb/LICENSE | 13 + dep/cubeb/README.md | 6 + dep/cubeb/TODO | 41 + dep/cubeb/cubeb.supp | 36 + dep/cubeb/cubeb.vcxproj | 406 ++ dep/cubeb/cubeb.vcxproj.filters | 59 + dep/cubeb/include/cubeb/cubeb.h | 658 +++ dep/cubeb/include/cubeb/cubeb_export.h | 12 + .../src/android/audiotrack_definitions.h | 81 + dep/cubeb/src/android/cubeb-output-latency.h | 76 + dep/cubeb/src/android/cubeb_media_library.h | 62 + dep/cubeb/src/android/sles_definitions.h | 102 + dep/cubeb/src/cubeb-internal.h | 79 + dep/cubeb/src/cubeb-jni-instances.h | 30 + dep/cubeb/src/cubeb-jni.cpp | 68 + dep/cubeb/src/cubeb-jni.h | 10 + dep/cubeb/src/cubeb-sles.h | 43 + dep/cubeb/src/cubeb-speex-resampler.h | 1 + dep/cubeb/src/cubeb.c | 665 +++ dep/cubeb/src/cubeb_alsa.c | 1452 +++++++ dep/cubeb/src/cubeb_array_queue.h | 97 + dep/cubeb/src/cubeb_assert.h | 26 + dep/cubeb/src/cubeb_audiotrack.c | 441 ++ dep/cubeb/src/cubeb_audiounit.cpp | 3620 +++++++++++++++++ dep/cubeb/src/cubeb_jack.cpp | 1075 +++++ dep/cubeb/src/cubeb_kai.c | 369 ++ dep/cubeb/src/cubeb_log.cpp | 144 + dep/cubeb/src/cubeb_log.h | 47 + dep/cubeb/src/cubeb_mixer.cpp | 663 +++ dep/cubeb/src/cubeb_mixer.h | 37 + dep/cubeb/src/cubeb_opensl.c | 1760 ++++++++ dep/cubeb/src/cubeb_osx_run_loop.cpp | 36 + dep/cubeb/src/cubeb_osx_run_loop.h | 22 + dep/cubeb/src/cubeb_pulse.c | 1605 ++++++++ dep/cubeb/src/cubeb_resampler.cpp | 372 ++ dep/cubeb/src/cubeb_resampler.h | 85 + dep/cubeb/src/cubeb_resampler_internal.h | 602 +++ dep/cubeb/src/cubeb_ring_array.h | 159 + dep/cubeb/src/cubeb_ringbuffer.h | 495 +++ dep/cubeb/src/cubeb_sndio.c | 669 +++ dep/cubeb/src/cubeb_strings.c | 155 + dep/cubeb/src/cubeb_strings.h | 44 + dep/cubeb/src/cubeb_sun.c | 752 ++++ dep/cubeb/src/cubeb_utils.cpp | 23 + dep/cubeb/src/cubeb_utils.h | 343 ++ dep/cubeb/src/cubeb_utils_unix.h | 89 + dep/cubeb/src/cubeb_utils_win.h | 71 + dep/cubeb/src/cubeb_wasapi.cpp | 2965 ++++++++++++++ dep/cubeb/src/cubeb_winmm.c | 1067 +++++ dep/cubeb/src/speex/arch.h | 235 ++ dep/cubeb/src/speex/fixed_generic.h | 110 + dep/cubeb/src/speex/resample.c | 1240 ++++++ dep/cubeb/src/speex/resample_neon.h | 201 + dep/cubeb/src/speex/resample_sse.h | 128 + dep/cubeb/src/speex/speex_config_types.h | 10 + dep/cubeb/src/speex/speex_resampler.h | 343 ++ dep/cubeb/src/speex/stack_alloc.h | 115 + duckstation.sln | 19 + 62 files changed, 24532 insertions(+) create mode 100644 dep/cubeb/AUTHORS create mode 100644 dep/cubeb/CMakeLists.txt create mode 100644 dep/cubeb/Config.cmake.in create mode 100644 dep/cubeb/INSTALL.md create mode 100644 dep/cubeb/LICENSE create mode 100644 dep/cubeb/README.md create mode 100644 dep/cubeb/TODO create mode 100644 dep/cubeb/cubeb.supp create mode 100644 dep/cubeb/cubeb.vcxproj create mode 100644 dep/cubeb/cubeb.vcxproj.filters create mode 100644 dep/cubeb/include/cubeb/cubeb.h create mode 100644 dep/cubeb/include/cubeb/cubeb_export.h create mode 100644 dep/cubeb/src/android/audiotrack_definitions.h create mode 100644 dep/cubeb/src/android/cubeb-output-latency.h create mode 100644 dep/cubeb/src/android/cubeb_media_library.h create mode 100644 dep/cubeb/src/android/sles_definitions.h create mode 100644 dep/cubeb/src/cubeb-internal.h create mode 100644 dep/cubeb/src/cubeb-jni-instances.h create mode 100644 dep/cubeb/src/cubeb-jni.cpp create mode 100644 dep/cubeb/src/cubeb-jni.h create mode 100644 dep/cubeb/src/cubeb-sles.h create mode 100644 dep/cubeb/src/cubeb-speex-resampler.h create mode 100644 dep/cubeb/src/cubeb.c create mode 100644 dep/cubeb/src/cubeb_alsa.c create mode 100644 dep/cubeb/src/cubeb_array_queue.h create mode 100644 dep/cubeb/src/cubeb_assert.h create mode 100644 dep/cubeb/src/cubeb_audiotrack.c create mode 100644 dep/cubeb/src/cubeb_audiounit.cpp create mode 100644 dep/cubeb/src/cubeb_jack.cpp create mode 100644 dep/cubeb/src/cubeb_kai.c create mode 100644 dep/cubeb/src/cubeb_log.cpp create mode 100644 dep/cubeb/src/cubeb_log.h create mode 100644 dep/cubeb/src/cubeb_mixer.cpp create mode 100644 dep/cubeb/src/cubeb_mixer.h create mode 100644 dep/cubeb/src/cubeb_opensl.c create mode 100644 dep/cubeb/src/cubeb_osx_run_loop.cpp create mode 100644 dep/cubeb/src/cubeb_osx_run_loop.h create mode 100644 dep/cubeb/src/cubeb_pulse.c create mode 100644 dep/cubeb/src/cubeb_resampler.cpp create mode 100644 dep/cubeb/src/cubeb_resampler.h create mode 100644 dep/cubeb/src/cubeb_resampler_internal.h create mode 100644 dep/cubeb/src/cubeb_ring_array.h create mode 100644 dep/cubeb/src/cubeb_ringbuffer.h create mode 100644 dep/cubeb/src/cubeb_sndio.c create mode 100644 dep/cubeb/src/cubeb_strings.c create mode 100644 dep/cubeb/src/cubeb_strings.h create mode 100644 dep/cubeb/src/cubeb_sun.c create mode 100644 dep/cubeb/src/cubeb_utils.cpp create mode 100644 dep/cubeb/src/cubeb_utils.h create mode 100644 dep/cubeb/src/cubeb_utils_unix.h create mode 100644 dep/cubeb/src/cubeb_utils_win.h create mode 100644 dep/cubeb/src/cubeb_wasapi.cpp create mode 100644 dep/cubeb/src/cubeb_winmm.c create mode 100644 dep/cubeb/src/speex/arch.h create mode 100644 dep/cubeb/src/speex/fixed_generic.h create mode 100644 dep/cubeb/src/speex/resample.c create mode 100644 dep/cubeb/src/speex/resample_neon.h create mode 100644 dep/cubeb/src/speex/resample_sse.h create mode 100644 dep/cubeb/src/speex/speex_config_types.h create mode 100644 dep/cubeb/src/speex/speex_resampler.h create mode 100644 dep/cubeb/src/speex/stack_alloc.h diff --git a/dep/cubeb/AUTHORS b/dep/cubeb/AUTHORS new file mode 100644 index 000000000..f0f959522 --- /dev/null +++ b/dep/cubeb/AUTHORS @@ -0,0 +1,16 @@ +Matthew Gregan +Alexandre Ratchov +Michael Wu +Paul Adenot +David Richards +Sebastien Alaiwan +KO Myung-Hun +Haakon Sporsheim +Alex Chronopoulos +Jan Beich +Vito Caputo +Landry Breuil +Jacek Caban +Paul Hancock +Ted Mielczarek +Chun-Min Chang diff --git a/dep/cubeb/CMakeLists.txt b/dep/cubeb/CMakeLists.txt new file mode 100644 index 000000000..c88579391 --- /dev/null +++ b/dep/cubeb/CMakeLists.txt @@ -0,0 +1,332 @@ +# TODO +# - backend selection via command line, rather than simply detecting headers. + +cmake_minimum_required(VERSION 3.1 FATAL_ERROR) +project(cubeb + VERSION 0.0.0) + +option(BUILD_SHARED_LIBS "Build shared libraries" OFF) +option(BUILD_TESTS "Build tests" ON) +option(BUILD_RUST_LIBS "Build rust backends" OFF) +option(BUILD_TOOLS "Build tools" ON) + +if(NOT CMAKE_BUILD_TYPE) + set(CMAKE_BUILD_TYPE "RelWithDebInfo" CACHE STRING + "Choose the type of build, options are: Debug Release RelWithDebInfo MinSizeRel." FORCE) +endif() + +if(POLICY CMP0063) + cmake_policy(SET CMP0063 NEW) +endif() +set(CMAKE_C_STANDARD 99) +set(CMAKE_CXX_STANDARD 11) +set(CMAKE_CXX_STANDARD_REQUIRED ON) + +if(NOT COMMAND add_sanitizers) + list(APPEND CMAKE_MODULE_PATH "${PROJECT_SOURCE_DIR}/cmake/sanitizers-cmake/cmake") + find_package(Sanitizers) + if(NOT COMMAND add_sanitizers) + message(FATAL_ERROR "Could not find sanitizers-cmake: run\n\tgit submodule update --init --recursive\nin base git checkout") + endif() +endif() + +if(BUILD_TESTS) + if(NOT TARGET gtest_main) + if(NOT EXISTS "${PROJECT_SOURCE_DIR}/googletest/CMakeLists.txt") + message(FATAL_ERROR "Could not find googletest: run\n\tgit submodule update --init --recursive\nin base git checkout") + endif() + add_definitions(-DGTEST_HAS_TR1_TUPLE=0) + set(gtest_force_shared_crt ON CACHE BOOL "") + add_subdirectory(googletest) + endif() +endif() + +if (BUILD_RUST_LIBS) + if(EXISTS "${PROJECT_SOURCE_DIR}/src/cubeb-pulse-rs") + set(USE_PULSE_RUST 1) + endif() + if(EXISTS "${PROJECT_SOURCE_DIR}/src/cubeb-coreaudio-rs") + set(USE_AUDIOUNIT_RUST 1) + endif() +endif() + +# On OS/2, visibility attribute is not supported. +if(NOT OS2) + set(CMAKE_C_VISIBILITY_PRESET hidden) + set(CMAKE_CXX_VISIBILITY_PRESET hidden) + set(CMAKE_VISIBILITY_INLINES_HIDDEN 1) +endif() + +set(CMAKE_CXX_WARNING_LEVEL 4) +if(NOT MSVC) + set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -Wall -Wextra -Wno-unused-parameter") + set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -Wall -Wextra -Wno-unused-parameter") +endif() + +add_library(cubeb + src/cubeb.c + src/cubeb_mixer.cpp + src/cubeb_resampler.cpp + src/cubeb_log.cpp + src/cubeb_strings.c + src/cubeb_utils.cpp + $) +target_include_directories(cubeb + PUBLIC $ $ +) +target_include_directories(cubeb PRIVATE src) +target_compile_definitions(cubeb PRIVATE OUTSIDE_SPEEX) +target_compile_definitions(cubeb PRIVATE FLOATING_POINT) +target_compile_definitions(cubeb PRIVATE EXPORT=) +target_compile_definitions(cubeb PRIVATE RANDOM_PREFIX=speex) + +add_sanitizers(cubeb) + +include(GenerateExportHeader) +generate_export_header(cubeb EXPORT_FILE_NAME ${CMAKE_BINARY_DIR}/exports/cubeb_export.h) +target_include_directories(cubeb + PUBLIC $ +) + +if(UNIX) + include(GNUInstallDirs) +else() + set(CMAKE_INSTALL_LIBDIR "lib") + set(CMAKE_INSTALL_BINDIR "bin") + set(CMAKE_INSTALL_DATADIR "share") + set(CMAKE_INSTALL_DOCDIR "${CMAKE_INSTALL_DATADIR}/doc") + set(CMAKE_INSTALL_INCLUDEDIR "include") +endif() + +install(DIRECTORY ${CMAKE_SOURCE_DIR}/include/${PROJECT_NAME} DESTINATION ${CMAKE_INSTALL_INCLUDEDIR}) +install(DIRECTORY ${CMAKE_BINARY_DIR}/exports/ DESTINATION ${CMAKE_INSTALL_INCLUDEDIR}/${PROJECT_NAME}) + +include(CMakePackageConfigHelpers) +write_basic_package_version_file( + "${PROJECT_BINARY_DIR}/${PROJECT_NAME}ConfigVersion.cmake" + COMPATIBILITY SameMajorVersion +) + +configure_package_config_file( + "Config.cmake.in" + "${PROJECT_BINARY_DIR}/${PROJECT_NAME}Config.cmake" + INSTALL_DESTINATION "${CMAKE_INSTALL_LIBDIR}/cmake/${PROJECT_NAME}" +) + +install(TARGETS cubeb + EXPORT "${PROJECT_NAME}Targets" + DESTINATION ${CMAKE_INSTALL_PREFIX} + LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} + ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} + RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} + INCLUDES DESTINATION ${CMAKE_INSTALL_INCLUDEDIR} +) +install( + FILES "${PROJECT_BINARY_DIR}/${PROJECT_NAME}Config.cmake" "${PROJECT_BINARY_DIR}/${PROJECT_NAME}ConfigVersion.cmake" + DESTINATION "${CMAKE_INSTALL_LIBDIR}/cmake/${PROJECT_NAME}" +) +install( + EXPORT "${PROJECT_NAME}Targets" + NAMESPACE "${PROJECT_NAME}::" + DESTINATION "${CMAKE_INSTALL_LIBDIR}/cmake/${PROJECT_NAME}" +) + +add_library(speex OBJECT + src/speex/resample.c) +set_target_properties(speex PROPERTIES POSITION_INDEPENDENT_CODE TRUE) +target_compile_definitions(speex PRIVATE OUTSIDE_SPEEX) +target_compile_definitions(speex PRIVATE FLOATING_POINT) +target_compile_definitions(speex PRIVATE EXPORT=) +target_compile_definitions(speex PRIVATE RANDOM_PREFIX=speex) + +include(CheckIncludeFiles) + +check_include_files(AudioUnit/AudioUnit.h USE_AUDIOUNIT) +if(USE_AUDIOUNIT) + target_sources(cubeb PRIVATE + src/cubeb_audiounit.cpp + src/cubeb_osx_run_loop.cpp) + target_compile_definitions(cubeb PRIVATE USE_AUDIOUNIT) + target_link_libraries(cubeb PRIVATE "-framework AudioUnit" "-framework CoreAudio" "-framework CoreServices") +endif() + +check_include_files(pulse/pulseaudio.h USE_PULSE) +if(USE_PULSE) + target_sources(cubeb PRIVATE + src/cubeb_pulse.c) + target_compile_definitions(cubeb PRIVATE USE_PULSE) + target_link_libraries(cubeb PRIVATE pthread ${CMAKE_DL_LIBS}) +endif() + +check_include_files(alsa/asoundlib.h USE_ALSA) +if(USE_ALSA) + target_sources(cubeb PRIVATE + src/cubeb_alsa.c) + target_compile_definitions(cubeb PRIVATE USE_ALSA) + target_link_libraries(cubeb PRIVATE pthread ${CMAKE_DL_LIBS}) +endif() + +check_include_files(jack/jack.h USE_JACK) +if(USE_JACK) + target_sources(cubeb PRIVATE + src/cubeb_jack.cpp) + target_compile_definitions(cubeb PRIVATE USE_JACK) + target_link_libraries(cubeb PRIVATE pthread ${CMAKE_DL_LIBS}) +endif() + +check_include_files(audioclient.h USE_WASAPI) +if(USE_WASAPI) + target_sources(cubeb PRIVATE + src/cubeb_wasapi.cpp) + target_compile_definitions(cubeb PRIVATE USE_WASAPI) + target_link_libraries(cubeb PRIVATE avrt ole32) +endif() + +check_include_files("windows.h;mmsystem.h" USE_WINMM) +if(USE_WINMM) + target_sources(cubeb PRIVATE + src/cubeb_winmm.c) + target_compile_definitions(cubeb PRIVATE USE_WINMM) + target_link_libraries(cubeb PRIVATE winmm) +endif() + +check_include_files(SLES/OpenSLES.h USE_OPENSL) +if(USE_OPENSL) + target_sources(cubeb PRIVATE + src/cubeb_opensl.c + src/cubeb-jni.cpp) + target_compile_definitions(cubeb PRIVATE USE_OPENSL) + target_link_libraries(cubeb PRIVATE OpenSLES) +endif() + +check_include_files(android/log.h USE_AUDIOTRACK) +if(USE_AUDIOTRACK) + target_sources(cubeb PRIVATE + src/cubeb_audiotrack.c) + target_compile_definitions(cubeb PRIVATE USE_AUDIOTRACK) + target_link_libraries(cubeb PRIVATE log) +endif() + +check_include_files(sndio.h USE_SNDIO) +if(USE_SNDIO) + target_sources(cubeb PRIVATE + src/cubeb_sndio.c) + target_compile_definitions(cubeb PRIVATE USE_SNDIO) + target_link_libraries(cubeb PRIVATE pthread ${CMAKE_DL_LIBS}) +endif() + +check_include_files(sys/audioio.h USE_SUN) +if(USE_SUN) + target_sources(cubeb PRIVATE + src/cubeb_sun.c) + target_compile_definitions(cubeb PRIVATE USE_SUN) + target_link_libraries(cubeb PRIVATE pthread) +endif() + +check_include_files(kai.h USE_KAI) +if(USE_KAI) + target_sources(cubeb PRIVATE + src/cubeb_kai.c) + target_compile_definitions(cubeb PRIVATE USE_KAI) + target_link_libraries(cubeb PRIVATE kai) +endif() + +if(USE_PULSE_RUST) + include(ExternalProject) + set_directory_properties(PROPERTIES EP_PREFIX ${CMAKE_BINARY_DIR}/rust) + ExternalProject_Add( + cubeb_pulse_rs + DOWNLOAD_COMMAND "" + CONFIGURE_COMMAND "" + BUILD_COMMAND cargo build COMMAND cargo build --release + BINARY_DIR "${CMAKE_SOURCE_DIR}/src/cubeb-pulse-rs" + INSTALL_COMMAND "" + LOG_BUILD ON) + add_dependencies(cubeb cubeb_pulse_rs) + target_compile_definitions(cubeb PRIVATE USE_PULSE_RUST) + target_link_libraries(cubeb PRIVATE + debug "${CMAKE_SOURCE_DIR}/src/cubeb-pulse-rs/target/debug/libcubeb_pulse.a" + optimized "${CMAKE_SOURCE_DIR}/src/cubeb-pulse-rs/target/release/libcubeb_pulse.a" pulse) +endif() + +if(USE_AUDIOUNIT_RUST) + include(ExternalProject) + set_directory_properties(PROPERTIES EP_PREFIX ${CMAKE_BINARY_DIR}/rust) + ExternalProject_Add( + cubeb_coreaudio_rs + DOWNLOAD_COMMAND "" + CONFIGURE_COMMAND "" + BUILD_COMMAND cargo build COMMAND cargo build --release + BINARY_DIR "${CMAKE_SOURCE_DIR}/src/cubeb-coreaudio-rs" + INSTALL_COMMAND "" + LOG_BUILD ON) + add_dependencies(cubeb cubeb_coreaudio_rs) + target_compile_definitions(cubeb PRIVATE USE_AUDIOUNIT_RUST) + target_link_libraries(cubeb PRIVATE + debug "${CMAKE_SOURCE_DIR}/src/cubeb-coreaudio-rs/target/debug/libcubeb_coreaudio.a" + optimized "${CMAKE_SOURCE_DIR}/src/cubeb-coreaudio-rs/target/release/libcubeb_coreaudio.a") +endif() + +find_package(Doxygen) +if(DOXYGEN_FOUND) + configure_file(${CMAKE_CURRENT_SOURCE_DIR}/docs/Doxyfile.in ${CMAKE_CURRENT_BINARY_DIR}/docs/Doxyfile @ONLY) + add_custom_target(doc ALL + ${DOXYGEN_EXECUTABLE} ${CMAKE_CURRENT_BINARY_DIR}/docs/Doxyfile + WORKING_DIRECTORY ${CMAKE_CURRENT_BINARY_DIR}/docs + COMMENT "Generating API documentation with Doxygen" VERBATIM) +endif() + +if(BUILD_TESTS) + enable_testing() + + macro(cubeb_add_test NAME) + add_executable(test_${NAME} test/test_${NAME}.cpp) + target_include_directories(test_${NAME} PRIVATE ${gtest_SOURCE_DIR}/include) + target_include_directories(test_${NAME} PRIVATE src) + target_link_libraries(test_${NAME} PRIVATE cubeb gtest_main) + add_test(${NAME} test_${NAME}) + add_sanitizers(test_${NAME}) + install(TARGETS test_${NAME} DESTINATION ${CMAKE_INSTALL_PREFIX}/${CMAKE_INSTALL_BINDIR}) + endmacro(cubeb_add_test) + + cubeb_add_test(sanity) + cubeb_add_test(tone) + cubeb_add_test(audio) + cubeb_add_test(record) + cubeb_add_test(devices) + cubeb_add_test(callback_ret) + + add_executable(test_resampler test/test_resampler.cpp src/cubeb_resampler.cpp $) + target_include_directories(test_resampler PRIVATE ${gtest_SOURCE_DIR}/include) + target_include_directories(test_resampler PRIVATE src) + target_compile_definitions(test_resampler PRIVATE OUTSIDE_SPEEX) + target_compile_definitions(test_resampler PRIVATE FLOATING_POINT) + target_compile_definitions(test_resampler PRIVATE EXPORT=) + target_compile_definitions(test_resampler PRIVATE RANDOM_PREFIX=speex) + target_link_libraries(test_resampler PRIVATE cubeb gtest_main) + add_test(resampler test_resampler) + add_sanitizers(test_resampler) + install(TARGETS test_resampler DESTINATION ${CMAKE_INSTALL_PREFIX}/${CMAKE_INSTALL_BINDIR}) + + cubeb_add_test(duplex) + + if (USE_WASAPI) + cubeb_add_test(overload_callback) + cubeb_add_test(loopback) + endif() + + cubeb_add_test(latency test_latency) + cubeb_add_test(ring_array) + + cubeb_add_test(utils) + cubeb_add_test(ring_buffer) + cubeb_add_test(device_changed_callback) +endif() + +if(BUILD_TOOLS) + add_executable(cubeb-test tools/cubeb-test.cpp) + target_include_directories(cubeb-test PRIVATE src) + target_link_libraries(cubeb-test PRIVATE cubeb) + add_sanitizers(cubeb-test) + install(TARGETS cubeb-test DESTINATION ${CMAKE_INSTALL_PREFIX}/${CMAKE_INSTALL_BINDIR}) +endif() diff --git a/dep/cubeb/Config.cmake.in b/dep/cubeb/Config.cmake.in new file mode 100644 index 000000000..c5326ef33 --- /dev/null +++ b/dep/cubeb/Config.cmake.in @@ -0,0 +1,4 @@ +@PACKAGE_INIT@ + +include("${CMAKE_CURRENT_LIST_DIR}/cubebTargets.cmake") +check_required_components(cubeb) \ No newline at end of file diff --git a/dep/cubeb/INSTALL.md b/dep/cubeb/INSTALL.md new file mode 100644 index 000000000..f8492243a --- /dev/null +++ b/dep/cubeb/INSTALL.md @@ -0,0 +1,46 @@ +# Build instructions for libcubeb + +You must have CMake v3.1 or later installed. + +1. `git clone --recursive https://github.com/kinetiknz/cubeb.git` +2. `mkdir cubeb-build` +3. `cd cubeb-build` +3. `cmake ../cubeb` +4. `cmake --build .` +5. `ctest` + +# Windows build notes + +Windows builds can use Microsoft Visual Studio 2015, Microsoft Visual Studio +2017, or MinGW-w64 with Win32 threads (by passing `cmake -G` to generate the +appropriate build configuration). + +## Microsoft Visual Studio 2015 or 2017 Command Line + +CMake can be used from the command line by following the build steps at the top +of this file. CMake will select a default generator based on the environment, +or one can be specified with the `-G` argument. + +## Microsoft Visual Studio 2017 IDE + +Visual Studio 2017 adds in built support for CMake. CMake can be used from +within the IDE via the following steps: + +- Navigate to `File -> Open -> Cmake...` +- Open `CMakeLists.txt` file in the root of the project. + +Note, to generate the build in the cubeb dir CMake settings need to be updated +via: `CMake -> Change CMake Settings -> CMakeLists.txt`. The default +configuration used by Visual Studio will place the build in a different location +than the steps detailed at the top of this file. + +## MinGW-w64 + +To build with MinGW-w64, install the following items: + +- Download and install MinGW-w64 with Win32 threads. +- Download and install CMake. +- Run MinGW-w64 Terminal from the Start Menu. +- Follow the build steps at the top of this file, but at step 3 run: + `cmake -G "MinGW Makefiles" ..` +- Continue the build steps at the top of this file. diff --git a/dep/cubeb/LICENSE b/dep/cubeb/LICENSE new file mode 100644 index 000000000..fffc9dc40 --- /dev/null +++ b/dep/cubeb/LICENSE @@ -0,0 +1,13 @@ +Copyright © 2011 Mozilla Foundation + +Permission to use, copy, modify, and distribute this software for any +purpose with or without fee is hereby granted, provided that the above +copyright notice and this permission notice appear in all copies. + +THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES +WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF +MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR +ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES +WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN +ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF +OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. diff --git a/dep/cubeb/README.md b/dep/cubeb/README.md new file mode 100644 index 000000000..d26b3b645 --- /dev/null +++ b/dep/cubeb/README.md @@ -0,0 +1,6 @@ +[![Build Status](https://travis-ci.org/kinetiknz/cubeb.svg?branch=master)](https://travis-ci.org/kinetiknz/cubeb) +[![Build status](https://ci.appveyor.com/api/projects/status/osv2r0m1j1nt9csr/branch/master?svg=true)](https://ci.appveyor.com/project/kinetiknz/cubeb/branch/master) + +See INSTALL.md for build instructions. + +Licensed under an ISC-style license. See LICENSE for details. diff --git a/dep/cubeb/TODO b/dep/cubeb/TODO new file mode 100644 index 000000000..57288a42e --- /dev/null +++ b/dep/cubeb/TODO @@ -0,0 +1,41 @@ +TODO: +- directsound: incomplete and somewhat broken +- osx: understand why AudioQueueGetCurrentTime can return negative mSampleTime +- test (and fix) sub-prefill size data playback +- report stream delay instead of position; leave position calculation to user +- capture support +- capture and output enumeration and configuration + - also expose default hardware config to allow decisions on speaker layout +- prefill occurs at different times in each backend: + - pulse prefills async off worker thread after init + - coreaudio prefills during init + - alsa prefills async after start +- expose configured prefill size; may differ from requested latency + - solved by exposing stream delay +- xruns may occur in user callback but also in audio hardware + may need to expose details of hardware xruns to user api +- document thread safety +- document which calls may block, and when effects take effect +- document what's permissible inside callbacks +- implement basic channel mapping for surround + - vorbis has documented mapping based on channel count (if mapping type == + 0) -- http://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9 + 1 -> M + 2 -> L, R + 3 -> L, C, R + 4 -> L, R, RL, RR + 5 -> L, C, R, RL, RR + 6 -> L, C, R, RL, RR, LFE + 7 -> L, C, R, SL, SR, RC, LFE + 8 -> L, C, R, SL, SR, RL, RR, LFE + >8 -> application defined + - wave files with channel count only + 3 -> L, R, C + 4 -> L, R, RL, RR + 5 -> L, R, C, RL, RR + 6 -> L, R, C, LFE, RL, RR + 7 -> L, R, C, LFE, RC, SL, SR + 8 -> L, R, C, LFE, RL, RR, SL, SR + - wave files with WAVE_FORMAT_EXTENSIBLE have explicitly mappings, can + extract these +- implement configurable channel mapping diff --git a/dep/cubeb/cubeb.supp b/dep/cubeb/cubeb.supp new file mode 100644 index 000000000..0012ea51e --- /dev/null +++ b/dep/cubeb/cubeb.supp @@ -0,0 +1,36 @@ +{ + snd_config_update-malloc + Memcheck:Leak + fun:malloc + ... + fun:snd_config_update_r +} +{ + snd1_dlobj_cache_get-malloc + Memcheck:Leak + fun:malloc + ... + fun:snd1_dlobj_cache_get +} +{ + parse_defs-malloc + Memcheck:Leak + fun:malloc + ... + fun:parse_defs +} +{ + parse_defs-calloc + Memcheck:Leak + fun:calloc + ... + fun:parse_defs +} +{ + pa_client_conf_from_x11-malloc + Memcheck:Leak + fun:malloc + ... + fun:pa_client_conf_from_x11 +} + diff --git a/dep/cubeb/cubeb.vcxproj b/dep/cubeb/cubeb.vcxproj new file mode 100644 index 000000000..fb8847881 --- /dev/null +++ b/dep/cubeb/cubeb.vcxproj @@ -0,0 +1,406 @@ + + + + + DebugFast + Win32 + + + DebugFast + x64 + + + Debug + Win32 + + + Debug + x64 + + + ReleaseLTCG + Win32 + + + ReleaseLTCG + x64 + + + Release + Win32 + + + Release + x64 + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + {72F9423C-91EE-4487-AAC6-555ED6F61AA1} + Win32Proj + inih + 10.0 + + + + StaticLibrary + true + v142 + NotSet + + + StaticLibrary + true + v142 + NotSet + + + StaticLibrary + true + v142 + NotSet + + + StaticLibrary + true + v142 + NotSet + + + StaticLibrary + false + v142 + true + NotSet + + + StaticLibrary + false + v142 + true + NotSet + + + StaticLibrary + false + v142 + true + NotSet + + + StaticLibrary + false + v142 + true + NotSet + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + true + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + + + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + true + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + + + true + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + + + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + true + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + + + false + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + + + false + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + + + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + false + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + + + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + $(ProjectName)-$(Platform)-$(Configuration) + false + $(SolutionDir)build\$(ProjectName)-$(Platform)-$(Configuration)\ + + + + + + TurnOffAllWarnings + Disabled + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_CRT_SECURE_NO_WARNINGS;WIN32;_DEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + true + ProgramDatabase + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + false + stdcpp17 + true + true + + + Windows + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib32-debug;%(AdditionalLibraryDirectories) + + + + + + + + TurnOffAllWarnings + Disabled + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_CRT_SECURE_NO_WARNINGS;WIN32;_DEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + true + ProgramDatabase + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + false + stdcpp17 + true + true + + + Windows + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib64-debug;%(AdditionalLibraryDirectories) + + + + + + + + TurnOffAllWarnings + Disabled + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_ITERATOR_DEBUG_LEVEL=1;_CRT_SECURE_NO_WARNINGS;WIN32;_DEBUGFAST;_DEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + true + ProgramDatabase + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + Default + false + stdcpp17 + false + true + OnlyExplicitInline + true + + + Windows + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib32-debug;%(AdditionalLibraryDirectories) + + + + + + + + TurnOffAllWarnings + Disabled + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_ITERATOR_DEBUG_LEVEL=1;_CRT_SECURE_NO_WARNINGS;WIN32;_DEBUGFAST;_DEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + true + ProgramDatabase + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + Default + false + stdcpp17 + false + true + OnlyExplicitInline + true + + + Windows + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib64-debug;%(AdditionalLibraryDirectories) + + + + + + TurnOffAllWarnings + + + MaxSpeed + true + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_CRT_SECURE_NO_WARNINGS;WIN32;NDEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + false + stdcpp17 + true + true + + + Windows + true + true + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib32;%(AdditionalLibraryDirectories) + + + + + + TurnOffAllWarnings + + + MaxSpeed + true + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_CRT_SECURE_NO_WARNINGS;WIN32;NDEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + true + stdcpp17 + true + true + true + + + Windows + true + true + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib32;%(AdditionalLibraryDirectories) + + + + + + TurnOffAllWarnings + + + MaxSpeed + true + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_CRT_SECURE_NO_WARNINGS;WIN32;NDEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + false + stdcpp17 + true + true + + + Windows + true + true + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib64;%(AdditionalLibraryDirectories) + + + + + + TurnOffAllWarnings + + + MaxSpeed + true + USE_WASAPI;USE_WINMM;OUTSIDE_SPEEX;FLOATING_POINT;RANDOM_PREFIX=speex;EXPORT=;_CRT_NONSTDC_NO_DEPRECATE;_CRT_SECURE_NO_WARNINGS;WIN32;NDEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + $(ProjectDir)include;$(ProjectDir)src;%(AdditionalIncludeDirectories) + true + stdcpp17 + true + true + true + + + Windows + true + true + true + SDL2.lib;SDL2main.lib;%(AdditionalDependencies) + $(SolutionDir)dep\lib64;%(AdditionalLibraryDirectories) + + + + + + + \ No newline at end of file diff --git a/dep/cubeb/cubeb.vcxproj.filters b/dep/cubeb/cubeb.vcxproj.filters new file mode 100644 index 000000000..3d0a6d76c --- /dev/null +++ b/dep/cubeb/cubeb.vcxproj.filters @@ -0,0 +1,59 @@ + + + + + + speex + + + speex + + + speex + + + speex + + + speex + + + speex + + + speex + + + + + + + + + + + + + + + + + + + {a22ed8dc-2384-4a96-bd3b-2370d6d7cd62} + + + + + speex + + + + + + + + + + + \ No newline at end of file diff --git a/dep/cubeb/include/cubeb/cubeb.h b/dep/cubeb/include/cubeb/cubeb.h new file mode 100644 index 000000000..b3fc56d2c --- /dev/null +++ b/dep/cubeb/include/cubeb/cubeb.h @@ -0,0 +1,658 @@ +/* + * Copyright © 2011 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#if !defined(CUBEB_c2f983e9_c96f_e71c_72c3_bbf62992a382) +#define CUBEB_c2f983e9_c96f_e71c_72c3_bbf62992a382 + +#include +#include +#include "cubeb_export.h" + +#if defined(__cplusplus) +extern "C" { +#endif + +/** @mainpage + + @section intro Introduction + + This is the documentation for the libcubeb C API. + libcubeb is a callback-based audio API library allowing the + authoring of portable multiplatform audio playback and recording. + + @section example Example code + + This example shows how to create a duplex stream that pipes the microphone + to the speakers, with minimal latency and the proper sample-rate for the + platform. + + @code + cubeb * app_ctx; + cubeb_init(&app_ctx, "Example Application", NULL); + int rv; + uint32_t rate; + uint32_t latency_frames; + uint64_t ts; + + rv = cubeb_get_preferred_sample_rate(app_ctx, &rate); + if (rv != CUBEB_OK) { + fprintf(stderr, "Could not get preferred sample-rate"); + return rv; + } + + cubeb_stream_params output_params; + output_params.format = CUBEB_SAMPLE_FLOAT32NE; + output_params.rate = rate; + output_params.channels = 2; + output_params.layout = CUBEB_LAYOUT_UNDEFINED; + output_params.prefs = CUBEB_STREAM_PREF_NONE; + + rv = cubeb_get_min_latency(app_ctx, &output_params, &latency_frames); + if (rv != CUBEB_OK) { + fprintf(stderr, "Could not get minimum latency"); + return rv; + } + + cubeb_stream_params input_params; + input_params.format = CUBEB_SAMPLE_FLOAT32NE; + input_params.rate = rate; + input_params.channels = 1; + input_params.layout = CUBEB_LAYOUT_UNDEFINED; + input_params.prefs = CUBEB_STREAM_PREF_NONE; + + cubeb_stream * stm; + rv = cubeb_stream_init(app_ctx, &stm, "Example Stream 1", + NULL, &input_params, + NULL, &output_params, + latency_frames, + data_cb, state_cb, + NULL); + if (rv != CUBEB_OK) { + fprintf(stderr, "Could not open the stream"); + return rv; + } + + rv = cubeb_stream_start(stm); + if (rv != CUBEB_OK) { + fprintf(stderr, "Could not start the stream"); + return rv; + } + for (;;) { + cubeb_stream_get_position(stm, &ts); + printf("time=%llu\n", ts); + sleep(1); + } + rv = cubeb_stream_stop(stm); + if (rv != CUBEB_OK) { + fprintf(stderr, "Could not stop the stream"); + return rv; + } + + cubeb_stream_destroy(stm); + cubeb_destroy(app_ctx); + @endcode + + @code + long data_cb(cubeb_stream * stm, void * user, + const void * input_buffer, void * output_buffer, long nframes) + { + const float * in = input_buffer; + float * out = output_buffer; + + for (int i = 0; i < nframes; ++i) { + for (int c = 0; c < 2; ++c) { + out[2 * i + c] = in[i]; + } + } + return nframes; + } + @endcode + + @code + void state_cb(cubeb_stream * stm, void * user, cubeb_state state) + { + printf("state=%d\n", state); + } + @endcode +*/ + +/** @file + The libcubeb C API. */ + +typedef struct cubeb cubeb; /**< Opaque handle referencing the application state. */ +typedef struct cubeb_stream cubeb_stream; /**< Opaque handle referencing the stream state. */ + +/** Sample format enumeration. */ +typedef enum { + /**< Little endian 16-bit signed PCM. */ + CUBEB_SAMPLE_S16LE, + /**< Big endian 16-bit signed PCM. */ + CUBEB_SAMPLE_S16BE, + /**< Little endian 32-bit IEEE floating point PCM. */ + CUBEB_SAMPLE_FLOAT32LE, + /**< Big endian 32-bit IEEE floating point PCM. */ + CUBEB_SAMPLE_FLOAT32BE, +#if defined(WORDS_BIGENDIAN) || defined(__BIG_ENDIAN__) + /**< Native endian 16-bit signed PCM. */ + CUBEB_SAMPLE_S16NE = CUBEB_SAMPLE_S16BE, + /**< Native endian 32-bit IEEE floating point PCM. */ + CUBEB_SAMPLE_FLOAT32NE = CUBEB_SAMPLE_FLOAT32BE +#else + /**< Native endian 16-bit signed PCM. */ + CUBEB_SAMPLE_S16NE = CUBEB_SAMPLE_S16LE, + /**< Native endian 32-bit IEEE floating point PCM. */ + CUBEB_SAMPLE_FLOAT32NE = CUBEB_SAMPLE_FLOAT32LE +#endif +} cubeb_sample_format; + +/** An opaque handle used to refer a particular input or output device + * across calls. */ +typedef void const * cubeb_devid; + +/** Level (verbosity) of logging for a particular cubeb context. */ +typedef enum { + CUBEB_LOG_DISABLED = 0, /** < Logging disabled */ + CUBEB_LOG_NORMAL = 1, /**< Logging lifetime operation (creation/destruction). */ + CUBEB_LOG_VERBOSE = 2, /**< Verbose logging of callbacks, can have performance implications. */ +} cubeb_log_level; + +typedef enum { + CHANNEL_UNKNOWN = 0, + CHANNEL_FRONT_LEFT = 1 << 0, + CHANNEL_FRONT_RIGHT = 1 << 1, + CHANNEL_FRONT_CENTER = 1 << 2, + CHANNEL_LOW_FREQUENCY = 1 << 3, + CHANNEL_BACK_LEFT = 1 << 4, + CHANNEL_BACK_RIGHT = 1 << 5, + CHANNEL_FRONT_LEFT_OF_CENTER = 1 << 6, + CHANNEL_FRONT_RIGHT_OF_CENTER = 1 << 7, + CHANNEL_BACK_CENTER = 1 << 8, + CHANNEL_SIDE_LEFT = 1 << 9, + CHANNEL_SIDE_RIGHT = 1 << 10, + CHANNEL_TOP_CENTER = 1 << 11, + CHANNEL_TOP_FRONT_LEFT = 1 << 12, + CHANNEL_TOP_FRONT_CENTER = 1 << 13, + CHANNEL_TOP_FRONT_RIGHT = 1 << 14, + CHANNEL_TOP_BACK_LEFT = 1 << 15, + CHANNEL_TOP_BACK_CENTER = 1 << 16, + CHANNEL_TOP_BACK_RIGHT = 1 << 17 +} cubeb_channel; + +typedef uint32_t cubeb_channel_layout; +// Some common layout definitions. +enum { + CUBEB_LAYOUT_UNDEFINED = 0, // Indicate the speaker's layout is undefined. + CUBEB_LAYOUT_MONO = CHANNEL_FRONT_CENTER, + CUBEB_LAYOUT_MONO_LFE = CUBEB_LAYOUT_MONO | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_STEREO = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT, + CUBEB_LAYOUT_STEREO_LFE = CUBEB_LAYOUT_STEREO | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_3F = + CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_FRONT_CENTER, + CUBEB_LAYOUT_3F_LFE = CUBEB_LAYOUT_3F | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_2F1 = + CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_BACK_CENTER, + CUBEB_LAYOUT_2F1_LFE = CUBEB_LAYOUT_2F1 | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_3F1 = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_FRONT_CENTER | CHANNEL_BACK_CENTER, + CUBEB_LAYOUT_3F1_LFE = CUBEB_LAYOUT_3F1 | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_2F2 = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT, + CUBEB_LAYOUT_2F2_LFE = CUBEB_LAYOUT_2F2 | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_QUAD = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT, + CUBEB_LAYOUT_QUAD_LFE = CUBEB_LAYOUT_QUAD | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_3F2 = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_FRONT_CENTER | CHANNEL_SIDE_LEFT | + CHANNEL_SIDE_RIGHT, + CUBEB_LAYOUT_3F2_LFE = CUBEB_LAYOUT_3F2 | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_3F2_BACK = CUBEB_LAYOUT_QUAD | CHANNEL_FRONT_CENTER, + CUBEB_LAYOUT_3F2_LFE_BACK = CUBEB_LAYOUT_3F2_BACK | CHANNEL_LOW_FREQUENCY, + CUBEB_LAYOUT_3F3R_LFE = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY | + CHANNEL_BACK_CENTER | CHANNEL_SIDE_LEFT | + CHANNEL_SIDE_RIGHT, + CUBEB_LAYOUT_3F4_LFE = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY | + CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT | + CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT, +}; + +/** Miscellaneous stream preferences. */ +typedef enum { + CUBEB_STREAM_PREF_NONE = 0x00, /**< No stream preferences are requested. */ + CUBEB_STREAM_PREF_LOOPBACK = 0x01, /**< Request a loopback stream. Should be + specified on the input params and an + output device to loopback from should + be passed in place of an input device. */ + CUBEB_STREAM_PREF_DISABLE_DEVICE_SWITCHING = 0x02, /**< Disable switching + default device on OS + changes. */ + CUBEB_STREAM_PREF_VOICE = 0x04 /**< This stream is going to transport voice data. + Depending on the backend and platform, this can + change the audio input or output devices + selected, as well as the quality of the stream, + for example to accomodate bluetooth SCO modes on + bluetooth devices. */ +} cubeb_stream_prefs; + +/** Stream format initialization parameters. */ +typedef struct { + cubeb_sample_format format; /**< Requested sample format. One of + #cubeb_sample_format. */ + uint32_t rate; /**< Requested sample rate. Valid range is [1000, 192000]. */ + uint32_t channels; /**< Requested channel count. Valid range is [1, 8]. */ + cubeb_channel_layout layout; /**< Requested channel layout. This must be consistent with the provided channels. CUBEB_LAYOUT_UNDEFINED if unknown */ + cubeb_stream_prefs prefs; /**< Requested preferences. */ +} cubeb_stream_params; + +/** Audio device description */ +typedef struct { + char * output_name; /**< The name of the output device */ + char * input_name; /**< The name of the input device */ +} cubeb_device; + +/** Stream states signaled via state_callback. */ +typedef enum { + CUBEB_STATE_STARTED, /**< Stream started. */ + CUBEB_STATE_STOPPED, /**< Stream stopped. */ + CUBEB_STATE_DRAINED, /**< Stream drained. */ + CUBEB_STATE_ERROR /**< Stream disabled due to error. */ +} cubeb_state; + +/** Result code enumeration. */ +enum { + CUBEB_OK = 0, /**< Success. */ + CUBEB_ERROR = -1, /**< Unclassified error. */ + CUBEB_ERROR_INVALID_FORMAT = -2, /**< Unsupported #cubeb_stream_params requested. */ + CUBEB_ERROR_INVALID_PARAMETER = -3, /**< Invalid parameter specified. */ + CUBEB_ERROR_NOT_SUPPORTED = -4, /**< Optional function not implemented in current backend. */ + CUBEB_ERROR_DEVICE_UNAVAILABLE = -5 /**< Device specified by #cubeb_devid not available. */ +}; + +/** + * Whether a particular device is an input device (e.g. a microphone), or an + * output device (e.g. headphones). */ +typedef enum { + CUBEB_DEVICE_TYPE_UNKNOWN, + CUBEB_DEVICE_TYPE_INPUT, + CUBEB_DEVICE_TYPE_OUTPUT +} cubeb_device_type; + +/** + * The state of a device. + */ +typedef enum { + CUBEB_DEVICE_STATE_DISABLED, /**< The device has been disabled at the system level. */ + CUBEB_DEVICE_STATE_UNPLUGGED, /**< The device is enabled, but nothing is plugged into it. */ + CUBEB_DEVICE_STATE_ENABLED /**< The device is enabled. */ +} cubeb_device_state; + +/** + * Architecture specific sample type. + */ +typedef enum { + CUBEB_DEVICE_FMT_S16LE = 0x0010, /**< 16-bit integers, Little Endian. */ + CUBEB_DEVICE_FMT_S16BE = 0x0020, /**< 16-bit integers, Big Endian. */ + CUBEB_DEVICE_FMT_F32LE = 0x1000, /**< 32-bit floating point, Little Endian. */ + CUBEB_DEVICE_FMT_F32BE = 0x2000 /**< 32-bit floating point, Big Endian. */ +} cubeb_device_fmt; + +#if defined(WORDS_BIGENDIAN) || defined(__BIG_ENDIAN__) +/** 16-bit integers, native endianess, when on a Big Endian environment. */ +#define CUBEB_DEVICE_FMT_S16NE CUBEB_DEVICE_FMT_S16BE +/** 32-bit floating points, native endianess, when on a Big Endian environment. */ +#define CUBEB_DEVICE_FMT_F32NE CUBEB_DEVICE_FMT_F32BE +#else +/** 16-bit integers, native endianess, when on a Little Endian environment. */ +#define CUBEB_DEVICE_FMT_S16NE CUBEB_DEVICE_FMT_S16LE +/** 32-bit floating points, native endianess, when on a Little Endian + * environment. */ +#define CUBEB_DEVICE_FMT_F32NE CUBEB_DEVICE_FMT_F32LE +#endif +/** All the 16-bit integers types. */ +#define CUBEB_DEVICE_FMT_S16_MASK (CUBEB_DEVICE_FMT_S16LE | CUBEB_DEVICE_FMT_S16BE) +/** All the 32-bit floating points types. */ +#define CUBEB_DEVICE_FMT_F32_MASK (CUBEB_DEVICE_FMT_F32LE | CUBEB_DEVICE_FMT_F32BE) +/** All the device formats types. */ +#define CUBEB_DEVICE_FMT_ALL (CUBEB_DEVICE_FMT_S16_MASK | CUBEB_DEVICE_FMT_F32_MASK) + +/** Channel type for a `cubeb_stream`. Depending on the backend and platform + * used, this can control inter-stream interruption, ducking, and volume + * control. + */ +typedef enum { + CUBEB_DEVICE_PREF_NONE = 0x00, + CUBEB_DEVICE_PREF_MULTIMEDIA = 0x01, + CUBEB_DEVICE_PREF_VOICE = 0x02, + CUBEB_DEVICE_PREF_NOTIFICATION = 0x04, + CUBEB_DEVICE_PREF_ALL = 0x0F +} cubeb_device_pref; + +/** This structure holds the characteristics + * of an input or output audio device. It is obtained using + * `cubeb_enumerate_devices`, which returns these structures via + * `cubeb_device_collection` and must be destroyed via + * `cubeb_device_collection_destroy`. */ +typedef struct { + cubeb_devid devid; /**< Device identifier handle. */ + char const * device_id; /**< Device identifier which might be presented in a UI. */ + char const * friendly_name; /**< Friendly device name which might be presented in a UI. */ + char const * group_id; /**< Two devices have the same group identifier if they belong to the same physical device; for example a headset and microphone. */ + char const * vendor_name; /**< Optional vendor name, may be NULL. */ + + cubeb_device_type type; /**< Type of device (Input/Output). */ + cubeb_device_state state; /**< State of device disabled/enabled/unplugged. */ + cubeb_device_pref preferred;/**< Preferred device. */ + + cubeb_device_fmt format; /**< Sample format supported. */ + cubeb_device_fmt default_format; /**< The default sample format for this device. */ + uint32_t max_channels; /**< Channels. */ + uint32_t default_rate; /**< Default/Preferred sample rate. */ + uint32_t max_rate; /**< Maximum sample rate supported. */ + uint32_t min_rate; /**< Minimum sample rate supported. */ + + uint32_t latency_lo; /**< Lowest possible latency in frames. */ + uint32_t latency_hi; /**< Higest possible latency in frames. */ +} cubeb_device_info; + +/** Device collection. + * Returned by `cubeb_enumerate_devices` and destroyed by + * `cubeb_device_collection_destroy`. */ +typedef struct { + cubeb_device_info * device; /**< Array of pointers to device info. */ + size_t count; /**< Device count in collection. */ +} cubeb_device_collection; + +/** User supplied data callback. + - Calling other cubeb functions from this callback is unsafe. + - The code in the callback should be non-blocking. + - Returning less than the number of frames this callback asks for or + provides puts the stream in drain mode. This callback will not be called + again, and the state callback will be called with CUBEB_STATE_DRAINED when + all the frames have been output. + @param stream The stream for which this callback fired. + @param user_ptr The pointer passed to cubeb_stream_init. + @param input_buffer A pointer containing the input data, or nullptr + if this is an output-only stream. + @param output_buffer A pointer to a buffer to be filled with audio samples, + or nullptr if this is an input-only stream. + @param nframes The number of frames of the two buffer. + @retval If the stream has output, this is the number of frames written to + the output buffer. In this case, if this number is less than + nframes then the stream will start to drain. If the stream is + input only, then returning nframes indicates data has been read. + In this case, a value less than nframes will result in the stream + being stopped. + @retval CUBEB_ERROR on error, in which case the data callback will stop + and the stream will enter a shutdown state. */ +typedef long (* cubeb_data_callback)(cubeb_stream * stream, + void * user_ptr, + void const * input_buffer, + void * output_buffer, + long nframes); + +/** User supplied state callback. + @param stream The stream for this this callback fired. + @param user_ptr The pointer passed to cubeb_stream_init. + @param state The new state of the stream. */ +typedef void (* cubeb_state_callback)(cubeb_stream * stream, + void * user_ptr, + cubeb_state state); + +/** + * User supplied callback called when the underlying device changed. + * @param user The pointer passed to cubeb_stream_init. */ +typedef void (* cubeb_device_changed_callback)(void * user_ptr); + +/** + * User supplied callback called when the underlying device collection changed. + * @param context A pointer to the cubeb context. + * @param user_ptr The pointer passed to cubeb_register_device_collection_changed. */ +typedef void (* cubeb_device_collection_changed_callback)(cubeb * context, + void * user_ptr); + +/** User supplied callback called when a message needs logging. */ +typedef void (* cubeb_log_callback)(char const * fmt, ...); + +/** Initialize an application context. This will perform any library or + application scoped initialization. + + Note: On Windows platforms, COM must be initialized in MTA mode on + any thread that will call the cubeb API. + + @param context A out param where an opaque pointer to the application + context will be returned. + @param context_name A name for the context. Depending on the platform this + can appear in different locations. + @param backend_name The name of the cubeb backend user desires to select. + Accepted values self-documented in cubeb.c: init_oneshot + If NULL, a default ordering is used for backend choice. + A valid choice overrides all other possible backends, + so long as the backend was included at compile time. + @retval CUBEB_OK in case of success. + @retval CUBEB_ERROR in case of error, for example because the host + has no audio hardware. */ +CUBEB_EXPORT int cubeb_init(cubeb ** context, char const * context_name, + char const * backend_name); + +/** Get a read-only string identifying this context's current backend. + @param context A pointer to the cubeb context. + @retval Read-only string identifying current backend. */ +CUBEB_EXPORT char const * cubeb_get_backend_id(cubeb * context); + +/** Get the maximum possible number of channels. + @param context A pointer to the cubeb context. + @param max_channels The maximum number of channels. + @retval CUBEB_OK + @retval CUBEB_ERROR_INVALID_PARAMETER + @retval CUBEB_ERROR_NOT_SUPPORTED + @retval CUBEB_ERROR */ +CUBEB_EXPORT int cubeb_get_max_channel_count(cubeb * context, uint32_t * max_channels); + +/** Get the minimal latency value, in frames, that is guaranteed to work + when creating a stream for the specified sample rate. This is platform, + hardware and backend dependent. + @param context A pointer to the cubeb context. + @param params On some backends, the minimum achievable latency depends on + the characteristics of the stream. + @param latency_frames The latency value, in frames, to pass to + cubeb_stream_init. + @retval CUBEB_OK + @retval CUBEB_ERROR_INVALID_PARAMETER + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_get_min_latency(cubeb * context, + cubeb_stream_params * params, + uint32_t * latency_frames); + +/** Get the preferred sample rate for this backend: this is hardware and + platform dependent, and can avoid resampling, and/or trigger fastpaths. + @param context A pointer to the cubeb context. + @param rate The samplerate (in Hz) the current configuration prefers. + @retval CUBEB_OK + @retval CUBEB_ERROR_INVALID_PARAMETER + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_get_preferred_sample_rate(cubeb * context, uint32_t * rate); + +/** Destroy an application context. This must be called after all stream have + * been destroyed. + @param context A pointer to the cubeb context.*/ +CUBEB_EXPORT void cubeb_destroy(cubeb * context); + +/** Initialize a stream associated with the supplied application context. + @param context A pointer to the cubeb context. + @param stream An out parameter to be filled with the an opaque pointer to a + cubeb stream. + @param stream_name A name for this stream. + @param input_device Device for the input side of the stream. If NULL the + default input device is used. + @param input_stream_params Parameters for the input side of the stream, or + NULL if this stream is output only. + @param output_device Device for the output side of the stream. If NULL the + default output device is used. + @param output_stream_params Parameters for the output side of the stream, or + NULL if this stream is input only. + @param latency_frames Stream latency in frames. Valid range + is [1, 96000]. + @param data_callback Will be called to preroll data before playback is + started by cubeb_stream_start. + @param state_callback A pointer to a state callback. + @param user_ptr A pointer that will be passed to the callbacks. This pointer + must outlive the life time of the stream. + @retval CUBEB_OK + @retval CUBEB_ERROR + @retval CUBEB_ERROR_INVALID_FORMAT + @retval CUBEB_ERROR_DEVICE_UNAVAILABLE */ +CUBEB_EXPORT int cubeb_stream_init(cubeb * context, + cubeb_stream ** stream, + char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + uint32_t latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr); + +/** Destroy a stream. `cubeb_stream_stop` MUST be called before destroying a + stream. + @param stream The stream to destroy. */ +CUBEB_EXPORT void cubeb_stream_destroy(cubeb_stream * stream); + +/** Start playback. + @param stream + @retval CUBEB_OK + @retval CUBEB_ERROR */ +CUBEB_EXPORT int cubeb_stream_start(cubeb_stream * stream); + +/** Stop playback. + @param stream + @retval CUBEB_OK + @retval CUBEB_ERROR */ +CUBEB_EXPORT int cubeb_stream_stop(cubeb_stream * stream); + +/** Reset stream to the default device. + @param stream + @retval CUBEB_OK + @retval CUBEB_ERROR_INVALID_PARAMETER + @retval CUBEB_ERROR_NOT_SUPPORTED + @retval CUBEB_ERROR */ +CUBEB_EXPORT int cubeb_stream_reset_default_device(cubeb_stream * stream); + +/** Get the current stream playback position. + @param stream + @param position Playback position in frames. + @retval CUBEB_OK + @retval CUBEB_ERROR */ +CUBEB_EXPORT int cubeb_stream_get_position(cubeb_stream * stream, uint64_t * position); + +/** Get the latency for this stream, in frames. This is the number of frames + between the time cubeb acquires the data in the callback and the listener + can hear the sound. + @param stream + @param latency Current approximate stream latency in frames. + @retval CUBEB_OK + @retval CUBEB_ERROR_NOT_SUPPORTED + @retval CUBEB_ERROR */ +CUBEB_EXPORT int cubeb_stream_get_latency(cubeb_stream * stream, uint32_t * latency); + +/** Set the volume for a stream. + @param stream the stream for which to adjust the volume. + @param volume a float between 0.0 (muted) and 1.0 (maximum volume) + @retval CUBEB_OK + @retval CUBEB_ERROR_INVALID_PARAMETER volume is outside [0.0, 1.0] or + stream is an invalid pointer + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_stream_set_volume(cubeb_stream * stream, float volume); + +/** Get the current output device for this stream. + @param stm the stream for which to query the current output device + @param device a pointer in which the current output device will be stored. + @retval CUBEB_OK in case of success + @retval CUBEB_ERROR_INVALID_PARAMETER if either stm, device or count are + invalid pointers + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_stream_get_current_device(cubeb_stream * stm, + cubeb_device ** const device); + +/** Destroy a cubeb_device structure. + @param stream the stream passed in cubeb_stream_get_current_device + @param devices the devices to destroy + @retval CUBEB_OK in case of success + @retval CUBEB_ERROR_INVALID_PARAMETER if devices is an invalid pointer + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_stream_device_destroy(cubeb_stream * stream, + cubeb_device * devices); + +/** Set a callback to be notified when the output device changes. + @param stream the stream for which to set the callback. + @param device_changed_callback a function called whenever the device has + changed. Passing NULL allow to unregister a function + @retval CUBEB_OK + @retval CUBEB_ERROR_INVALID_PARAMETER if either stream or + device_changed_callback are invalid pointers. + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_stream_register_device_changed_callback(cubeb_stream * stream, + cubeb_device_changed_callback device_changed_callback); + +/** Return the user data pointer registered with the stream with cubeb_stream_init. + @param stream the stream for which to retrieve user data pointer. + @retval user data pointer */ +CUBEB_EXPORT void * cubeb_stream_user_ptr(cubeb_stream * stream); + +/** Returns enumerated devices. + @param context + @param devtype device type to include + @param collection output collection. Must be destroyed with cubeb_device_collection_destroy + @retval CUBEB_OK in case of success + @retval CUBEB_ERROR_INVALID_PARAMETER if collection is an invalid pointer + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_enumerate_devices(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection * collection); + +/** Destroy a cubeb_device_collection, and its `cubeb_device_info`. + @param context + @param collection collection to destroy + @retval CUBEB_OK + @retval CUBEB_ERROR_INVALID_PARAMETER if collection is an invalid pointer */ +CUBEB_EXPORT int cubeb_device_collection_destroy(cubeb * context, + cubeb_device_collection * collection); + +/** Registers a callback which is called when the system detects + a new device or a device is removed. + @param context + @param devtype device type to include. Different callbacks and user pointers + can be registered for each devtype. The hybrid devtype + `CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_OUTPUT` is also valid + and will register the provided callback and user pointer in both sides. + @param callback a function called whenever the system device list changes. + Passing NULL allow to unregister a function. You have to unregister + first before you register a new callback. + @param user_ptr pointer to user specified data which will be present in + subsequent callbacks. + @retval CUBEB_ERROR_NOT_SUPPORTED */ +CUBEB_EXPORT int cubeb_register_device_collection_changed(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection_changed_callback callback, + void * user_ptr); + +/** Set a callback to be called with a message. + @param log_level CUBEB_LOG_VERBOSE, CUBEB_LOG_NORMAL. + @param log_callback A function called with a message when there is + something to log. Pass NULL to unregister. + @retval CUBEB_OK in case of success. + @retval CUBEB_ERROR_INVALID_PARAMETER if either context or log_callback are + invalid pointers, or if level is not + in cubeb_log_level. */ +CUBEB_EXPORT int cubeb_set_log_callback(cubeb_log_level log_level, + cubeb_log_callback log_callback); + +#if defined(__cplusplus) +} +#endif + +#endif /* CUBEB_c2f983e9_c96f_e71c_72c3_bbf62992a382 */ diff --git a/dep/cubeb/include/cubeb/cubeb_export.h b/dep/cubeb/include/cubeb/cubeb_export.h new file mode 100644 index 000000000..fa4cf0ca0 --- /dev/null +++ b/dep/cubeb/include/cubeb/cubeb_export.h @@ -0,0 +1,12 @@ +#ifndef CUBEB_EXPORT_H +#define CUBEB_EXPORT_H + +#define CUBEB_EXPORT +#define CUBEB_NO_EXPORT + +#ifdef WIN32 +#pragma comment(lib, "winmm.lib") +#pragma comment(lib, "avrt.lib") +#endif + +#endif diff --git a/dep/cubeb/src/android/audiotrack_definitions.h b/dep/cubeb/src/android/audiotrack_definitions.h new file mode 100644 index 000000000..cd501533d --- /dev/null +++ b/dep/cubeb/src/android/audiotrack_definitions.h @@ -0,0 +1,81 @@ +/* + * Copyright (C) 2008 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +/* + * The following definitions are copied from the android sources. Only the + * relevant enum member and values needed are copied. + */ + +/* + * From https://android.googlesource.com/platform/frameworks/base/+/android-2.2.3_r2.1/include/utils/Errors.h + */ +typedef int32_t status_t; + +/* + * From https://android.googlesource.com/platform/frameworks/base/+/android-2.2.3_r2.1/include/media/AudioTrack.h + */ +struct Buffer { + uint32_t flags; + int channelCount; + int format; + size_t frameCount; + size_t size; + union { + void* raw; + short* i16; + int8_t* i8; + }; +}; + +enum event_type { + EVENT_MORE_DATA = 0, + EVENT_UNDERRUN = 1, + EVENT_LOOP_END = 2, + EVENT_MARKER = 3, + EVENT_NEW_POS = 4, + EVENT_BUFFER_END = 5 +}; + +/** + * From https://android.googlesource.com/platform/frameworks/base/+/android-2.2.3_r2.1/include/media/AudioSystem.h + * and + * https://android.googlesource.com/platform/system/core/+/android-4.2.2_r1/include/system/audio.h + */ + +#define AUDIO_STREAM_TYPE_MUSIC 3 + +enum { + AUDIO_CHANNEL_OUT_FRONT_LEFT_ICS = 0x1, + AUDIO_CHANNEL_OUT_FRONT_RIGHT_ICS = 0x2, + AUDIO_CHANNEL_OUT_MONO_ICS = AUDIO_CHANNEL_OUT_FRONT_LEFT_ICS, + AUDIO_CHANNEL_OUT_STEREO_ICS = (AUDIO_CHANNEL_OUT_FRONT_LEFT_ICS | AUDIO_CHANNEL_OUT_FRONT_RIGHT_ICS) +} AudioTrack_ChannelMapping_ICS; + +enum { + AUDIO_CHANNEL_OUT_FRONT_LEFT_Legacy = 0x4, + AUDIO_CHANNEL_OUT_FRONT_RIGHT_Legacy = 0x8, + AUDIO_CHANNEL_OUT_MONO_Legacy = AUDIO_CHANNEL_OUT_FRONT_LEFT_Legacy, + AUDIO_CHANNEL_OUT_STEREO_Legacy = (AUDIO_CHANNEL_OUT_FRONT_LEFT_Legacy | AUDIO_CHANNEL_OUT_FRONT_RIGHT_Legacy) +} AudioTrack_ChannelMapping_Legacy; + +typedef enum { + AUDIO_FORMAT_PCM = 0x00000000, + AUDIO_FORMAT_PCM_SUB_16_BIT = 0x1, + AUDIO_FORMAT_PCM_16_BIT = (AUDIO_FORMAT_PCM | AUDIO_FORMAT_PCM_SUB_16_BIT), +} AudioTrack_SampleType; + diff --git a/dep/cubeb/src/android/cubeb-output-latency.h b/dep/cubeb/src/android/cubeb-output-latency.h new file mode 100644 index 000000000..2128cd176 --- /dev/null +++ b/dep/cubeb/src/android/cubeb-output-latency.h @@ -0,0 +1,76 @@ +#ifndef _CUBEB_OUTPUT_LATENCY_H_ +#define _CUBEB_OUTPUT_LATENCY_H_ + +#include +#include "cubeb_media_library.h" +#include "../cubeb-jni.h" + +struct output_latency_function { + media_lib * from_lib; + cubeb_jni * from_jni; + int version; +}; + +typedef struct output_latency_function output_latency_function; + +const int ANDROID_JELLY_BEAN_MR1_4_2 = 17; + +output_latency_function * +cubeb_output_latency_load_method(int version) +{ + output_latency_function * ol = NULL; + ol = calloc(1, sizeof(output_latency_function)); + + ol->version = version; + + if (ol->version > ANDROID_JELLY_BEAN_MR1_4_2){ + ol->from_jni = cubeb_jni_init(); + return ol; + } + + ol->from_lib = cubeb_load_media_library(); + return ol; +} + +bool +cubeb_output_latency_method_is_loaded(output_latency_function * ol) +{ + assert(ol); + if (ol->version > ANDROID_JELLY_BEAN_MR1_4_2){ + return !!ol->from_jni; + } + + return !!ol->from_lib; +} + +void +cubeb_output_latency_unload_method(output_latency_function * ol) +{ + if (!ol) { + return; + } + + if (ol->version > ANDROID_JELLY_BEAN_MR1_4_2 && ol->from_jni) { + cubeb_jni_destroy(ol->from_jni); + } + + if (ol->version <= ANDROID_JELLY_BEAN_MR1_4_2 && ol->from_lib) { + cubeb_close_media_library(ol->from_lib); + } + + free(ol); +} + +uint32_t +cubeb_get_output_latency(output_latency_function * ol) +{ + assert(cubeb_output_latency_method_is_loaded(ol)); + + if (ol->version > ANDROID_JELLY_BEAN_MR1_4_2){ + return cubeb_get_output_latency_from_jni(ol->from_jni); + } + + return cubeb_get_output_latency_from_media_library(ol->from_lib); +} + +#endif // _CUBEB_OUTPUT_LATENCY_H_ diff --git a/dep/cubeb/src/android/cubeb_media_library.h b/dep/cubeb/src/android/cubeb_media_library.h new file mode 100644 index 000000000..ab21b779d --- /dev/null +++ b/dep/cubeb/src/android/cubeb_media_library.h @@ -0,0 +1,62 @@ +#ifndef _CUBEB_MEDIA_LIBRARY_H_ +#define _CUBEB_MEDIA_LIBRARY_H_ + +struct media_lib { + void * libmedia; + int32_t (* get_output_latency)(uint32_t * latency, int stream_type); +}; + +typedef struct media_lib media_lib; + +media_lib * +cubeb_load_media_library() +{ + media_lib ml = {0}; + ml.libmedia = dlopen("libmedia.so", RTLD_LAZY); + if (!ml.libmedia) { + return NULL; + } + + // Get the latency, in ms, from AudioFlinger. First, try the most recent signature. + // status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t streamType) + ml.get_output_latency = + dlsym(ml.libmedia, "_ZN7android11AudioSystem16getOutputLatencyEPj19audio_stream_type_t"); + if (!ml.get_output_latency) { + // In case of failure, try the signature from legacy version. + // status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType) + ml.get_output_latency = + dlsym(ml.libmedia, "_ZN7android11AudioSystem16getOutputLatencyEPji"); + if (!ml.get_output_latency) { + return NULL; + } + } + + media_lib * rv = NULL; + rv = calloc(1, sizeof(media_lib)); + assert(rv); + *rv = ml; + return rv; +} + +void +cubeb_close_media_library(media_lib * ml) +{ + dlclose(ml->libmedia); + ml->libmedia = NULL; + ml->get_output_latency = NULL; + free(ml); +} + +uint32_t +cubeb_get_output_latency_from_media_library(media_lib * ml) +{ + uint32_t latency = 0; + const int audio_stream_type_music = 3; + int32_t r = ml->get_output_latency(&latency, audio_stream_type_music); + if (r) { + return 0; + } + return latency; +} + +#endif // _CUBEB_MEDIA_LIBRARY_H_ diff --git a/dep/cubeb/src/android/sles_definitions.h b/dep/cubeb/src/android/sles_definitions.h new file mode 100644 index 000000000..06d2e8d49 --- /dev/null +++ b/dep/cubeb/src/android/sles_definitions.h @@ -0,0 +1,102 @@ +/* + * Copyright (C) 2010 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +/** + * This file is similar to the file "OpenSLES_AndroidConfiguration.h" found in + * the Android NDK, but removes the #ifdef __cplusplus defines, so we can keep + * using a C compiler in cubeb. + */ + +#ifndef OPENSL_ES_ANDROIDCONFIGURATION_H_ +#define OPENSL_ES_ANDROIDCONFIGURATION_H_ + +/*---------------------------------------------------------------------------*/ +/* Android AudioRecorder configuration */ +/*---------------------------------------------------------------------------*/ + +/** Audio recording preset */ +/** Audio recording preset key */ +#define SL_ANDROID_KEY_RECORDING_PRESET ((const SLchar*) "androidRecordingPreset") +/** Audio recording preset values */ +/** preset "none" cannot be set, it is used to indicate the current settings + * do not match any of the presets. */ +#define SL_ANDROID_RECORDING_PRESET_NONE ((SLuint32) 0x00000000) +/** generic recording configuration on the platform */ +#define SL_ANDROID_RECORDING_PRESET_GENERIC ((SLuint32) 0x00000001) +/** uses the microphone audio source with the same orientation as the camera + * if available, the main device microphone otherwise */ +#define SL_ANDROID_RECORDING_PRESET_CAMCORDER ((SLuint32) 0x00000002) +/** uses the main microphone tuned for voice recognition */ +#define SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION ((SLuint32) 0x00000003) +/** uses the main microphone tuned for audio communications */ +#define SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION ((SLuint32) 0x00000004) +/** uses the main microphone unprocessed */ +#define SL_ANDROID_RECORDING_PRESET_UNPROCESSED ((SLuint32) 0x00000005) + + +/*---------------------------------------------------------------------------*/ +/* Android AudioPlayer configuration */ +/*---------------------------------------------------------------------------*/ + +/** Audio playback stream type */ +/** Audio playback stream type key */ +#define SL_ANDROID_KEY_STREAM_TYPE ((const SLchar*) "androidPlaybackStreamType") + +/** Audio playback stream type values */ +/* same as android.media.AudioManager.STREAM_VOICE_CALL */ +#define SL_ANDROID_STREAM_VOICE ((SLint32) 0x00000000) +/* same as android.media.AudioManager.STREAM_SYSTEM */ +#define SL_ANDROID_STREAM_SYSTEM ((SLint32) 0x00000001) +/* same as android.media.AudioManager.STREAM_RING */ +#define SL_ANDROID_STREAM_RING ((SLint32) 0x00000002) +/* same as android.media.AudioManager.STREAM_MUSIC */ +#define SL_ANDROID_STREAM_MEDIA ((SLint32) 0x00000003) +/* same as android.media.AudioManager.STREAM_ALARM */ +#define SL_ANDROID_STREAM_ALARM ((SLint32) 0x00000004) +/* same as android.media.AudioManager.STREAM_NOTIFICATION */ +#define SL_ANDROID_STREAM_NOTIFICATION ((SLint32) 0x00000005) + + +/*---------------------------------------------------------------------------*/ +/* Android AudioPlayer and AudioRecorder configuration */ +/*---------------------------------------------------------------------------*/ + +/** Audio Performance mode. + * Performance mode tells the framework how to configure the audio path + * for a player or recorder according to application performance and + * functional requirements. + * It affects the output or input latency based on acceptable tradeoffs on + * battery drain and use of pre or post processing effects. + * Performance mode should be set before realizing the object and should be + * read after realizing the object to check if the requested mode could be + * granted or not. + */ +/** Audio Performance mode key */ +#define SL_ANDROID_KEY_PERFORMANCE_MODE ((const SLchar*) "androidPerformanceMode") + +/** Audio performance values */ +/* No specific performance requirement. Allows HW and SW pre/post processing. */ +#define SL_ANDROID_PERFORMANCE_NONE ((SLuint32) 0x00000000) +/* Priority given to latency. No HW or software pre/post processing. + * This is the default if no performance mode is specified. */ +#define SL_ANDROID_PERFORMANCE_LATENCY ((SLuint32) 0x00000001) +/* Priority given to latency while still allowing HW pre and post processing. */ +#define SL_ANDROID_PERFORMANCE_LATENCY_EFFECTS ((SLuint32) 0x00000002) +/* Priority given to power saving if latency is not a concern. + * Allows HW and SW pre/post processing. */ +#define SL_ANDROID_PERFORMANCE_POWER_SAVING ((SLuint32) 0x00000003) + +#endif /* OPENSL_ES_ANDROIDCONFIGURATION_H_ */ diff --git a/dep/cubeb/src/cubeb-internal.h b/dep/cubeb/src/cubeb-internal.h new file mode 100644 index 000000000..312a9ea3a --- /dev/null +++ b/dep/cubeb/src/cubeb-internal.h @@ -0,0 +1,79 @@ +/* + * Copyright © 2013 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#if !defined(CUBEB_INTERNAL_0eb56756_4e20_4404_a76d_42bf88cd15a5) +#define CUBEB_INTERNAL_0eb56756_4e20_4404_a76d_42bf88cd15a5 + +#include "cubeb/cubeb.h" +#include "cubeb_log.h" +#include "cubeb_assert.h" +#include +#include + +#ifdef __clang__ +#ifndef CLANG_ANALYZER_NORETURN +#if __has_feature(attribute_analyzer_noreturn) +#define CLANG_ANALYZER_NORETURN __attribute__((analyzer_noreturn)) +#else +#define CLANG_ANALYZER_NORETURN +#endif // ifndef CLANG_ANALYZER_NORETURN +#endif // __has_feature(attribute_analyzer_noreturn) +#else // __clang__ +#define CLANG_ANALYZER_NORETURN +#endif + +#if defined(__cplusplus) +extern "C" { +#endif + +#if defined(__cplusplus) +} +#endif + +struct cubeb_ops { + int (* init)(cubeb ** context, char const * context_name); + char const * (* get_backend_id)(cubeb * context); + int (* get_max_channel_count)(cubeb * context, uint32_t * max_channels); + int (* get_min_latency)(cubeb * context, + cubeb_stream_params params, + uint32_t * latency_ms); + int (* get_preferred_sample_rate)(cubeb * context, uint32_t * rate); + int (* enumerate_devices)(cubeb * context, cubeb_device_type type, + cubeb_device_collection * collection); + int (* device_collection_destroy)(cubeb * context, + cubeb_device_collection * collection); + void (* destroy)(cubeb * context); + int (* stream_init)(cubeb * context, + cubeb_stream ** stream, + char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr); + void (* stream_destroy)(cubeb_stream * stream); + int (* stream_start)(cubeb_stream * stream); + int (* stream_stop)(cubeb_stream * stream); + int (* stream_reset_default_device)(cubeb_stream * stream); + int (* stream_get_position)(cubeb_stream * stream, uint64_t * position); + int (* stream_get_latency)(cubeb_stream * stream, uint32_t * latency); + int (* stream_set_volume)(cubeb_stream * stream, float volumes); + int (* stream_get_current_device)(cubeb_stream * stream, + cubeb_device ** const device); + int (* stream_device_destroy)(cubeb_stream * stream, + cubeb_device * device); + int (* stream_register_device_changed_callback)(cubeb_stream * stream, + cubeb_device_changed_callback device_changed_callback); + int (* register_device_collection_changed)(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection_changed_callback callback, + void * user_ptr); +}; + +#endif /* CUBEB_INTERNAL_0eb56756_4e20_4404_a76d_42bf88cd15a5 */ diff --git a/dep/cubeb/src/cubeb-jni-instances.h b/dep/cubeb/src/cubeb-jni-instances.h new file mode 100644 index 000000000..4fed04659 --- /dev/null +++ b/dep/cubeb/src/cubeb-jni-instances.h @@ -0,0 +1,30 @@ +#ifndef _CUBEB_JNI_INSTANCES_H_ +#define _CUBEB_JNI_INSTANCES_H_ + +/* + * The methods in this file offer a way to pass in the required + * JNI instances in the cubeb library. By default they return NULL. + * In this case part of the cubeb API that depends on JNI + * will return CUBEB_ERROR_NOT_SUPPORTED. Currently only one + * method depends on that: + * + * cubeb_stream_get_position() + * + * Users that want to use that cubeb API method must "override" + * the methods bellow to return a valid instance of JavaVM + * and application's Context object. + * */ + +JNIEnv * +cubeb_get_jni_env_for_thread() +{ + return nullptr; +} + +jobject +cubeb_jni_get_context_instance() +{ + return nullptr; +} + +#endif //_CUBEB_JNI_INSTANCES_H_ diff --git a/dep/cubeb/src/cubeb-jni.cpp b/dep/cubeb/src/cubeb-jni.cpp new file mode 100644 index 000000000..a5066967a --- /dev/null +++ b/dep/cubeb/src/cubeb-jni.cpp @@ -0,0 +1,68 @@ +#include "jni.h" +#include +#include "cubeb-jni-instances.h" + +#define AUDIO_STREAM_TYPE_MUSIC 3 + +struct cubeb_jni { + jobject s_audio_manager_obj = nullptr; + jclass s_audio_manager_class = nullptr; + jmethodID s_get_output_latency_id = nullptr; +}; + +extern "C" +cubeb_jni * +cubeb_jni_init() +{ + jobject ctx_obj = cubeb_jni_get_context_instance(); + JNIEnv * jni_env = cubeb_get_jni_env_for_thread(); + if (!jni_env || !ctx_obj) { + return nullptr; + } + + cubeb_jni * cubeb_jni_ptr = new cubeb_jni; + assert(cubeb_jni_ptr); + + // Find the audio manager object and make it global to call it from another method + jclass context_class = jni_env->FindClass("android/content/Context"); + jfieldID audio_service_field = jni_env->GetStaticFieldID(context_class, "AUDIO_SERVICE", "Ljava/lang/String;"); + jstring jstr = (jstring)jni_env->GetStaticObjectField(context_class, audio_service_field); + jmethodID get_system_service_id = jni_env->GetMethodID(context_class, "getSystemService", "(Ljava/lang/String;)Ljava/lang/Object;"); + jobject audio_manager_obj = jni_env->CallObjectMethod(ctx_obj, get_system_service_id, jstr); + cubeb_jni_ptr->s_audio_manager_obj = reinterpret_cast(jni_env->NewGlobalRef(audio_manager_obj)); + + // Make the audio manager class a global reference in order to preserve method id + jclass audio_manager_class = jni_env->FindClass("android/media/AudioManager"); + cubeb_jni_ptr->s_audio_manager_class = reinterpret_cast(jni_env->NewGlobalRef(audio_manager_class)); + cubeb_jni_ptr->s_get_output_latency_id = jni_env->GetMethodID (audio_manager_class, "getOutputLatency", "(I)I"); + + jni_env->DeleteLocalRef(ctx_obj); + jni_env->DeleteLocalRef(context_class); + jni_env->DeleteLocalRef(jstr); + jni_env->DeleteLocalRef(audio_manager_obj); + jni_env->DeleteLocalRef(audio_manager_class); + + return cubeb_jni_ptr; +} + +extern "C" +int cubeb_get_output_latency_from_jni(cubeb_jni * cubeb_jni_ptr) +{ + assert(cubeb_jni_ptr); + JNIEnv * jni_env = cubeb_get_jni_env_for_thread(); + return jni_env->CallIntMethod(cubeb_jni_ptr->s_audio_manager_obj, cubeb_jni_ptr->s_get_output_latency_id, AUDIO_STREAM_TYPE_MUSIC); //param: AudioManager.STREAM_MUSIC +} + +extern "C" +void cubeb_jni_destroy(cubeb_jni * cubeb_jni_ptr) +{ + assert(cubeb_jni_ptr); + + JNIEnv * jni_env = cubeb_get_jni_env_for_thread(); + assert(jni_env); + + jni_env->DeleteGlobalRef(cubeb_jni_ptr->s_audio_manager_obj); + jni_env->DeleteGlobalRef(cubeb_jni_ptr->s_audio_manager_class); + + delete cubeb_jni_ptr; +} diff --git a/dep/cubeb/src/cubeb-jni.h b/dep/cubeb/src/cubeb-jni.h new file mode 100644 index 000000000..8c7ddb6ac --- /dev/null +++ b/dep/cubeb/src/cubeb-jni.h @@ -0,0 +1,10 @@ +#ifndef _CUBEB_JNI_H_ +#define _CUBEB_JNI_H_ + +typedef struct cubeb_jni cubeb_jni; + +cubeb_jni * cubeb_jni_init(); +int cubeb_get_output_latency_from_jni(cubeb_jni * cubeb_jni_ptr); +void cubeb_jni_destroy(cubeb_jni * cubeb_jni_ptr); + +#endif // _CUBEB_JNI_H_ diff --git a/dep/cubeb/src/cubeb-sles.h b/dep/cubeb/src/cubeb-sles.h new file mode 100644 index 000000000..ac22150e1 --- /dev/null +++ b/dep/cubeb/src/cubeb-sles.h @@ -0,0 +1,43 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef _CUBEB_SLES_H_ +#define _CUBEB_SLES_H_ +#include + +static SLresult +cubeb_get_sles_engine(SLObjectItf * pEngine, + SLuint32 numOptions, + const SLEngineOption * pEngineOptions, + SLuint32 numInterfaces, + const SLInterfaceID * pInterfaceIds, + const SLboolean * pInterfaceRequired) +{ + return slCreateEngine(pEngine, + numOptions, + pEngineOptions, + numInterfaces, + pInterfaceIds, + pInterfaceRequired); +} + +static void +cubeb_destroy_sles_engine(SLObjectItf * self) +{ + if (*self != NULL) { + (**self)->Destroy(*self); + *self = NULL; + } +} + +static SLresult +cubeb_realize_sles_engine(SLObjectItf self) +{ + return (*self)->Realize(self, SL_BOOLEAN_FALSE); +} + +#endif diff --git a/dep/cubeb/src/cubeb-speex-resampler.h b/dep/cubeb/src/cubeb-speex-resampler.h new file mode 100644 index 000000000..9ecf747cb --- /dev/null +++ b/dep/cubeb/src/cubeb-speex-resampler.h @@ -0,0 +1 @@ +#include diff --git a/dep/cubeb/src/cubeb.c b/dep/cubeb/src/cubeb.c new file mode 100644 index 000000000..a43d012a4 --- /dev/null +++ b/dep/cubeb/src/cubeb.c @@ -0,0 +1,665 @@ +/* + * Copyright © 2013 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#undef NDEBUG +#include +#include +#include +#include +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" + +#define NELEMS(x) ((int) (sizeof(x) / sizeof(x[0]))) + +struct cubeb { + struct cubeb_ops * ops; +}; + +struct cubeb_stream { + /* + * Note: All implementations of cubeb_stream must keep the following + * layout. + */ + struct cubeb * context; + void * user_ptr; +}; + +#if defined(USE_PULSE) +int pulse_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_PULSE_RUST) +int pulse_rust_init(cubeb ** contet, char const * context_name); +#endif +#if defined(USE_JACK) +int jack_init (cubeb ** context, char const * context_name); +#endif +#if defined(USE_ALSA) +int alsa_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_AUDIOUNIT) +int audiounit_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_AUDIOUNIT_RUST) +int audiounit_rust_init(cubeb ** contet, char const * context_name); +#endif +#if defined(USE_WINMM) +int winmm_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_WASAPI) +int wasapi_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_SNDIO) +int sndio_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_SUN) +int sun_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_OPENSL) +int opensl_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_AUDIOTRACK) +int audiotrack_init(cubeb ** context, char const * context_name); +#endif +#if defined(USE_KAI) +int kai_init(cubeb ** context, char const * context_name); +#endif + +static int +validate_stream_params(cubeb_stream_params * input_stream_params, + cubeb_stream_params * output_stream_params) +{ + XASSERT(input_stream_params || output_stream_params); + if (output_stream_params) { + if (output_stream_params->rate < 1000 || output_stream_params->rate > 192000 || + output_stream_params->channels < 1 || output_stream_params->channels > UINT8_MAX) { + return CUBEB_ERROR_INVALID_FORMAT; + } + } + if (input_stream_params) { + if (input_stream_params->rate < 1000 || input_stream_params->rate > 192000 || + input_stream_params->channels < 1 || input_stream_params->channels > 8) { + return CUBEB_ERROR_INVALID_FORMAT; + } + } + // Rate and sample format must be the same for input and output, if using a + // duplex stream + if (input_stream_params && output_stream_params) { + if (input_stream_params->rate != output_stream_params->rate || + input_stream_params->format != output_stream_params->format) { + return CUBEB_ERROR_INVALID_FORMAT; + } + } + + cubeb_stream_params * params = input_stream_params ? + input_stream_params : output_stream_params; + + switch (params->format) { + case CUBEB_SAMPLE_S16LE: + case CUBEB_SAMPLE_S16BE: + case CUBEB_SAMPLE_FLOAT32LE: + case CUBEB_SAMPLE_FLOAT32BE: + return CUBEB_OK; + } + + return CUBEB_ERROR_INVALID_FORMAT; +} + +static int +validate_latency(int latency) +{ + if (latency < 1 || latency > 96000) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + return CUBEB_OK; +} + +int +cubeb_init(cubeb ** context, char const * context_name, char const * backend_name) +{ + int (* init_oneshot)(cubeb **, char const *) = NULL; + + if (backend_name != NULL) { + if (!strcmp(backend_name, "pulse")) { +#if defined(USE_PULSE) + init_oneshot = pulse_init; +#endif + } else if (!strcmp(backend_name, "pulse-rust")) { +#if defined(USE_PULSE_RUST) + init_oneshot = pulse_rust_init; +#endif + } else if (!strcmp(backend_name, "jack")) { +#if defined(USE_JACK) + init_oneshot = jack_init; +#endif + } else if (!strcmp(backend_name, "alsa")) { +#if defined(USE_ALSA) + init_oneshot = alsa_init; +#endif + } else if (!strcmp(backend_name, "audiounit")) { +#if defined(USE_AUDIOUNIT) + init_oneshot = audiounit_init; +#endif + } else if (!strcmp(backend_name, "audiounit-rust")) { +#if defined(USE_AUDIOUNIT_RUST) + init_oneshot = audiounit_rust_init; +#endif + } else if (!strcmp(backend_name, "wasapi")) { +#if defined(USE_WASAPI) + init_oneshot = wasapi_init; +#endif + } else if (!strcmp(backend_name, "winmm")) { +#if defined(USE_WINMM) + init_oneshot = winmm_init; +#endif + } else if (!strcmp(backend_name, "sndio")) { +#if defined(USE_SNDIO) + init_oneshot = sndio_init; +#endif + } else if (!strcmp(backend_name, "sun")) { +#if defined(USE_SUN) + init_oneshot = sun_init; +#endif + } else if (!strcmp(backend_name, "opensl")) { +#if defined(USE_OPENSL) + init_oneshot = opensl_init; +#endif + } else if (!strcmp(backend_name, "audiotrack")) { +#if defined(USE_AUDIOTRACK) + init_oneshot = audiotrack_init; +#endif + } else if (!strcmp(backend_name, "kai")) { +#if defined(USE_KAI) + init_oneshot = kai_init; +#endif + } else { + /* Already set */ + } + } + + int (* default_init[])(cubeb **, char const *) = { + /* + * init_oneshot must be at the top to allow user + * to override all other choices + */ + init_oneshot, +#if defined(USE_PULSE_RUST) + pulse_rust_init, +#endif +#if defined(USE_PULSE) + pulse_init, +#endif +#if defined(USE_JACK) + jack_init, +#endif +#if defined(USE_SNDIO) + sndio_init, +#endif +#if defined(USE_ALSA) + alsa_init, +#endif +#if defined(USE_AUDIOUNIT) + audiounit_init, +#endif +#if defined(USE_AUDIOUNIT_RUST) + audiounit_rust_init, +#endif +#if defined(USE_WASAPI) + wasapi_init, +#endif +#if defined(USE_WINMM) + winmm_init, +#endif +#if defined(USE_SUN) + sun_init, +#endif +#if defined(USE_OPENSL) + opensl_init, +#endif +#if defined(USE_AUDIOTRACK) + audiotrack_init, +#endif +#if defined(USE_KAI) + kai_init, +#endif + }; + int i; + + if (!context) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + +#define OK(fn) assert((* context)->ops->fn) + for (i = 0; i < NELEMS(default_init); ++i) { + if (default_init[i] && default_init[i](context, context_name) == CUBEB_OK) { + /* Assert that the minimal API is implemented. */ + OK(get_backend_id); + OK(destroy); + OK(stream_init); + OK(stream_destroy); + OK(stream_start); + OK(stream_stop); + OK(stream_get_position); + return CUBEB_OK; + } + } + return CUBEB_ERROR; +} + +char const * +cubeb_get_backend_id(cubeb * context) +{ + if (!context) { + return NULL; + } + + return context->ops->get_backend_id(context); +} + +int +cubeb_get_max_channel_count(cubeb * context, uint32_t * max_channels) +{ + if (!context || !max_channels) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!context->ops->get_max_channel_count) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return context->ops->get_max_channel_count(context, max_channels); +} + +int +cubeb_get_min_latency(cubeb * context, cubeb_stream_params * params, uint32_t * latency_ms) +{ + if (!context || !params || !latency_ms) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!context->ops->get_min_latency) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return context->ops->get_min_latency(context, *params, latency_ms); +} + +int +cubeb_get_preferred_sample_rate(cubeb * context, uint32_t * rate) +{ + if (!context || !rate) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!context->ops->get_preferred_sample_rate) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return context->ops->get_preferred_sample_rate(context, rate); +} + +void +cubeb_destroy(cubeb * context) +{ + if (!context) { + return; + } + + context->ops->destroy(context); +} + +int +cubeb_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + int r; + + if (!context || !stream || !data_callback || !state_callback) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if ((r = validate_stream_params(input_stream_params, output_stream_params)) != CUBEB_OK || + (r = validate_latency(latency)) != CUBEB_OK) { + return r; + } + + r = context->ops->stream_init(context, stream, stream_name, + input_device, + input_stream_params, + output_device, + output_stream_params, + latency, + data_callback, + state_callback, + user_ptr); + + if (r == CUBEB_ERROR_INVALID_FORMAT) { + LOG("Invalid format, %p %p %d %d", + output_stream_params, input_stream_params, + output_stream_params && output_stream_params->format, + input_stream_params && input_stream_params->format); + } + + return r; +} + +void +cubeb_stream_destroy(cubeb_stream * stream) +{ + if (!stream) { + return; + } + + stream->context->ops->stream_destroy(stream); +} + +int +cubeb_stream_start(cubeb_stream * stream) +{ + if (!stream) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + return stream->context->ops->stream_start(stream); +} + +int +cubeb_stream_stop(cubeb_stream * stream) +{ + if (!stream) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + return stream->context->ops->stream_stop(stream); +} + +int +cubeb_stream_reset_default_device(cubeb_stream * stream) +{ + if (!stream) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!stream->context->ops->stream_reset_default_device) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return stream->context->ops->stream_reset_default_device(stream); +} + +int +cubeb_stream_get_position(cubeb_stream * stream, uint64_t * position) +{ + if (!stream || !position) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + return stream->context->ops->stream_get_position(stream, position); +} + +int +cubeb_stream_get_latency(cubeb_stream * stream, uint32_t * latency) +{ + if (!stream || !latency) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!stream->context->ops->stream_get_latency) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return stream->context->ops->stream_get_latency(stream, latency); +} + +int +cubeb_stream_set_volume(cubeb_stream * stream, float volume) +{ + if (!stream || volume > 1.0 || volume < 0.0) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!stream->context->ops->stream_set_volume) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return stream->context->ops->stream_set_volume(stream, volume); +} + +int cubeb_stream_get_current_device(cubeb_stream * stream, + cubeb_device ** const device) +{ + if (!stream || !device) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!stream->context->ops->stream_get_current_device) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return stream->context->ops->stream_get_current_device(stream, device); +} + +int cubeb_stream_device_destroy(cubeb_stream * stream, + cubeb_device * device) +{ + if (!stream || !device) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!stream->context->ops->stream_device_destroy) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return stream->context->ops->stream_device_destroy(stream, device); +} + +int cubeb_stream_register_device_changed_callback(cubeb_stream * stream, + cubeb_device_changed_callback device_changed_callback) +{ + if (!stream) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (!stream->context->ops->stream_register_device_changed_callback) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return stream->context->ops->stream_register_device_changed_callback(stream, device_changed_callback); +} + +void * cubeb_stream_user_ptr(cubeb_stream * stream) +{ + if (!stream) { + return NULL; + } + + return stream->user_ptr; +} + +static +void log_device(cubeb_device_info * device_info) +{ + char devfmts[128] = ""; + const char * devtype, * devstate, * devdeffmt; + + switch (device_info->type) { + case CUBEB_DEVICE_TYPE_INPUT: + devtype = "input"; + break; + case CUBEB_DEVICE_TYPE_OUTPUT: + devtype = "output"; + break; + case CUBEB_DEVICE_TYPE_UNKNOWN: + default: + devtype = "unknown?"; + break; + }; + + switch (device_info->state) { + case CUBEB_DEVICE_STATE_DISABLED: + devstate = "disabled"; + break; + case CUBEB_DEVICE_STATE_UNPLUGGED: + devstate = "unplugged"; + break; + case CUBEB_DEVICE_STATE_ENABLED: + devstate = "enabled"; + break; + default: + devstate = "unknown?"; + break; + }; + + switch (device_info->default_format) { + case CUBEB_DEVICE_FMT_S16LE: + devdeffmt = "S16LE"; + break; + case CUBEB_DEVICE_FMT_S16BE: + devdeffmt = "S16BE"; + break; + case CUBEB_DEVICE_FMT_F32LE: + devdeffmt = "F32LE"; + break; + case CUBEB_DEVICE_FMT_F32BE: + devdeffmt = "F32BE"; + break; + default: + devdeffmt = "unknown?"; + break; + }; + + if (device_info->format & CUBEB_DEVICE_FMT_S16LE) { + strcat(devfmts, " S16LE"); + } + if (device_info->format & CUBEB_DEVICE_FMT_S16BE) { + strcat(devfmts, " S16BE"); + } + if (device_info->format & CUBEB_DEVICE_FMT_F32LE) { + strcat(devfmts, " F32LE"); + } + if (device_info->format & CUBEB_DEVICE_FMT_F32BE) { + strcat(devfmts, " F32BE"); + } + + LOG("DeviceID: \"%s\"%s\n" + "\tName:\t\"%s\"\n" + "\tGroup:\t\"%s\"\n" + "\tVendor:\t\"%s\"\n" + "\tType:\t%s\n" + "\tState:\t%s\n" + "\tMaximum channels:\t%u\n" + "\tFormat:\t%s (0x%x) (default: %s)\n" + "\tRate:\t[%u, %u] (default: %u)\n" + "\tLatency: lo %u frames, hi %u frames", + device_info->device_id, device_info->preferred ? " (PREFERRED)" : "", + device_info->friendly_name, + device_info->group_id, + device_info->vendor_name, + devtype, + devstate, + device_info->max_channels, + (devfmts[0] == '\0') ? devfmts : devfmts + 1, (unsigned int)device_info->format, devdeffmt, + device_info->min_rate, device_info->max_rate, device_info->default_rate, + device_info->latency_lo, device_info->latency_hi); +} + +int cubeb_enumerate_devices(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection * collection) +{ + int rv; + if ((devtype & (CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_OUTPUT)) == 0) + return CUBEB_ERROR_INVALID_PARAMETER; + if (collection == NULL) + return CUBEB_ERROR_INVALID_PARAMETER; + if (!context->ops->enumerate_devices) + return CUBEB_ERROR_NOT_SUPPORTED; + + rv = context->ops->enumerate_devices(context, devtype, collection); + + if (g_cubeb_log_callback) { + for (size_t i = 0; i < collection->count; i++) { + log_device(&collection->device[i]); + } + } + + return rv; +} + +int cubeb_device_collection_destroy(cubeb * context, + cubeb_device_collection * collection) +{ + int r; + + if (context == NULL || collection == NULL) + return CUBEB_ERROR_INVALID_PARAMETER; + + if (!context->ops->device_collection_destroy) + return CUBEB_ERROR_NOT_SUPPORTED; + + if (!collection->device) + return CUBEB_OK; + + r = context->ops->device_collection_destroy(context, collection); + if (r == CUBEB_OK) { + collection->device = NULL; + collection->count = 0; + } + + return r; +} + +int cubeb_register_device_collection_changed(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection_changed_callback callback, + void * user_ptr) +{ + if (context == NULL || (devtype & (CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_OUTPUT)) == 0) + return CUBEB_ERROR_INVALID_PARAMETER; + + if (!context->ops->register_device_collection_changed) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + return context->ops->register_device_collection_changed(context, devtype, callback, user_ptr); +} + +int cubeb_set_log_callback(cubeb_log_level log_level, + cubeb_log_callback log_callback) +{ + if (log_level < CUBEB_LOG_DISABLED || log_level > CUBEB_LOG_VERBOSE) { + return CUBEB_ERROR_INVALID_FORMAT; + } + + if (!log_callback && log_level != CUBEB_LOG_DISABLED) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (g_cubeb_log_callback && log_callback) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + g_cubeb_log_callback = log_callback; + g_cubeb_log_level = log_level; + + // Logging a message here allows to initialize the asynchronous logger from a + // thread that is not the audio rendering thread, and especially to not + // initialize it the first time we find a verbose log, which is often in the + // audio rendering callback, that runs from the audio rendering thread, and + // that is high priority, and that we don't want to block. + if (log_level >= CUBEB_LOG_VERBOSE) { + ALOGV("Starting cubeb log"); + } + + return CUBEB_OK; +} + diff --git a/dep/cubeb/src/cubeb_alsa.c b/dep/cubeb/src/cubeb_alsa.c new file mode 100644 index 000000000..a564fbfc6 --- /dev/null +++ b/dep/cubeb/src/cubeb_alsa.c @@ -0,0 +1,1452 @@ +/* + * Copyright © 2011 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#undef NDEBUG +#define _DEFAULT_SOURCE +#define _BSD_SOURCE +#define _XOPEN_SOURCE 500 +#include +#include +#include +#include +#include +#include +#include +#include +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" + +#ifdef DISABLE_LIBASOUND_DLOPEN +#define WRAP(x) x +#else +#define WRAP(x) cubeb_##x +#define LIBASOUND_API_VISIT(X) \ + X(snd_config) \ + X(snd_config_add) \ + X(snd_config_copy) \ + X(snd_config_delete) \ + X(snd_config_get_id) \ + X(snd_config_get_string) \ + X(snd_config_imake_integer) \ + X(snd_config_search) \ + X(snd_config_search_definition) \ + X(snd_lib_error_set_handler) \ + X(snd_pcm_avail_update) \ + X(snd_pcm_close) \ + X(snd_pcm_delay) \ + X(snd_pcm_drain) \ + X(snd_pcm_frames_to_bytes) \ + X(snd_pcm_get_params) \ + X(snd_pcm_hw_params_any) \ + X(snd_pcm_hw_params_get_channels_max) \ + X(snd_pcm_hw_params_get_rate) \ + X(snd_pcm_hw_params_set_rate_near) \ + X(snd_pcm_hw_params_sizeof) \ + X(snd_pcm_nonblock) \ + X(snd_pcm_open) \ + X(snd_pcm_open_lconf) \ + X(snd_pcm_pause) \ + X(snd_pcm_poll_descriptors) \ + X(snd_pcm_poll_descriptors_count) \ + X(snd_pcm_poll_descriptors_revents) \ + X(snd_pcm_readi) \ + X(snd_pcm_recover) \ + X(snd_pcm_set_params) \ + X(snd_pcm_start) \ + X(snd_pcm_state) \ + X(snd_pcm_writei) \ + +#define MAKE_TYPEDEF(x) static typeof(x) * cubeb_##x; +LIBASOUND_API_VISIT(MAKE_TYPEDEF); +#undef MAKE_TYPEDEF +/* snd_pcm_hw_params_alloca is actually a macro */ +#define snd_pcm_hw_params_sizeof cubeb_snd_pcm_hw_params_sizeof +#endif + +#define CUBEB_STREAM_MAX 16 +#define CUBEB_WATCHDOG_MS 10000 + +#define CUBEB_ALSA_PCM_NAME "default" + +#define ALSA_PA_PLUGIN "ALSA <-> PulseAudio PCM I/O Plugin" + +/* ALSA is not thread-safe. snd_pcm_t instances are individually protected + by the owning cubeb_stream's mutex. snd_pcm_t creation and destruction + is not thread-safe until ALSA 1.0.24 (see alsa-lib.git commit 91c9c8f1), + so those calls must be wrapped in the following mutex. */ +static pthread_mutex_t cubeb_alsa_mutex = PTHREAD_MUTEX_INITIALIZER; +static int cubeb_alsa_error_handler_set = 0; + +static struct cubeb_ops const alsa_ops; + +struct cubeb { + struct cubeb_ops const * ops; + void * libasound; + + pthread_t thread; + + /* Mutex for streams array, must not be held while blocked in poll(2). */ + pthread_mutex_t mutex; + + /* Sparse array of streams managed by this context. */ + cubeb_stream * streams[CUBEB_STREAM_MAX]; + + /* fds and nfds are only updated by alsa_run when rebuild is set. */ + struct pollfd * fds; + nfds_t nfds; + int rebuild; + + int shutdown; + + /* Control pipe for forcing poll to wake and rebuild fds or recalculate the timeout. */ + int control_fd_read; + int control_fd_write; + + /* Track number of active streams. This is limited to CUBEB_STREAM_MAX + due to resource contraints. */ + unsigned int active_streams; + + /* Local configuration with handle_underrun workaround set for PulseAudio + ALSA plugin. Will be NULL if the PA ALSA plugin is not in use or the + workaround is not required. */ + snd_config_t * local_config; + int is_pa; +}; + +enum stream_state { + INACTIVE, + RUNNING, + DRAINING, + PROCESSING, + ERROR +}; + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr; + /**/ + pthread_mutex_t mutex; + snd_pcm_t * pcm; + cubeb_data_callback data_callback; + cubeb_state_callback state_callback; + snd_pcm_uframes_t stream_position; + snd_pcm_uframes_t last_position; + snd_pcm_uframes_t buffer_size; + cubeb_stream_params params; + + /* Every member after this comment is protected by the owning context's + mutex rather than the stream's mutex, or is only used on the context's + run thread. */ + pthread_cond_t cond; /* Signaled when the stream's state is changed. */ + + enum stream_state state; + + struct pollfd * saved_fds; /* A copy of the pollfds passed in at init time. */ + struct pollfd * fds; /* Pointer to this waitable's pollfds within struct cubeb's fds. */ + nfds_t nfds; + + struct timeval drain_timeout; + + /* XXX: Horrible hack -- if an active stream has been idle for + CUBEB_WATCHDOG_MS it will be disabled and the error callback will be + called. This works around a bug seen with older versions of ALSA and + PulseAudio where streams would stop requesting new data despite still + being logically active and playing. */ + struct timeval last_activity; + float volume; + + char * buffer; + snd_pcm_uframes_t bufframes; + snd_pcm_stream_t stream_type; + + struct cubeb_stream * other_stream; +}; + +static int +any_revents(struct pollfd * fds, nfds_t nfds) +{ + nfds_t i; + + for (i = 0; i < nfds; ++i) { + if (fds[i].revents) { + return 1; + } + } + + return 0; +} + +static int +cmp_timeval(struct timeval * a, struct timeval * b) +{ + if (a->tv_sec == b->tv_sec) { + if (a->tv_usec == b->tv_usec) { + return 0; + } + return a->tv_usec > b->tv_usec ? 1 : -1; + } + return a->tv_sec > b->tv_sec ? 1 : -1; +} + +static int +timeval_to_relative_ms(struct timeval * tv) +{ + struct timeval now; + struct timeval dt; + long long t; + int r; + + gettimeofday(&now, NULL); + r = cmp_timeval(tv, &now); + if (r >= 0) { + timersub(tv, &now, &dt); + } else { + timersub(&now, tv, &dt); + } + t = dt.tv_sec; + t *= 1000; + t += (dt.tv_usec + 500) / 1000; + + if (t > INT_MAX) { + t = INT_MAX; + } else if (t < INT_MIN) { + t = INT_MIN; + } + + return r >= 0 ? t : -t; +} + +static int +ms_until(struct timeval * tv) +{ + return timeval_to_relative_ms(tv); +} + +static int +ms_since(struct timeval * tv) +{ + return -timeval_to_relative_ms(tv); +} + +static void +rebuild(cubeb * ctx) +{ + nfds_t nfds; + int i; + nfds_t j; + cubeb_stream * stm; + + assert(ctx->rebuild); + + /* Always count context's control pipe fd. */ + nfds = 1; + for (i = 0; i < CUBEB_STREAM_MAX; ++i) { + stm = ctx->streams[i]; + if (stm) { + stm->fds = NULL; + if (stm->state == RUNNING) { + nfds += stm->nfds; + } + } + } + + free(ctx->fds); + ctx->fds = calloc(nfds, sizeof(struct pollfd)); + assert(ctx->fds); + ctx->nfds = nfds; + + /* Include context's control pipe fd. */ + ctx->fds[0].fd = ctx->control_fd_read; + ctx->fds[0].events = POLLIN | POLLERR; + + for (i = 0, j = 1; i < CUBEB_STREAM_MAX; ++i) { + stm = ctx->streams[i]; + if (stm && stm->state == RUNNING) { + memcpy(&ctx->fds[j], stm->saved_fds, stm->nfds * sizeof(struct pollfd)); + stm->fds = &ctx->fds[j]; + j += stm->nfds; + } + } + + ctx->rebuild = 0; +} + +static void +poll_wake(cubeb * ctx) +{ + if (write(ctx->control_fd_write, "x", 1) < 0) { + /* ignore write error */ + } +} + +static void +set_timeout(struct timeval * timeout, unsigned int ms) +{ + gettimeofday(timeout, NULL); + timeout->tv_sec += ms / 1000; + timeout->tv_usec += (ms % 1000) * 1000; +} + +static void +stream_buffer_decrement(cubeb_stream * stm, long count) +{ + char * bufremains = stm->buffer + WRAP(snd_pcm_frames_to_bytes)(stm->pcm, count); + memmove(stm->buffer, bufremains, WRAP(snd_pcm_frames_to_bytes)(stm->pcm, stm->bufframes - count)); + stm->bufframes -= count; +} + +static void +alsa_set_stream_state(cubeb_stream * stm, enum stream_state state) +{ + cubeb * ctx; + int r; + + ctx = stm->context; + stm->state = state; + r = pthread_cond_broadcast(&stm->cond); + assert(r == 0); + ctx->rebuild = 1; + poll_wake(ctx); +} + +static enum stream_state +alsa_process_stream(cubeb_stream * stm) +{ + unsigned short revents; + snd_pcm_sframes_t avail; + int draining; + + draining = 0; + + pthread_mutex_lock(&stm->mutex); + + /* Call _poll_descriptors_revents() even if we don't use it + to let underlying plugins clear null events. Otherwise poll() + may wake up again and again, producing unnecessary CPU usage. */ + WRAP(snd_pcm_poll_descriptors_revents)(stm->pcm, stm->fds, stm->nfds, &revents); + + avail = WRAP(snd_pcm_avail_update)(stm->pcm); + + /* Got null event? Bail and wait for another wakeup. */ + if (avail == 0) { + pthread_mutex_unlock(&stm->mutex); + return RUNNING; + } + + /* This could happen if we were suspended with SIGSTOP/Ctrl+Z for a long time. */ + if ((unsigned int) avail > stm->buffer_size) { + avail = stm->buffer_size; + } + + /* Capture: Read available frames */ + if (stm->stream_type == SND_PCM_STREAM_CAPTURE && avail > 0) { + snd_pcm_sframes_t got; + + if (avail + stm->bufframes > stm->buffer_size) { + /* Buffer overflow. Skip and overwrite with new data. */ + stm->bufframes = 0; + // TODO: should it be marked as DRAINING? + } + + got = WRAP(snd_pcm_readi)(stm->pcm, stm->buffer+stm->bufframes, avail); + + if (got < 0) { + avail = got; // the error handler below will recover us + } else { + stm->bufframes += got; + stm->stream_position += got; + + gettimeofday(&stm->last_activity, NULL); + } + } + + /* Capture: Pass read frames to callback function */ + if (stm->stream_type == SND_PCM_STREAM_CAPTURE && stm->bufframes > 0 && + (!stm->other_stream || stm->other_stream->bufframes < stm->other_stream->buffer_size)) { + snd_pcm_sframes_t wrote = stm->bufframes; + struct cubeb_stream * mainstm = stm->other_stream ? stm->other_stream : stm; + void * other_buffer = stm->other_stream ? stm->other_stream->buffer + stm->other_stream->bufframes : NULL; + + /* Correct write size to the other stream available space */ + if (stm->other_stream && wrote > (snd_pcm_sframes_t) (stm->other_stream->buffer_size - stm->other_stream->bufframes)) { + wrote = stm->other_stream->buffer_size - stm->other_stream->bufframes; + } + + pthread_mutex_unlock(&stm->mutex); + wrote = stm->data_callback(mainstm, stm->user_ptr, stm->buffer, other_buffer, wrote); + pthread_mutex_lock(&stm->mutex); + + if (wrote < 0) { + avail = wrote; // the error handler below will recover us + } else { + stream_buffer_decrement(stm, wrote); + + if (stm->other_stream) { + stm->other_stream->bufframes += wrote; + } + } + } + + /* Playback: Don't have enough data? Let's ask for more. */ + if (stm->stream_type == SND_PCM_STREAM_PLAYBACK && avail > (snd_pcm_sframes_t) stm->bufframes && + (!stm->other_stream || stm->other_stream->bufframes > 0)) { + long got = avail - stm->bufframes; + void * other_buffer = stm->other_stream ? stm->other_stream->buffer : NULL; + char * buftail = stm->buffer + WRAP(snd_pcm_frames_to_bytes)(stm->pcm, stm->bufframes); + + /* Correct read size to the other stream available frames */ + if (stm->other_stream && got > (snd_pcm_sframes_t) stm->other_stream->bufframes) { + got = stm->other_stream->bufframes; + } + + pthread_mutex_unlock(&stm->mutex); + got = stm->data_callback(stm, stm->user_ptr, other_buffer, buftail, got); + pthread_mutex_lock(&stm->mutex); + + if (got < 0) { + avail = got; // the error handler below will recover us + } else { + stm->bufframes += got; + + if (stm->other_stream) { + stream_buffer_decrement(stm->other_stream, got); + } + } + } + + /* Playback: Still don't have enough data? Add some silence. */ + if (stm->stream_type == SND_PCM_STREAM_PLAYBACK && avail > (snd_pcm_sframes_t) stm->bufframes) { + long drain_frames = avail - stm->bufframes; + double drain_time = (double) drain_frames / stm->params.rate; + + char * buftail = stm->buffer + WRAP(snd_pcm_frames_to_bytes)(stm->pcm, stm->bufframes); + memset(buftail, 0, WRAP(snd_pcm_frames_to_bytes)(stm->pcm, drain_frames)); + stm->bufframes = avail; + + /* Mark as draining, unless we're waiting for capture */ + if (!stm->other_stream || stm->other_stream->bufframes > 0) { + set_timeout(&stm->drain_timeout, drain_time * 1000); + + draining = 1; + } + } + + /* Playback: Have enough data and no errors. Let's write it out. */ + if (stm->stream_type == SND_PCM_STREAM_PLAYBACK && avail > 0) { + snd_pcm_sframes_t wrote; + + if (stm->params.format == CUBEB_SAMPLE_FLOAT32NE) { + float * b = (float *) stm->buffer; + for (uint32_t i = 0; i < avail * stm->params.channels; i++) { + b[i] *= stm->volume; + } + } else { + short * b = (short *) stm->buffer; + for (uint32_t i = 0; i < avail * stm->params.channels; i++) { + b[i] *= stm->volume; + } + } + + wrote = WRAP(snd_pcm_writei)(stm->pcm, stm->buffer, avail); + if (wrote < 0) { + avail = wrote; // the error handler below will recover us + } else { + stream_buffer_decrement(stm, wrote); + + stm->stream_position += wrote; + gettimeofday(&stm->last_activity, NULL); + } + } + + /* Got some error? Let's try to recover the stream. */ + if (avail < 0) { + avail = WRAP(snd_pcm_recover)(stm->pcm, avail, 0); + + /* Capture pcm must be started after initial setup/recover */ + if (avail >= 0 && + stm->stream_type == SND_PCM_STREAM_CAPTURE && + WRAP(snd_pcm_state)(stm->pcm) == SND_PCM_STATE_PREPARED) { + avail = WRAP(snd_pcm_start)(stm->pcm); + } + } + + /* Failed to recover, this stream must be broken. */ + if (avail < 0) { + pthread_mutex_unlock(&stm->mutex); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + return ERROR; + } + + pthread_mutex_unlock(&stm->mutex); + return draining ? DRAINING : RUNNING; +} + +static int +alsa_run(cubeb * ctx) +{ + int r; + int timeout; + int i; + char dummy; + cubeb_stream * stm; + enum stream_state state; + + pthread_mutex_lock(&ctx->mutex); + + if (ctx->rebuild) { + rebuild(ctx); + } + + /* Wake up at least once per second for the watchdog. */ + timeout = 1000; + for (i = 0; i < CUBEB_STREAM_MAX; ++i) { + stm = ctx->streams[i]; + if (stm && stm->state == DRAINING) { + r = ms_until(&stm->drain_timeout); + if (r >= 0 && timeout > r) { + timeout = r; + } + } + } + + pthread_mutex_unlock(&ctx->mutex); + r = poll(ctx->fds, ctx->nfds, timeout); + pthread_mutex_lock(&ctx->mutex); + + if (r > 0) { + if (ctx->fds[0].revents & POLLIN) { + if (read(ctx->control_fd_read, &dummy, 1) < 0) { + /* ignore read error */ + } + + if (ctx->shutdown) { + pthread_mutex_unlock(&ctx->mutex); + return -1; + } + } + + for (i = 0; i < CUBEB_STREAM_MAX; ++i) { + stm = ctx->streams[i]; + /* We can't use snd_pcm_poll_descriptors_revents here because of + https://github.com/kinetiknz/cubeb/issues/135. */ + if (stm && stm->state == RUNNING && stm->fds && any_revents(stm->fds, stm->nfds)) { + alsa_set_stream_state(stm, PROCESSING); + pthread_mutex_unlock(&ctx->mutex); + state = alsa_process_stream(stm); + pthread_mutex_lock(&ctx->mutex); + alsa_set_stream_state(stm, state); + } + } + } else if (r == 0) { + for (i = 0; i < CUBEB_STREAM_MAX; ++i) { + stm = ctx->streams[i]; + if (stm) { + if (stm->state == DRAINING && ms_since(&stm->drain_timeout) >= 0) { + alsa_set_stream_state(stm, INACTIVE); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + } else if (stm->state == RUNNING && ms_since(&stm->last_activity) > CUBEB_WATCHDOG_MS) { + alsa_set_stream_state(stm, ERROR); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + } + } + } + } + + pthread_mutex_unlock(&ctx->mutex); + + return 0; +} + +static void * +alsa_run_thread(void * context) +{ + cubeb * ctx = context; + int r; + + do { + r = alsa_run(ctx); + } while (r >= 0); + + return NULL; +} + +static snd_config_t * +get_slave_pcm_node(snd_config_t * lconf, snd_config_t * root_pcm) +{ + int r; + snd_config_t * slave_pcm; + snd_config_t * slave_def; + snd_config_t * pcm; + char const * string; + char node_name[64]; + + slave_def = NULL; + + r = WRAP(snd_config_search)(root_pcm, "slave", &slave_pcm); + if (r < 0) { + return NULL; + } + + r = WRAP(snd_config_get_string)(slave_pcm, &string); + if (r >= 0) { + r = WRAP(snd_config_search_definition)(lconf, "pcm_slave", string, &slave_def); + if (r < 0) { + return NULL; + } + } + + do { + r = WRAP(snd_config_search)(slave_def ? slave_def : slave_pcm, "pcm", &pcm); + if (r < 0) { + break; + } + + r = WRAP(snd_config_get_string)(slave_def ? slave_def : slave_pcm, &string); + if (r < 0) { + break; + } + + r = snprintf(node_name, sizeof(node_name), "pcm.%s", string); + if (r < 0 || r > (int) sizeof(node_name)) { + break; + } + r = WRAP(snd_config_search)(lconf, node_name, &pcm); + if (r < 0) { + break; + } + + return pcm; + } while (0); + + if (slave_def) { + WRAP(snd_config_delete)(slave_def); + } + + return NULL; +} + +/* Work around PulseAudio ALSA plugin bug where the PA server forces a + higher than requested latency, but the plugin does not update its (and + ALSA's) internal state to reflect that, leading to an immediate underrun + situation. Inspired by WINE's make_handle_underrun_config. + Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-July/05 */ +static snd_config_t * +init_local_config_with_workaround(char const * pcm_name) +{ + int r; + snd_config_t * lconf; + snd_config_t * pcm_node; + snd_config_t * node; + char const * string; + char node_name[64]; + + lconf = NULL; + + if (*WRAP(snd_config) == NULL) { + return NULL; + } + + r = WRAP(snd_config_copy)(&lconf, *WRAP(snd_config)); + if (r < 0) { + return NULL; + } + + do { + r = WRAP(snd_config_search_definition)(lconf, "pcm", pcm_name, &pcm_node); + if (r < 0) { + break; + } + + r = WRAP(snd_config_get_id)(pcm_node, &string); + if (r < 0) { + break; + } + + r = snprintf(node_name, sizeof(node_name), "pcm.%s", string); + if (r < 0 || r > (int) sizeof(node_name)) { + break; + } + r = WRAP(snd_config_search)(lconf, node_name, &pcm_node); + if (r < 0) { + break; + } + + /* If this PCM has a slave, walk the slave configurations until we reach the bottom. */ + while ((node = get_slave_pcm_node(lconf, pcm_node)) != NULL) { + pcm_node = node; + } + + /* Fetch the PCM node's type, and bail out if it's not the PulseAudio plugin. */ + r = WRAP(snd_config_search)(pcm_node, "type", &node); + if (r < 0) { + break; + } + + r = WRAP(snd_config_get_string)(node, &string); + if (r < 0) { + break; + } + + if (strcmp(string, "pulse") != 0) { + break; + } + + /* Don't clobber an explicit existing handle_underrun value, set it only + if it doesn't already exist. */ + r = WRAP(snd_config_search)(pcm_node, "handle_underrun", &node); + if (r != -ENOENT) { + break; + } + + /* Disable pcm_pulse's asynchronous underrun handling. */ + r = WRAP(snd_config_imake_integer)(&node, "handle_underrun", 0); + if (r < 0) { + break; + } + + r = WRAP(snd_config_add)(pcm_node, node); + if (r < 0) { + break; + } + + return lconf; + } while (0); + + WRAP(snd_config_delete)(lconf); + + return NULL; +} + +static int +alsa_locked_pcm_open(snd_pcm_t ** pcm, char const * pcm_name, snd_pcm_stream_t stream, snd_config_t * local_config) +{ + int r; + + pthread_mutex_lock(&cubeb_alsa_mutex); + if (local_config) { + r = WRAP(snd_pcm_open_lconf)(pcm, pcm_name, stream, SND_PCM_NONBLOCK, local_config); + } else { + r = WRAP(snd_pcm_open)(pcm, pcm_name, stream, SND_PCM_NONBLOCK); + } + pthread_mutex_unlock(&cubeb_alsa_mutex); + + return r; +} + +static int +alsa_locked_pcm_close(snd_pcm_t * pcm) +{ + int r; + + pthread_mutex_lock(&cubeb_alsa_mutex); + r = WRAP(snd_pcm_close)(pcm); + pthread_mutex_unlock(&cubeb_alsa_mutex); + + return r; +} + +static int +alsa_register_stream(cubeb * ctx, cubeb_stream * stm) +{ + int i; + + pthread_mutex_lock(&ctx->mutex); + for (i = 0; i < CUBEB_STREAM_MAX; ++i) { + if (!ctx->streams[i]) { + ctx->streams[i] = stm; + break; + } + } + pthread_mutex_unlock(&ctx->mutex); + + return i == CUBEB_STREAM_MAX; +} + +static void +alsa_unregister_stream(cubeb_stream * stm) +{ + cubeb * ctx; + int i; + + ctx = stm->context; + + pthread_mutex_lock(&ctx->mutex); + for (i = 0; i < CUBEB_STREAM_MAX; ++i) { + if (ctx->streams[i] == stm) { + ctx->streams[i] = NULL; + break; + } + } + pthread_mutex_unlock(&ctx->mutex); +} + +static void +silent_error_handler(char const * file, int line, char const * function, + int err, char const * fmt, ...) +{ + (void)file; + (void)line; + (void)function; + (void)err; + (void)fmt; +} + +/*static*/ int +alsa_init(cubeb ** context, char const * context_name) +{ + (void)context_name; + void * libasound = NULL; + cubeb * ctx; + int r; + int i; + int fd[2]; + pthread_attr_t attr; + snd_pcm_t * dummy; + + assert(context); + *context = NULL; + +#ifndef DISABLE_LIBASOUND_DLOPEN + libasound = dlopen("libasound.so.2", RTLD_LAZY); + if (!libasound) { + libasound = dlopen("libasound.so", RTLD_LAZY); + if (!libasound) { + return CUBEB_ERROR; + } + } + +#define LOAD(x) { \ + cubeb_##x = dlsym(libasound, #x); \ + if (!cubeb_##x) { \ + dlclose(libasound); \ + return CUBEB_ERROR; \ + } \ + } + + LIBASOUND_API_VISIT(LOAD); +#undef LOAD +#endif + + pthread_mutex_lock(&cubeb_alsa_mutex); + if (!cubeb_alsa_error_handler_set) { + WRAP(snd_lib_error_set_handler)(silent_error_handler); + cubeb_alsa_error_handler_set = 1; + } + pthread_mutex_unlock(&cubeb_alsa_mutex); + + ctx = calloc(1, sizeof(*ctx)); + assert(ctx); + + ctx->ops = &alsa_ops; + ctx->libasound = libasound; + + r = pthread_mutex_init(&ctx->mutex, NULL); + assert(r == 0); + + r = pipe(fd); + assert(r == 0); + + for (i = 0; i < 2; ++i) { + fcntl(fd[i], F_SETFD, fcntl(fd[i], F_GETFD) | FD_CLOEXEC); + fcntl(fd[i], F_SETFL, fcntl(fd[i], F_GETFL) | O_NONBLOCK); + } + + ctx->control_fd_read = fd[0]; + ctx->control_fd_write = fd[1]; + + /* Force an early rebuild when alsa_run is first called to ensure fds and + nfds have been initialized. */ + ctx->rebuild = 1; + + r = pthread_attr_init(&attr); + assert(r == 0); + + r = pthread_attr_setstacksize(&attr, 256 * 1024); + assert(r == 0); + + r = pthread_create(&ctx->thread, &attr, alsa_run_thread, ctx); + assert(r == 0); + + r = pthread_attr_destroy(&attr); + assert(r == 0); + + /* Open a dummy PCM to force the configuration space to be evaluated so that + init_local_config_with_workaround can find and modify the default node. */ + r = alsa_locked_pcm_open(&dummy, CUBEB_ALSA_PCM_NAME, SND_PCM_STREAM_PLAYBACK, NULL); + if (r >= 0) { + alsa_locked_pcm_close(dummy); + } + ctx->is_pa = 0; + pthread_mutex_lock(&cubeb_alsa_mutex); + ctx->local_config = init_local_config_with_workaround(CUBEB_ALSA_PCM_NAME); + pthread_mutex_unlock(&cubeb_alsa_mutex); + if (ctx->local_config) { + ctx->is_pa = 1; + r = alsa_locked_pcm_open(&dummy, CUBEB_ALSA_PCM_NAME, SND_PCM_STREAM_PLAYBACK, ctx->local_config); + /* If we got a local_config, we found a PA PCM. If opening a PCM with that + config fails with EINVAL, the PA PCM is too old for this workaround. */ + if (r == -EINVAL) { + pthread_mutex_lock(&cubeb_alsa_mutex); + WRAP(snd_config_delete)(ctx->local_config); + pthread_mutex_unlock(&cubeb_alsa_mutex); + ctx->local_config = NULL; + } else if (r >= 0) { + alsa_locked_pcm_close(dummy); + } + } + + *context = ctx; + + return CUBEB_OK; +} + +static char const * +alsa_get_backend_id(cubeb * ctx) +{ + (void)ctx; + return "alsa"; +} + +static void +alsa_destroy(cubeb * ctx) +{ + int r; + + assert(ctx); + + pthread_mutex_lock(&ctx->mutex); + ctx->shutdown = 1; + poll_wake(ctx); + pthread_mutex_unlock(&ctx->mutex); + + r = pthread_join(ctx->thread, NULL); + assert(r == 0); + + close(ctx->control_fd_read); + close(ctx->control_fd_write); + pthread_mutex_destroy(&ctx->mutex); + free(ctx->fds); + + if (ctx->local_config) { + pthread_mutex_lock(&cubeb_alsa_mutex); + WRAP(snd_config_delete)(ctx->local_config); + pthread_mutex_unlock(&cubeb_alsa_mutex); + } + + if (ctx->libasound) { + dlclose(ctx->libasound); + } + + free(ctx); +} + +static void alsa_stream_destroy(cubeb_stream * stm); + +static int +alsa_stream_init_single(cubeb * ctx, cubeb_stream ** stream, char const * stream_name, + snd_pcm_stream_t stream_type, + cubeb_devid deviceid, + cubeb_stream_params * stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + (void)stream_name; + cubeb_stream * stm; + int r; + snd_pcm_format_t format; + snd_pcm_uframes_t period_size; + int latency_us = 0; + char const * pcm_name = deviceid ? (char const *) deviceid : CUBEB_ALSA_PCM_NAME; + + assert(ctx && stream); + + *stream = NULL; + + if (stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + switch (stream_params->format) { + case CUBEB_SAMPLE_S16LE: + format = SND_PCM_FORMAT_S16_LE; + break; + case CUBEB_SAMPLE_S16BE: + format = SND_PCM_FORMAT_S16_BE; + break; + case CUBEB_SAMPLE_FLOAT32LE: + format = SND_PCM_FORMAT_FLOAT_LE; + break; + case CUBEB_SAMPLE_FLOAT32BE: + format = SND_PCM_FORMAT_FLOAT_BE; + break; + default: + return CUBEB_ERROR_INVALID_FORMAT; + } + + pthread_mutex_lock(&ctx->mutex); + if (ctx->active_streams >= CUBEB_STREAM_MAX) { + pthread_mutex_unlock(&ctx->mutex); + return CUBEB_ERROR; + } + ctx->active_streams += 1; + pthread_mutex_unlock(&ctx->mutex); + + stm = calloc(1, sizeof(*stm)); + assert(stm); + + stm->context = ctx; + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + stm->params = *stream_params; + stm->state = INACTIVE; + stm->volume = 1.0; + stm->buffer = NULL; + stm->bufframes = 0; + stm->stream_type = stream_type; + stm->other_stream = NULL; + + r = pthread_mutex_init(&stm->mutex, NULL); + assert(r == 0); + + r = pthread_cond_init(&stm->cond, NULL); + assert(r == 0); + + r = alsa_locked_pcm_open(&stm->pcm, pcm_name, stm->stream_type, ctx->local_config); + if (r < 0) { + alsa_stream_destroy(stm); + return CUBEB_ERROR; + } + + r = WRAP(snd_pcm_nonblock)(stm->pcm, 1); + assert(r == 0); + + latency_us = latency_frames * 1e6 / stm->params.rate; + + /* Ugly hack: the PA ALSA plugin allows buffer configurations that can't + possibly work. See https://bugzilla.mozilla.org/show_bug.cgi?id=761274. + Only resort to this hack if the handle_underrun workaround failed. */ + if (!ctx->local_config && ctx->is_pa) { + const int min_latency = 5e5; + latency_us = latency_us < min_latency ? min_latency: latency_us; + } + + r = WRAP(snd_pcm_set_params)(stm->pcm, format, SND_PCM_ACCESS_RW_INTERLEAVED, + stm->params.channels, stm->params.rate, 1, + latency_us); + if (r < 0) { + alsa_stream_destroy(stm); + return CUBEB_ERROR_INVALID_FORMAT; + } + + r = WRAP(snd_pcm_get_params)(stm->pcm, &stm->buffer_size, &period_size); + assert(r == 0); + + /* Double internal buffer size to have enough space when waiting for the other side of duplex connection */ + stm->buffer_size *= 2; + stm->buffer = calloc(1, WRAP(snd_pcm_frames_to_bytes)(stm->pcm, stm->buffer_size)); + assert(stm->buffer); + + stm->nfds = WRAP(snd_pcm_poll_descriptors_count)(stm->pcm); + assert(stm->nfds > 0); + + stm->saved_fds = calloc(stm->nfds, sizeof(struct pollfd)); + assert(stm->saved_fds); + r = WRAP(snd_pcm_poll_descriptors)(stm->pcm, stm->saved_fds, stm->nfds); + assert((nfds_t) r == stm->nfds); + + if (alsa_register_stream(ctx, stm) != 0) { + alsa_stream_destroy(stm); + return CUBEB_ERROR; + } + + *stream = stm; + + return CUBEB_OK; +} + +static int +alsa_stream_init(cubeb * ctx, cubeb_stream ** stream, char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, cubeb_state_callback state_callback, + void * user_ptr) +{ + int result = CUBEB_OK; + cubeb_stream * instm = NULL, * outstm = NULL; + + if (result == CUBEB_OK && input_stream_params) { + result = alsa_stream_init_single(ctx, &instm, stream_name, SND_PCM_STREAM_CAPTURE, + input_device, input_stream_params, latency_frames, + data_callback, state_callback, user_ptr); + } + + if (result == CUBEB_OK && output_stream_params) { + result = alsa_stream_init_single(ctx, &outstm, stream_name, SND_PCM_STREAM_PLAYBACK, + output_device, output_stream_params, latency_frames, + data_callback, state_callback, user_ptr); + } + + if (result == CUBEB_OK && input_stream_params && output_stream_params) { + instm->other_stream = outstm; + outstm->other_stream = instm; + } + + if (result != CUBEB_OK && instm) { + alsa_stream_destroy(instm); + } + + *stream = outstm ? outstm : instm; + + return result; +} + +static void +alsa_stream_destroy(cubeb_stream * stm) +{ + int r; + cubeb * ctx; + + assert(stm && (stm->state == INACTIVE || + stm->state == ERROR || + stm->state == DRAINING)); + + ctx = stm->context; + + if (stm->other_stream) { + stm->other_stream->other_stream = NULL; // to stop infinite recursion + alsa_stream_destroy(stm->other_stream); + } + + pthread_mutex_lock(&stm->mutex); + if (stm->pcm) { + if (stm->state == DRAINING) { + WRAP(snd_pcm_drain)(stm->pcm); + } + alsa_locked_pcm_close(stm->pcm); + stm->pcm = NULL; + } + free(stm->saved_fds); + pthread_mutex_unlock(&stm->mutex); + pthread_mutex_destroy(&stm->mutex); + + r = pthread_cond_destroy(&stm->cond); + assert(r == 0); + + alsa_unregister_stream(stm); + + pthread_mutex_lock(&ctx->mutex); + assert(ctx->active_streams >= 1); + ctx->active_streams -= 1; + pthread_mutex_unlock(&ctx->mutex); + + free(stm->buffer); + + free(stm); +} + +static int +alsa_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + int r; + cubeb_stream * stm; + snd_pcm_hw_params_t* hw_params; + cubeb_stream_params params; + params.rate = 44100; + params.format = CUBEB_SAMPLE_FLOAT32NE; + params.channels = 2; + + snd_pcm_hw_params_alloca(&hw_params); + + assert(ctx); + + r = alsa_stream_init(ctx, &stm, "", NULL, NULL, NULL, ¶ms, 100, NULL, NULL, NULL); + if (r != CUBEB_OK) { + return CUBEB_ERROR; + } + + assert(stm); + + r = WRAP(snd_pcm_hw_params_any)(stm->pcm, hw_params); + if (r < 0) { + return CUBEB_ERROR; + } + + r = WRAP(snd_pcm_hw_params_get_channels_max)(hw_params, max_channels); + if (r < 0) { + return CUBEB_ERROR; + } + + alsa_stream_destroy(stm); + + return CUBEB_OK; +} + +static int +alsa_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) { + (void)ctx; + int r, dir; + snd_pcm_t * pcm; + snd_pcm_hw_params_t * hw_params; + + snd_pcm_hw_params_alloca(&hw_params); + + /* get a pcm, disabling resampling, so we get a rate the + * hardware/dmix/pulse/etc. supports. */ + r = WRAP(snd_pcm_open)(&pcm, CUBEB_ALSA_PCM_NAME, SND_PCM_STREAM_PLAYBACK, SND_PCM_NO_AUTO_RESAMPLE); + if (r < 0) { + return CUBEB_ERROR; + } + + r = WRAP(snd_pcm_hw_params_any)(pcm, hw_params); + if (r < 0) { + WRAP(snd_pcm_close)(pcm); + return CUBEB_ERROR; + } + + r = WRAP(snd_pcm_hw_params_get_rate)(hw_params, rate, &dir); + if (r >= 0) { + /* There is a default rate: use it. */ + WRAP(snd_pcm_close)(pcm); + return CUBEB_OK; + } + + /* Use a common rate, alsa may adjust it based on hw/etc. capabilities. */ + *rate = 44100; + + r = WRAP(snd_pcm_hw_params_set_rate_near)(pcm, hw_params, rate, NULL); + if (r < 0) { + WRAP(snd_pcm_close)(pcm); + return CUBEB_ERROR; + } + + WRAP(snd_pcm_close)(pcm); + + return CUBEB_OK; +} + +static int +alsa_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames) +{ + (void)ctx; + /* 40ms is found to be an acceptable minimum, even on a super low-end + * machine. */ + *latency_frames = 40 * params.rate / 1000; + + return CUBEB_OK; +} + +static int +alsa_stream_start(cubeb_stream * stm) +{ + cubeb * ctx; + + assert(stm); + ctx = stm->context; + + if (stm->stream_type == SND_PCM_STREAM_PLAYBACK && stm->other_stream) { + int r = alsa_stream_start(stm->other_stream); + if (r != CUBEB_OK) + return r; + } + + pthread_mutex_lock(&stm->mutex); + /* Capture pcm must be started after initial setup/recover */ + if (stm->stream_type == SND_PCM_STREAM_CAPTURE && + WRAP(snd_pcm_state)(stm->pcm) == SND_PCM_STATE_PREPARED) { + WRAP(snd_pcm_start)(stm->pcm); + } + WRAP(snd_pcm_pause)(stm->pcm, 0); + gettimeofday(&stm->last_activity, NULL); + pthread_mutex_unlock(&stm->mutex); + + pthread_mutex_lock(&ctx->mutex); + if (stm->state != INACTIVE) { + pthread_mutex_unlock(&ctx->mutex); + return CUBEB_ERROR; + } + alsa_set_stream_state(stm, RUNNING); + pthread_mutex_unlock(&ctx->mutex); + + return CUBEB_OK; +} + +static int +alsa_stream_stop(cubeb_stream * stm) +{ + cubeb * ctx; + int r; + + assert(stm); + ctx = stm->context; + + if (stm->stream_type == SND_PCM_STREAM_PLAYBACK && stm->other_stream) { + int r = alsa_stream_stop(stm->other_stream); + if (r != CUBEB_OK) + return r; + } + + pthread_mutex_lock(&ctx->mutex); + while (stm->state == PROCESSING) { + r = pthread_cond_wait(&stm->cond, &ctx->mutex); + assert(r == 0); + } + + alsa_set_stream_state(stm, INACTIVE); + pthread_mutex_unlock(&ctx->mutex); + + pthread_mutex_lock(&stm->mutex); + WRAP(snd_pcm_pause)(stm->pcm, 1); + pthread_mutex_unlock(&stm->mutex); + + return CUBEB_OK; +} + +static int +alsa_stream_get_position(cubeb_stream * stm, uint64_t * position) +{ + snd_pcm_sframes_t delay; + + assert(stm && position); + + pthread_mutex_lock(&stm->mutex); + + delay = -1; + if (WRAP(snd_pcm_state)(stm->pcm) != SND_PCM_STATE_RUNNING || + WRAP(snd_pcm_delay)(stm->pcm, &delay) != 0) { + *position = stm->last_position; + pthread_mutex_unlock(&stm->mutex); + return CUBEB_OK; + } + + assert(delay >= 0); + + *position = 0; + if (stm->stream_position >= (snd_pcm_uframes_t) delay) { + *position = stm->stream_position - delay; + } + + stm->last_position = *position; + + pthread_mutex_unlock(&stm->mutex); + return CUBEB_OK; +} + +static int +alsa_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ + snd_pcm_sframes_t delay; + /* This function returns the delay in frames until a frame written using + snd_pcm_writei is sent to the DAC. The DAC delay should be < 1ms anyways. */ + if (WRAP(snd_pcm_delay)(stm->pcm, &delay)) { + return CUBEB_ERROR; + } + + *latency = delay; + + return CUBEB_OK; +} + +static int +alsa_stream_set_volume(cubeb_stream * stm, float volume) +{ + /* setting the volume using an API call does not seem very stable/supported */ + pthread_mutex_lock(&stm->mutex); + stm->volume = volume; + pthread_mutex_unlock(&stm->mutex); + + return CUBEB_OK; +} + +static int +alsa_enumerate_devices(cubeb * context, cubeb_device_type type, + cubeb_device_collection * collection) +{ + cubeb_device_info* device = NULL; + + if (!context) + return CUBEB_ERROR; + + uint32_t rate, max_channels; + int r; + + r = alsa_get_preferred_sample_rate(context, &rate); + if (r != CUBEB_OK) { + return CUBEB_ERROR; + } + + r = alsa_get_max_channel_count(context, &max_channels); + if (r != CUBEB_OK) { + return CUBEB_ERROR; + } + + char const * a_name = "default"; + device = (cubeb_device_info *) calloc(1, sizeof(cubeb_device_info)); + assert(device); + if (!device) + return CUBEB_ERROR; + + device->device_id = a_name; + device->devid = (cubeb_devid) device->device_id; + device->friendly_name = a_name; + device->group_id = a_name; + device->vendor_name = a_name; + device->type = type; + device->state = CUBEB_DEVICE_STATE_ENABLED; + device->preferred = CUBEB_DEVICE_PREF_ALL; + device->format = CUBEB_DEVICE_FMT_S16NE; + device->default_format = CUBEB_DEVICE_FMT_S16NE; + device->max_channels = max_channels; + device->min_rate = rate; + device->max_rate = rate; + device->default_rate = rate; + device->latency_lo = 0; + device->latency_hi = 0; + + collection->device = device; + collection->count = 1; + + return CUBEB_OK; +} + +static int +alsa_device_collection_destroy(cubeb * context, + cubeb_device_collection * collection) +{ + assert(collection->count == 1); + (void) context; + free(collection->device); + return CUBEB_OK; +} + +static struct cubeb_ops const alsa_ops = { + .init = alsa_init, + .get_backend_id = alsa_get_backend_id, + .get_max_channel_count = alsa_get_max_channel_count, + .get_min_latency = alsa_get_min_latency, + .get_preferred_sample_rate = alsa_get_preferred_sample_rate, + .enumerate_devices = alsa_enumerate_devices, + .device_collection_destroy = alsa_device_collection_destroy, + .destroy = alsa_destroy, + .stream_init = alsa_stream_init, + .stream_destroy = alsa_stream_destroy, + .stream_start = alsa_stream_start, + .stream_stop = alsa_stream_stop, + .stream_reset_default_device = NULL, + .stream_get_position = alsa_stream_get_position, + .stream_get_latency = alsa_stream_get_latency, + .stream_set_volume = alsa_stream_set_volume, + .stream_get_current_device = NULL, + .stream_device_destroy = NULL, + .stream_register_device_changed_callback = NULL, + .register_device_collection_changed = NULL +}; diff --git a/dep/cubeb/src/cubeb_array_queue.h b/dep/cubeb/src/cubeb_array_queue.h new file mode 100644 index 000000000..a8ea4cd17 --- /dev/null +++ b/dep/cubeb/src/cubeb_array_queue.h @@ -0,0 +1,97 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef CUBEB_ARRAY_QUEUE_H +#define CUBEB_ARRAY_QUEUE_H + +#include +#include +#include + +#if defined(__cplusplus) +extern "C" { +#endif + +typedef struct +{ + void ** buf; + size_t num; + size_t writePos; + size_t readPos; + pthread_mutex_t mutex; +} array_queue; + +array_queue * array_queue_create(size_t num) +{ + assert(num != 0); + array_queue * new_queue = (array_queue*)calloc(1, sizeof(array_queue)); + new_queue->buf = (void **)calloc(1, sizeof(void *) * num); + new_queue->readPos = 0; + new_queue->writePos = 0; + new_queue->num = num; + + pthread_mutex_init(&new_queue->mutex, NULL); + + return new_queue; +} + +void array_queue_destroy(array_queue * aq) +{ + assert(aq); + + free(aq->buf); + pthread_mutex_destroy(&aq->mutex); + free(aq); +} + +int array_queue_push(array_queue * aq, void * item) +{ + assert(item); + + pthread_mutex_lock(&aq->mutex); + int ret = -1; + if(aq->buf[aq->writePos % aq->num] == NULL) + { + aq->buf[aq->writePos % aq->num] = item; + aq->writePos = (aq->writePos + 1) % aq->num; + ret = 0; + } + // else queue is full + pthread_mutex_unlock(&aq->mutex); + return ret; +} + +void* array_queue_pop(array_queue * aq) +{ + pthread_mutex_lock(&aq->mutex); + void * value = aq->buf[aq->readPos % aq->num]; + if(value) + { + aq->buf[aq->readPos % aq->num] = NULL; + aq->readPos = (aq->readPos + 1) % aq->num; + } + pthread_mutex_unlock(&aq->mutex); + return value; +} + +size_t array_queue_get_size(array_queue * aq) +{ + pthread_mutex_lock(&aq->mutex); + ssize_t r = aq->writePos - aq->readPos; + if (r < 0) { + r = aq->num + r; + assert(r >= 0); + } + pthread_mutex_unlock(&aq->mutex); + return (size_t)r; +} + +#if defined(__cplusplus) +} +#endif + +#endif //CUBE_ARRAY_QUEUE_H diff --git a/dep/cubeb/src/cubeb_assert.h b/dep/cubeb/src/cubeb_assert.h new file mode 100644 index 000000000..9257a2c86 --- /dev/null +++ b/dep/cubeb/src/cubeb_assert.h @@ -0,0 +1,26 @@ +/* + * Copyright © 2017 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef CUBEB_ASSERT +#define CUBEB_ASSERT + +#include +#include + +/** + * This allow using an external release assert method. This file should only + * export a function or macro called XASSERT that aborts the program. + */ + +#define XASSERT(expr) do { \ + if (!(expr)) { \ + fprintf(stderr, "%s:%d - fatal error: %s\n", __FILE__, __LINE__, #expr); \ + abort(); \ + } \ + } while (0) + +#endif diff --git a/dep/cubeb/src/cubeb_audiotrack.c b/dep/cubeb/src/cubeb_audiotrack.c new file mode 100644 index 000000000..22f1fe0bc --- /dev/null +++ b/dep/cubeb/src/cubeb_audiotrack.c @@ -0,0 +1,441 @@ +/* + * Copyright © 2013 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#if !defined(NDEBUG) +#define NDEBUG +#endif +#include +#include +#include +#include +#include +#include + +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" +#include "android/audiotrack_definitions.h" + +#ifndef ALOG +#if defined(DEBUG) || defined(FORCE_ALOG) +#define ALOG(args...) __android_log_print(ANDROID_LOG_INFO, "Gecko - Cubeb" , ## args) +#else +#define ALOG(args...) +#endif +#endif + +/** + * A lot of bytes for safety. It should be possible to bring this down a bit. */ +#define SIZE_AUDIOTRACK_INSTANCE 256 + +/** + * call dlsym to get the symbol |mangled_name|, handle the error and store the + * pointer in |pointer|. Because depending on Android version, we want different + * symbols, not finding a symbol is not an error. */ +#define DLSYM_DLERROR(mangled_name, pointer, lib) \ + do { \ + pointer = dlsym(lib, mangled_name); \ + if (!pointer) { \ + ALOG("error while loading %stm: %stm\n", mangled_name, dlerror()); \ + } else { \ + ALOG("%stm: OK", mangled_name); \ + } \ + } while(0); + +static struct cubeb_ops const audiotrack_ops; +void audiotrack_destroy(cubeb * context); +void audiotrack_stream_destroy(cubeb_stream * stream); + +struct AudioTrack { + /* only available on ICS and later. The second int paramter is in fact of type audio_stream_type_t. */ + /* static */ status_t (*get_min_frame_count)(int* frame_count, int stream_type, uint32_t rate); + /* if we have a recent ctor, but can't find the above symbol, we + * can get the minimum frame count with this signature, and we are + * running gingerbread. */ + /* static */ status_t (*get_min_frame_count_gingerbread)(int* frame_count, int stream_type, uint32_t rate); + void* (*ctor)(void* instance, int, unsigned int, int, int, int, unsigned int, void (*)(int, void*, void*), void*, int, int); + void* (*dtor)(void* instance); + void (*start)(void* instance); + void (*pause)(void* instance); + uint32_t (*latency)(void* instance); + status_t (*check)(void* instance); + status_t (*get_position)(void* instance, uint32_t* position); + /* static */ int (*get_output_samplingrate)(int* samplerate, int stream); + status_t (*set_marker_position)(void* instance, unsigned int); + status_t (*set_volume)(void* instance, float left, float right); +}; + +struct cubeb { + struct cubeb_ops const * ops; + void * library; + struct AudioTrack klass; +}; + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr; + /**/ + cubeb_stream_params params; + cubeb_data_callback data_callback; + cubeb_state_callback state_callback; + void * instance; + /* Number of frames that have been passed to the AudioTrack callback */ + long unsigned written; + int draining; +}; + +static void +audiotrack_refill(int event, void* user, void* info) +{ + cubeb_stream * stream = user; + switch (event) { + case EVENT_MORE_DATA: { + long got = 0; + struct Buffer * b = (struct Buffer*)info; + + if (stream->draining) { + return; + } + + got = stream->data_callback(stream, stream->user_ptr, NULL, b->raw, b->frameCount); + + stream->written += got; + + if (got != (long)b->frameCount) { + stream->draining = 1; + /* set a marker so we are notified when the are done draining, that is, + * when every frame has been played by android. */ + stream->context->klass.set_marker_position(stream->instance, stream->written); + } + + break; + } + case EVENT_UNDERRUN: + ALOG("underrun in cubeb backend."); + break; + case EVENT_LOOP_END: + assert(0 && "We don't support the loop feature of audiotrack."); + break; + case EVENT_MARKER: + assert(stream->draining); + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_DRAINED); + break; + case EVENT_NEW_POS: + assert(0 && "We don't support the setPositionUpdatePeriod feature of audiotrack."); + break; + case EVENT_BUFFER_END: + assert(0 && "Should not happen."); + break; + } +} + +/* We are running on gingerbread if we found the gingerbread signature for + * getMinFrameCount */ +static int +audiotrack_version_is_gingerbread(cubeb * ctx) +{ + return ctx->klass.get_min_frame_count_gingerbread != NULL; +} + +int +audiotrack_get_min_frame_count(cubeb * ctx, cubeb_stream_params * params, int * min_frame_count) +{ + status_t status; + /* Recent Android have a getMinFrameCount method. */ + if (!audiotrack_version_is_gingerbread(ctx)) { + status = ctx->klass.get_min_frame_count(min_frame_count, AUDIO_STREAM_TYPE_MUSIC, params->rate); + } else { + status = ctx->klass.get_min_frame_count_gingerbread(min_frame_count, AUDIO_STREAM_TYPE_MUSIC, params->rate); + } + if (status != 0) { + ALOG("error getting the min frame count"); + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +int +audiotrack_init(cubeb ** context, char const * context_name) +{ + cubeb * ctx; + struct AudioTrack* c; + + assert(context); + *context = NULL; + + ctx = calloc(1, sizeof(*ctx)); + assert(ctx); + + /* If we use an absolute path here ("/system/lib/libmedia.so"), and on Android + * 2.2, the dlopen succeeds, all the dlsym succeed, but a segfault happens on + * the first call to a dlsym'ed function. Somehow this does not happen when + * using only the name of the library. */ + ctx->library = dlopen("libmedia.so", RTLD_LAZY); + if (!ctx->library) { + ALOG("dlopen error: %s.", dlerror()); + free(ctx); + return CUBEB_ERROR; + } + + /* Recent Android first, then Gingerbread. */ + DLSYM_DLERROR("_ZN7android10AudioTrackC1EijiiijPFviPvS1_ES1_ii", ctx->klass.ctor, ctx->library); + DLSYM_DLERROR("_ZN7android10AudioTrackD1Ev", ctx->klass.dtor, ctx->library); + + DLSYM_DLERROR("_ZNK7android10AudioTrack7latencyEv", ctx->klass.latency, ctx->library); + DLSYM_DLERROR("_ZNK7android10AudioTrack9initCheckEv", ctx->klass.check, ctx->library); + + DLSYM_DLERROR("_ZN7android11AudioSystem21getOutputSamplingRateEPii", ctx->klass.get_output_samplingrate, ctx->library); + + /* |getMinFrameCount| is available on gingerbread and ICS with different signatures. */ + DLSYM_DLERROR("_ZN7android10AudioTrack16getMinFrameCountEPi19audio_stream_type_tj", ctx->klass.get_min_frame_count, ctx->library); + if (!ctx->klass.get_min_frame_count) { + DLSYM_DLERROR("_ZN7android10AudioTrack16getMinFrameCountEPiij", ctx->klass.get_min_frame_count_gingerbread, ctx->library); + } + + DLSYM_DLERROR("_ZN7android10AudioTrack5startEv", ctx->klass.start, ctx->library); + DLSYM_DLERROR("_ZN7android10AudioTrack5pauseEv", ctx->klass.pause, ctx->library); + DLSYM_DLERROR("_ZN7android10AudioTrack11getPositionEPj", ctx->klass.get_position, ctx->library); + DLSYM_DLERROR("_ZN7android10AudioTrack17setMarkerPositionEj", ctx->klass.set_marker_position, ctx->library); + DLSYM_DLERROR("_ZN7android10AudioTrack9setVolumeEff", ctx->klass.set_volume, ctx->library); + + /* check that we have a combination of symbol that makes sense */ + c = &ctx->klass; + if(!(c->ctor && + c->dtor && c->latency && c->check && + /* at least one way to get the minimum frame count to request. */ + (c->get_min_frame_count || + c->get_min_frame_count_gingerbread) && + c->start && c->pause && c->get_position && c->set_marker_position)) { + ALOG("Could not find all the symbols we need."); + audiotrack_destroy(ctx); + return CUBEB_ERROR; + } + + ctx->ops = &audiotrack_ops; + + *context = ctx; + + return CUBEB_OK; +} + +char const * +audiotrack_get_backend_id(cubeb * context) +{ + return "audiotrack"; +} + +static int +audiotrack_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + assert(ctx && max_channels); + + /* The android mixer handles up to two channels, see + http://androidxref.com/4.2.2_r1/xref/frameworks/av/services/audioflinger/AudioFlinger.h#67 */ + *max_channels = 2; + + return CUBEB_OK; +} + +static int +audiotrack_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_ms) +{ + /* We always use the lowest latency possible when using this backend (see + * audiotrack_stream_init), so this value is not going to be used. */ + int r; + + r = audiotrack_get_min_frame_count(ctx, ¶ms, (int *)latency_ms); + if (r != CUBEB_OK) { + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +static int +audiotrack_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) +{ + status_t r; + + r = ctx->klass.get_output_samplingrate((int32_t *)rate, 3 /* MUSIC */); + + return r == 0 ? CUBEB_OK : CUBEB_ERROR; +} + +void +audiotrack_destroy(cubeb * context) +{ + assert(context); + + dlclose(context->library); + + free(context); +} + +int +audiotrack_stream_init(cubeb * ctx, cubeb_stream ** stream, char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + cubeb_stream * stm; + int32_t channels; + uint32_t min_frame_count; + + assert(ctx && stream); + + assert(!input_stream_params && "not supported"); + if (input_device || output_device) { + /* Device selection not yet implemented. */ + return CUBEB_ERROR_DEVICE_UNAVAILABLE; + } + + if (output_stream_params->format == CUBEB_SAMPLE_FLOAT32LE || + output_stream_params->format == CUBEB_SAMPLE_FLOAT32BE) { + return CUBEB_ERROR_INVALID_FORMAT; + } + + if (audiotrack_get_min_frame_count(ctx, output_stream_params, (int *)&min_frame_count)) { + return CUBEB_ERROR; + } + + stm = calloc(1, sizeof(*stm)); + assert(stm); + + stm->context = ctx; + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + stm->params = *output_stream_params; + + stm->instance = calloc(SIZE_AUDIOTRACK_INSTANCE, 1); + (*(uint32_t*)((intptr_t)stm->instance + SIZE_AUDIOTRACK_INSTANCE - 4)) = 0xbaadbaad; + assert(stm->instance && "cubeb: EOM"); + + /* gingerbread uses old channel layout enum */ + if (audiotrack_version_is_gingerbread(ctx)) { + channels = stm->params.channels == 2 ? AUDIO_CHANNEL_OUT_STEREO_Legacy : AUDIO_CHANNEL_OUT_MONO_Legacy; + } else { + channels = stm->params.channels == 2 ? AUDIO_CHANNEL_OUT_STEREO_ICS : AUDIO_CHANNEL_OUT_MONO_ICS; + } + + ctx->klass.ctor(stm->instance, AUDIO_STREAM_TYPE_MUSIC, stm->params.rate, + AUDIO_FORMAT_PCM_16_BIT, channels, min_frame_count, 0, + audiotrack_refill, stm, 0, 0); + + assert((*(uint32_t*)((intptr_t)stm->instance + SIZE_AUDIOTRACK_INSTANCE - 4)) == 0xbaadbaad); + + if (ctx->klass.check(stm->instance)) { + ALOG("stream not initialized properly."); + audiotrack_stream_destroy(stm); + return CUBEB_ERROR; + } + + *stream = stm; + + return CUBEB_OK; +} + +void +audiotrack_stream_destroy(cubeb_stream * stream) +{ + assert(stream->context); + + stream->context->klass.dtor(stream->instance); + + free(stream->instance); + stream->instance = NULL; + free(stream); +} + +int +audiotrack_stream_start(cubeb_stream * stream) +{ + assert(stream->instance); + + stream->context->klass.start(stream->instance); + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_STARTED); + + return CUBEB_OK; +} + +int +audiotrack_stream_stop(cubeb_stream * stream) +{ + assert(stream->instance); + + stream->context->klass.pause(stream->instance); + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_STOPPED); + + return CUBEB_OK; +} + +int +audiotrack_stream_get_position(cubeb_stream * stream, uint64_t * position) +{ + uint32_t p; + + assert(stream->instance && position); + stream->context->klass.get_position(stream->instance, &p); + *position = p; + + return CUBEB_OK; +} + +int +audiotrack_stream_get_latency(cubeb_stream * stream, uint32_t * latency) +{ + assert(stream->instance && latency); + + /* Android returns the latency in ms, we want it in frames. */ + *latency = stream->context->klass.latency(stream->instance); + /* with rate <= 96000, we won't overflow until 44.739 seconds of latency */ + *latency = (*latency * stream->params.rate) / 1000; + + return 0; +} + +int +audiotrack_stream_set_volume(cubeb_stream * stream, float volume) +{ + status_t status; + + status = stream->context->klass.set_volume(stream->instance, volume, volume); + + if (status) { + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +static struct cubeb_ops const audiotrack_ops = { + .init = audiotrack_init, + .get_backend_id = audiotrack_get_backend_id, + .get_max_channel_count = audiotrack_get_max_channel_count, + .get_min_latency = audiotrack_get_min_latency, + .get_preferred_sample_rate = audiotrack_get_preferred_sample_rate, + .enumerate_devices = NULL, + .device_collection_destroy = NULL, + .destroy = audiotrack_destroy, + .stream_init = audiotrack_stream_init, + .stream_destroy = audiotrack_stream_destroy, + .stream_start = audiotrack_stream_start, + .stream_stop = audiotrack_stream_stop, + .stream_reset_default_device = NULL, + .stream_get_position = audiotrack_stream_get_position, + .stream_get_latency = audiotrack_stream_get_latency, + .stream_set_volume = audiotrack_stream_set_volume, + .stream_get_current_device = NULL, + .stream_device_destroy = NULL, + .stream_register_device_changed_callback = NULL, + .register_device_collection_changed = NULL +}; diff --git a/dep/cubeb/src/cubeb_audiounit.cpp b/dep/cubeb/src/cubeb_audiounit.cpp new file mode 100644 index 000000000..4e89a7d5a --- /dev/null +++ b/dep/cubeb/src/cubeb_audiounit.cpp @@ -0,0 +1,3620 @@ +/* + * Copyright © 2011 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#undef NDEBUG + +#include +#include +#include +#include +#include +#include +#if !TARGET_OS_IPHONE +#include +#include +#include +#include +#endif +#include +#include +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" +#include "cubeb_mixer.h" +#if !TARGET_OS_IPHONE +#include "cubeb_osx_run_loop.h" +#endif +#include "cubeb_resampler.h" +#include "cubeb_ring_array.h" +#include +#include +#include +#include +#include +#include + +using namespace std; + +#if MAC_OS_X_VERSION_MIN_REQUIRED < 101000 +typedef UInt32 AudioFormatFlags; +#endif + +#define AU_OUT_BUS 0 +#define AU_IN_BUS 1 + +const char * DISPATCH_QUEUE_LABEL = "org.mozilla.cubeb"; +const char * PRIVATE_AGGREGATE_DEVICE_NAME = "CubebAggregateDevice"; + +#ifdef ALOGV +#undef ALOGV +#endif +#define ALOGV(msg, ...) dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_HIGH, 0), ^{LOGV(msg, ##__VA_ARGS__);}) + +#ifdef ALOG +#undef ALOG +#endif +#define ALOG(msg, ...) dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_HIGH, 0), ^{LOG(msg, ##__VA_ARGS__);}) + +/* Testing empirically, some headsets report a minimal latency that is very + * low, but this does not work in practice. Lie and say the minimum is 256 + * frames. */ +const uint32_t SAFE_MIN_LATENCY_FRAMES = 128; +const uint32_t SAFE_MAX_LATENCY_FRAMES = 512; + +const AudioObjectPropertyAddress DEFAULT_INPUT_DEVICE_PROPERTY_ADDRESS = { + kAudioHardwarePropertyDefaultInputDevice, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster +}; + +const AudioObjectPropertyAddress DEFAULT_OUTPUT_DEVICE_PROPERTY_ADDRESS = { + kAudioHardwarePropertyDefaultOutputDevice, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster +}; + +const AudioObjectPropertyAddress DEVICE_IS_ALIVE_PROPERTY_ADDRESS = { + kAudioDevicePropertyDeviceIsAlive, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster +}; + +const AudioObjectPropertyAddress DEVICES_PROPERTY_ADDRESS = { + kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster +}; + +const AudioObjectPropertyAddress INPUT_DATA_SOURCE_PROPERTY_ADDRESS = { + kAudioDevicePropertyDataSource, + kAudioDevicePropertyScopeInput, + kAudioObjectPropertyElementMaster +}; + +const AudioObjectPropertyAddress OUTPUT_DATA_SOURCE_PROPERTY_ADDRESS = { + kAudioDevicePropertyDataSource, + kAudioDevicePropertyScopeOutput, + kAudioObjectPropertyElementMaster +}; + +typedef uint32_t device_flags_value; + +enum device_flags { + DEV_UNKNOWN = 0x00, /* Unknown */ + DEV_INPUT = 0x01, /* Record device like mic */ + DEV_OUTPUT = 0x02, /* Playback device like speakers */ + DEV_SYSTEM_DEFAULT = 0x04, /* System default device */ + DEV_SELECTED_DEFAULT = 0x08, /* User selected to use the system default device */ +}; + +void audiounit_stream_stop_internal(cubeb_stream * stm); +static int audiounit_stream_start_internal(cubeb_stream * stm); +static void audiounit_close_stream(cubeb_stream *stm); +static int audiounit_setup_stream(cubeb_stream *stm); +static vector +audiounit_get_devices_of_type(cubeb_device_type devtype); +static UInt32 audiounit_get_device_presentation_latency(AudioObjectID devid, AudioObjectPropertyScope scope); + +#if !TARGET_OS_IPHONE +static AudioObjectID audiounit_get_default_device_id(cubeb_device_type type); +static int audiounit_uninstall_device_changed_callback(cubeb_stream * stm); +static int audiounit_uninstall_system_changed_callback(cubeb_stream * stm); +static void audiounit_reinit_stream_async(cubeb_stream * stm, device_flags_value flags); +#endif + +extern cubeb_ops const audiounit_ops; + +struct cubeb { + cubeb_ops const * ops = &audiounit_ops; + owned_critical_section mutex; + int active_streams = 0; + uint32_t global_latency_frames = 0; + cubeb_device_collection_changed_callback input_collection_changed_callback = nullptr; + void * input_collection_changed_user_ptr = nullptr; + cubeb_device_collection_changed_callback output_collection_changed_callback = nullptr; + void * output_collection_changed_user_ptr = nullptr; + // Store list of devices to detect changes + vector input_device_array; + vector output_device_array; + // The queue should be released when it’s no longer needed. + dispatch_queue_t serial_queue = dispatch_queue_create(DISPATCH_QUEUE_LABEL, DISPATCH_QUEUE_SERIAL); + // Current used channel layout + atomic layout{ CUBEB_LAYOUT_UNDEFINED }; + uint32_t channels = 0; +}; + +static unique_ptr +make_sized_audio_channel_layout(size_t sz) +{ + assert(sz >= sizeof(AudioChannelLayout)); + AudioChannelLayout * acl = reinterpret_cast(calloc(1, sz)); + assert(acl); // Assert the allocation works. + return unique_ptr(acl, free); +} + +enum class io_side { + INPUT, + OUTPUT, +}; + +static char const * +to_string(io_side side) +{ + switch (side) { + case io_side::INPUT: + return "input"; + case io_side::OUTPUT: + return "output"; + } +} + +struct device_info { + AudioDeviceID id = kAudioObjectUnknown; + device_flags_value flags = DEV_UNKNOWN; +}; + +struct property_listener { + AudioDeviceID device_id; + const AudioObjectPropertyAddress * property_address; + AudioObjectPropertyListenerProc callback; + cubeb_stream * stream; + + property_listener(AudioDeviceID id, + const AudioObjectPropertyAddress * address, + AudioObjectPropertyListenerProc proc, + cubeb_stream * stm) + : device_id(id) + , property_address(address) + , callback(proc) + , stream(stm) + {} +}; + +struct cubeb_stream { + explicit cubeb_stream(cubeb * context); + + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr = nullptr; + /**/ + + cubeb_data_callback data_callback = nullptr; + cubeb_state_callback state_callback = nullptr; + cubeb_device_changed_callback device_changed_callback = nullptr; + owned_critical_section device_changed_callback_lock; + /* Stream creation parameters */ + cubeb_stream_params input_stream_params = { CUBEB_SAMPLE_FLOAT32NE, 0, 0, CUBEB_LAYOUT_UNDEFINED, CUBEB_STREAM_PREF_NONE }; + cubeb_stream_params output_stream_params = { CUBEB_SAMPLE_FLOAT32NE, 0, 0, CUBEB_LAYOUT_UNDEFINED, CUBEB_STREAM_PREF_NONE }; + device_info input_device; + device_info output_device; + /* Format descriptions */ + AudioStreamBasicDescription input_desc; + AudioStreamBasicDescription output_desc; + /* I/O AudioUnits */ + AudioUnit input_unit = nullptr; + AudioUnit output_unit = nullptr; + /* I/O device sample rate */ + Float64 input_hw_rate = 0; + Float64 output_hw_rate = 0; + /* Expected I/O thread interleave, + * calculated from I/O hw rate. */ + int expected_output_callbacks_in_a_row = 0; + owned_critical_section mutex; + // Hold the input samples in every input callback iteration. + // Only accessed on input/output callback thread and during initial configure. + unique_ptr input_linear_buffer; + /* Frame counters */ + atomic frames_played{ 0 }; + uint64_t frames_queued = 0; + // How many frames got read from the input since the stream started (includes + // padded silence) + atomic frames_read{ 0 }; + // How many frames got written to the output device since the stream started + atomic frames_written{ 0 }; + atomic shutdown{ true }; + atomic draining{ false }; + atomic reinit_pending { false }; + atomic destroy_pending{ false }; + /* Latency requested by the user. */ + uint32_t latency_frames = 0; + atomic current_latency_frames{ 0 }; + atomic total_output_latency_frames { 0 }; + unique_ptr resampler; + /* This is true if a device change callback is currently running. */ + atomic switching_device{ false }; + atomic buffer_size_change_state{ false }; + AudioDeviceID aggregate_device_id = kAudioObjectUnknown; // the aggregate device id + AudioObjectID plugin_id = kAudioObjectUnknown; // used to create aggregate device + /* Mixer interface */ + unique_ptr mixer; + /* Buffer where remixing/resampling will occur when upmixing is required */ + /* Only accessed from callback thread */ + unique_ptr temp_buffer; + size_t temp_buffer_size = 0; // size in bytes. + /* Listeners indicating what system events are monitored. */ + unique_ptr default_input_listener; + unique_ptr default_output_listener; + unique_ptr input_alive_listener; + unique_ptr input_source_listener; + unique_ptr output_source_listener; +}; + +bool has_input(cubeb_stream * stm) +{ + return stm->input_stream_params.rate != 0; +} + +bool has_output(cubeb_stream * stm) +{ + return stm->output_stream_params.rate != 0; +} + +cubeb_channel +channel_label_to_cubeb_channel(UInt32 label) +{ + switch (label) { + case kAudioChannelLabel_Left: + return CHANNEL_FRONT_LEFT; + case kAudioChannelLabel_Right: + return CHANNEL_FRONT_RIGHT; + case kAudioChannelLabel_Center: + return CHANNEL_FRONT_CENTER; + case kAudioChannelLabel_LFEScreen: + return CHANNEL_LOW_FREQUENCY; + case kAudioChannelLabel_LeftSurround: + return CHANNEL_BACK_LEFT; + case kAudioChannelLabel_RightSurround: + return CHANNEL_BACK_RIGHT; + case kAudioChannelLabel_LeftCenter: + return CHANNEL_FRONT_LEFT_OF_CENTER; + case kAudioChannelLabel_RightCenter: + return CHANNEL_FRONT_RIGHT_OF_CENTER; + case kAudioChannelLabel_CenterSurround: + return CHANNEL_BACK_CENTER; + case kAudioChannelLabel_LeftSurroundDirect: + return CHANNEL_SIDE_LEFT; + case kAudioChannelLabel_RightSurroundDirect: + return CHANNEL_SIDE_RIGHT; + case kAudioChannelLabel_TopCenterSurround: + return CHANNEL_TOP_CENTER; + case kAudioChannelLabel_VerticalHeightLeft: + return CHANNEL_TOP_FRONT_LEFT; + case kAudioChannelLabel_VerticalHeightCenter: + return CHANNEL_TOP_FRONT_CENTER; + case kAudioChannelLabel_VerticalHeightRight: + return CHANNEL_TOP_FRONT_RIGHT; + case kAudioChannelLabel_TopBackLeft: + return CHANNEL_TOP_BACK_LEFT; + case kAudioChannelLabel_TopBackCenter: + return CHANNEL_TOP_BACK_CENTER; + case kAudioChannelLabel_TopBackRight: + return CHANNEL_TOP_BACK_RIGHT; + default: + return CHANNEL_UNKNOWN; + } +} + +AudioChannelLabel +cubeb_channel_to_channel_label(cubeb_channel channel) +{ + switch (channel) { + case CHANNEL_FRONT_LEFT: + return kAudioChannelLabel_Left; + case CHANNEL_FRONT_RIGHT: + return kAudioChannelLabel_Right; + case CHANNEL_FRONT_CENTER: + return kAudioChannelLabel_Center; + case CHANNEL_LOW_FREQUENCY: + return kAudioChannelLabel_LFEScreen; + case CHANNEL_BACK_LEFT: + return kAudioChannelLabel_LeftSurround; + case CHANNEL_BACK_RIGHT: + return kAudioChannelLabel_RightSurround; + case CHANNEL_FRONT_LEFT_OF_CENTER: + return kAudioChannelLabel_LeftCenter; + case CHANNEL_FRONT_RIGHT_OF_CENTER: + return kAudioChannelLabel_RightCenter; + case CHANNEL_BACK_CENTER: + return kAudioChannelLabel_CenterSurround; + case CHANNEL_SIDE_LEFT: + return kAudioChannelLabel_LeftSurroundDirect; + case CHANNEL_SIDE_RIGHT: + return kAudioChannelLabel_RightSurroundDirect; + case CHANNEL_TOP_CENTER: + return kAudioChannelLabel_TopCenterSurround; + case CHANNEL_TOP_FRONT_LEFT: + return kAudioChannelLabel_VerticalHeightLeft; + case CHANNEL_TOP_FRONT_CENTER: + return kAudioChannelLabel_VerticalHeightCenter; + case CHANNEL_TOP_FRONT_RIGHT: + return kAudioChannelLabel_VerticalHeightRight; + case CHANNEL_TOP_BACK_LEFT: + return kAudioChannelLabel_TopBackLeft; + case CHANNEL_TOP_BACK_CENTER: + return kAudioChannelLabel_TopBackCenter; + case CHANNEL_TOP_BACK_RIGHT: + return kAudioChannelLabel_TopBackRight; + default: + return kAudioChannelLabel_Unknown; + } +} + +#if TARGET_OS_IPHONE +typedef UInt32 AudioDeviceID; +typedef UInt32 AudioObjectID; + +#define AudioGetCurrentHostTime mach_absolute_time + +#endif + +uint64_t +ConvertHostTimeToNanos(uint64_t host_time) +{ + static struct mach_timebase_info timebase_info; + static bool initialized = false; + if (!initialized) { + mach_timebase_info(&timebase_info); + initialized = true; + } + + long double answer = host_time; + if (timebase_info.numer != timebase_info.denom) { + answer *= timebase_info.numer; + answer /= timebase_info.denom; + } + return (uint64_t)answer; +} + +static void +audiounit_increment_active_streams(cubeb * ctx) +{ + ctx->mutex.assert_current_thread_owns(); + ctx->active_streams += 1; +} + +static void +audiounit_decrement_active_streams(cubeb * ctx) +{ + ctx->mutex.assert_current_thread_owns(); + ctx->active_streams -= 1; +} + +static int +audiounit_active_streams(cubeb * ctx) +{ + ctx->mutex.assert_current_thread_owns(); + return ctx->active_streams; +} + +static void +audiounit_set_global_latency(cubeb * ctx, uint32_t latency_frames) +{ + ctx->mutex.assert_current_thread_owns(); + assert(audiounit_active_streams(ctx) == 1); + ctx->global_latency_frames = latency_frames; +} + +static void +audiounit_make_silent(AudioBuffer * ioData) +{ + assert(ioData); + assert(ioData->mData); + memset(ioData->mData, 0, ioData->mDataByteSize); +} + +static OSStatus +audiounit_render_input(cubeb_stream * stm, + AudioUnitRenderActionFlags * flags, + AudioTimeStamp const * tstamp, + UInt32 bus, + UInt32 input_frames) +{ + /* Create the AudioBufferList to store input. */ + AudioBufferList input_buffer_list; + input_buffer_list.mBuffers[0].mDataByteSize = + stm->input_desc.mBytesPerFrame * input_frames; + input_buffer_list.mBuffers[0].mData = nullptr; + input_buffer_list.mBuffers[0].mNumberChannels = stm->input_desc.mChannelsPerFrame; + input_buffer_list.mNumberBuffers = 1; + + /* Render input samples */ + OSStatus r = AudioUnitRender(stm->input_unit, + flags, + tstamp, + bus, + input_frames, + &input_buffer_list); + + if (r != noErr) { + LOG("AudioUnitRender rv=%d", r); + if (r != kAudioUnitErr_CannotDoInCurrentContext) { + return r; + } + if (stm->output_unit) { + // kAudioUnitErr_CannotDoInCurrentContext is returned when using a BT + // headset and the profile is changed from A2DP to HFP/HSP. The previous + // output device is no longer valid and must be reset. + audiounit_reinit_stream_async(stm, DEV_INPUT | DEV_OUTPUT); + } + // For now state that no error occurred and feed silence, stream will be + // resumed once reinit has completed. + ALOGV("(%p) input: reinit pending feeding silence instead", stm); + stm->input_linear_buffer->push_silence(input_frames * stm->input_desc.mChannelsPerFrame); + } else { + /* Copy input data in linear buffer. */ + stm->input_linear_buffer->push(input_buffer_list.mBuffers[0].mData, + input_frames * stm->input_desc.mChannelsPerFrame); + } + + /* Advance input frame counter. */ + assert(input_frames > 0); + stm->frames_read += input_frames; + + ALOGV("(%p) input: buffers %u, size %u, channels %u, rendered frames %d, total frames %lu.", + stm, + (unsigned int) input_buffer_list.mNumberBuffers, + (unsigned int) input_buffer_list.mBuffers[0].mDataByteSize, + (unsigned int) input_buffer_list.mBuffers[0].mNumberChannels, + (unsigned int) input_frames, + stm->input_linear_buffer->length() / stm->input_desc.mChannelsPerFrame); + + return noErr; +} + +static OSStatus +audiounit_input_callback(void * user_ptr, + AudioUnitRenderActionFlags * flags, + AudioTimeStamp const * tstamp, + UInt32 bus, + UInt32 input_frames, + AudioBufferList * /* bufs */) +{ + cubeb_stream * stm = static_cast(user_ptr); + + assert(stm->input_unit != NULL); + assert(AU_IN_BUS == bus); + + if (stm->shutdown) { + ALOG("(%p) input shutdown", stm); + return noErr; + } + + if (stm->draining) { + OSStatus r = AudioOutputUnitStop(stm->input_unit); + assert(r == 0); + // Only fire state callback in input-only stream. For duplex stream, + // the state callback will be fired in output callback. + if (stm->output_unit == NULL) { + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + } + return noErr; + } + + OSStatus r = audiounit_render_input(stm, flags, tstamp, bus, input_frames); + if (r != noErr) { + return r; + } + + // Full Duplex. We'll call data_callback in the AudioUnit output callback. + if (stm->output_unit != NULL) { + return noErr; + } + + /* Input only. Call the user callback through resampler. + Resampler will deliver input buffer in the correct rate. */ + assert(input_frames <= stm->input_linear_buffer->length() / stm->input_desc.mChannelsPerFrame); + long total_input_frames = stm->input_linear_buffer->length() / stm->input_desc.mChannelsPerFrame; + long outframes = cubeb_resampler_fill(stm->resampler.get(), + stm->input_linear_buffer->data(), + &total_input_frames, + NULL, + 0); + stm->draining = outframes < total_input_frames; + + // Reset input buffer + stm->input_linear_buffer->clear(); + + return noErr; +} + +static void +audiounit_mix_output_buffer(cubeb_stream * stm, + size_t output_frames, + void * input_buffer, + size_t input_buffer_size, + void * output_buffer, + size_t output_buffer_size) +{ + assert(input_buffer_size >= + cubeb_sample_size(stm->output_stream_params.format) * + stm->output_stream_params.channels * output_frames); + assert(output_buffer_size >= stm->output_desc.mBytesPerFrame * output_frames); + + int r = cubeb_mixer_mix(stm->mixer.get(), + output_frames, + input_buffer, + input_buffer_size, + output_buffer, + output_buffer_size); + if (r != 0) { + LOG("Remix error = %d", r); + } +} + +// Return how many input frames (sampled at input_hw_rate) are needed to provide +// output_frames (sampled at output_stream_params.rate) +static int64_t +minimum_resampling_input_frames(cubeb_stream * stm, uint32_t output_frames) +{ + if (stm->input_hw_rate == stm->output_stream_params.rate) { + // Fast path. + return output_frames; + } + return ceil(stm->input_hw_rate * output_frames / + stm->output_stream_params.rate); +} + +static OSStatus +audiounit_output_callback(void * user_ptr, + AudioUnitRenderActionFlags * /* flags */, + AudioTimeStamp const * tstamp, + UInt32 bus, + UInt32 output_frames, + AudioBufferList * outBufferList) +{ + assert(AU_OUT_BUS == bus); + assert(outBufferList->mNumberBuffers == 1); + + cubeb_stream * stm = static_cast(user_ptr); + + uint64_t now = ConvertHostTimeToNanos(mach_absolute_time()); + uint64_t audio_output_time = ConvertHostTimeToNanos(tstamp->mHostTime); + uint64_t output_latency_ns = audio_output_time - now; + + const int ns2s = 1e9; + // The total output latency is the timestamp difference + the stream latency + + // the hardware latency. + stm->total_output_latency_frames = output_latency_ns * stm->output_hw_rate / ns2s + stm->current_latency_frames; + + ALOGV("(%p) output: buffers %u, size %u, channels %u, frames %u, total input frames %lu.", + stm, + (unsigned int) outBufferList->mNumberBuffers, + (unsigned int) outBufferList->mBuffers[0].mDataByteSize, + (unsigned int) outBufferList->mBuffers[0].mNumberChannels, + (unsigned int) output_frames, + has_input(stm) ? stm->input_linear_buffer->length() / stm->input_desc.mChannelsPerFrame : 0); + + long input_frames = 0; + void * output_buffer = NULL, * input_buffer = NULL; + + if (stm->shutdown) { + ALOG("(%p) output shutdown.", stm); + audiounit_make_silent(&outBufferList->mBuffers[0]); + return noErr; + } + + if (stm->draining) { + OSStatus r = AudioOutputUnitStop(stm->output_unit); + assert(r == 0); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + audiounit_make_silent(&outBufferList->mBuffers[0]); + return noErr; + } + + /* Get output buffer. */ + if (stm->mixer) { + // If remixing needs to occur, we can't directly work in our final + // destination buffer as data may be overwritten or too small to start with. + size_t size_needed = output_frames * stm->output_stream_params.channels * + cubeb_sample_size(stm->output_stream_params.format); + if (stm->temp_buffer_size < size_needed) { + stm->temp_buffer.reset(new uint8_t[size_needed]); + stm->temp_buffer_size = size_needed; + } + output_buffer = stm->temp_buffer.get(); + } else { + output_buffer = outBufferList->mBuffers[0].mData; + } + + stm->frames_written += output_frames; + + /* If Full duplex get also input buffer */ + if (stm->input_unit != NULL) { + /* If the output callback came first and this is a duplex stream, we need to + * fill in some additional silence in the resampler. + * Otherwise, if we had more than expected callbacks in a row, or we're + * currently switching, we add some silence as well to compensate for the + * fact that we're lacking some input data. */ + uint32_t input_frames_needed = + minimum_resampling_input_frames(stm, stm->frames_written); + long missing_frames = input_frames_needed - stm->frames_read; + if (missing_frames > 0) { + stm->input_linear_buffer->push_silence(missing_frames * stm->input_desc.mChannelsPerFrame); + stm->frames_read = input_frames_needed; + + ALOG("(%p) %s pushed %ld frames of input silence.", stm, stm->frames_read == 0 ? "Input hasn't started," : + stm->switching_device ? "Device switching," : "Drop out,", missing_frames); + } + input_buffer = stm->input_linear_buffer->data(); + // Number of input frames in the buffer. It will change to actually used frames + // inside fill + input_frames = stm->input_linear_buffer->length() / stm->input_desc.mChannelsPerFrame; + } + + /* Call user callback through resampler. */ + long outframes = cubeb_resampler_fill(stm->resampler.get(), + input_buffer, + input_buffer ? &input_frames : NULL, + output_buffer, + output_frames); + + if (input_buffer) { + // Pop from the buffer the frames used by the the resampler. + stm->input_linear_buffer->pop(input_frames * stm->input_desc.mChannelsPerFrame); + } + + if (outframes < 0 || outframes > output_frames) { + stm->shutdown = true; + OSStatus r = AudioOutputUnitStop(stm->output_unit); + assert(r == 0); + if (stm->input_unit) { + r = AudioOutputUnitStop(stm->input_unit); + assert(r == 0); + } + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + audiounit_make_silent(&outBufferList->mBuffers[0]); + return noErr; + } + + stm->draining = (UInt32) outframes < output_frames; + stm->frames_played = stm->frames_queued; + stm->frames_queued += outframes; + + /* Post process output samples. */ + if (stm->draining) { + /* Clear missing frames (silence) */ + size_t channels = stm->output_stream_params.channels; + size_t missing_samples = (output_frames - outframes) * channels; + size_t size_sample = cubeb_sample_size(stm->output_stream_params.format); + /* number of bytes that have been filled with valid audio by the callback. */ + size_t audio_byte_count = outframes * channels * size_sample; + PodZero((uint8_t*)output_buffer + audio_byte_count, + missing_samples * size_sample); + } + + /* Mixing */ + if (stm->mixer) { + audiounit_mix_output_buffer(stm, + output_frames, + output_buffer, + stm->temp_buffer_size, + outBufferList->mBuffers[0].mData, + outBufferList->mBuffers[0].mDataByteSize); + } + + return noErr; +} + +extern "C" { +int +audiounit_init(cubeb ** context, char const * /* context_name */) +{ +#if !TARGET_OS_IPHONE + cubeb_set_coreaudio_notification_runloop(); +#endif + + *context = new cubeb; + + return CUBEB_OK; +} +} + +static char const * +audiounit_get_backend_id(cubeb * /* ctx */) +{ + return "audiounit"; +} + +#if !TARGET_OS_IPHONE + +static int audiounit_stream_get_volume(cubeb_stream * stm, float * volume); +static int audiounit_stream_set_volume(cubeb_stream * stm, float volume); + +static int +audiounit_set_device_info(cubeb_stream * stm, AudioDeviceID id, io_side side) +{ + assert(stm); + + device_info * info = nullptr; + cubeb_device_type type = CUBEB_DEVICE_TYPE_UNKNOWN; + + if (side == io_side::INPUT) { + info = &stm->input_device; + type = CUBEB_DEVICE_TYPE_INPUT; + } else if (side == io_side::OUTPUT) { + info = &stm->output_device; + type = CUBEB_DEVICE_TYPE_OUTPUT; + } + memset(info, 0, sizeof(device_info)); + info->id = id; + + if (side == io_side::INPUT) { + info->flags |= DEV_INPUT; + } else if (side == io_side::OUTPUT) { + info->flags |= DEV_OUTPUT; + } + + AudioDeviceID default_device_id = audiounit_get_default_device_id(type); + if (default_device_id == kAudioObjectUnknown) { + return CUBEB_ERROR; + } + if (id == kAudioObjectUnknown) { + info->id = default_device_id; + info->flags |= DEV_SELECTED_DEFAULT; + } + + if (info->id == default_device_id) { + info->flags |= DEV_SYSTEM_DEFAULT; + } + + assert(info->id); + assert(info->flags & DEV_INPUT && !(info->flags & DEV_OUTPUT) || + !(info->flags & DEV_INPUT) && info->flags & DEV_OUTPUT); + + return CUBEB_OK; +} + + +static int +audiounit_reinit_stream(cubeb_stream * stm, device_flags_value flags) +{ + auto_lock context_lock(stm->context->mutex); + assert((flags & DEV_INPUT && stm->input_unit) || + (flags & DEV_OUTPUT && stm->output_unit)); + if (!stm->shutdown) { + audiounit_stream_stop_internal(stm); + } + + int r = audiounit_uninstall_device_changed_callback(stm); + if (r != CUBEB_OK) { + LOG("(%p) Could not uninstall all device change listeners.", stm); + } + + { + auto_lock lock(stm->mutex); + float volume = 0.0; + int vol_rv = CUBEB_ERROR; + if (stm->output_unit) { + vol_rv = audiounit_stream_get_volume(stm, &volume); + } + + audiounit_close_stream(stm); + + /* Reinit occurs in one of the following case: + * - When the device is not alive any more + * - When the default system device change. + * - The bluetooth device changed from A2DP to/from HFP/HSP profile + * We first attempt to re-use the same device id, should that fail we will + * default to the (potentially new) default device. */ + AudioDeviceID input_device = flags & DEV_INPUT ? stm->input_device.id : kAudioObjectUnknown; + if (flags & DEV_INPUT) { + r = audiounit_set_device_info(stm, input_device, io_side::INPUT); + if (r != CUBEB_OK) { + LOG("(%p) Set input device info failed. This can happen when last media device is unplugged", stm); + return CUBEB_ERROR; + } + } + + /* Always use the default output on reinit. This is not correct in every + * case but it is sufficient for Firefox and prevent reinit from reporting + * failures. It will change soon when reinit mechanism will be updated. */ + r = audiounit_set_device_info(stm, kAudioObjectUnknown, io_side::OUTPUT); + if (r != CUBEB_OK) { + LOG("(%p) Set output device info failed. This can happen when last media device is unplugged", stm); + return CUBEB_ERROR; + } + + if (audiounit_setup_stream(stm) != CUBEB_OK) { + LOG("(%p) Stream reinit failed.", stm); + if (flags & DEV_INPUT && input_device != kAudioObjectUnknown) { + // Attempt to re-use the same device-id failed, so attempt again with + // default input device. + audiounit_close_stream(stm); + if (audiounit_set_device_info(stm, kAudioObjectUnknown, io_side::INPUT) != CUBEB_OK || + audiounit_setup_stream(stm) != CUBEB_OK) { + LOG("(%p) Second stream reinit failed.", stm); + return CUBEB_ERROR; + } + } + } + + if (vol_rv == CUBEB_OK) { + audiounit_stream_set_volume(stm, volume); + } + + // If the stream was running, start it again. + if (!stm->shutdown) { + r = audiounit_stream_start_internal(stm); + if (r != CUBEB_OK) { + return CUBEB_ERROR; + } + } + } + return CUBEB_OK; +} + +static void +audiounit_reinit_stream_async(cubeb_stream * stm, device_flags_value flags) +{ + if (std::atomic_exchange(&stm->reinit_pending, true)) { + // A reinit task is already pending, nothing more to do. + ALOG("(%p) re-init stream task already pending, cancelling request", stm); + return; + } + + // Use a new thread, through the queue, to avoid deadlock when calling + // Get/SetProperties method from inside notify callback + dispatch_async(stm->context->serial_queue, ^() { + if (stm->destroy_pending) { + ALOG("(%p) stream pending destroy, cancelling reinit task", stm); + return; + } + + if (audiounit_reinit_stream(stm, flags) != CUBEB_OK) { + if (audiounit_uninstall_system_changed_callback(stm) != CUBEB_OK) { + LOG("(%p) Could not uninstall system changed callback", stm); + } + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + LOG("(%p) Could not reopen the stream after switching.", stm); + } + stm->switching_device = false; + stm->reinit_pending = false; + }); +} + +static char const * +event_addr_to_string(AudioObjectPropertySelector selector) +{ + switch(selector) { + case kAudioHardwarePropertyDefaultOutputDevice: + return "kAudioHardwarePropertyDefaultOutputDevice"; + case kAudioHardwarePropertyDefaultInputDevice: + return "kAudioHardwarePropertyDefaultInputDevice"; + case kAudioDevicePropertyDeviceIsAlive: + return "kAudioDevicePropertyDeviceIsAlive"; + case kAudioDevicePropertyDataSource: + return "kAudioDevicePropertyDataSource"; + default: + return "Unknown"; + } +} + +static OSStatus +audiounit_property_listener_callback(AudioObjectID id, UInt32 address_count, + const AudioObjectPropertyAddress * addresses, + void * user) +{ + cubeb_stream * stm = (cubeb_stream*) user; + if (stm->switching_device) { + LOG("Switching is already taking place. Skip Event %s for id=%d", event_addr_to_string(addresses[0].mSelector), id); + return noErr; + } + stm->switching_device = true; + + LOG("(%p) Audio device changed, %u events.", stm, (unsigned int) address_count); + for (UInt32 i = 0; i < address_count; i++) { + switch(addresses[i].mSelector) { + case kAudioHardwarePropertyDefaultOutputDevice: { + LOG("Event[%u] - mSelector == kAudioHardwarePropertyDefaultOutputDevice for id=%d", (unsigned int) i, id); + } + break; + case kAudioHardwarePropertyDefaultInputDevice: { + LOG("Event[%u] - mSelector == kAudioHardwarePropertyDefaultInputDevice for id=%d", (unsigned int) i, id); + } + break; + case kAudioDevicePropertyDeviceIsAlive: { + LOG("Event[%u] - mSelector == kAudioDevicePropertyDeviceIsAlive for id=%d", (unsigned int) i, id); + // If this is the default input device ignore the event, + // kAudioHardwarePropertyDefaultInputDevice will take care of the switch + if (stm->input_device.flags & DEV_SYSTEM_DEFAULT) { + LOG("It's the default input device, ignore the event"); + stm->switching_device = false; + return noErr; + } + } + break; + case kAudioDevicePropertyDataSource: { + LOG("Event[%u] - mSelector == kAudioDevicePropertyDataSource for id=%d", (unsigned int) i, id); + } + break; + default: + LOG("Event[%u] - mSelector == Unexpected Event id %d, return", (unsigned int) i, addresses[i].mSelector); + stm->switching_device = false; + return noErr; + } + } + + // Allow restart to choose the new default + device_flags_value switch_side = DEV_UNKNOWN; + if (has_input(stm)) { + switch_side |= DEV_INPUT; + } + if (has_output(stm)) { + switch_side |= DEV_OUTPUT; + } + + for (UInt32 i = 0; i < address_count; i++) { + switch(addresses[i].mSelector) { + case kAudioHardwarePropertyDefaultOutputDevice: + case kAudioHardwarePropertyDefaultInputDevice: + case kAudioDevicePropertyDeviceIsAlive: + /* fall through */ + case kAudioDevicePropertyDataSource: { + auto_lock dev_cb_lock(stm->device_changed_callback_lock); + if (stm->device_changed_callback) { + stm->device_changed_callback(stm->user_ptr); + } + break; + } + } + } + + audiounit_reinit_stream_async(stm, switch_side); + + return noErr; +} + +OSStatus +audiounit_add_listener(const property_listener * listener) +{ + assert(listener); + return AudioObjectAddPropertyListener(listener->device_id, + listener->property_address, + listener->callback, + listener->stream); +} + +OSStatus +audiounit_remove_listener(const property_listener * listener) +{ + assert(listener); + return AudioObjectRemovePropertyListener(listener->device_id, + listener->property_address, + listener->callback, + listener->stream); +} + +static int +audiounit_install_device_changed_callback(cubeb_stream * stm) +{ + OSStatus rv; + int r = CUBEB_OK; + + if (stm->output_unit) { + /* This event will notify us when the data source on the same device changes, + * for example when the user plugs in a normal (non-usb) headset in the + * headphone jack. */ + stm->output_source_listener.reset(new property_listener( + stm->output_device.id, &OUTPUT_DATA_SOURCE_PROPERTY_ADDRESS, + &audiounit_property_listener_callback, stm)); + rv = audiounit_add_listener(stm->output_source_listener.get()); + if (rv != noErr) { + stm->output_source_listener.reset(); + LOG("AudioObjectAddPropertyListener/output/kAudioDevicePropertyDataSource rv=%d, device id=%d", rv, stm->output_device.id); + r = CUBEB_ERROR; + } + } + + if (stm->input_unit) { + /* This event will notify us when the data source on the input device changes. */ + stm->input_source_listener.reset(new property_listener( + stm->input_device.id, &INPUT_DATA_SOURCE_PROPERTY_ADDRESS, + &audiounit_property_listener_callback, stm)); + rv = audiounit_add_listener(stm->input_source_listener.get()); + if (rv != noErr) { + stm->input_source_listener.reset(); + LOG("AudioObjectAddPropertyListener/input/kAudioDevicePropertyDataSource rv=%d, device id=%d", rv, stm->input_device.id); + r = CUBEB_ERROR; + } + + /* Event to notify when the input is going away. */ + stm->input_alive_listener.reset(new property_listener( + stm->input_device.id, &DEVICE_IS_ALIVE_PROPERTY_ADDRESS, + &audiounit_property_listener_callback, stm)); + rv = audiounit_add_listener(stm->input_alive_listener.get()); + if (rv != noErr) { + stm->input_alive_listener.reset(); + LOG("AudioObjectAddPropertyListener/input/kAudioDevicePropertyDeviceIsAlive rv=%d, device id =%d", rv, stm->input_device.id); + r = CUBEB_ERROR; + } + } + + return r; +} + +static int +audiounit_install_system_changed_callback(cubeb_stream * stm) +{ + OSStatus r; + + if (stm->output_unit) { + /* This event will notify us when the default audio device changes, + * for example when the user plugs in a USB headset and the system chooses it + * automatically as the default, or when another device is chosen in the + * dropdown list. */ + stm->default_output_listener.reset(new property_listener( + kAudioObjectSystemObject, &DEFAULT_OUTPUT_DEVICE_PROPERTY_ADDRESS, + &audiounit_property_listener_callback, stm)); + r = audiounit_add_listener(stm->default_output_listener.get()); + if (r != noErr) { + stm->default_output_listener.reset(); + LOG("AudioObjectAddPropertyListener/output/kAudioHardwarePropertyDefaultOutputDevice rv=%d", r); + return CUBEB_ERROR; + } + } + + if (stm->input_unit) { + /* This event will notify us when the default input device changes. */ + stm->default_input_listener.reset(new property_listener( + kAudioObjectSystemObject, &DEFAULT_INPUT_DEVICE_PROPERTY_ADDRESS, + &audiounit_property_listener_callback, stm)); + r = audiounit_add_listener(stm->default_input_listener.get()); + if (r != noErr) { + stm->default_input_listener.reset(); + LOG("AudioObjectAddPropertyListener/input/kAudioHardwarePropertyDefaultInputDevice rv=%d", r); + return CUBEB_ERROR; + } + } + + return CUBEB_OK; +} + +static int +audiounit_uninstall_device_changed_callback(cubeb_stream * stm) +{ + OSStatus rv; + // Failing to uninstall listeners is not a fatal error. + int r = CUBEB_OK; + + if (stm->output_source_listener) { + rv = audiounit_remove_listener(stm->output_source_listener.get()); + if (rv != noErr) { + LOG("AudioObjectRemovePropertyListener/output/kAudioDevicePropertyDataSource rv=%d, device id=%d", rv, stm->output_device.id); + r = CUBEB_ERROR; + } + stm->output_source_listener.reset(); + } + + if (stm->input_source_listener) { + rv = audiounit_remove_listener(stm->input_source_listener.get()); + if (rv != noErr) { + LOG("AudioObjectRemovePropertyListener/input/kAudioDevicePropertyDataSource rv=%d, device id=%d", rv, stm->input_device.id); + r = CUBEB_ERROR; + } + stm->input_source_listener.reset(); + } + + if (stm->input_alive_listener) { + rv = audiounit_remove_listener(stm->input_alive_listener.get()); + if (rv != noErr) { + LOG("AudioObjectRemovePropertyListener/input/kAudioDevicePropertyDeviceIsAlive rv=%d, device id=%d", rv, stm->input_device.id); + r = CUBEB_ERROR; + } + stm->input_alive_listener.reset(); + } + + return r; +} + +static int +audiounit_uninstall_system_changed_callback(cubeb_stream * stm) +{ + OSStatus r; + + if (stm->default_output_listener) { + r = audiounit_remove_listener(stm->default_output_listener.get()); + if (r != noErr) { + return CUBEB_ERROR; + } + stm->default_output_listener.reset(); + } + + if (stm->default_input_listener) { + r = audiounit_remove_listener(stm->default_input_listener.get()); + if (r != noErr) { + return CUBEB_ERROR; + } + stm->default_input_listener.reset(); + } + return CUBEB_OK; +} + +/* Get the acceptable buffer size (in frames) that this device can work with. */ +static int +audiounit_get_acceptable_latency_range(AudioValueRange * latency_range) +{ + UInt32 size; + OSStatus r; + AudioDeviceID output_device_id; + AudioObjectPropertyAddress output_device_buffer_size_range = { + kAudioDevicePropertyBufferFrameSizeRange, + kAudioDevicePropertyScopeOutput, + kAudioObjectPropertyElementMaster + }; + + output_device_id = audiounit_get_default_device_id(CUBEB_DEVICE_TYPE_OUTPUT); + if (output_device_id == kAudioObjectUnknown) { + LOG("Could not get default output device id."); + return CUBEB_ERROR; + } + + /* Get the buffer size range this device supports */ + size = sizeof(*latency_range); + + r = AudioObjectGetPropertyData(output_device_id, + &output_device_buffer_size_range, + 0, + NULL, + &size, + latency_range); + if (r != noErr) { + LOG("AudioObjectGetPropertyData/buffer size range rv=%d", r); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} +#endif /* !TARGET_OS_IPHONE */ + +static AudioObjectID +audiounit_get_default_device_id(cubeb_device_type type) +{ + const AudioObjectPropertyAddress * adr; + if (type == CUBEB_DEVICE_TYPE_OUTPUT) { + adr = &DEFAULT_OUTPUT_DEVICE_PROPERTY_ADDRESS; + } else if (type == CUBEB_DEVICE_TYPE_INPUT) { + adr = &DEFAULT_INPUT_DEVICE_PROPERTY_ADDRESS; + } else { + return kAudioObjectUnknown; + } + + AudioDeviceID devid; + UInt32 size = sizeof(AudioDeviceID); + if (AudioObjectGetPropertyData(kAudioObjectSystemObject, + adr, 0, NULL, &size, &devid) != noErr) { + return kAudioObjectUnknown; + } + + return devid; +} + +int +audiounit_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ +#if TARGET_OS_IPHONE + //TODO: [[AVAudioSession sharedInstance] maximumOutputNumberOfChannels] + *max_channels = 2; +#else + UInt32 size; + OSStatus r; + AudioDeviceID output_device_id; + AudioStreamBasicDescription stream_format; + AudioObjectPropertyAddress stream_format_address = { + kAudioDevicePropertyStreamFormat, + kAudioDevicePropertyScopeOutput, + kAudioObjectPropertyElementMaster + }; + + assert(ctx && max_channels); + + output_device_id = audiounit_get_default_device_id(CUBEB_DEVICE_TYPE_OUTPUT); + if (output_device_id == kAudioObjectUnknown) { + return CUBEB_ERROR; + } + + size = sizeof(stream_format); + + r = AudioObjectGetPropertyData(output_device_id, + &stream_format_address, + 0, + NULL, + &size, + &stream_format); + if (r != noErr) { + LOG("AudioObjectPropertyAddress/StreamFormat rv=%d", r); + return CUBEB_ERROR; + } + + *max_channels = stream_format.mChannelsPerFrame; +#endif + return CUBEB_OK; +} + +static int +audiounit_get_min_latency(cubeb * /* ctx */, + cubeb_stream_params /* params */, + uint32_t * latency_frames) +{ +#if TARGET_OS_IPHONE + //TODO: [[AVAudioSession sharedInstance] inputLatency] + return CUBEB_ERROR_NOT_SUPPORTED; +#else + AudioValueRange latency_range; + if (audiounit_get_acceptable_latency_range(&latency_range) != CUBEB_OK) { + LOG("Could not get acceptable latency range."); + return CUBEB_ERROR; + } + + *latency_frames = max(latency_range.mMinimum, + SAFE_MIN_LATENCY_FRAMES); +#endif + + return CUBEB_OK; +} + +static int +audiounit_get_preferred_sample_rate(cubeb * /* ctx */, uint32_t * rate) +{ +#if TARGET_OS_IPHONE + //TODO + return CUBEB_ERROR_NOT_SUPPORTED; +#else + UInt32 size; + OSStatus r; + Float64 fsamplerate; + AudioDeviceID output_device_id; + AudioObjectPropertyAddress samplerate_address = { + kAudioDevicePropertyNominalSampleRate, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster + }; + + output_device_id = audiounit_get_default_device_id(CUBEB_DEVICE_TYPE_OUTPUT); + if (output_device_id == kAudioObjectUnknown) { + return CUBEB_ERROR; + } + + size = sizeof(fsamplerate); + r = AudioObjectGetPropertyData(output_device_id, + &samplerate_address, + 0, + NULL, + &size, + &fsamplerate); + + if (r != noErr) { + return CUBEB_ERROR; + } + + *rate = static_cast(fsamplerate); +#endif + return CUBEB_OK; +} + +static cubeb_channel_layout +audiounit_convert_channel_layout(AudioChannelLayout * layout) +{ + // When having one or two channel, force mono or stereo. Some devices (namely, + // Bose QC35, mark 1 and 2), expose a single channel mapped to the right for + // some reason. + if (layout->mNumberChannelDescriptions == 1) { + return CUBEB_LAYOUT_MONO; + } else if (layout->mNumberChannelDescriptions == 2) { + return CUBEB_LAYOUT_STEREO; + } + + if (layout->mChannelLayoutTag != kAudioChannelLayoutTag_UseChannelDescriptions) { + // kAudioChannelLayoutTag_UseChannelBitmap + // kAudioChannelLayoutTag_Mono + // kAudioChannelLayoutTag_Stereo + // .... + LOG("Only handle UseChannelDescriptions for now.\n"); + return CUBEB_LAYOUT_UNDEFINED; + } + + cubeb_channel_layout cl = 0; + for (UInt32 i = 0; i < layout->mNumberChannelDescriptions; ++i) { + cubeb_channel cc = channel_label_to_cubeb_channel( + layout->mChannelDescriptions[i].mChannelLabel); + if (cc == CHANNEL_UNKNOWN) { + return CUBEB_LAYOUT_UNDEFINED; + } + cl |= cc; + } + + return cl; +} + +static cubeb_channel_layout +audiounit_get_preferred_channel_layout(AudioUnit output_unit) +{ + OSStatus rv = noErr; + UInt32 size = 0; + rv = AudioUnitGetPropertyInfo(output_unit, + kAudioDevicePropertyPreferredChannelLayout, + kAudioUnitScope_Output, + AU_OUT_BUS, + &size, + nullptr); + if (rv != noErr) { + LOG("AudioUnitGetPropertyInfo/kAudioDevicePropertyPreferredChannelLayout rv=%d", rv); + return CUBEB_LAYOUT_UNDEFINED; + } + assert(size > 0); + + auto layout = make_sized_audio_channel_layout(size); + rv = AudioUnitGetProperty(output_unit, + kAudioDevicePropertyPreferredChannelLayout, + kAudioUnitScope_Output, + AU_OUT_BUS, + layout.get(), + &size); + if (rv != noErr) { + LOG("AudioUnitGetProperty/kAudioDevicePropertyPreferredChannelLayout rv=%d", rv); + return CUBEB_LAYOUT_UNDEFINED; + } + + return audiounit_convert_channel_layout(layout.get()); +} + +static cubeb_channel_layout +audiounit_get_current_channel_layout(AudioUnit output_unit) +{ + OSStatus rv = noErr; + UInt32 size = 0; + rv = AudioUnitGetPropertyInfo(output_unit, + kAudioUnitProperty_AudioChannelLayout, + kAudioUnitScope_Output, + AU_OUT_BUS, + &size, + nullptr); + if (rv != noErr) { + LOG("AudioUnitGetPropertyInfo/kAudioUnitProperty_AudioChannelLayout rv=%d", rv); + // This property isn't known before macOS 10.12, attempt another method. + return audiounit_get_preferred_channel_layout(output_unit); + } + assert(size > 0); + + auto layout = make_sized_audio_channel_layout(size); + rv = AudioUnitGetProperty(output_unit, + kAudioUnitProperty_AudioChannelLayout, + kAudioUnitScope_Output, + AU_OUT_BUS, + layout.get(), + &size); + if (rv != noErr) { + LOG("AudioUnitGetProperty/kAudioUnitProperty_AudioChannelLayout rv=%d", rv); + return CUBEB_LAYOUT_UNDEFINED; + } + + return audiounit_convert_channel_layout(layout.get()); +} + +static int audiounit_create_unit(AudioUnit * unit, device_info * device); + +static OSStatus audiounit_remove_device_listener(cubeb * context, cubeb_device_type devtype); + +static void +audiounit_destroy(cubeb * ctx) +{ + { + auto_lock lock(ctx->mutex); + + // Disabling this assert for bug 1083664 -- we seem to leak a stream + // assert(ctx->active_streams == 0); + if (audiounit_active_streams(ctx) > 0) { + LOG("(%p) API misuse, %d streams active when context destroyed!", ctx, audiounit_active_streams(ctx)); + } + + /* Unregister the callback if necessary. */ + if (ctx->input_collection_changed_callback) { + audiounit_remove_device_listener(ctx, CUBEB_DEVICE_TYPE_INPUT); + } + if (ctx->output_collection_changed_callback) { + audiounit_remove_device_listener(ctx, CUBEB_DEVICE_TYPE_OUTPUT); + } + } + + dispatch_release(ctx->serial_queue); + + delete ctx; +} + +static void audiounit_stream_destroy(cubeb_stream * stm); + +static int +audio_stream_desc_init(AudioStreamBasicDescription * ss, + const cubeb_stream_params * stream_params) +{ + switch (stream_params->format) { + case CUBEB_SAMPLE_S16LE: + ss->mBitsPerChannel = 16; + ss->mFormatFlags = kAudioFormatFlagIsSignedInteger; + break; + case CUBEB_SAMPLE_S16BE: + ss->mBitsPerChannel = 16; + ss->mFormatFlags = kAudioFormatFlagIsSignedInteger | + kAudioFormatFlagIsBigEndian; + break; + case CUBEB_SAMPLE_FLOAT32LE: + ss->mBitsPerChannel = 32; + ss->mFormatFlags = kAudioFormatFlagIsFloat; + break; + case CUBEB_SAMPLE_FLOAT32BE: + ss->mBitsPerChannel = 32; + ss->mFormatFlags = kAudioFormatFlagIsFloat | + kAudioFormatFlagIsBigEndian; + break; + default: + return CUBEB_ERROR_INVALID_FORMAT; + } + + ss->mFormatID = kAudioFormatLinearPCM; + ss->mFormatFlags |= kLinearPCMFormatFlagIsPacked; + ss->mSampleRate = stream_params->rate; + ss->mChannelsPerFrame = stream_params->channels; + + ss->mBytesPerFrame = (ss->mBitsPerChannel / 8) * ss->mChannelsPerFrame; + ss->mFramesPerPacket = 1; + ss->mBytesPerPacket = ss->mBytesPerFrame * ss->mFramesPerPacket; + + ss->mReserved = 0; + + return CUBEB_OK; +} + +void +audiounit_init_mixer(cubeb_stream * stm) +{ + // We can't rely on macOS' AudioUnit to properly downmix (or upmix) the audio + // data, it silently drop the channels so we need to remix the + // audio data by ourselves to keep all the information. + stm->mixer.reset(cubeb_mixer_create(stm->output_stream_params.format, + stm->output_stream_params.channels, + stm->output_stream_params.layout, + stm->context->channels, + stm->context->layout)); + assert(stm->mixer); +} + +static int +audiounit_set_channel_layout(AudioUnit unit, + io_side side, + cubeb_channel_layout layout) +{ + if (side != io_side::OUTPUT) { + return CUBEB_ERROR; + } + + if (layout == CUBEB_LAYOUT_UNDEFINED) { + // We leave everything as-is... + return CUBEB_OK; + } + + + OSStatus r; + uint32_t nb_channels = cubeb_channel_layout_nb_channels(layout); + + // We do not use CoreAudio standard layout for lack of documentation on what + // the actual channel orders are. So we set a custom layout. + size_t size = offsetof(AudioChannelLayout, mChannelDescriptions[nb_channels]); + auto au_layout = make_sized_audio_channel_layout(size); + au_layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions; + au_layout->mNumberChannelDescriptions = nb_channels; + + uint32_t channels = 0; + cubeb_channel_layout channelMap = layout; + for (uint32_t i = 0; channelMap != 0; ++i) { + XASSERT(channels < nb_channels); + uint32_t channel = (channelMap & 1) << i; + if (channel != 0) { + au_layout->mChannelDescriptions[channels].mChannelLabel = + cubeb_channel_to_channel_label(static_cast(channel)); + au_layout->mChannelDescriptions[channels].mChannelFlags = kAudioChannelFlags_AllOff; + channels++; + } + channelMap = channelMap >> 1; + } + + r = AudioUnitSetProperty(unit, + kAudioUnitProperty_AudioChannelLayout, + kAudioUnitScope_Input, + AU_OUT_BUS, + au_layout.get(), + size); + if (r != noErr) { + LOG("AudioUnitSetProperty/%s/kAudioUnitProperty_AudioChannelLayout rv=%d", to_string(side), r); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +void +audiounit_layout_init(cubeb_stream * stm, io_side side) +{ + // We currently don't support the input layout setting. + if (side == io_side::INPUT) { + return; + } + + stm->context->layout = audiounit_get_current_channel_layout(stm->output_unit); + + audiounit_set_channel_layout(stm->output_unit, io_side::OUTPUT, stm->context->layout); +} + +static vector +audiounit_get_sub_devices(AudioDeviceID device_id) +{ + vector sub_devices; + AudioObjectPropertyAddress property_address = { kAudioAggregateDevicePropertyActiveSubDeviceList, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + UInt32 size = 0; + OSStatus rv = AudioObjectGetPropertyDataSize(device_id, + &property_address, + 0, + nullptr, + &size); + + if (rv != noErr) { + sub_devices.push_back(device_id); + return sub_devices; + } + + uint32_t count = static_cast(size / sizeof(AudioObjectID)); + sub_devices.resize(count); + rv = AudioObjectGetPropertyData(device_id, + &property_address, + 0, + nullptr, + &size, + sub_devices.data()); + if (rv != noErr) { + sub_devices.clear(); + sub_devices.push_back(device_id); + } else { + LOG("Found %u sub-devices", count); + } + return sub_devices; +} + +static int +audiounit_create_blank_aggregate_device(AudioObjectID * plugin_id, AudioDeviceID * aggregate_device_id) +{ + AudioObjectPropertyAddress address_plugin_bundle_id = { kAudioHardwarePropertyPlugInForBundleID, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + UInt32 size = 0; + OSStatus r = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, + &address_plugin_bundle_id, + 0, NULL, + &size); + if (r != noErr) { + LOG("AudioObjectGetPropertyDataSize/kAudioHardwarePropertyPlugInForBundleID, rv=%d", r); + return CUBEB_ERROR; + } + + AudioValueTranslation translation_value; + CFStringRef in_bundle_ref = CFSTR("com.apple.audio.CoreAudio"); + translation_value.mInputData = &in_bundle_ref; + translation_value.mInputDataSize = sizeof(in_bundle_ref); + translation_value.mOutputData = plugin_id; + translation_value.mOutputDataSize = sizeof(*plugin_id); + + r = AudioObjectGetPropertyData(kAudioObjectSystemObject, + &address_plugin_bundle_id, + 0, + nullptr, + &size, + &translation_value); + if (r != noErr) { + LOG("AudioObjectGetPropertyData/kAudioHardwarePropertyPlugInForBundleID, rv=%d", r); + return CUBEB_ERROR; + } + + AudioObjectPropertyAddress create_aggregate_device_address = { kAudioPlugInCreateAggregateDevice, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + r = AudioObjectGetPropertyDataSize(*plugin_id, + &create_aggregate_device_address, + 0, + nullptr, + &size); + if (r != noErr) { + LOG("AudioObjectGetPropertyDataSize/kAudioPlugInCreateAggregateDevice, rv=%d", r); + return CUBEB_ERROR; + } + + CFMutableDictionaryRef aggregate_device_dict = CFDictionaryCreateMutable(kCFAllocatorDefault, 0, + &kCFTypeDictionaryKeyCallBacks, + &kCFTypeDictionaryValueCallBacks); + struct timeval timestamp; + gettimeofday(×tamp, NULL); + long long int time_id = timestamp.tv_sec * 1000000LL + timestamp.tv_usec; + CFStringRef aggregate_device_name = CFStringCreateWithFormat(NULL, NULL, CFSTR("%s_%llx"), PRIVATE_AGGREGATE_DEVICE_NAME, time_id); + CFDictionaryAddValue(aggregate_device_dict, CFSTR(kAudioAggregateDeviceNameKey), aggregate_device_name); + CFRelease(aggregate_device_name); + + CFStringRef aggregate_device_UID = CFStringCreateWithFormat(NULL, NULL, CFSTR("org.mozilla.%s_%llx"), PRIVATE_AGGREGATE_DEVICE_NAME, time_id); + CFDictionaryAddValue(aggregate_device_dict, CFSTR(kAudioAggregateDeviceUIDKey), aggregate_device_UID); + CFRelease(aggregate_device_UID); + + int private_value = 1; + CFNumberRef aggregate_device_private_key = CFNumberCreate(kCFAllocatorDefault, kCFNumberIntType, &private_value); + CFDictionaryAddValue(aggregate_device_dict, CFSTR(kAudioAggregateDeviceIsPrivateKey), aggregate_device_private_key); + CFRelease(aggregate_device_private_key); + + int stacked_value = 0; + CFNumberRef aggregate_device_stacked_key = CFNumberCreate(kCFAllocatorDefault, kCFNumberIntType, &stacked_value); + CFDictionaryAddValue(aggregate_device_dict, CFSTR(kAudioAggregateDeviceIsStackedKey), aggregate_device_stacked_key); + CFRelease(aggregate_device_stacked_key); + + r = AudioObjectGetPropertyData(*plugin_id, + &create_aggregate_device_address, + sizeof(aggregate_device_dict), + &aggregate_device_dict, + &size, + aggregate_device_id); + CFRelease(aggregate_device_dict); + if (r != noErr) { + LOG("AudioObjectGetPropertyData/kAudioPlugInCreateAggregateDevice, rv=%d", r); + return CUBEB_ERROR; + } + LOG("New aggregate device %u", *aggregate_device_id); + + return CUBEB_OK; +} + +// The returned CFStringRef object needs to be released (via CFRelease) +// if it's not NULL, since the reference count of the returned CFStringRef +// object is increased. +static CFStringRef +get_device_name(AudioDeviceID id) +{ + UInt32 size = sizeof(CFStringRef); + CFStringRef UIname = nullptr; + AudioObjectPropertyAddress address_uuid = { kAudioDevicePropertyDeviceUID, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus err = AudioObjectGetPropertyData(id, &address_uuid, 0, nullptr, &size, &UIname); + return (err == noErr) ? UIname : NULL; +} + +static int +audiounit_set_aggregate_sub_device_list(AudioDeviceID aggregate_device_id, + AudioDeviceID input_device_id, + AudioDeviceID output_device_id) +{ + LOG("Add devices input %u and output %u into aggregate device %u", + input_device_id, output_device_id, aggregate_device_id); + const vector output_sub_devices = audiounit_get_sub_devices(output_device_id); + const vector input_sub_devices = audiounit_get_sub_devices(input_device_id); + + CFMutableArrayRef aggregate_sub_devices_array = CFArrayCreateMutable(NULL, 0, &kCFTypeArrayCallBacks); + /* The order of the items in the array is significant and is used to determine the order of the streams + of the AudioAggregateDevice. */ + for (UInt32 i = 0; i < output_sub_devices.size(); i++) { + CFStringRef ref = get_device_name(output_sub_devices[i]); + if (ref == NULL) { + CFRelease(aggregate_sub_devices_array); + return CUBEB_ERROR; + } + CFArrayAppendValue(aggregate_sub_devices_array, ref); + CFRelease(ref); + } + for (UInt32 i = 0; i < input_sub_devices.size(); i++) { + CFStringRef ref = get_device_name(input_sub_devices[i]); + if (ref == NULL) { + CFRelease(aggregate_sub_devices_array); + return CUBEB_ERROR; + } + CFArrayAppendValue(aggregate_sub_devices_array, ref); + CFRelease(ref); + } + + AudioObjectPropertyAddress aggregate_sub_device_list = { kAudioAggregateDevicePropertyFullSubDeviceList, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + UInt32 size = sizeof(CFMutableArrayRef); + OSStatus rv = AudioObjectSetPropertyData(aggregate_device_id, + &aggregate_sub_device_list, + 0, + nullptr, + size, + &aggregate_sub_devices_array); + CFRelease(aggregate_sub_devices_array); + if (rv != noErr) { + LOG("AudioObjectSetPropertyData/kAudioAggregateDevicePropertyFullSubDeviceList, rv=%d", rv); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +static int +audiounit_set_master_aggregate_device(const AudioDeviceID aggregate_device_id) +{ + assert(aggregate_device_id != kAudioObjectUnknown); + AudioObjectPropertyAddress master_aggregate_sub_device = { kAudioAggregateDevicePropertyMasterSubDevice, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + // Master become the 1st output sub device + AudioDeviceID output_device_id = audiounit_get_default_device_id(CUBEB_DEVICE_TYPE_OUTPUT); + const vector output_sub_devices = audiounit_get_sub_devices(output_device_id); + CFStringRef master_sub_device = get_device_name(output_sub_devices[0]); + + UInt32 size = sizeof(CFStringRef); + OSStatus rv = AudioObjectSetPropertyData(aggregate_device_id, + &master_aggregate_sub_device, + 0, + NULL, + size, + &master_sub_device); + if (master_sub_device) { + CFRelease(master_sub_device); + } + if (rv != noErr) { + LOG("AudioObjectSetPropertyData/kAudioAggregateDevicePropertyMasterSubDevice, rv=%d", rv); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +static int +audiounit_activate_clock_drift_compensation(const AudioDeviceID aggregate_device_id) +{ + assert(aggregate_device_id != kAudioObjectUnknown); + AudioObjectPropertyAddress address_owned = { kAudioObjectPropertyOwnedObjects, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + UInt32 qualifier_data_size = sizeof(AudioObjectID); + AudioClassID class_id = kAudioSubDeviceClassID; + void * qualifier_data = &class_id; + UInt32 size = 0; + OSStatus rv = AudioObjectGetPropertyDataSize(aggregate_device_id, + &address_owned, + qualifier_data_size, + qualifier_data, + &size); + if (rv != noErr) { + LOG("AudioObjectGetPropertyDataSize/kAudioObjectPropertyOwnedObjects, rv=%d", rv); + return CUBEB_ERROR; + } + + UInt32 subdevices_num = 0; + subdevices_num = size / sizeof(AudioObjectID); + AudioObjectID sub_devices[subdevices_num]; + size = sizeof(sub_devices); + + rv = AudioObjectGetPropertyData(aggregate_device_id, + &address_owned, + qualifier_data_size, + qualifier_data, + &size, + sub_devices); + if (rv != noErr) { + LOG("AudioObjectGetPropertyData/kAudioObjectPropertyOwnedObjects, rv=%d", rv); + return CUBEB_ERROR; + } + + AudioObjectPropertyAddress address_drift = { kAudioSubDevicePropertyDriftCompensation, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + // Start from the second device since the first is the master clock + for (UInt32 i = 1; i < subdevices_num; ++i) { + UInt32 drift_compensation_value = 1; + rv = AudioObjectSetPropertyData(sub_devices[i], + &address_drift, + 0, + nullptr, + sizeof(UInt32), + &drift_compensation_value); + if (rv != noErr) { + LOG("AudioObjectSetPropertyData/kAudioSubDevicePropertyDriftCompensation, rv=%d", rv); + return CUBEB_OK; + } + } + return CUBEB_OK; +} + +static int audiounit_destroy_aggregate_device(AudioObjectID plugin_id, AudioDeviceID * aggregate_device_id); +static void audiounit_get_available_samplerate(AudioObjectID devid, AudioObjectPropertyScope scope, + uint32_t * min, uint32_t * max, uint32_t * def); +static int +audiounit_create_device_from_hwdev(cubeb_device_info * dev_info, AudioObjectID devid, cubeb_device_type type); +static void audiounit_device_destroy(cubeb_device_info * device); + +static void +audiounit_workaround_for_airpod(cubeb_stream * stm) +{ + cubeb_device_info input_device_info; + audiounit_create_device_from_hwdev(&input_device_info, stm->input_device.id, CUBEB_DEVICE_TYPE_INPUT); + + cubeb_device_info output_device_info; + audiounit_create_device_from_hwdev(&output_device_info, stm->output_device.id, CUBEB_DEVICE_TYPE_OUTPUT); + + std::string input_name_str(input_device_info.friendly_name); + std::string output_name_str(output_device_info.friendly_name); + + if(input_name_str.find("AirPods") != std::string::npos && + output_name_str.find("AirPods") != std::string::npos) { + uint32_t input_min_rate = 0; + uint32_t input_max_rate = 0; + uint32_t input_nominal_rate = 0; + audiounit_get_available_samplerate(stm->input_device.id, kAudioObjectPropertyScopeGlobal, + &input_min_rate, &input_max_rate, &input_nominal_rate); + LOG("(%p) Input device %u, name: %s, min: %u, max: %u, nominal rate: %u", stm, stm->input_device.id + , input_device_info.friendly_name, input_min_rate, input_max_rate, input_nominal_rate); + uint32_t output_min_rate = 0; + uint32_t output_max_rate = 0; + uint32_t output_nominal_rate = 0; + audiounit_get_available_samplerate(stm->output_device.id, kAudioObjectPropertyScopeGlobal, + &output_min_rate, &output_max_rate, &output_nominal_rate); + LOG("(%p) Output device %u, name: %s, min: %u, max: %u, nominal rate: %u", stm, stm->output_device.id + , output_device_info.friendly_name, output_min_rate, output_max_rate, output_nominal_rate); + + Float64 rate = input_nominal_rate; + AudioObjectPropertyAddress addr = {kAudioDevicePropertyNominalSampleRate, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster}; + + OSStatus rv = AudioObjectSetPropertyData(stm->aggregate_device_id, + &addr, + 0, + nullptr, + sizeof(Float64), + &rate); + if (rv != noErr) { + LOG("Non fatal error, AudioObjectSetPropertyData/kAudioDevicePropertyNominalSampleRate, rv=%d", rv); + } + } + audiounit_device_destroy(&input_device_info); + audiounit_device_destroy(&output_device_info); +} + +/* + * Aggregate Device is a virtual audio interface which utilizes inputs and outputs + * of one or more physical audio interfaces. It is possible to use the clock of + * one of the devices as a master clock for all the combined devices and enable + * drift compensation for the devices that are not designated clock master. + * + * Creating a new aggregate device programmatically requires [0][1]: + * 1. Locate the base plug-in ("com.apple.audio.CoreAudio") + * 2. Create a dictionary that describes the aggregate device + * (don't add sub-devices in that step, prone to fail [0]) + * 3. Ask the base plug-in to create the aggregate device (blank) + * 4. Add the array of sub-devices. + * 5. Set the master device (1st output device in our case) + * 6. Enable drift compensation for the non-master devices + * + * [0] https://lists.apple.com/archives/coreaudio-api/2006/Apr/msg00092.html + * [1] https://lists.apple.com/archives/coreaudio-api/2005/Jul/msg00150.html + * [2] CoreAudio.framework/Headers/AudioHardware.h + * */ +static int +audiounit_create_aggregate_device(cubeb_stream * stm) +{ + int r = audiounit_create_blank_aggregate_device(&stm->plugin_id, &stm->aggregate_device_id); + if (r != CUBEB_OK) { + LOG("(%p) Failed to create blank aggregate device", stm); + return CUBEB_ERROR; + } + + r = audiounit_set_aggregate_sub_device_list(stm->aggregate_device_id, stm->input_device.id, stm->output_device.id); + if (r != CUBEB_OK) { + LOG("(%p) Failed to set aggregate sub-device list", stm); + audiounit_destroy_aggregate_device(stm->plugin_id, &stm->aggregate_device_id); + return CUBEB_ERROR; + } + + r = audiounit_set_master_aggregate_device(stm->aggregate_device_id); + if (r != CUBEB_OK) { + LOG("(%p) Failed to set master sub-device for aggregate device", stm); + audiounit_destroy_aggregate_device(stm->plugin_id, &stm->aggregate_device_id); + return CUBEB_ERROR; + } + + r = audiounit_activate_clock_drift_compensation(stm->aggregate_device_id); + if (r != CUBEB_OK) { + LOG("(%p) Failed to activate clock drift compensation for aggregate device", stm); + audiounit_destroy_aggregate_device(stm->plugin_id, &stm->aggregate_device_id); + return CUBEB_ERROR; + } + + audiounit_workaround_for_airpod(stm); + + return CUBEB_OK; +} + +static int +audiounit_destroy_aggregate_device(AudioObjectID plugin_id, AudioDeviceID * aggregate_device_id) +{ + assert(aggregate_device_id && + *aggregate_device_id != kAudioDeviceUnknown && + plugin_id != kAudioObjectUnknown); + AudioObjectPropertyAddress destroy_aggregate_device_addr = { kAudioPlugInDestroyAggregateDevice, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster}; + UInt32 size; + OSStatus rv = AudioObjectGetPropertyDataSize(plugin_id, + &destroy_aggregate_device_addr, + 0, + NULL, + &size); + if (rv != noErr) { + LOG("AudioObjectGetPropertyDataSize/kAudioPlugInDestroyAggregateDevice, rv=%d", rv); + return CUBEB_ERROR; + } + + rv = AudioObjectGetPropertyData(plugin_id, + &destroy_aggregate_device_addr, + 0, + NULL, + &size, + aggregate_device_id); + if (rv != noErr) { + LOG("AudioObjectGetPropertyData/kAudioPlugInDestroyAggregateDevice, rv=%d", rv); + return CUBEB_ERROR; + } + + LOG("Destroyed aggregate device %d", *aggregate_device_id); + *aggregate_device_id = kAudioObjectUnknown; + return CUBEB_OK; +} + +static int +audiounit_new_unit_instance(AudioUnit * unit, device_info * device) +{ + AudioComponentDescription desc; + AudioComponent comp; + OSStatus rv; + + desc.componentType = kAudioUnitType_Output; +#if TARGET_OS_IPHONE + desc.componentSubType = kAudioUnitSubType_RemoteIO; +#else + // Use the DefaultOutputUnit for output when no device is specified + // so we retain automatic output device switching when the default + // changes. Once we have complete support for device notifications + // and switching, we can use the AUHAL for everything. + if ((device->flags & DEV_SYSTEM_DEFAULT) && + (device->flags & DEV_OUTPUT)) { + desc.componentSubType = kAudioUnitSubType_DefaultOutput; + } else { + desc.componentSubType = kAudioUnitSubType_HALOutput; + } +#endif + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + comp = AudioComponentFindNext(NULL, &desc); + if (comp == NULL) { + LOG("Could not find matching audio hardware."); + return CUBEB_ERROR; + } + + rv = AudioComponentInstanceNew(comp, unit); + if (rv != noErr) { + LOG("AudioComponentInstanceNew rv=%d", rv); + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +enum enable_state { + DISABLE, + ENABLE, +}; + +static int +audiounit_enable_unit_scope(AudioUnit * unit, io_side side, enable_state state) +{ + OSStatus rv; + UInt32 enable = state; + rv = AudioUnitSetProperty(*unit, kAudioOutputUnitProperty_EnableIO, + (side == io_side::INPUT) ? kAudioUnitScope_Input : kAudioUnitScope_Output, + (side == io_side::INPUT) ? AU_IN_BUS : AU_OUT_BUS, + &enable, + sizeof(UInt32)); + if (rv != noErr) { + LOG("AudioUnitSetProperty/kAudioOutputUnitProperty_EnableIO rv=%d", rv); + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +static int +audiounit_create_unit(AudioUnit * unit, device_info * device) +{ + assert(*unit == nullptr); + assert(device); + + OSStatus rv; + int r; + + r = audiounit_new_unit_instance(unit, device); + if (r != CUBEB_OK) { + return r; + } + assert(*unit); + + if ((device->flags & DEV_SYSTEM_DEFAULT) && + (device->flags & DEV_OUTPUT)) { + return CUBEB_OK; + } + + + if (device->flags & DEV_INPUT) { + r = audiounit_enable_unit_scope(unit, io_side::INPUT, ENABLE); + if (r != CUBEB_OK) { + LOG("Failed to enable audiounit input scope"); + return r; + } + r = audiounit_enable_unit_scope(unit, io_side::OUTPUT, DISABLE); + if (r != CUBEB_OK) { + LOG("Failed to disable audiounit output scope"); + return r; + } + } else if (device->flags & DEV_OUTPUT) { + r = audiounit_enable_unit_scope(unit, io_side::OUTPUT, ENABLE); + if (r != CUBEB_OK) { + LOG("Failed to enable audiounit output scope"); + return r; + } + r = audiounit_enable_unit_scope(unit, io_side::INPUT, DISABLE); + if (r != CUBEB_OK) { + LOG("Failed to disable audiounit input scope"); + return r; + } + } else { + assert(false); + } + + rv = AudioUnitSetProperty(*unit, + kAudioOutputUnitProperty_CurrentDevice, + kAudioUnitScope_Global, + 0, + &device->id, sizeof(AudioDeviceID)); + if (rv != noErr) { + LOG("AudioUnitSetProperty/kAudioOutputUnitProperty_CurrentDevice rv=%d", rv); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +static int +audiounit_init_input_linear_buffer(cubeb_stream * stream, uint32_t capacity) +{ + uint32_t size = capacity * stream->latency_frames * stream->input_desc.mChannelsPerFrame; + if (stream->input_desc.mFormatFlags & kAudioFormatFlagIsSignedInteger) { + stream->input_linear_buffer.reset(new auto_array_wrapper_impl(size)); + } else { + stream->input_linear_buffer.reset(new auto_array_wrapper_impl(size)); + } + assert(stream->input_linear_buffer->length() == 0); + + return CUBEB_OK; +} + +static uint32_t +audiounit_clamp_latency(cubeb_stream * stm, uint32_t latency_frames) +{ + // For the 1st stream set anything within safe min-max + assert(audiounit_active_streams(stm->context) > 0); + if (audiounit_active_streams(stm->context) == 1) { + return max(min(latency_frames, SAFE_MAX_LATENCY_FRAMES), + SAFE_MIN_LATENCY_FRAMES); + } + assert(stm->output_unit); + + // If more than one stream operates in parallel + // allow only lower values of latency + int r; + UInt32 output_buffer_size = 0; + UInt32 size = sizeof(output_buffer_size); + if (stm->output_unit) { + r = AudioUnitGetProperty(stm->output_unit, + kAudioDevicePropertyBufferFrameSize, + kAudioUnitScope_Output, + AU_OUT_BUS, + &output_buffer_size, + &size); + if (r != noErr) { + LOG("AudioUnitGetProperty/output/kAudioDevicePropertyBufferFrameSize rv=%d", r); + return 0; + } + + output_buffer_size = max(min(output_buffer_size, SAFE_MAX_LATENCY_FRAMES), + SAFE_MIN_LATENCY_FRAMES); + } + + UInt32 input_buffer_size = 0; + if (stm->input_unit) { + r = AudioUnitGetProperty(stm->input_unit, + kAudioDevicePropertyBufferFrameSize, + kAudioUnitScope_Input, + AU_IN_BUS, + &input_buffer_size, + &size); + if (r != noErr) { + LOG("AudioUnitGetProperty/input/kAudioDevicePropertyBufferFrameSize rv=%d", r); + return 0; + } + + input_buffer_size = max(min(input_buffer_size, SAFE_MAX_LATENCY_FRAMES), + SAFE_MIN_LATENCY_FRAMES); + } + + // Every following active streams can only set smaller latency + UInt32 upper_latency_limit = 0; + if (input_buffer_size != 0 && output_buffer_size != 0) { + upper_latency_limit = min(input_buffer_size, output_buffer_size); + } else if (input_buffer_size != 0) { + upper_latency_limit = input_buffer_size; + } else if (output_buffer_size != 0) { + upper_latency_limit = output_buffer_size; + } else { + upper_latency_limit = SAFE_MAX_LATENCY_FRAMES; + } + + return max(min(latency_frames, upper_latency_limit), + SAFE_MIN_LATENCY_FRAMES); +} + +/* + * Change buffer size is prone to deadlock thus we change it + * following the steps: + * - register a listener for the buffer size property + * - change the property + * - wait until the listener is executed + * - property has changed, remove the listener + * */ +static void +buffer_size_changed_callback(void * inClientData, + AudioUnit inUnit, + AudioUnitPropertyID inPropertyID, + AudioUnitScope inScope, + AudioUnitElement inElement) +{ + cubeb_stream * stm = (cubeb_stream *)inClientData; + + AudioUnit au = inUnit; + AudioUnitScope au_scope = kAudioUnitScope_Input; + AudioUnitElement au_element = inElement; + char const * au_type = "output"; + + if (AU_IN_BUS == inElement) { + au_scope = kAudioUnitScope_Output; + au_type = "input"; + } + + switch (inPropertyID) { + + case kAudioDevicePropertyBufferFrameSize: { + if (inScope != au_scope) { + break; + } + UInt32 new_buffer_size; + UInt32 outSize = sizeof(UInt32); + OSStatus r = AudioUnitGetProperty(au, + kAudioDevicePropertyBufferFrameSize, + au_scope, + au_element, + &new_buffer_size, + &outSize); + if (r != noErr) { + LOG("(%p) Event: kAudioDevicePropertyBufferFrameSize: Cannot get current buffer size", stm); + } else { + LOG("(%p) Event: kAudioDevicePropertyBufferFrameSize: New %s buffer size = %d for scope %d", stm, + au_type, new_buffer_size, inScope); + } + stm->buffer_size_change_state = true; + break; + } + } +} + +static int +audiounit_set_buffer_size(cubeb_stream * stm, uint32_t new_size_frames, io_side side) +{ + AudioUnit au = stm->output_unit; + AudioUnitScope au_scope = kAudioUnitScope_Input; + AudioUnitElement au_element = AU_OUT_BUS; + + if (side == io_side::INPUT) { + au = stm->input_unit; + au_scope = kAudioUnitScope_Output; + au_element = AU_IN_BUS; + } + + uint32_t buffer_frames = 0; + UInt32 size = sizeof(buffer_frames); + int r = AudioUnitGetProperty(au, + kAudioDevicePropertyBufferFrameSize, + au_scope, + au_element, + &buffer_frames, + &size); + if (r != noErr) { + LOG("AudioUnitGetProperty/%s/kAudioDevicePropertyBufferFrameSize rv=%d", to_string(side), r); + return CUBEB_ERROR; + } + + if (new_size_frames == buffer_frames) { + LOG("(%p) No need to update %s buffer size already %u frames", stm, to_string(side), buffer_frames); + return CUBEB_OK; + } + + r = AudioUnitAddPropertyListener(au, + kAudioDevicePropertyBufferFrameSize, + buffer_size_changed_callback, + stm); + if (r != noErr) { + LOG("AudioUnitAddPropertyListener/%s/kAudioDevicePropertyBufferFrameSize rv=%d", to_string(side), r); + return CUBEB_ERROR; + } + + stm->buffer_size_change_state = false; + + r = AudioUnitSetProperty(au, + kAudioDevicePropertyBufferFrameSize, + au_scope, + au_element, + &new_size_frames, + sizeof(new_size_frames)); + if (r != noErr) { + LOG("AudioUnitSetProperty/%s/kAudioDevicePropertyBufferFrameSize rv=%d", to_string(side), r); + + r = AudioUnitRemovePropertyListenerWithUserData(au, + kAudioDevicePropertyBufferFrameSize, + buffer_size_changed_callback, + stm); + if (r != noErr) { + LOG("AudioUnitAddPropertyListener/%s/kAudioDevicePropertyBufferFrameSize rv=%d", to_string(side), r); + } + + return CUBEB_ERROR; + } + + int count = 0; + while (!stm->buffer_size_change_state && count++ < 30) { + struct timespec req, rem; + req.tv_sec = 0; + req.tv_nsec = 100000000L; // 0.1 sec + if (nanosleep(&req , &rem) < 0 ) { + LOG("(%p) Warning: nanosleep call failed or interrupted. Remaining time %ld nano secs \n", stm, rem.tv_nsec); + } + LOG("(%p) audiounit_set_buffer_size : wait count = %d", stm, count); + } + + r = AudioUnitRemovePropertyListenerWithUserData(au, + kAudioDevicePropertyBufferFrameSize, + buffer_size_changed_callback, + stm); + if (r != noErr) { + LOG("AudioUnitAddPropertyListener/%s/kAudioDevicePropertyBufferFrameSize rv=%d", to_string(side), r); + return CUBEB_ERROR; + } + + if (!stm->buffer_size_change_state && count >= 30) { + LOG("(%p) Error, did not get buffer size change callback ...", stm); + return CUBEB_ERROR; + } + + LOG("(%p) %s buffer size changed to %u frames.", stm, to_string(side), new_size_frames); + return CUBEB_OK; +} + +static int +audiounit_configure_input(cubeb_stream * stm) +{ + assert(stm && stm->input_unit); + + int r = 0; + UInt32 size; + AURenderCallbackStruct aurcbs_in; + + LOG("(%p) Opening input side: rate %u, channels %u, format %d, latency in frames %u.", + stm, stm->input_stream_params.rate, stm->input_stream_params.channels, + stm->input_stream_params.format, stm->latency_frames); + + /* Get input device sample rate. */ + AudioStreamBasicDescription input_hw_desc; + size = sizeof(AudioStreamBasicDescription); + r = AudioUnitGetProperty(stm->input_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + AU_IN_BUS, + &input_hw_desc, + &size); + if (r != noErr) { + LOG("AudioUnitGetProperty/input/kAudioUnitProperty_StreamFormat rv=%d", r); + return CUBEB_ERROR; + } + stm->input_hw_rate = input_hw_desc.mSampleRate; + LOG("(%p) Input device sampling rate: %.2f", stm, stm->input_hw_rate); + + /* Set format description according to the input params. */ + r = audio_stream_desc_init(&stm->input_desc, &stm->input_stream_params); + if (r != CUBEB_OK) { + LOG("(%p) Setting format description for input failed.", stm); + return r; + } + + // Use latency to set buffer size + r = audiounit_set_buffer_size(stm, stm->latency_frames, io_side::INPUT); + if (r != CUBEB_OK) { + LOG("(%p) Error in change input buffer size.", stm); + return CUBEB_ERROR; + } + + AudioStreamBasicDescription src_desc = stm->input_desc; + /* Input AudioUnit must be configured with device's sample rate. + we will resample inside input callback. */ + src_desc.mSampleRate = stm->input_hw_rate; + + r = AudioUnitSetProperty(stm->input_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, + AU_IN_BUS, + &src_desc, + sizeof(AudioStreamBasicDescription)); + if (r != noErr) { + LOG("AudioUnitSetProperty/input/kAudioUnitProperty_StreamFormat rv=%d", r); + return CUBEB_ERROR; + } + + /* Frames per buffer in the input callback. */ + r = AudioUnitSetProperty(stm->input_unit, + kAudioUnitProperty_MaximumFramesPerSlice, + kAudioUnitScope_Global, + AU_IN_BUS, + &stm->latency_frames, + sizeof(UInt32)); + if (r != noErr) { + LOG("AudioUnitSetProperty/input/kAudioUnitProperty_MaximumFramesPerSlice rv=%d", r); + return CUBEB_ERROR; + } + + // Input only capacity + unsigned int array_capacity = 1; + if (has_output(stm)) { + // Full-duplex increase capacity + array_capacity = 8; + } + if (audiounit_init_input_linear_buffer(stm, array_capacity) != CUBEB_OK) { + return CUBEB_ERROR; + } + + aurcbs_in.inputProc = audiounit_input_callback; + aurcbs_in.inputProcRefCon = stm; + + r = AudioUnitSetProperty(stm->input_unit, + kAudioOutputUnitProperty_SetInputCallback, + kAudioUnitScope_Global, + AU_OUT_BUS, + &aurcbs_in, + sizeof(aurcbs_in)); + if (r != noErr) { + LOG("AudioUnitSetProperty/input/kAudioOutputUnitProperty_SetInputCallback rv=%d", r); + return CUBEB_ERROR; + } + + stm->frames_read = 0; + + LOG("(%p) Input audiounit init successfully.", stm); + + return CUBEB_OK; +} + +static int +audiounit_configure_output(cubeb_stream * stm) +{ + assert(stm && stm->output_unit); + + int r; + AURenderCallbackStruct aurcbs_out; + UInt32 size; + + + LOG("(%p) Opening output side: rate %u, channels %u, format %d, latency in frames %u.", + stm, stm->output_stream_params.rate, stm->output_stream_params.channels, + stm->output_stream_params.format, stm->latency_frames); + + r = audio_stream_desc_init(&stm->output_desc, &stm->output_stream_params); + if (r != CUBEB_OK) { + LOG("(%p) Could not initialize the audio stream description.", stm); + return r; + } + + /* Get output device sample rate. */ + AudioStreamBasicDescription output_hw_desc; + size = sizeof(AudioStreamBasicDescription); + memset(&output_hw_desc, 0, size); + r = AudioUnitGetProperty(stm->output_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, + AU_OUT_BUS, + &output_hw_desc, + &size); + if (r != noErr) { + LOG("AudioUnitGetProperty/output/kAudioUnitProperty_StreamFormat rv=%d", r); + return CUBEB_ERROR; + } + stm->output_hw_rate = output_hw_desc.mSampleRate; + LOG("(%p) Output device sampling rate: %.2f", stm, output_hw_desc.mSampleRate); + stm->context->channels = output_hw_desc.mChannelsPerFrame; + + // Set the input layout to match the output device layout. + audiounit_layout_init(stm, io_side::OUTPUT); + if (stm->context->channels != stm->output_stream_params.channels || + stm->context->layout != stm->output_stream_params.layout) { + LOG("Incompatible channel layouts detected, setting up remixer"); + audiounit_init_mixer(stm); + // We will be remixing the data before it reaches the output device. + // We need to adjust the number of channels and other + // AudioStreamDescription details. + stm->output_desc.mChannelsPerFrame = stm->context->channels; + stm->output_desc.mBytesPerFrame = (stm->output_desc.mBitsPerChannel / 8) * + stm->output_desc.mChannelsPerFrame; + stm->output_desc.mBytesPerPacket = + stm->output_desc.mBytesPerFrame * stm->output_desc.mFramesPerPacket; + } else { + stm->mixer = nullptr; + } + + r = AudioUnitSetProperty(stm->output_unit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + AU_OUT_BUS, + &stm->output_desc, + sizeof(AudioStreamBasicDescription)); + if (r != noErr) { + LOG("AudioUnitSetProperty/output/kAudioUnitProperty_StreamFormat rv=%d", r); + return CUBEB_ERROR; + } + + r = audiounit_set_buffer_size(stm, stm->latency_frames, io_side::OUTPUT); + if (r != CUBEB_OK) { + LOG("(%p) Error in change output buffer size.", stm); + return CUBEB_ERROR; + } + + /* Frames per buffer in the input callback. */ + r = AudioUnitSetProperty(stm->output_unit, + kAudioUnitProperty_MaximumFramesPerSlice, + kAudioUnitScope_Global, + AU_OUT_BUS, + &stm->latency_frames, + sizeof(UInt32)); + if (r != noErr) { + LOG("AudioUnitSetProperty/output/kAudioUnitProperty_MaximumFramesPerSlice rv=%d", r); + return CUBEB_ERROR; + } + + aurcbs_out.inputProc = audiounit_output_callback; + aurcbs_out.inputProcRefCon = stm; + r = AudioUnitSetProperty(stm->output_unit, + kAudioUnitProperty_SetRenderCallback, + kAudioUnitScope_Global, + AU_OUT_BUS, + &aurcbs_out, + sizeof(aurcbs_out)); + if (r != noErr) { + LOG("AudioUnitSetProperty/output/kAudioUnitProperty_SetRenderCallback rv=%d", r); + return CUBEB_ERROR; + } + + stm->frames_written = 0; + + LOG("(%p) Output audiounit init successfully.", stm); + return CUBEB_OK; +} + +static int +audiounit_setup_stream(cubeb_stream * stm) +{ + stm->mutex.assert_current_thread_owns(); + + if ((stm->input_stream_params.prefs & CUBEB_STREAM_PREF_LOOPBACK) || + (stm->output_stream_params.prefs & CUBEB_STREAM_PREF_LOOPBACK)) { + LOG("(%p) Loopback not supported for audiounit.", stm); + return CUBEB_ERROR_NOT_SUPPORTED; + } + + int r = 0; + + device_info in_dev_info = stm->input_device; + device_info out_dev_info = stm->output_device; + + if (has_input(stm) && has_output(stm) && + stm->input_device.id != stm->output_device.id) { + r = audiounit_create_aggregate_device(stm); + if (r != CUBEB_OK) { + stm->aggregate_device_id = kAudioObjectUnknown; + LOG("(%p) Create aggregate devices failed.", stm); + // !!!NOTE: It is not necessary to return here. If it does not + // return it will fallback to the old implementation. The intention + // is to investigate how often it fails. I plan to remove + // it after a couple of weeks. + return r; + } else { + in_dev_info.id = out_dev_info.id = stm->aggregate_device_id; + in_dev_info.flags = DEV_INPUT; + out_dev_info.flags = DEV_OUTPUT; + } + } + + if (has_input(stm)) { + r = audiounit_create_unit(&stm->input_unit, &in_dev_info); + if (r != CUBEB_OK) { + LOG("(%p) AudioUnit creation for input failed.", stm); + return r; + } + } + + if (has_output(stm)) { + r = audiounit_create_unit(&stm->output_unit, &out_dev_info); + if (r != CUBEB_OK) { + LOG("(%p) AudioUnit creation for output failed.", stm); + return r; + } + } + + /* Latency cannot change if another stream is operating in parallel. In this case + * latency is set to the other stream value. */ + if (audiounit_active_streams(stm->context) > 1) { + LOG("(%p) More than one active stream, use global latency.", stm); + stm->latency_frames = stm->context->global_latency_frames; + } else { + /* Silently clamp the latency down to the platform default, because we + * synthetize the clock from the callbacks, and we want the clock to update + * often. */ + stm->latency_frames = audiounit_clamp_latency(stm, stm->latency_frames); + assert(stm->latency_frames); // Ugly error check + audiounit_set_global_latency(stm->context, stm->latency_frames); + } + + /* Configure I/O stream */ + if (has_input(stm)) { + r = audiounit_configure_input(stm); + if (r != CUBEB_OK) { + LOG("(%p) Configure audiounit input failed.", stm); + return r; + } + } + + if (has_output(stm)) { + r = audiounit_configure_output(stm); + if (r != CUBEB_OK) { + LOG("(%p) Configure audiounit output failed.", stm); + return r; + } + } + + // Setting the latency doesn't work well for USB headsets (eg. plantronics). + // Keep the default latency for now. +#if 0 + buffer_size = latency; + + /* Get the range of latency this particular device can work with, and clamp + * the requested latency to this acceptable range. */ +#if !TARGET_OS_IPHONE + if (audiounit_get_acceptable_latency_range(&latency_range) != CUBEB_OK) { + return CUBEB_ERROR; + } + + if (buffer_size < (unsigned int) latency_range.mMinimum) { + buffer_size = (unsigned int) latency_range.mMinimum; + } else if (buffer_size > (unsigned int) latency_range.mMaximum) { + buffer_size = (unsigned int) latency_range.mMaximum; + } + + /** + * Get the default buffer size. If our latency request is below the default, + * set it. Otherwise, use the default latency. + **/ + size = sizeof(default_buffer_size); + if (AudioUnitGetProperty(stm->output_unit, kAudioDevicePropertyBufferFrameSize, + kAudioUnitScope_Output, 0, &default_buffer_size, &size) != 0) { + return CUBEB_ERROR; + } + + if (buffer_size < default_buffer_size) { + /* Set the maximum number of frame that the render callback will ask for, + * effectively setting the latency of the stream. This is process-wide. */ + if (AudioUnitSetProperty(stm->output_unit, kAudioDevicePropertyBufferFrameSize, + kAudioUnitScope_Output, 0, &buffer_size, sizeof(buffer_size)) != 0) { + return CUBEB_ERROR; + } + } +#else // TARGET_OS_IPHONE + //TODO: [[AVAudioSession sharedInstance] inputLatency] + // http://stackoverflow.com/questions/13157523/kaudiodevicepropertybufferframesize-replacement-for-ios +#endif +#endif + + /* We use a resampler because input AudioUnit operates + * reliable only in the capture device sample rate. + * Resampler will convert it to the user sample rate + * and deliver it to the callback. */ + uint32_t target_sample_rate; + if (has_input(stm)) { + target_sample_rate = stm->input_stream_params.rate; + } else { + assert(has_output(stm)); + target_sample_rate = stm->output_stream_params.rate; + } + + cubeb_stream_params input_unconverted_params; + if (has_input(stm)) { + input_unconverted_params = stm->input_stream_params; + /* Use the rate of the input device. */ + input_unconverted_params.rate = stm->input_hw_rate; + } + + /* Create resampler. Output params are unchanged + * because we do not need conversion on the output. */ + stm->resampler.reset(cubeb_resampler_create(stm, + has_input(stm) ? &input_unconverted_params : NULL, + has_output(stm) ? &stm->output_stream_params : NULL, + target_sample_rate, + stm->data_callback, + stm->user_ptr, + CUBEB_RESAMPLER_QUALITY_DESKTOP)); + if (!stm->resampler) { + LOG("(%p) Could not create resampler.", stm); + return CUBEB_ERROR; + } + + if (stm->input_unit != NULL) { + r = AudioUnitInitialize(stm->input_unit); + if (r != noErr) { + LOG("AudioUnitInitialize/input rv=%d", r); + return CUBEB_ERROR; + } + } + + if (stm->output_unit != NULL) { + r = AudioUnitInitialize(stm->output_unit); + if (r != noErr) { + LOG("AudioUnitInitialize/output rv=%d", r); + return CUBEB_ERROR; + } + + stm->current_latency_frames = audiounit_get_device_presentation_latency(stm->output_device.id, kAudioDevicePropertyScopeOutput); + + Float64 unit_s; + UInt32 size = sizeof(unit_s); + if (AudioUnitGetProperty(stm->output_unit, kAudioUnitProperty_Latency, kAudioUnitScope_Global, 0, &unit_s, &size) == noErr) { + stm->current_latency_frames += static_cast(unit_s * stm->output_desc.mSampleRate); + } + } + + if (stm->input_unit && stm->output_unit) { + // According to the I/O hardware rate it is expected a specific pattern of callbacks + // for example is input is 44100 and output is 48000 we expected no more than 2 + // out callback in a row. + stm->expected_output_callbacks_in_a_row = ceilf(stm->output_hw_rate / stm->input_hw_rate); + } + + r = audiounit_install_device_changed_callback(stm); + if (r != CUBEB_OK) { + LOG("(%p) Could not install all device change callback.", stm); + } + + + return CUBEB_OK; +} + +cubeb_stream::cubeb_stream(cubeb * context) + : context(context) + , resampler(nullptr, cubeb_resampler_destroy) + , mixer(nullptr, cubeb_mixer_destroy) +{ + PodZero(&input_desc, 1); + PodZero(&output_desc, 1); +} + +static void audiounit_stream_destroy_internal(cubeb_stream * stm); + +static int +audiounit_stream_init(cubeb * context, + cubeb_stream ** stream, + char const * /* stream_name */, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + assert(context); + auto_lock context_lock(context->mutex); + audiounit_increment_active_streams(context); + unique_ptr stm(new cubeb_stream(context), + audiounit_stream_destroy_internal); + int r; + *stream = NULL; + assert(latency_frames > 0); + + /* These could be different in the future if we have both + * full-duplex stream and different devices for input vs output. */ + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + stm->latency_frames = latency_frames; + + if ((input_device && !input_stream_params) || + (output_device && !output_stream_params)) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + if (input_stream_params) { + stm->input_stream_params = *input_stream_params; + r = audiounit_set_device_info(stm.get(), reinterpret_cast(input_device), io_side::INPUT); + if (r != CUBEB_OK) { + LOG("(%p) Fail to set device info for input.", stm.get()); + return r; + } + } + if (output_stream_params) { + stm->output_stream_params = *output_stream_params; + r = audiounit_set_device_info(stm.get(), reinterpret_cast(output_device), io_side::OUTPUT); + if (r != CUBEB_OK) { + LOG("(%p) Fail to set device info for output.", stm.get()); + return r; + } + } + + { + // It's not critical to lock here, because no other thread has been started + // yet, but it allows to assert that the lock has been taken in + // `audiounit_setup_stream`. + auto_lock lock(stm->mutex); + r = audiounit_setup_stream(stm.get()); + } + + if (r != CUBEB_OK) { + LOG("(%p) Could not setup the audiounit stream.", stm.get()); + return r; + } + + r = audiounit_install_system_changed_callback(stm.get()); + if (r != CUBEB_OK) { + LOG("(%p) Could not install the device change callback.", stm.get()); + return r; + } + + *stream = stm.release(); + LOG("(%p) Cubeb stream init successful.", *stream); + return CUBEB_OK; +} + +static void +audiounit_close_stream(cubeb_stream *stm) +{ + stm->mutex.assert_current_thread_owns(); + + if (stm->input_unit) { + AudioUnitUninitialize(stm->input_unit); + AudioComponentInstanceDispose(stm->input_unit); + stm->input_unit = nullptr; + } + + stm->input_linear_buffer.reset(); + + if (stm->output_unit) { + AudioUnitUninitialize(stm->output_unit); + AudioComponentInstanceDispose(stm->output_unit); + stm->output_unit = nullptr; + } + + stm->resampler.reset(); + stm->mixer.reset(); + + if (stm->aggregate_device_id != kAudioObjectUnknown) { + audiounit_destroy_aggregate_device(stm->plugin_id, &stm->aggregate_device_id); + stm->aggregate_device_id = kAudioObjectUnknown; + } +} + +static void +audiounit_stream_destroy_internal(cubeb_stream *stm) +{ + stm->context->mutex.assert_current_thread_owns(); + + int r = audiounit_uninstall_system_changed_callback(stm); + if (r != CUBEB_OK) { + LOG("(%p) Could not uninstall the device changed callback", stm); + } + r = audiounit_uninstall_device_changed_callback(stm); + if (r != CUBEB_OK) { + LOG("(%p) Could not uninstall all device change listeners", stm); + } + + auto_lock lock(stm->mutex); + audiounit_close_stream(stm); + assert(audiounit_active_streams(stm->context) >= 1); + audiounit_decrement_active_streams(stm->context); +} + +static void +audiounit_stream_destroy(cubeb_stream * stm) +{ + if (!stm->shutdown.load()){ + auto_lock context_lock(stm->context->mutex); + audiounit_stream_stop_internal(stm); + stm->shutdown = true; + } + + stm->destroy_pending = true; + // Execute close in serial queue to avoid collision + // with reinit when un/plug devices + dispatch_sync(stm->context->serial_queue, ^() { + auto_lock context_lock(stm->context->mutex); + audiounit_stream_destroy_internal(stm); + }); + + LOG("Cubeb stream (%p) destroyed successful.", stm); + delete stm; +} + +static int +audiounit_stream_start_internal(cubeb_stream * stm) +{ + OSStatus r; + if (stm->input_unit != NULL) { + r = AudioOutputUnitStart(stm->input_unit); + if (r != noErr) { + LOG("AudioOutputUnitStart (input) rv=%d", r); + return CUBEB_ERROR; + } + } + if (stm->output_unit != NULL) { + r = AudioOutputUnitStart(stm->output_unit); + if (r != noErr) { + LOG("AudioOutputUnitStart (output) rv=%d", r); + return CUBEB_ERROR; + } + } + return CUBEB_OK; +} + +static int +audiounit_stream_start(cubeb_stream * stm) +{ + auto_lock context_lock(stm->context->mutex); + stm->shutdown = false; + stm->draining = false; + + int r = audiounit_stream_start_internal(stm); + if (r != CUBEB_OK) { + return r; + } + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED); + + LOG("Cubeb stream (%p) started successfully.", stm); + return CUBEB_OK; +} + +void +audiounit_stream_stop_internal(cubeb_stream * stm) +{ + OSStatus r; + if (stm->input_unit != NULL) { + r = AudioOutputUnitStop(stm->input_unit); + assert(r == 0); + } + if (stm->output_unit != NULL) { + r = AudioOutputUnitStop(stm->output_unit); + assert(r == 0); + } +} + +static int +audiounit_stream_stop(cubeb_stream * stm) +{ + auto_lock context_lock(stm->context->mutex); + stm->shutdown = true; + + audiounit_stream_stop_internal(stm); + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED); + + LOG("Cubeb stream (%p) stopped successfully.", stm); + return CUBEB_OK; +} + +static int +audiounit_stream_get_position(cubeb_stream * stm, uint64_t * position) +{ + assert(stm); + if (stm->current_latency_frames > stm->frames_played) { + *position = 0; + } else { + *position = stm->frames_played - stm->current_latency_frames; + } + return CUBEB_OK; +} + +int +audiounit_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ +#if TARGET_OS_IPHONE + //TODO + return CUBEB_ERROR_NOT_SUPPORTED; +#else + *latency = stm->total_output_latency_frames; + return CUBEB_OK; +#endif +} + +static int +audiounit_stream_get_volume(cubeb_stream * stm, float * volume) +{ + assert(stm->output_unit); + OSStatus r = AudioUnitGetParameter(stm->output_unit, + kHALOutputParam_Volume, + kAudioUnitScope_Global, + 0, volume); + if (r != noErr) { + LOG("AudioUnitGetParameter/kHALOutputParam_Volume rv=%d", r); + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +static int +audiounit_stream_set_volume(cubeb_stream * stm, float volume) +{ + assert(stm->output_unit); + OSStatus r; + r = AudioUnitSetParameter(stm->output_unit, + kHALOutputParam_Volume, + kAudioUnitScope_Global, + 0, volume, 0); + + if (r != noErr) { + LOG("AudioUnitSetParameter/kHALOutputParam_Volume rv=%d", r); + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +unique_ptr convert_uint32_into_string(UInt32 data) +{ + // Simply create an empty string if no data. + size_t size = data == 0 ? 0 : 4; // 4 bytes for uint32. + auto str = unique_ptr { new char[size + 1] }; // + 1 for '\0'. + str[size] = '\0'; + if (size < 4) { + return str; + } + + // Reverse 0xWXYZ into 0xZYXW. + str[0] = (char)(data >> 24); + str[1] = (char)(data >> 16); + str[2] = (char)(data >> 8); + str[3] = (char)(data); + return str; +} + +int audiounit_get_default_device_datasource(cubeb_device_type type, + UInt32 * data) +{ + AudioDeviceID id = audiounit_get_default_device_id(type); + if (id == kAudioObjectUnknown) { + return CUBEB_ERROR; + } + + UInt32 size = sizeof(*data); + /* This fails with some USB headsets (e.g., Plantronic .Audio 628). */ + OSStatus r = AudioObjectGetPropertyData(id, + type == CUBEB_DEVICE_TYPE_INPUT ? + &INPUT_DATA_SOURCE_PROPERTY_ADDRESS : + &OUTPUT_DATA_SOURCE_PROPERTY_ADDRESS, + 0, NULL, &size, data); + if (r != noErr) { + *data = 0; + } + + return CUBEB_OK; +} + +int audiounit_get_default_device_name(cubeb_stream * stm, + cubeb_device * const device, + cubeb_device_type type) +{ + assert(stm); + assert(device); + + UInt32 data; + int r = audiounit_get_default_device_datasource(type, &data); + if (r != CUBEB_OK) { + return r; + } + char ** name = type == CUBEB_DEVICE_TYPE_INPUT ? + &device->input_name : &device->output_name; + *name = convert_uint32_into_string(data).release(); + if (!strlen(*name)) { // empty string. + LOG("(%p) name of %s device is empty!", stm, + type == CUBEB_DEVICE_TYPE_INPUT ? "input" : "output"); + } + return CUBEB_OK; +} + + +int audiounit_stream_get_current_device(cubeb_stream * stm, + cubeb_device ** const device) +{ +#if TARGET_OS_IPHONE + //TODO + return CUBEB_ERROR_NOT_SUPPORTED; +#else + *device = new cubeb_device; + if (!*device) { + return CUBEB_ERROR; + } + PodZero(*device, 1); + + int r = audiounit_get_default_device_name(stm, *device, + CUBEB_DEVICE_TYPE_OUTPUT); + if (r != CUBEB_OK) { + return r; + } + + r = audiounit_get_default_device_name(stm, *device, + CUBEB_DEVICE_TYPE_INPUT); + if (r != CUBEB_OK) { + return r; + } + + return CUBEB_OK; +#endif +} + +int audiounit_stream_device_destroy(cubeb_stream * /* stream */, + cubeb_device * device) +{ + delete [] device->output_name; + delete [] device->input_name; + delete device; + return CUBEB_OK; +} + +int audiounit_stream_register_device_changed_callback(cubeb_stream * stream, + cubeb_device_changed_callback device_changed_callback) +{ + auto_lock dev_cb_lock(stream->device_changed_callback_lock); + /* Note: second register without unregister first causes 'nope' error. + * Current implementation requires unregister before register a new cb. */ + assert(!device_changed_callback || !stream->device_changed_callback); + stream->device_changed_callback = device_changed_callback; + return CUBEB_OK; +} + +static char * +audiounit_strref_to_cstr_utf8(CFStringRef strref) +{ + CFIndex len, size; + char * ret; + if (strref == NULL) { + return NULL; + } + + len = CFStringGetLength(strref); + // Add 1 to size to allow for '\0' termination character. + size = CFStringGetMaximumSizeForEncoding(len, kCFStringEncodingUTF8) + 1; + ret = new char[size]; + + if (!CFStringGetCString(strref, ret, size, kCFStringEncodingUTF8)) { + delete [] ret; + ret = NULL; + } + + return ret; +} + +static uint32_t +audiounit_get_channel_count(AudioObjectID devid, AudioObjectPropertyScope scope) +{ + AudioObjectPropertyAddress adr = { 0, scope, kAudioObjectPropertyElementMaster }; + UInt32 size = 0; + uint32_t i, ret = 0; + + adr.mSelector = kAudioDevicePropertyStreamConfiguration; + + if (AudioObjectGetPropertyDataSize(devid, &adr, 0, NULL, &size) == noErr && size > 0) { + AudioBufferList * list = static_cast(alloca(size)); + if (AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, list) == noErr) { + for (i = 0; i < list->mNumberBuffers; i++) + ret += list->mBuffers[i].mNumberChannels; + } + } + + return ret; +} + +static void +audiounit_get_available_samplerate(AudioObjectID devid, AudioObjectPropertyScope scope, + uint32_t * min, uint32_t * max, uint32_t * def) +{ + AudioObjectPropertyAddress adr = { 0, scope, kAudioObjectPropertyElementMaster }; + + adr.mSelector = kAudioDevicePropertyNominalSampleRate; + if (AudioObjectHasProperty(devid, &adr)) { + UInt32 size = sizeof(Float64); + Float64 fvalue = 0.0; + if (AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &fvalue) == noErr) { + *def = fvalue; + } + } + + adr.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; + UInt32 size = 0; + AudioValueRange range; + if (AudioObjectHasProperty(devid, &adr) && + AudioObjectGetPropertyDataSize(devid, &adr, 0, NULL, &size) == noErr) { + uint32_t count = size / sizeof(AudioValueRange); + vector ranges(count); + range.mMinimum = 9999999999.0; + range.mMaximum = 0.0; + if (AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, ranges.data()) == noErr) { + for (uint32_t i = 0; i < count; i++) { + if (ranges[i].mMaximum > range.mMaximum) + range.mMaximum = ranges[i].mMaximum; + if (ranges[i].mMinimum < range.mMinimum) + range.mMinimum = ranges[i].mMinimum; + } + } + *max = static_cast(range.mMaximum); + *min = static_cast(range.mMinimum); + } else { + *min = *max = 0; + } + +} + +static UInt32 +audiounit_get_device_presentation_latency(AudioObjectID devid, AudioObjectPropertyScope scope) +{ + AudioObjectPropertyAddress adr = { 0, scope, kAudioObjectPropertyElementMaster }; + UInt32 size, dev, stream = 0; + AudioStreamID sid[1]; + + adr.mSelector = kAudioDevicePropertyLatency; + size = sizeof(UInt32); + if (AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &dev) != noErr) { + dev = 0; + } + + adr.mSelector = kAudioDevicePropertyStreams; + size = sizeof(sid); + if (AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, sid) == noErr) { + adr.mSelector = kAudioStreamPropertyLatency; + size = sizeof(UInt32); + AudioObjectGetPropertyData(sid[0], &adr, 0, NULL, &size, &stream); + } + + return dev + stream; +} + +static int +audiounit_create_device_from_hwdev(cubeb_device_info * dev_info, AudioObjectID devid, cubeb_device_type type) +{ + AudioObjectPropertyAddress adr = { 0, 0, kAudioObjectPropertyElementMaster }; + UInt32 size; + + if (type == CUBEB_DEVICE_TYPE_OUTPUT) { + adr.mScope = kAudioDevicePropertyScopeOutput; + } else if (type == CUBEB_DEVICE_TYPE_INPUT) { + adr.mScope = kAudioDevicePropertyScopeInput; + } else { + return CUBEB_ERROR; + } + + UInt32 ch = audiounit_get_channel_count(devid, adr.mScope); + if (ch == 0) { + return CUBEB_ERROR; + } + + PodZero(dev_info, 1); + + CFStringRef device_id_str = nullptr; + size = sizeof(CFStringRef); + adr.mSelector = kAudioDevicePropertyDeviceUID; + OSStatus ret = AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &device_id_str); + if ( ret == noErr && device_id_str != NULL) { + dev_info->device_id = audiounit_strref_to_cstr_utf8(device_id_str); + static_assert(sizeof(cubeb_devid) >= sizeof(decltype(devid)), "cubeb_devid can't represent devid"); + dev_info->devid = reinterpret_cast(devid); + dev_info->group_id = dev_info->device_id; + CFRelease(device_id_str); + } + + CFStringRef friendly_name_str = nullptr; + UInt32 ds; + size = sizeof(UInt32); + adr.mSelector = kAudioDevicePropertyDataSource; + ret = AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &ds); + if (ret == noErr) { + AudioValueTranslation trl = { &ds, sizeof(ds), &friendly_name_str, sizeof(CFStringRef) }; + adr.mSelector = kAudioDevicePropertyDataSourceNameForIDCFString; + size = sizeof(AudioValueTranslation); + AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &trl); + } + + // If there is no datasource for this device, fall back to the + // device name. + if (!friendly_name_str) { + size = sizeof(CFStringRef); + adr.mSelector = kAudioObjectPropertyName; + AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &friendly_name_str); + } + + if (friendly_name_str) { + dev_info->friendly_name = audiounit_strref_to_cstr_utf8(friendly_name_str); + CFRelease(friendly_name_str); + } else { + // Couldn't get a datasource name nor a device name, return a + // valid string of length 0. + char * fallback_name = new char[1]; + fallback_name[0] = '\0'; + dev_info->friendly_name = fallback_name; + } + + CFStringRef vendor_name_str = nullptr; + size = sizeof(CFStringRef); + adr.mSelector = kAudioObjectPropertyManufacturer; + ret = AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &vendor_name_str); + if (ret == noErr && vendor_name_str != NULL) { + dev_info->vendor_name = audiounit_strref_to_cstr_utf8(vendor_name_str); + CFRelease(vendor_name_str); + } + + dev_info->type = type; + dev_info->state = CUBEB_DEVICE_STATE_ENABLED; + dev_info->preferred = (devid == audiounit_get_default_device_id(type)) ? + CUBEB_DEVICE_PREF_ALL : CUBEB_DEVICE_PREF_NONE; + + dev_info->max_channels = ch; + dev_info->format = (cubeb_device_fmt)CUBEB_DEVICE_FMT_ALL; /* CoreAudio supports All! */ + /* kAudioFormatFlagsAudioUnitCanonical is deprecated, prefer floating point */ + dev_info->default_format = CUBEB_DEVICE_FMT_F32NE; + audiounit_get_available_samplerate(devid, adr.mScope, + &dev_info->min_rate, &dev_info->max_rate, &dev_info->default_rate); + + UInt32 latency = audiounit_get_device_presentation_latency(devid, adr.mScope); + + AudioValueRange range; + adr.mSelector = kAudioDevicePropertyBufferFrameSizeRange; + size = sizeof(AudioValueRange); + ret = AudioObjectGetPropertyData(devid, &adr, 0, NULL, &size, &range); + if (ret == noErr) { + dev_info->latency_lo = latency + range.mMinimum; + dev_info->latency_hi = latency + range.mMaximum; + } else { + dev_info->latency_lo = 10 * dev_info->default_rate / 1000; /* Default to 10ms */ + dev_info->latency_hi = 100 * dev_info->default_rate / 1000; /* Default to 100ms */ + } + + return CUBEB_OK; +} + +bool +is_aggregate_device(cubeb_device_info * device_info) +{ + assert(device_info->friendly_name); + return !strncmp(device_info->friendly_name, PRIVATE_AGGREGATE_DEVICE_NAME, + strlen(PRIVATE_AGGREGATE_DEVICE_NAME)); +} + +static int +audiounit_enumerate_devices(cubeb * /* context */, cubeb_device_type type, + cubeb_device_collection * collection) +{ + vector input_devs; + vector output_devs; + + // Count number of input and output devices. This is not + // necessarily the same as the count of raw devices supported by the + // system since, for example, with Soundflower installed, some + // devices may report as being both input *and* output and cubeb + // separates those into two different devices. + + if (type & CUBEB_DEVICE_TYPE_OUTPUT) { + output_devs = audiounit_get_devices_of_type(CUBEB_DEVICE_TYPE_OUTPUT); + } + + if (type & CUBEB_DEVICE_TYPE_INPUT) { + input_devs = audiounit_get_devices_of_type(CUBEB_DEVICE_TYPE_INPUT); + } + + auto devices = new cubeb_device_info[output_devs.size() + input_devs.size()]; + collection->count = 0; + + if (type & CUBEB_DEVICE_TYPE_OUTPUT) { + for (auto dev: output_devs) { + auto device = &devices[collection->count]; + auto err = audiounit_create_device_from_hwdev(device, dev, CUBEB_DEVICE_TYPE_OUTPUT); + if (err != CUBEB_OK || is_aggregate_device(device)) { + continue; + } + collection->count += 1; + } + } + + if (type & CUBEB_DEVICE_TYPE_INPUT) { + for (auto dev: input_devs) { + auto device = &devices[collection->count]; + auto err = audiounit_create_device_from_hwdev(device, dev, CUBEB_DEVICE_TYPE_INPUT); + if (err != CUBEB_OK || is_aggregate_device(device)) { + continue; + } + collection->count += 1; + } + } + + if (collection->count > 0) { + collection->device = devices; + } else { + delete [] devices; + collection->device = NULL; + } + + return CUBEB_OK; +} + +static void +audiounit_device_destroy(cubeb_device_info * device) +{ + delete [] device->device_id; + delete [] device->friendly_name; + delete [] device->vendor_name; +} + +static int +audiounit_device_collection_destroy(cubeb * /* context */, + cubeb_device_collection * collection) +{ + for (size_t i = 0; i < collection->count; i++) { + audiounit_device_destroy(&collection->device[i]); + } + delete [] collection->device; + + return CUBEB_OK; +} + +static vector +audiounit_get_devices_of_type(cubeb_device_type devtype) +{ + UInt32 size = 0; + OSStatus ret = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, + &DEVICES_PROPERTY_ADDRESS, 0, + NULL, &size); + if (ret != noErr) { + return vector(); + } + vector devices(size / sizeof(AudioObjectID)); + ret = AudioObjectGetPropertyData(kAudioObjectSystemObject, + &DEVICES_PROPERTY_ADDRESS, 0, NULL, &size, + devices.data()); + if (ret != noErr) { + return vector(); + } + + // Remove the aggregate device from the list of devices (if any). + for (auto it = devices.begin(); it != devices.end();) { + CFStringRef name = get_device_name(*it); + if (name && CFStringFind(name, CFSTR("CubebAggregateDevice"), 0).location != + kCFNotFound) { + it = devices.erase(it); + } else { + it++; + } + if (name) { + CFRelease(name); + } + } + + /* Expected sorted but did not find anything in the docs. */ + sort(devices.begin(), devices.end(), [](AudioObjectID a, AudioObjectID b) { + return a < b; + }); + + if (devtype == (CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_OUTPUT)) { + return devices; + } + + AudioObjectPropertyScope scope = (devtype == CUBEB_DEVICE_TYPE_INPUT) ? + kAudioDevicePropertyScopeInput : + kAudioDevicePropertyScopeOutput; + + vector devices_in_scope; + for (uint32_t i = 0; i < devices.size(); ++i) { + /* For device in the given scope channel must be > 0. */ + if (audiounit_get_channel_count(devices[i], scope) > 0) { + devices_in_scope.push_back(devices[i]); + } + } + + return devices_in_scope; +} + +static OSStatus +audiounit_collection_changed_callback(AudioObjectID /* inObjectID */, + UInt32 /* inNumberAddresses */, + const AudioObjectPropertyAddress * /* inAddresses */, + void * inClientData) +{ + cubeb * context = static_cast(inClientData); + + // This can be called from inside an AudioUnit function, dispatch to another queue. + dispatch_async(context->serial_queue, ^() { + auto_lock lock(context->mutex); + if (!context->input_collection_changed_callback && + !context->output_collection_changed_callback) { + /* Listener removed while waiting in mutex, abort. */ + return; + } + if (context->input_collection_changed_callback) { + vector devices = audiounit_get_devices_of_type(CUBEB_DEVICE_TYPE_INPUT); + /* Elements in the vector expected sorted. */ + if (context->input_device_array != devices) { + context->input_device_array = devices; + context->input_collection_changed_callback(context, context->input_collection_changed_user_ptr); + } + } + if (context->output_collection_changed_callback) { + vector devices = audiounit_get_devices_of_type(CUBEB_DEVICE_TYPE_OUTPUT); + /* Elements in the vector expected sorted. */ + if (context->output_device_array != devices) { + context->output_device_array = devices; + context->output_collection_changed_callback(context, context->output_collection_changed_user_ptr); + } + } + }); + return noErr; +} + +static OSStatus +audiounit_add_device_listener(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection_changed_callback collection_changed_callback, + void * user_ptr) +{ + context->mutex.assert_current_thread_owns(); + assert(devtype & (CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_OUTPUT)); + /* Note: second register without unregister first causes 'nope' error. + * Current implementation requires unregister before register a new cb. */ + assert((devtype & CUBEB_DEVICE_TYPE_INPUT) && !context->input_collection_changed_callback || + (devtype & CUBEB_DEVICE_TYPE_OUTPUT) && !context->output_collection_changed_callback); + + if (!context->input_collection_changed_callback && + !context->output_collection_changed_callback) { + OSStatus ret = AudioObjectAddPropertyListener(kAudioObjectSystemObject, + &DEVICES_PROPERTY_ADDRESS, + audiounit_collection_changed_callback, + context); + if (ret != noErr) { + return ret; + } + } + if (devtype & CUBEB_DEVICE_TYPE_INPUT) { + /* Expected empty after unregister. */ + assert(context->input_device_array.empty()); + context->input_device_array = audiounit_get_devices_of_type(CUBEB_DEVICE_TYPE_INPUT); + context->input_collection_changed_callback = collection_changed_callback; + context->input_collection_changed_user_ptr = user_ptr; + } + if (devtype & CUBEB_DEVICE_TYPE_OUTPUT) { + /* Expected empty after unregister. */ + assert(context->output_device_array.empty()); + context->output_device_array = audiounit_get_devices_of_type(CUBEB_DEVICE_TYPE_OUTPUT); + context->output_collection_changed_callback = collection_changed_callback; + context->output_collection_changed_user_ptr = user_ptr; + } + return noErr; +} + +static OSStatus +audiounit_remove_device_listener(cubeb * context, cubeb_device_type devtype) +{ + context->mutex.assert_current_thread_owns(); + + if (devtype & CUBEB_DEVICE_TYPE_INPUT) { + context->input_collection_changed_callback = nullptr; + context->input_collection_changed_user_ptr = nullptr; + context->input_device_array.clear(); + } + if (devtype & CUBEB_DEVICE_TYPE_OUTPUT) { + context->output_collection_changed_callback = nullptr; + context->output_collection_changed_user_ptr = nullptr; + context->output_device_array.clear(); + } + + if (context->input_collection_changed_callback || + context->output_collection_changed_callback) { + return noErr; + } + /* Note: unregister a non registered cb is not a problem, not checking. */ + return AudioObjectRemovePropertyListener(kAudioObjectSystemObject, + &DEVICES_PROPERTY_ADDRESS, + audiounit_collection_changed_callback, + context); +} + +int audiounit_register_device_collection_changed(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection_changed_callback collection_changed_callback, + void * user_ptr) +{ + if (devtype == CUBEB_DEVICE_TYPE_UNKNOWN) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + OSStatus ret; + auto_lock lock(context->mutex); + if (collection_changed_callback) { + ret = audiounit_add_device_listener(context, + devtype, + collection_changed_callback, + user_ptr); + } else { + ret = audiounit_remove_device_listener(context, devtype); + } + return (ret == noErr) ? CUBEB_OK : CUBEB_ERROR; +} + +cubeb_ops const audiounit_ops = { + /*.init =*/ audiounit_init, + /*.get_backend_id =*/ audiounit_get_backend_id, + /*.get_max_channel_count =*/ audiounit_get_max_channel_count, + /*.get_min_latency =*/ audiounit_get_min_latency, + /*.get_preferred_sample_rate =*/ audiounit_get_preferred_sample_rate, + /*.enumerate_devices =*/ audiounit_enumerate_devices, + /*.device_collection_destroy =*/ audiounit_device_collection_destroy, + /*.destroy =*/ audiounit_destroy, + /*.stream_init =*/ audiounit_stream_init, + /*.stream_destroy =*/ audiounit_stream_destroy, + /*.stream_start =*/ audiounit_stream_start, + /*.stream_stop =*/ audiounit_stream_stop, + /*.stream_reset_default_device =*/ nullptr, + /*.stream_get_position =*/ audiounit_stream_get_position, + /*.stream_get_latency =*/ audiounit_stream_get_latency, + /*.stream_set_volume =*/ audiounit_stream_set_volume, + /*.stream_get_current_device =*/ audiounit_stream_get_current_device, + /*.stream_device_destroy =*/ audiounit_stream_device_destroy, + /*.stream_register_device_changed_callback =*/ audiounit_stream_register_device_changed_callback, + /*.register_device_collection_changed =*/ audiounit_register_device_collection_changed +}; diff --git a/dep/cubeb/src/cubeb_jack.cpp b/dep/cubeb/src/cubeb_jack.cpp new file mode 100644 index 000000000..9fc97a4b8 --- /dev/null +++ b/dep/cubeb/src/cubeb_jack.cpp @@ -0,0 +1,1075 @@ +/* + * Copyright © 2012 David Richards + * Copyright © 2013 Sebastien Alaiwan + * Copyright © 2016 Damien Zammit + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#define _DEFAULT_SOURCE +#define _BSD_SOURCE +#ifndef __FreeBSD__ +#define _POSIX_SOURCE +#endif +#include +#include +#include +#include +#include +#include +#include +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" +#include "cubeb_resampler.h" +#include "cubeb_utils.h" + +#include +#include + +#define JACK_API_VISIT(X) \ + X(jack_activate) \ + X(jack_client_close) \ + X(jack_client_open) \ + X(jack_connect) \ + X(jack_free) \ + X(jack_get_ports) \ + X(jack_get_sample_rate) \ + X(jack_get_xrun_delayed_usecs) \ + X(jack_get_buffer_size) \ + X(jack_port_get_buffer) \ + X(jack_port_name) \ + X(jack_port_register) \ + X(jack_port_unregister) \ + X(jack_port_get_latency_range) \ + X(jack_set_process_callback) \ + X(jack_set_xrun_callback) \ + X(jack_set_graph_order_callback) \ + X(jack_set_error_function) \ + X(jack_set_info_function) + +#define IMPORT_FUNC(x) static decltype(x) * api_##x; +JACK_API_VISIT(IMPORT_FUNC); + +#define JACK_DEFAULT_IN "JACK capture" +#define JACK_DEFAULT_OUT "JACK playback" + +static const int MAX_STREAMS = 16; +static const int MAX_CHANNELS = 8; +static const int FIFO_SIZE = 4096 * sizeof(float); + +enum devstream { + NONE = 0, + IN_ONLY, + OUT_ONLY, + DUPLEX, +}; + +static void +s16ne_to_float(float * dst, const int16_t * src, size_t n) +{ + for (size_t i = 0; i < n; i++) + *(dst++) = (float)((float)*(src++) / 32767.0f); +} + +static void +float_to_s16ne(int16_t * dst, float * src, size_t n) +{ + for (size_t i = 0; i < n; i++) { + if (*src > 1.f) *src = 1.f; + if (*src < -1.f) *src = -1.f; + *(dst++) = (int16_t)((int16_t)(*(src++) * 32767)); + } +} + +extern "C" +{ +/*static*/ int jack_init (cubeb ** context, char const * context_name); +} +static char const * cbjack_get_backend_id(cubeb * context); +static int cbjack_get_max_channel_count(cubeb * ctx, uint32_t * max_channels); +static int cbjack_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames); +static int cbjack_get_latency(cubeb_stream * stm, unsigned int * latency_frames); +static int cbjack_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate); +static void cbjack_destroy(cubeb * context); +static void cbjack_interleave_capture(cubeb_stream * stream, float **in, jack_nframes_t nframes, bool format_mismatch); +static void cbjack_deinterleave_playback_refill_s16ne(cubeb_stream * stream, short **bufs_in, float **bufs_out, jack_nframes_t nframes); +static void cbjack_deinterleave_playback_refill_float(cubeb_stream * stream, float **bufs_in, float **bufs_out, jack_nframes_t nframes); +static int cbjack_stream_device_destroy(cubeb_stream * stream, + cubeb_device * device); +static int cbjack_stream_get_current_device(cubeb_stream * stm, cubeb_device ** const device); +static int cbjack_enumerate_devices(cubeb * context, cubeb_device_type type, + cubeb_device_collection * collection); +static int cbjack_device_collection_destroy(cubeb * context, + cubeb_device_collection * collection); +static int cbjack_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr); +static void cbjack_stream_destroy(cubeb_stream * stream); +static int cbjack_stream_start(cubeb_stream * stream); +static int cbjack_stream_stop(cubeb_stream * stream); +static int cbjack_stream_get_position(cubeb_stream * stream, uint64_t * position); +static int cbjack_stream_set_volume(cubeb_stream * stm, float volume); + +static struct cubeb_ops const cbjack_ops = { + .init = jack_init, + .get_backend_id = cbjack_get_backend_id, + .get_max_channel_count = cbjack_get_max_channel_count, + .get_min_latency = cbjack_get_min_latency, + .get_preferred_sample_rate = cbjack_get_preferred_sample_rate, + .enumerate_devices = cbjack_enumerate_devices, + .device_collection_destroy = cbjack_device_collection_destroy, + .destroy = cbjack_destroy, + .stream_init = cbjack_stream_init, + .stream_destroy = cbjack_stream_destroy, + .stream_start = cbjack_stream_start, + .stream_stop = cbjack_stream_stop, + .stream_reset_default_device = NULL, + .stream_get_position = cbjack_stream_get_position, + .stream_get_latency = cbjack_get_latency, + .stream_set_volume = cbjack_stream_set_volume, + .stream_get_current_device = cbjack_stream_get_current_device, + .stream_device_destroy = cbjack_stream_device_destroy, + .stream_register_device_changed_callback = NULL, + .register_device_collection_changed = NULL +}; + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr; + /**/ + + /**< Mutex for each stream */ + pthread_mutex_t mutex; + + bool in_use; /**< Set to false iff the stream is free */ + bool ports_ready; /**< Set to true iff the JACK ports are ready */ + + cubeb_data_callback data_callback; + cubeb_state_callback state_callback; + cubeb_stream_params in_params; + cubeb_stream_params out_params; + + cubeb_resampler * resampler; + + uint64_t position; + bool pause; + float ratio; + enum devstream devs; + char stream_name[256]; + jack_port_t * output_ports[MAX_CHANNELS]; + jack_port_t * input_ports[MAX_CHANNELS]; + float volume; +}; + +struct cubeb { + struct cubeb_ops const * ops; + void * libjack; + + /**< Mutex for whole context */ + pthread_mutex_t mutex; + + /**< Audio buffers, converted to float */ + float in_float_interleaved_buffer[FIFO_SIZE * MAX_CHANNELS]; + float out_float_interleaved_buffer[FIFO_SIZE * MAX_CHANNELS]; + + /**< Audio buffer, at the sampling rate of the output */ + float in_resampled_interleaved_buffer_float[FIFO_SIZE * MAX_CHANNELS * 3]; + int16_t in_resampled_interleaved_buffer_s16ne[FIFO_SIZE * MAX_CHANNELS * 3]; + float out_resampled_interleaved_buffer_float[FIFO_SIZE * MAX_CHANNELS * 3]; + int16_t out_resampled_interleaved_buffer_s16ne[FIFO_SIZE * MAX_CHANNELS * 3]; + + cubeb_stream streams[MAX_STREAMS]; + unsigned int active_streams; + + cubeb_device_collection_changed_callback collection_changed_callback; + + bool active; + unsigned int jack_sample_rate; + unsigned int jack_latency; + unsigned int jack_xruns; + unsigned int jack_buffer_size; + unsigned int fragment_size; + unsigned int output_bytes_per_frame; + jack_client_t * jack_client; +}; + +static int +load_jack_lib(cubeb * context) +{ +#ifdef __APPLE__ + context->libjack = dlopen("libjack.0.dylib", RTLD_LAZY); + context->libjack = dlopen("/usr/local/lib/libjack.0.dylib", RTLD_LAZY); +#elif defined(__WIN32__) +# ifdef _WIN64 + context->libjack = LoadLibrary("libjack64.dll"); +# else + context->libjack = LoadLibrary("libjack.dll"); +# endif +#else + context->libjack = dlopen("libjack.so.0", RTLD_LAZY); + if (!context->libjack) { + context->libjack = dlopen("libjack.so", RTLD_LAZY); + } +#endif + if (!context->libjack) { + return CUBEB_ERROR; + } + +#define LOAD(x) \ + { \ + api_##x = (decltype(x)*)dlsym(context->libjack, #x); \ + if (!api_##x) { \ + dlclose(context->libjack); \ + return CUBEB_ERROR; \ + } \ + } + + JACK_API_VISIT(LOAD); +#undef LOAD + + return CUBEB_OK; +} + +static int +cbjack_connect_ports (cubeb_stream * stream) +{ + int r = CUBEB_ERROR; + const char ** phys_in_ports = api_jack_get_ports (stream->context->jack_client, + NULL, NULL, + JackPortIsInput + | JackPortIsPhysical); + const char ** phys_out_ports = api_jack_get_ports (stream->context->jack_client, + NULL, NULL, + JackPortIsOutput + | JackPortIsPhysical); + + if (phys_in_ports == NULL || *phys_in_ports == NULL) { + goto skipplayback; + } + + // Connect outputs to playback + for (unsigned int c = 0; c < stream->out_params.channels && phys_in_ports[c] != NULL; c++) { + const char *src_port = api_jack_port_name (stream->output_ports[c]); + + api_jack_connect (stream->context->jack_client, src_port, phys_in_ports[c]); + } + r = CUBEB_OK; + +skipplayback: + if (phys_out_ports == NULL || *phys_out_ports == NULL) { + goto end; + } + // Connect inputs to capture + for (unsigned int c = 0; c < stream->in_params.channels && phys_out_ports[c] != NULL; c++) { + const char *src_port = api_jack_port_name (stream->input_ports[c]); + + api_jack_connect (stream->context->jack_client, phys_out_ports[c], src_port); + } + r = CUBEB_OK; +end: + if (phys_out_ports) { + api_jack_free(phys_out_ports); + } + if (phys_in_ports) { + api_jack_free(phys_in_ports); + } + return r; +} + +static int +cbjack_xrun_callback(void * arg) +{ + cubeb * ctx = (cubeb *)arg; + + float delay = api_jack_get_xrun_delayed_usecs(ctx->jack_client); + int fragments = (int)ceilf( ((delay / 1000000.0) * ctx->jack_sample_rate ) + / (float)(ctx->jack_buffer_size) ); + ctx->jack_xruns += fragments; + return 0; +} + +static int +cbjack_graph_order_callback(void * arg) +{ + cubeb * ctx = (cubeb *)arg; + int i; + jack_latency_range_t latency_range; + jack_nframes_t port_latency, max_latency = 0; + + for (int j = 0; j < MAX_STREAMS; j++) { + cubeb_stream *stm = &ctx->streams[j]; + + if (!stm->in_use) + continue; + if (!stm->ports_ready) + continue; + + for (i = 0; i < (int)stm->out_params.channels; ++i) { + api_jack_port_get_latency_range(stm->output_ports[i], JackPlaybackLatency, &latency_range); + port_latency = latency_range.max; + if (port_latency > max_latency) + max_latency = port_latency; + } + /* Cap minimum latency to 128 frames */ + if (max_latency < 128) + max_latency = 128; + } + + ctx->jack_latency = max_latency; + + return 0; +} + +static int +cbjack_process(jack_nframes_t nframes, void * arg) +{ + cubeb * ctx = (cubeb *)arg; + int t_jack_xruns = ctx->jack_xruns; + int i; + + for (int j = 0; j < MAX_STREAMS; j++) { + cubeb_stream *stm = &ctx->streams[j]; + float *bufs_out[stm->out_params.channels]; + float *bufs_in[stm->in_params.channels]; + + if (!stm->in_use) + continue; + + // handle xruns by skipping audio that should have been played + for (i = 0; i < t_jack_xruns; i++) { + stm->position += ctx->fragment_size * stm->ratio; + } + ctx->jack_xruns -= t_jack_xruns; + + if (!stm->ports_ready) + continue; + + if (stm->devs & OUT_ONLY) { + // get jack output buffers + for (i = 0; i < (int)stm->out_params.channels; i++) + bufs_out[i] = (float*)api_jack_port_get_buffer(stm->output_ports[i], nframes); + } + if (stm->devs & IN_ONLY) { + // get jack input buffers + for (i = 0; i < (int)stm->in_params.channels; i++) + bufs_in[i] = (float*)api_jack_port_get_buffer(stm->input_ports[i], nframes); + } + if (stm->pause) { + // paused, play silence on output + if (stm->devs & OUT_ONLY) { + for (unsigned int c = 0; c < stm->out_params.channels; c++) { + float* buffer_out = bufs_out[c]; + for (long f = 0; f < nframes; f++) { + buffer_out[f] = 0.f; + } + } + } + if (stm->devs & IN_ONLY) { + // paused, capture silence + for (unsigned int c = 0; c < stm->in_params.channels; c++) { + float* buffer_in = bufs_in[c]; + for (long f = 0; f < nframes; f++) { + buffer_in[f] = 0.f; + } + } + } + } else { + + // try to lock stream mutex + if (pthread_mutex_trylock(&stm->mutex) == 0) { + + int16_t *in_s16ne = stm->context->in_resampled_interleaved_buffer_s16ne; + float *in_float = stm->context->in_resampled_interleaved_buffer_float; + + // unpaused, play audio + if (stm->devs == DUPLEX) { + if (stm->out_params.format == CUBEB_SAMPLE_S16NE) { + cbjack_interleave_capture(stm, bufs_in, nframes, true); + cbjack_deinterleave_playback_refill_s16ne(stm, &in_s16ne, bufs_out, nframes); + } else if (stm->out_params.format == CUBEB_SAMPLE_FLOAT32NE) { + cbjack_interleave_capture(stm, bufs_in, nframes, false); + cbjack_deinterleave_playback_refill_float(stm, &in_float, bufs_out, nframes); + } + } else if (stm->devs == IN_ONLY) { + if (stm->in_params.format == CUBEB_SAMPLE_S16NE) { + cbjack_interleave_capture(stm, bufs_in, nframes, true); + cbjack_deinterleave_playback_refill_s16ne(stm, &in_s16ne, nullptr, nframes); + } else if (stm->in_params.format == CUBEB_SAMPLE_FLOAT32NE) { + cbjack_interleave_capture(stm, bufs_in, nframes, false); + cbjack_deinterleave_playback_refill_float(stm, &in_float, nullptr, nframes); + } + } else if (stm->devs == OUT_ONLY) { + if (stm->out_params.format == CUBEB_SAMPLE_S16NE) { + cbjack_deinterleave_playback_refill_s16ne(stm, nullptr, bufs_out, nframes); + } else if (stm->out_params.format == CUBEB_SAMPLE_FLOAT32NE) { + cbjack_deinterleave_playback_refill_float(stm, nullptr, bufs_out, nframes); + } + } + // unlock stream mutex + pthread_mutex_unlock(&stm->mutex); + + } else { + // could not lock mutex + // output silence + if (stm->devs & OUT_ONLY) { + for (unsigned int c = 0; c < stm->out_params.channels; c++) { + float* buffer_out = bufs_out[c]; + for (long f = 0; f < nframes; f++) { + buffer_out[f] = 0.f; + } + } + } + if (stm->devs & IN_ONLY) { + // capture silence + for (unsigned int c = 0; c < stm->in_params.channels; c++) { + float* buffer_in = bufs_in[c]; + for (long f = 0; f < nframes; f++) { + buffer_in[f] = 0.f; + } + } + } + } + } + } + return 0; +} + +static void +cbjack_deinterleave_playback_refill_float(cubeb_stream * stream, float ** in, float ** bufs_out, jack_nframes_t nframes) +{ + float * out_interleaved_buffer = nullptr; + + float * inptr = (in != NULL) ? *in : nullptr; + float * outptr = (bufs_out != NULL) ? *bufs_out : nullptr; + + long needed_frames = (bufs_out != NULL) ? nframes : 0; + long done_frames = 0; + long input_frames_count = (in != NULL) ? nframes : 0; + + done_frames = cubeb_resampler_fill(stream->resampler, + inptr, + &input_frames_count, + (bufs_out != NULL) ? stream->context->out_resampled_interleaved_buffer_float : NULL, + needed_frames); + + out_interleaved_buffer = stream->context->out_resampled_interleaved_buffer_float; + + if (outptr) { + // convert interleaved output buffers to contiguous buffers + for (unsigned int c = 0; c < stream->out_params.channels; c++) { + float* buffer = bufs_out[c]; + for (long f = 0; f < done_frames; f++) { + buffer[f] = out_interleaved_buffer[(f * stream->out_params.channels) + c] * stream->volume; + } + if (done_frames < needed_frames) { + // draining + for (long f = done_frames; f < needed_frames; f++) { + buffer[f] = 0.f; + } + } + if (done_frames == 0) { + // stop, but first zero out the existing buffer + for (long f = 0; f < needed_frames; f++) { + buffer[f] = 0.f; + } + } + } + } + + if (done_frames >= 0 && done_frames < needed_frames) { + // set drained + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_DRAINED); + // stop stream + cbjack_stream_stop(stream); + } + if (done_frames > 0 && done_frames <= needed_frames) { + // advance stream position + stream->position += done_frames * stream->ratio; + } + if (done_frames < 0 || done_frames > needed_frames) { + // stream error + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_ERROR); + } +} + +static void +cbjack_deinterleave_playback_refill_s16ne(cubeb_stream * stream, short ** in, float ** bufs_out, jack_nframes_t nframes) +{ + float * out_interleaved_buffer = nullptr; + + short * inptr = (in != NULL) ? *in : nullptr; + float * outptr = (bufs_out != NULL) ? *bufs_out : nullptr; + + long needed_frames = (bufs_out != NULL) ? nframes : 0; + long done_frames = 0; + long input_frames_count = (in != NULL) ? nframes : 0; + + done_frames = cubeb_resampler_fill(stream->resampler, + inptr, + &input_frames_count, + (bufs_out != NULL) ? stream->context->out_resampled_interleaved_buffer_s16ne : NULL, + needed_frames); + + s16ne_to_float(stream->context->out_resampled_interleaved_buffer_float, stream->context->out_resampled_interleaved_buffer_s16ne, done_frames * stream->out_params.channels); + + out_interleaved_buffer = stream->context->out_resampled_interleaved_buffer_float; + + if (outptr) { + // convert interleaved output buffers to contiguous buffers + for (unsigned int c = 0; c < stream->out_params.channels; c++) { + float* buffer = bufs_out[c]; + for (long f = 0; f < done_frames; f++) { + buffer[f] = out_interleaved_buffer[(f * stream->out_params.channels) + c] * stream->volume; + } + if (done_frames < needed_frames) { + // draining + for (long f = done_frames; f < needed_frames; f++) { + buffer[f] = 0.f; + } + } + if (done_frames == 0) { + // stop, but first zero out the existing buffer + for (long f = 0; f < needed_frames; f++) { + buffer[f] = 0.f; + } + } + } + } + + if (done_frames >= 0 && done_frames < needed_frames) { + // set drained + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_DRAINED); + // stop stream + cbjack_stream_stop(stream); + } + if (done_frames > 0 && done_frames <= needed_frames) { + // advance stream position + stream->position += done_frames * stream->ratio; + } + if (done_frames < 0 || done_frames > needed_frames) { + // stream error + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_ERROR); + } +} + +static void +cbjack_interleave_capture(cubeb_stream * stream, float **in, jack_nframes_t nframes, bool format_mismatch) +{ + float *in_buffer = stream->context->in_float_interleaved_buffer; + + for (unsigned int c = 0; c < stream->in_params.channels; c++) { + for (long f = 0; f < nframes; f++) { + in_buffer[(f * stream->in_params.channels) + c] = in[c][f] * stream->volume; + } + } + if (format_mismatch) { + float_to_s16ne(stream->context->in_resampled_interleaved_buffer_s16ne, in_buffer, nframes * stream->in_params.channels); + } else { + memset(stream->context->in_resampled_interleaved_buffer_float, 0, (FIFO_SIZE * MAX_CHANNELS * 3) * sizeof(float)); + memcpy(stream->context->in_resampled_interleaved_buffer_float, in_buffer, (FIFO_SIZE * MAX_CHANNELS * 2) * sizeof(float)); + } +} + +static void +silent_jack_error_callback(char const * /*msg*/) +{ +} + +/*static*/ int +jack_init (cubeb ** context, char const * context_name) +{ + int r; + + *context = NULL; + + cubeb * ctx = (cubeb *)calloc(1, sizeof(*ctx)); + if (ctx == NULL) { + return CUBEB_ERROR; + } + + r = load_jack_lib(ctx); + if (r != 0) { + cbjack_destroy(ctx); + return CUBEB_ERROR; + } + + api_jack_set_error_function(silent_jack_error_callback); + api_jack_set_info_function(silent_jack_error_callback); + + ctx->ops = &cbjack_ops; + + ctx->mutex = PTHREAD_MUTEX_INITIALIZER; + for (r = 0; r < MAX_STREAMS; r++) { + ctx->streams[r].mutex = PTHREAD_MUTEX_INITIALIZER; + } + + const char * jack_client_name = "cubeb"; + if (context_name) + jack_client_name = context_name; + + ctx->jack_client = api_jack_client_open(jack_client_name, + JackNoStartServer, + NULL); + + if (ctx->jack_client == NULL) { + cbjack_destroy(ctx); + return CUBEB_ERROR; + } + + ctx->jack_xruns = 0; + + api_jack_set_process_callback (ctx->jack_client, cbjack_process, ctx); + api_jack_set_xrun_callback (ctx->jack_client, cbjack_xrun_callback, ctx); + api_jack_set_graph_order_callback (ctx->jack_client, cbjack_graph_order_callback, ctx); + + if (api_jack_activate (ctx->jack_client)) { + cbjack_destroy(ctx); + return CUBEB_ERROR; + } + + ctx->jack_sample_rate = api_jack_get_sample_rate(ctx->jack_client); + ctx->jack_latency = 128 * 1000 / ctx->jack_sample_rate; + + ctx->active = true; + *context = ctx; + + return CUBEB_OK; +} + +static char const * +cbjack_get_backend_id(cubeb * /*context*/) +{ + return "jack"; +} + +static int +cbjack_get_max_channel_count(cubeb * /*ctx*/, uint32_t * max_channels) +{ + *max_channels = MAX_CHANNELS; + return CUBEB_OK; +} + +static int +cbjack_get_latency(cubeb_stream * stm, unsigned int * latency_ms) +{ + *latency_ms = stm->context->jack_latency; + return CUBEB_OK; +} + +static int +cbjack_get_min_latency(cubeb * ctx, cubeb_stream_params /*params*/, uint32_t * latency_ms) +{ + *latency_ms = ctx->jack_latency; + return CUBEB_OK; +} + +static int +cbjack_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) +{ + if (!ctx->jack_client) { + jack_client_t * testclient = api_jack_client_open("test-samplerate", + JackNoStartServer, + NULL); + if (!testclient) { + return CUBEB_ERROR; + } + + *rate = api_jack_get_sample_rate(testclient); + api_jack_client_close(testclient); + + } else { + *rate = api_jack_get_sample_rate(ctx->jack_client); + } + return CUBEB_OK; +} + +static void +cbjack_destroy(cubeb * context) +{ + context->active = false; + + if (context->jack_client != NULL) + api_jack_client_close (context->jack_client); + + if (context->libjack) + dlclose(context->libjack); + + free(context); +} + +static cubeb_stream * +context_alloc_stream(cubeb * context, char const * stream_name) +{ + for (int i = 0; i < MAX_STREAMS; i++) { + if (!context->streams[i].in_use) { + cubeb_stream * stm = &context->streams[i]; + stm->in_use = true; + snprintf(stm->stream_name, 255, "%s_%u", stream_name, i); + return stm; + } + } + return NULL; +} + +static int +cbjack_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int /*latency_frames*/, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + int stream_actual_rate = 0; + int jack_rate = api_jack_get_sample_rate(context->jack_client); + + if (output_stream_params + && (output_stream_params->format != CUBEB_SAMPLE_FLOAT32NE && + output_stream_params->format != CUBEB_SAMPLE_S16NE) + ) { + return CUBEB_ERROR_INVALID_FORMAT; + } + + if (input_stream_params + && (input_stream_params->format != CUBEB_SAMPLE_FLOAT32NE && + input_stream_params->format != CUBEB_SAMPLE_S16NE) + ) { + return CUBEB_ERROR_INVALID_FORMAT; + } + + if ((input_device && input_device != JACK_DEFAULT_IN) || + (output_device && output_device != JACK_DEFAULT_OUT)) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + // Loopback is unsupported + if ((input_stream_params && (input_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK)) || + (output_stream_params && (output_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK))) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + *stream = NULL; + + // Find a free stream. + pthread_mutex_lock(&context->mutex); + cubeb_stream * stm = context_alloc_stream(context, stream_name); + + // No free stream? + if (stm == NULL) { + pthread_mutex_unlock(&context->mutex); + return CUBEB_ERROR; + } + + // unlock context mutex + pthread_mutex_unlock(&context->mutex); + + // Lock active stream + pthread_mutex_lock(&stm->mutex); + + stm->ports_ready = false; + stm->user_ptr = user_ptr; + stm->context = context; + stm->devs = NONE; + if (output_stream_params && !input_stream_params) { + stm->out_params = *output_stream_params; + stream_actual_rate = stm->out_params.rate; + stm->out_params.rate = jack_rate; + stm->devs = OUT_ONLY; + if (stm->out_params.format == CUBEB_SAMPLE_FLOAT32NE) { + context->output_bytes_per_frame = sizeof(float); + } else { + context->output_bytes_per_frame = sizeof(short); + } + } + if (input_stream_params && output_stream_params) { + stm->in_params = *input_stream_params; + stm->out_params = *output_stream_params; + stream_actual_rate = stm->out_params.rate; + stm->in_params.rate = jack_rate; + stm->out_params.rate = jack_rate; + stm->devs = DUPLEX; + if (stm->out_params.format == CUBEB_SAMPLE_FLOAT32NE) { + context->output_bytes_per_frame = sizeof(float); + stm->in_params.format = CUBEB_SAMPLE_FLOAT32NE; + } else { + context->output_bytes_per_frame = sizeof(short); + stm->in_params.format = CUBEB_SAMPLE_S16NE; + } + } else if (input_stream_params && !output_stream_params) { + stm->in_params = *input_stream_params; + stream_actual_rate = stm->in_params.rate; + stm->in_params.rate = jack_rate; + stm->devs = IN_ONLY; + if (stm->in_params.format == CUBEB_SAMPLE_FLOAT32NE) { + context->output_bytes_per_frame = sizeof(float); + } else { + context->output_bytes_per_frame = sizeof(short); + } + } + + stm->ratio = (float)stream_actual_rate / (float)jack_rate; + + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->position = 0; + stm->volume = 1.0f; + context->jack_buffer_size = api_jack_get_buffer_size(context->jack_client); + context->fragment_size = context->jack_buffer_size; + + if (stm->devs == NONE) { + pthread_mutex_unlock(&stm->mutex); + return CUBEB_ERROR; + } + + stm->resampler = NULL; + + if (stm->devs == DUPLEX) { + stm->resampler = cubeb_resampler_create(stm, + &stm->in_params, + &stm->out_params, + stream_actual_rate, + stm->data_callback, + stm->user_ptr, + CUBEB_RESAMPLER_QUALITY_DESKTOP); + } else if (stm->devs == IN_ONLY) { + stm->resampler = cubeb_resampler_create(stm, + &stm->in_params, + nullptr, + stream_actual_rate, + stm->data_callback, + stm->user_ptr, + CUBEB_RESAMPLER_QUALITY_DESKTOP); + } else if (stm->devs == OUT_ONLY) { + stm->resampler = cubeb_resampler_create(stm, + nullptr, + &stm->out_params, + stream_actual_rate, + stm->data_callback, + stm->user_ptr, + CUBEB_RESAMPLER_QUALITY_DESKTOP); + } + + if (!stm->resampler) { + stm->in_use = false; + pthread_mutex_unlock(&stm->mutex); + return CUBEB_ERROR; + } + + if (stm->devs == DUPLEX || stm->devs == OUT_ONLY) { + for (unsigned int c = 0; c < stm->out_params.channels; c++) { + char portname[256]; + snprintf(portname, 255, "%s_out_%d", stm->stream_name, c); + stm->output_ports[c] = api_jack_port_register(stm->context->jack_client, + portname, + JACK_DEFAULT_AUDIO_TYPE, + JackPortIsOutput, + 0); + } + } + + if (stm->devs == DUPLEX || stm->devs == IN_ONLY) { + for (unsigned int c = 0; c < stm->in_params.channels; c++) { + char portname[256]; + snprintf(portname, 255, "%s_in_%d", stm->stream_name, c); + stm->input_ports[c] = api_jack_port_register(stm->context->jack_client, + portname, + JACK_DEFAULT_AUDIO_TYPE, + JackPortIsInput, + 0); + } + } + + if (cbjack_connect_ports(stm) != CUBEB_OK) { + pthread_mutex_unlock(&stm->mutex); + cbjack_stream_destroy(stm); + return CUBEB_ERROR; + } + + *stream = stm; + + stm->ports_ready = true; + stm->pause = true; + pthread_mutex_unlock(&stm->mutex); + + return CUBEB_OK; +} + +static void +cbjack_stream_destroy(cubeb_stream * stream) +{ + pthread_mutex_lock(&stream->mutex); + stream->ports_ready = false; + + if (stream->devs == DUPLEX || stream->devs == OUT_ONLY) { + for (unsigned int c = 0; c < stream->out_params.channels; c++) { + if (stream->output_ports[c]) { + api_jack_port_unregister (stream->context->jack_client, stream->output_ports[c]); + stream->output_ports[c] = NULL; + } + } + } + + if (stream->devs == DUPLEX || stream->devs == IN_ONLY) { + for (unsigned int c = 0; c < stream->in_params.channels; c++) { + if (stream->input_ports[c]) { + api_jack_port_unregister (stream->context->jack_client, stream->input_ports[c]); + stream->input_ports[c] = NULL; + } + } + } + + if (stream->resampler) { + cubeb_resampler_destroy(stream->resampler); + stream->resampler = NULL; + } + stream->in_use = false; + pthread_mutex_unlock(&stream->mutex); +} + +static int +cbjack_stream_start(cubeb_stream * stream) +{ + stream->pause = false; + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_STARTED); + return CUBEB_OK; +} + +static int +cbjack_stream_stop(cubeb_stream * stream) +{ + stream->pause = true; + stream->state_callback(stream, stream->user_ptr, CUBEB_STATE_STOPPED); + return CUBEB_OK; +} + +static int +cbjack_stream_get_position(cubeb_stream * stream, uint64_t * position) +{ + *position = stream->position; + return CUBEB_OK; +} + +static int +cbjack_stream_set_volume(cubeb_stream * stm, float volume) +{ + stm->volume = volume; + return CUBEB_OK; +} + +static int +cbjack_stream_get_current_device(cubeb_stream * stm, cubeb_device ** const device) +{ + *device = (cubeb_device *)calloc(1, sizeof(cubeb_device)); + if (*device == NULL) + return CUBEB_ERROR; + + const char * j_in = JACK_DEFAULT_IN; + const char * j_out = JACK_DEFAULT_OUT; + const char * empty = ""; + + if (stm->devs == DUPLEX) { + (*device)->input_name = strdup(j_in); + (*device)->output_name = strdup(j_out); + } else if (stm->devs == IN_ONLY) { + (*device)->input_name = strdup(j_in); + (*device)->output_name = strdup(empty); + } else if (stm->devs == OUT_ONLY) { + (*device)->input_name = strdup(empty); + (*device)->output_name = strdup(j_out); + } + + return CUBEB_OK; +} + +static int +cbjack_stream_device_destroy(cubeb_stream * /*stream*/, + cubeb_device * device) +{ + if (device->input_name) + free(device->input_name); + if (device->output_name) + free(device->output_name); + free(device); + return CUBEB_OK; +} + +static int +cbjack_enumerate_devices(cubeb * context, cubeb_device_type type, + cubeb_device_collection * collection) +{ + if (!context) + return CUBEB_ERROR; + + uint32_t rate; + cbjack_get_preferred_sample_rate(context, &rate); + + cubeb_device_info * devices = new cubeb_device_info[2]; + if (!devices) + return CUBEB_ERROR; + PodZero(devices, 2); + collection->count = 0; + + if (type & CUBEB_DEVICE_TYPE_OUTPUT) { + cubeb_device_info * cur = &devices[collection->count]; + cur->device_id = JACK_DEFAULT_OUT; + cur->devid = (cubeb_devid) cur->device_id; + cur->friendly_name = JACK_DEFAULT_OUT; + cur->group_id = JACK_DEFAULT_OUT; + cur->vendor_name = JACK_DEFAULT_OUT; + cur->type = CUBEB_DEVICE_TYPE_OUTPUT; + cur->state = CUBEB_DEVICE_STATE_ENABLED; + cur->preferred = CUBEB_DEVICE_PREF_ALL; + cur->format = CUBEB_DEVICE_FMT_F32NE; + cur->default_format = CUBEB_DEVICE_FMT_F32NE; + cur->max_channels = MAX_CHANNELS; + cur->min_rate = rate; + cur->max_rate = rate; + cur->default_rate = rate; + cur->latency_lo = 0; + cur->latency_hi = 0; + collection->count +=1 ; + } + + if (type & CUBEB_DEVICE_TYPE_INPUT) { + cubeb_device_info * cur = &devices[collection->count]; + cur->device_id = JACK_DEFAULT_IN; + cur->devid = (cubeb_devid) cur->device_id; + cur->friendly_name = JACK_DEFAULT_IN; + cur->group_id = JACK_DEFAULT_IN; + cur->vendor_name = JACK_DEFAULT_IN; + cur->type = CUBEB_DEVICE_TYPE_INPUT; + cur->state = CUBEB_DEVICE_STATE_ENABLED; + cur->preferred = CUBEB_DEVICE_PREF_ALL; + cur->format = CUBEB_DEVICE_FMT_F32NE; + cur->default_format = CUBEB_DEVICE_FMT_F32NE; + cur->max_channels = MAX_CHANNELS; + cur->min_rate = rate; + cur->max_rate = rate; + cur->default_rate = rate; + cur->latency_lo = 0; + cur->latency_hi = 0; + collection->count += 1; + } + + collection->device = devices; + + return CUBEB_OK; +} + +static int +cbjack_device_collection_destroy(cubeb * /*ctx*/, + cubeb_device_collection * collection) +{ + XASSERT(collection); + delete [] collection->device; + return CUBEB_OK; +} diff --git a/dep/cubeb/src/cubeb_kai.c b/dep/cubeb/src/cubeb_kai.c new file mode 100644 index 000000000..5333968c8 --- /dev/null +++ b/dep/cubeb/src/cubeb_kai.c @@ -0,0 +1,369 @@ +/* + * Copyright © 2015 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#include +#include +#include +#include + +#include + +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" + +/* We don't support more than 2 channels in KAI */ +#define MAX_CHANNELS 2 + +#define NBUFS 2 +#define FRAME_SIZE 2048 + +struct cubeb_stream_item { + cubeb_stream * stream; +}; + +static struct cubeb_ops const kai_ops; + +struct cubeb { + struct cubeb_ops const * ops; +}; + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr; + /**/ + cubeb_stream_params params; + cubeb_data_callback data_callback; + cubeb_state_callback state_callback; + + HKAI hkai; + KAISPEC spec; + uint64_t total_frames; + float soft_volume; + _fmutex mutex; + float float_buffer[FRAME_SIZE * MAX_CHANNELS]; +}; + +static inline long +frames_to_bytes(long frames, cubeb_stream_params params) +{ + return frames * 2 * params.channels; /* 2 bytes per frame */ +} + +static inline long +bytes_to_frames(long bytes, cubeb_stream_params params) +{ + return bytes / 2 / params.channels; /* 2 bytes per frame */ +} + +static void kai_destroy(cubeb * ctx); + +/*static*/ int +kai_init(cubeb ** context, char const * context_name) +{ + cubeb * ctx; + + XASSERT(context); + *context = NULL; + + if (kaiInit(KAIM_AUTO)) + return CUBEB_ERROR; + + ctx = calloc(1, sizeof(*ctx)); + XASSERT(ctx); + + ctx->ops = &kai_ops; + + *context = ctx; + + return CUBEB_OK; +} + +static char const * +kai_get_backend_id(cubeb * ctx) +{ + return "kai"; +} + +static void +kai_destroy(cubeb * ctx) +{ + kaiDone(); + + free(ctx); +} + +static void +float_to_s16ne(int16_t *dst, float *src, size_t n) +{ + long l; + + while (n--) { + l = lrintf(*src++ * 0x8000); + if (l > 32767) + l = 32767; + if (l < -32768) + l = -32768; + *dst++ = (int16_t)l; + } +} + +static ULONG APIENTRY +kai_callback(PVOID cbdata, PVOID buffer, ULONG len) +{ + cubeb_stream * stm = cbdata; + void *p; + long wanted_frames; + long frames; + float soft_volume; + int elements = len / sizeof(int16_t); + + p = stm->params.format == CUBEB_SAMPLE_FLOAT32NE + ? stm->float_buffer : buffer; + + wanted_frames = bytes_to_frames(len, stm->params); + frames = stm->data_callback(stm, stm->user_ptr, NULL, p, wanted_frames); + + _fmutex_request(&stm->mutex, 0); + stm->total_frames += frames; + soft_volume = stm->soft_volume; + _fmutex_release(&stm->mutex); + + if (frames < wanted_frames) + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + + if (stm->params.format == CUBEB_SAMPLE_FLOAT32NE) + float_to_s16ne(buffer, p, elements); + + if (soft_volume != -1.0f) { + int16_t *b = buffer; + int i; + + for (i = 0; i < elements; i++) + *b++ *= soft_volume; + } + + return frames_to_bytes(frames, stm->params); +} + +static void kai_stream_destroy(cubeb_stream * stm); + +static int +kai_stream_init(cubeb * context, cubeb_stream ** stream, + char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency, cubeb_data_callback data_callback, + cubeb_state_callback state_callback, void * user_ptr) +{ + cubeb_stream * stm; + KAISPEC wanted_spec; + + XASSERT(!input_stream_params && "not supported."); + if (input_device || output_device) { + /* Device selection not yet implemented. */ + return CUBEB_ERROR_DEVICE_UNAVAILABLE; + } + + if (!output_stream_params) + return CUBEB_ERROR_INVALID_PARAMETER; + + // Loopback is unsupported + if (output_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + if (output_stream_params->channels < 1 || + output_stream_params->channels > MAX_CHANNELS) + return CUBEB_ERROR_INVALID_FORMAT; + + XASSERT(context); + XASSERT(stream); + + *stream = NULL; + + stm = calloc(1, sizeof(*stm)); + XASSERT(stm); + + stm->context = context; + stm->params = *output_stream_params; + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + stm->soft_volume = -1.0f; + + if (_fmutex_create(&stm->mutex, 0)) { + free(stm); + return CUBEB_ERROR; + } + + wanted_spec.usDeviceIndex = 0; + wanted_spec.ulType = KAIT_PLAY; + wanted_spec.ulBitsPerSample = BPS_16; + wanted_spec.ulSamplingRate = stm->params.rate; + wanted_spec.ulDataFormat = MCI_WAVE_FORMAT_PCM; + wanted_spec.ulChannels = stm->params.channels; + wanted_spec.ulNumBuffers = NBUFS; + wanted_spec.ulBufferSize = frames_to_bytes(FRAME_SIZE, stm->params); + wanted_spec.fShareable = TRUE; + wanted_spec.pfnCallBack = kai_callback; + wanted_spec.pCallBackData = stm; + + if (kaiOpen(&wanted_spec, &stm->spec, &stm->hkai)) { + _fmutex_close(&stm->mutex); + free(stm); + return CUBEB_ERROR; + } + + *stream = stm; + + return CUBEB_OK; +} + +static void +kai_stream_destroy(cubeb_stream * stm) +{ + kaiClose(stm->hkai); + _fmutex_close(&stm->mutex); + free(stm); +} + +static int +kai_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + XASSERT(ctx && max_channels); + + *max_channels = MAX_CHANNELS; + + return CUBEB_OK; +} + +static int +kai_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency) +{ + /* We have at least two buffers. One is being played, the other one is being + filled. So there is as much latency as one buffer. */ + *latency = FRAME_SIZE; + + return CUBEB_OK; +} + +static int +kai_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) +{ + cubeb_stream_params params; + KAISPEC wanted_spec; + KAISPEC spec; + HKAI hkai; + + params.format = CUBEB_SAMPLE_S16NE; + params.rate = 48000; + params.channels = 2; + + wanted_spec.usDeviceIndex = 0; + wanted_spec.ulType = KAIT_PLAY; + wanted_spec.ulBitsPerSample = BPS_16; + wanted_spec.ulSamplingRate = params.rate; + wanted_spec.ulDataFormat = MCI_WAVE_FORMAT_PCM; + wanted_spec.ulChannels = params.channels; + wanted_spec.ulNumBuffers = NBUFS; + wanted_spec.ulBufferSize = frames_to_bytes(FRAME_SIZE, params); + wanted_spec.fShareable = TRUE; + wanted_spec.pfnCallBack = kai_callback; + wanted_spec.pCallBackData = NULL; + + /* Test 48KHz */ + if (kaiOpen(&wanted_spec, &spec, &hkai)) { + /* Not supported. Fall back to 44.1KHz */ + params.rate = 44100; + } else { + /* Supported. Use 48KHz */ + kaiClose(hkai); + } + + *rate = params.rate; + + return CUBEB_OK; +} + +static int +kai_stream_start(cubeb_stream * stm) +{ + if (kaiPlay(stm->hkai)) + return CUBEB_ERROR; + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED); + + return CUBEB_OK; +} + +static int +kai_stream_stop(cubeb_stream * stm) +{ + if (kaiStop(stm->hkai)) + return CUBEB_ERROR; + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED); + + return CUBEB_OK; +} + +static int +kai_stream_get_position(cubeb_stream * stm, uint64_t * position) +{ + _fmutex_request(&stm->mutex, 0); + *position = stm->total_frames; + _fmutex_release(&stm->mutex); + + return CUBEB_OK; +} + +static int +kai_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ + /* Out of buffers, one is being played, the others are being filled. + So there is as much latency as total buffers - 1. */ + *latency = bytes_to_frames(stm->spec.ulBufferSize, stm->params) + * (stm->spec.ulNumBuffers - 1); + + return CUBEB_OK; +} + +static int +kai_stream_set_volume(cubeb_stream * stm, float volume) +{ + _fmutex_request(&stm->mutex, 0); + stm->soft_volume = volume; + _fmutex_release(&stm->mutex); + + return CUBEB_OK; +} + +static struct cubeb_ops const kai_ops = { + /*.init =*/ kai_init, + /*.get_backend_id =*/ kai_get_backend_id, + /*.get_max_channel_count=*/ kai_get_max_channel_count, + /*.get_min_latency=*/ kai_get_min_latency, + /*.get_preferred_sample_rate =*/ kai_get_preferred_sample_rate, + /*.get_preferred_channel_layout =*/ NULL, + /*.enumerate_devices =*/ NULL, + /*.device_collection_destroy =*/ NULL, + /*.destroy =*/ kai_destroy, + /*.stream_init =*/ kai_stream_init, + /*.stream_destroy =*/ kai_stream_destroy, + /*.stream_start =*/ kai_stream_start, + /*.stream_stop =*/ kai_stream_stop, + /*.stream_reset_default_device =*/ NULL, + /*.stream_get_position =*/ kai_stream_get_position, + /*.stream_get_latency = */ kai_stream_get_latency, + /*.stream_set_volume =*/ kai_stream_set_volume, + /*.stream_get_current_device =*/ NULL, + /*.stream_device_destroy =*/ NULL, + /*.stream_register_device_changed_callback=*/ NULL, + /*.register_device_collection_changed=*/ NULL +}; diff --git a/dep/cubeb/src/cubeb_log.cpp b/dep/cubeb/src/cubeb_log.cpp new file mode 100644 index 000000000..54c7f4a15 --- /dev/null +++ b/dep/cubeb/src/cubeb_log.cpp @@ -0,0 +1,144 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#define NOMINMAX + +#include "cubeb_log.h" +#include "cubeb_ringbuffer.h" +#include +#ifdef _WIN32 +#include +#else +#include +#endif + +cubeb_log_level g_cubeb_log_level; +cubeb_log_callback g_cubeb_log_callback; + +/** The maximum size of a log message, after having been formatted. */ +const size_t CUBEB_LOG_MESSAGE_MAX_SIZE = 256; +/** The maximum number of log messages that can be queued before dropping + * messages. */ +const size_t CUBEB_LOG_MESSAGE_QUEUE_DEPTH = 40; +/** Number of milliseconds to wait before dequeuing log messages. */ +#define CUBEB_LOG_BATCH_PRINT_INTERVAL_MS 10 + +/** + * This wraps an inline buffer, that represents a log message, that must be + * null-terminated. + * This class should not use system calls or other potentially blocking code. + */ +class cubeb_log_message +{ +public: + cubeb_log_message() + { + *storage = '\0'; + } + cubeb_log_message(char const str[CUBEB_LOG_MESSAGE_MAX_SIZE]) + { + size_t length = strlen(str); + /* paranoia against malformed message */ + assert(length < CUBEB_LOG_MESSAGE_MAX_SIZE); + if (length > CUBEB_LOG_MESSAGE_MAX_SIZE - 1) { + return; + } + PodCopy(storage, str, length); + storage[length] = '\0'; + } + char const * get() { + return storage; + } +private: + char storage[CUBEB_LOG_MESSAGE_MAX_SIZE]; +}; + +/** Lock-free asynchronous logger, made so that logging from a + * real-time audio callback does not block the audio thread. */ +class cubeb_async_logger +{ +public: + /* This is thread-safe since C++11 */ + static cubeb_async_logger & get() { + static cubeb_async_logger instance; + return instance; + } + void push(char const str[CUBEB_LOG_MESSAGE_MAX_SIZE]) + { + cubeb_log_message msg(str); + msg_queue.enqueue(msg); + } + void run() + { + std::thread([this]() { + while (true) { + cubeb_log_message msg; + while (msg_queue.dequeue(&msg, 1)) { + LOGV("%s", msg.get()); + } +#ifdef _WIN32 + Sleep(CUBEB_LOG_BATCH_PRINT_INTERVAL_MS); +#else + timespec sleep_duration = sleep_for; + timespec remainder; + do { + if (nanosleep(&sleep_duration, &remainder) == 0 || + errno != EINTR) { + break; + } + sleep_duration = remainder; + } while (remainder.tv_sec || remainder.tv_nsec); +#endif + } + }).detach(); + } + // Tell the underlying queue the producer thread has changed, so it does not + // assert in debug. This should be called with the thread stopped. + void reset_producer_thread() + { + msg_queue.reset_thread_ids(); + } +private: +#ifndef _WIN32 + const struct timespec sleep_for = { + CUBEB_LOG_BATCH_PRINT_INTERVAL_MS/1000, + (CUBEB_LOG_BATCH_PRINT_INTERVAL_MS%1000)*1000*1000 + }; +#endif + cubeb_async_logger() + : msg_queue(CUBEB_LOG_MESSAGE_QUEUE_DEPTH) + { + run(); + } + /** This is quite a big data structure, but is only instantiated if the + * asynchronous logger is used.*/ + lock_free_queue msg_queue; +}; + + +void cubeb_async_log(char const * fmt, ...) +{ + if (!g_cubeb_log_callback) { + return; + } + // This is going to copy a 256 bytes array around, which is fine. + // We don't want to allocate memory here, because this is made to + // be called from a real-time callback. + va_list args; + va_start(args, fmt); + char msg[CUBEB_LOG_MESSAGE_MAX_SIZE]; + vsnprintf(msg, CUBEB_LOG_MESSAGE_MAX_SIZE, fmt, args); + cubeb_async_logger::get().push(msg); + va_end(args); +} + +void cubeb_async_log_reset_threads() +{ + if (!g_cubeb_log_callback) { + return; + } + cubeb_async_logger::get().reset_producer_thread(); +} diff --git a/dep/cubeb/src/cubeb_log.h b/dep/cubeb/src/cubeb_log.h new file mode 100644 index 000000000..a79976bb3 --- /dev/null +++ b/dep/cubeb/src/cubeb_log.h @@ -0,0 +1,47 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef CUBEB_LOG +#define CUBEB_LOG + +#include "cubeb/cubeb.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#if defined(__GNUC__) || defined(__clang__) +#define PRINTF_FORMAT(fmt, args) __attribute__((format(printf, fmt, args))) +#else +#define PRINTF_FORMAT(fmt, args) +#endif + +extern cubeb_log_level g_cubeb_log_level; +extern cubeb_log_callback g_cubeb_log_callback PRINTF_FORMAT(1, 2); +void cubeb_async_log(const char * fmt, ...); +void cubeb_async_log_reset_threads(); + +#ifdef __cplusplus +} +#endif + +#define LOGV(msg, ...) LOG_INTERNAL(CUBEB_LOG_VERBOSE, msg, ##__VA_ARGS__) +#define LOG(msg, ...) LOG_INTERNAL(CUBEB_LOG_NORMAL, msg, ##__VA_ARGS__) + +#define LOG_INTERNAL(level, fmt, ...) do { \ + if (g_cubeb_log_callback && level <= g_cubeb_log_level) { \ + g_cubeb_log_callback("%s:%d: " fmt "\n", __FILE__, __LINE__, ##__VA_ARGS__); \ + } \ + } while(0) + +/* Asynchronous verbose logging, to log in real-time callbacks. */ +#define ALOGV(fmt, ...) \ +do { \ + cubeb_async_log(fmt, ##__VA_ARGS__); \ +} while(0) + +#endif // CUBEB_LOG diff --git a/dep/cubeb/src/cubeb_mixer.cpp b/dep/cubeb/src/cubeb_mixer.cpp new file mode 100644 index 000000000..2ab7f673a --- /dev/null +++ b/dep/cubeb/src/cubeb_mixer.cpp @@ -0,0 +1,663 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + * + * Adapted from code based on libswresample's rematrix.c + */ + +#define NOMINMAX + +#include +#include +#include +#include +#include +#include +#include +#include "cubeb-internal.h" +#include "cubeb_mixer.h" +#include "cubeb_utils.h" + +#ifndef FF_ARRAY_ELEMS +#define FF_ARRAY_ELEMS(a) (sizeof(a) / sizeof((a)[0])) +#endif + +#define CHANNELS_MAX 32 +#define FRONT_LEFT 0 +#define FRONT_RIGHT 1 +#define FRONT_CENTER 2 +#define LOW_FREQUENCY 3 +#define BACK_LEFT 4 +#define BACK_RIGHT 5 +#define FRONT_LEFT_OF_CENTER 6 +#define FRONT_RIGHT_OF_CENTER 7 +#define BACK_CENTER 8 +#define SIDE_LEFT 9 +#define SIDE_RIGHT 10 +#define TOP_CENTER 11 +#define TOP_FRONT_LEFT 12 +#define TOP_FRONT_CENTER 13 +#define TOP_FRONT_RIGHT 14 +#define TOP_BACK_LEFT 15 +#define TOP_BACK_CENTER 16 +#define TOP_BACK_RIGHT 17 +#define NUM_NAMED_CHANNELS 18 + +#ifndef M_SQRT1_2 +#define M_SQRT1_2 0.70710678118654752440 /* 1/sqrt(2) */ +#endif +#ifndef M_SQRT2 +#define M_SQRT2 1.41421356237309504880 /* sqrt(2) */ +#endif +#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */ + +#define C30DB M_SQRT2 +#define C15DB 1.189207115 +#define C__0DB 1.0 +#define C_15DB 0.840896415 +#define C_30DB M_SQRT1_2 +#define C_45DB 0.594603558 +#define C_60DB 0.5 + +static cubeb_channel_layout +cubeb_channel_layout_check(cubeb_channel_layout l, uint32_t c) +{ + if (l == CUBEB_LAYOUT_UNDEFINED) { + switch (c) { + case 1: return CUBEB_LAYOUT_MONO; + case 2: return CUBEB_LAYOUT_STEREO; + } + } + return l; +} + +unsigned int cubeb_channel_layout_nb_channels(cubeb_channel_layout x) +{ +#if __GNUC__ || __clang__ + return __builtin_popcount (x); +#else + x -= (x >> 1) & 0x55555555; + x = (x & 0x33333333) + ((x >> 2) & 0x33333333); + x = (x + (x >> 4)) & 0x0F0F0F0F; + x += x >> 8; + return (x + (x >> 16)) & 0x3F; +#endif +} + +struct MixerContext { + MixerContext(cubeb_sample_format f, + uint32_t in_channels, + cubeb_channel_layout in, + uint32_t out_channels, + cubeb_channel_layout out) + : _format(f) + , _in_ch_layout(cubeb_channel_layout_check(in, in_channels)) + , _out_ch_layout(cubeb_channel_layout_check(out, out_channels)) + , _in_ch_count(in_channels) + , _out_ch_count(out_channels) + { + if (in_channels != cubeb_channel_layout_nb_channels(in) || + out_channels != cubeb_channel_layout_nb_channels(out)) { + // Mismatch between channels and layout, aborting. + return; + } + _valid = init() >= 0; + } + + static bool even(cubeb_channel_layout layout) + { + if (!layout) { + return true; + } + if (layout & (layout - 1)) { + return true; + } + return false; + } + + // Ensure that the layout is sane (that is have symmetrical left/right + // channels), if not, layout will be treated as mono. + static cubeb_channel_layout clean_layout(cubeb_channel_layout layout) + { + if (layout && layout != CHANNEL_FRONT_LEFT && !(layout & (layout - 1))) { + LOG("Treating layout as mono"); + return CHANNEL_FRONT_CENTER; + } + + return layout; + } + + static bool sane_layout(cubeb_channel_layout layout) + { + if (!(layout & CUBEB_LAYOUT_3F)) { // at least 1 front speaker + return false; + } + if (!even(layout & (CHANNEL_FRONT_LEFT | + CHANNEL_FRONT_RIGHT))) { // no asymetric front + return false; + } + if (!even(layout & + (CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT))) { // no asymetric side + return false; + } + if (!even(layout & (CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT))) { + return false; + } + if (!even(layout & + (CHANNEL_FRONT_LEFT_OF_CENTER | CHANNEL_FRONT_RIGHT_OF_CENTER))) { + return false; + } + if (cubeb_channel_layout_nb_channels(layout) >= CHANNELS_MAX) { + return false; + } + return true; + } + + int auto_matrix(); + int init(); + + const cubeb_sample_format _format; + const cubeb_channel_layout _in_ch_layout; ///< input channel layout + const cubeb_channel_layout _out_ch_layout; ///< output channel layout + const uint32_t _in_ch_count; ///< input channel count + const uint32_t _out_ch_count; ///< output channel count + const float _surround_mix_level = C_30DB; ///< surround mixing level + const float _center_mix_level = C_30DB; ///< center mixing level + const float _lfe_mix_level = 1; ///< LFE mixing level + double _matrix[CHANNELS_MAX][CHANNELS_MAX] = {{ 0 }}; ///< floating point rematrixing coefficients + float _matrix_flt[CHANNELS_MAX][CHANNELS_MAX] = {{ 0 }}; ///< single precision floating point rematrixing coefficients + int32_t _matrix32[CHANNELS_MAX][CHANNELS_MAX] = {{ 0 }}; ///< 17.15 fixed point rematrixing coefficients + uint8_t _matrix_ch[CHANNELS_MAX][CHANNELS_MAX+1] = {{ 0 }}; ///< Lists of input channels per output channel that have non zero rematrixing coefficients + bool _clipping = false; ///< Set to true if clipping detection is required + bool _valid = false; ///< Set to true if context is valid. +}; + +int MixerContext::auto_matrix() +{ + double matrix[NUM_NAMED_CHANNELS][NUM_NAMED_CHANNELS] = { { 0 } }; + double maxcoef = 0; + float maxval; + + cubeb_channel_layout in_ch_layout = clean_layout(_in_ch_layout); + cubeb_channel_layout out_ch_layout = clean_layout(_out_ch_layout); + + if (!sane_layout(in_ch_layout)) { + // Channel Not Supported + LOG("Input Layout %x is not supported", _in_ch_layout); + return -1; + } + + if (!sane_layout(out_ch_layout)) { + LOG("Output Layout %x is not supported", _out_ch_layout); + return -1; + } + + for (uint32_t i = 0; i < FF_ARRAY_ELEMS(matrix); i++) { + if (in_ch_layout & out_ch_layout & (1U << i)) { + matrix[i][i] = 1.0; + } + } + + cubeb_channel_layout unaccounted = in_ch_layout & ~out_ch_layout; + + // Rematrixing is done via a matrix of coefficient that should be applied to + // all channels. Channels are treated as pair and must be symmetrical (if a + // left channel exists, the corresponding right should exist too) unless the + // output layout has similar layout. Channels are then mixed toward the front + // center or back center if they exist with a slight bias toward the front. + + if (unaccounted & CHANNEL_FRONT_CENTER) { + if ((out_ch_layout & CUBEB_LAYOUT_STEREO) == CUBEB_LAYOUT_STEREO) { + if (in_ch_layout & CUBEB_LAYOUT_STEREO) { + matrix[FRONT_LEFT][FRONT_CENTER] += _center_mix_level; + matrix[FRONT_RIGHT][FRONT_CENTER] += _center_mix_level; + } else { + matrix[FRONT_LEFT][FRONT_CENTER] += M_SQRT1_2; + matrix[FRONT_RIGHT][FRONT_CENTER] += M_SQRT1_2; + } + } + } + if (unaccounted & CUBEB_LAYOUT_STEREO) { + if (out_ch_layout & CHANNEL_FRONT_CENTER) { + matrix[FRONT_CENTER][FRONT_LEFT] += M_SQRT1_2; + matrix[FRONT_CENTER][FRONT_RIGHT] += M_SQRT1_2; + if (in_ch_layout & CHANNEL_FRONT_CENTER) + matrix[FRONT_CENTER][FRONT_CENTER] = _center_mix_level * M_SQRT2; + } + } + + if (unaccounted & CHANNEL_BACK_CENTER) { + if (out_ch_layout & CHANNEL_BACK_LEFT) { + matrix[BACK_LEFT][BACK_CENTER] += M_SQRT1_2; + matrix[BACK_RIGHT][BACK_CENTER] += M_SQRT1_2; + } else if (out_ch_layout & CHANNEL_SIDE_LEFT) { + matrix[SIDE_LEFT][BACK_CENTER] += M_SQRT1_2; + matrix[SIDE_RIGHT][BACK_CENTER] += M_SQRT1_2; + } else if (out_ch_layout & CHANNEL_FRONT_LEFT) { + matrix[FRONT_LEFT][BACK_CENTER] += _surround_mix_level * M_SQRT1_2; + matrix[FRONT_RIGHT][BACK_CENTER] += _surround_mix_level * M_SQRT1_2; + } else if (out_ch_layout & CHANNEL_FRONT_CENTER) { + matrix[FRONT_CENTER][BACK_CENTER] += + _surround_mix_level * M_SQRT1_2; + } + } + if (unaccounted & CHANNEL_BACK_LEFT) { + if (out_ch_layout & CHANNEL_BACK_CENTER) { + matrix[BACK_CENTER][BACK_LEFT] += M_SQRT1_2; + matrix[BACK_CENTER][BACK_RIGHT] += M_SQRT1_2; + } else if (out_ch_layout & CHANNEL_SIDE_LEFT) { + if (in_ch_layout & CHANNEL_SIDE_LEFT) { + matrix[SIDE_LEFT][BACK_LEFT] += M_SQRT1_2; + matrix[SIDE_RIGHT][BACK_RIGHT] += M_SQRT1_2; + } else { + matrix[SIDE_LEFT][BACK_LEFT] += 1.0; + matrix[SIDE_RIGHT][BACK_RIGHT] += 1.0; + } + } else if (out_ch_layout & CHANNEL_FRONT_LEFT) { + matrix[FRONT_LEFT][BACK_LEFT] += _surround_mix_level; + matrix[FRONT_RIGHT][BACK_RIGHT] += _surround_mix_level; + } else if (out_ch_layout & CHANNEL_FRONT_CENTER) { + matrix[FRONT_CENTER][BACK_LEFT] += _surround_mix_level * M_SQRT1_2; + matrix[FRONT_CENTER][BACK_RIGHT] += _surround_mix_level * M_SQRT1_2; + } + } + + if (unaccounted & CHANNEL_SIDE_LEFT) { + if (out_ch_layout & CHANNEL_BACK_LEFT) { + /* if back channels do not exist in the input, just copy side + channels to back channels, otherwise mix side into back */ + if (in_ch_layout & CHANNEL_BACK_LEFT) { + matrix[BACK_LEFT][SIDE_LEFT] += M_SQRT1_2; + matrix[BACK_RIGHT][SIDE_RIGHT] += M_SQRT1_2; + } else { + matrix[BACK_LEFT][SIDE_LEFT] += 1.0; + matrix[BACK_RIGHT][SIDE_RIGHT] += 1.0; + } + } else if (out_ch_layout & CHANNEL_BACK_CENTER) { + matrix[BACK_CENTER][SIDE_LEFT] += M_SQRT1_2; + matrix[BACK_CENTER][SIDE_RIGHT] += M_SQRT1_2; + } else if (out_ch_layout & CHANNEL_FRONT_LEFT) { + matrix[FRONT_LEFT][SIDE_LEFT] += _surround_mix_level; + matrix[FRONT_RIGHT][SIDE_RIGHT] += _surround_mix_level; + } else if (out_ch_layout & CHANNEL_FRONT_CENTER) { + matrix[FRONT_CENTER][SIDE_LEFT] += _surround_mix_level * M_SQRT1_2; + matrix[FRONT_CENTER][SIDE_RIGHT] += _surround_mix_level * M_SQRT1_2; + } + } + + if (unaccounted & CHANNEL_FRONT_LEFT_OF_CENTER) { + if (out_ch_layout & CHANNEL_FRONT_LEFT) { + matrix[FRONT_LEFT][FRONT_LEFT_OF_CENTER] += 1.0; + matrix[FRONT_RIGHT][FRONT_RIGHT_OF_CENTER] += 1.0; + } else if (out_ch_layout & CHANNEL_FRONT_CENTER) { + matrix[FRONT_CENTER][FRONT_LEFT_OF_CENTER] += M_SQRT1_2; + matrix[FRONT_CENTER][FRONT_RIGHT_OF_CENTER] += M_SQRT1_2; + } + } + /* mix LFE into front left/right or center */ + if (unaccounted & CHANNEL_LOW_FREQUENCY) { + if (out_ch_layout & CHANNEL_FRONT_CENTER) { + matrix[FRONT_CENTER][LOW_FREQUENCY] += _lfe_mix_level; + } else if (out_ch_layout & CHANNEL_FRONT_LEFT) { + matrix[FRONT_LEFT][LOW_FREQUENCY] += _lfe_mix_level * M_SQRT1_2; + matrix[FRONT_RIGHT][LOW_FREQUENCY] += _lfe_mix_level * M_SQRT1_2; + } + } + + // Normalize the conversion matrix. + for (uint32_t out_i = 0, i = 0; i < CHANNELS_MAX; i++) { + double sum = 0; + int in_i = 0; + if ((out_ch_layout & (1U << i)) == 0) { + continue; + } + for (uint32_t j = 0; j < CHANNELS_MAX; j++) { + if ((in_ch_layout & (1U << j)) == 0) { + continue; + } + if (i < FF_ARRAY_ELEMS(matrix) && j < FF_ARRAY_ELEMS(matrix[0])) { + _matrix[out_i][in_i] = matrix[i][j]; + } else { + _matrix[out_i][in_i] = + i == j && (in_ch_layout & out_ch_layout & (1U << i)); + } + sum += fabs(_matrix[out_i][in_i]); + in_i++; + } + maxcoef = std::max(maxcoef, sum); + out_i++; + } + + if (_format == CUBEB_SAMPLE_S16NE) { + maxval = 1.0; + } else { + maxval = INT_MAX; + } + + // Normalize matrix if needed. + if (maxcoef > maxval) { + maxcoef /= maxval; + for (uint32_t i = 0; i < CHANNELS_MAX; i++) + for (uint32_t j = 0; j < CHANNELS_MAX; j++) { + _matrix[i][j] /= maxcoef; + } + } + + if (_format == CUBEB_SAMPLE_FLOAT32NE) { + for (uint32_t i = 0; i < FF_ARRAY_ELEMS(_matrix); i++) { + for (uint32_t j = 0; j < FF_ARRAY_ELEMS(_matrix[0]); j++) { + _matrix_flt[i][j] = _matrix[i][j]; + } + } + } + + return 0; +} + +int MixerContext::init() +{ + int r = auto_matrix(); + if (r) { + return r; + } + + // Determine if matrix operation would overflow + if (_format == CUBEB_SAMPLE_S16NE) { + int maxsum = 0; + for (uint32_t i = 0; i < _out_ch_count; i++) { + double rem = 0; + int sum = 0; + + for (uint32_t j = 0; j < _in_ch_count; j++) { + double target = _matrix[i][j] * 32768 + rem; + int value = lrintf(target); + rem += target - value; + sum += std::abs(value); + } + maxsum = std::max(maxsum, sum); + } + if (maxsum > 32768) { + _clipping = true; + } + } + + // FIXME quantize for integers + for (uint32_t i = 0; i < CHANNELS_MAX; i++) { + int ch_in = 0; + for (uint32_t j = 0; j < CHANNELS_MAX; j++) { + _matrix32[i][j] = lrintf(_matrix[i][j] * 32768); + if (_matrix[i][j]) { + _matrix_ch[i][++ch_in] = j; + } + } + _matrix_ch[i][0] = ch_in; + } + + return 0; +} + +template +void +sum2(TYPE_SAMPLE * out, + uint32_t stride_out, + const TYPE_SAMPLE * in1, + const TYPE_SAMPLE * in2, + uint32_t stride_in, + TYPE_COEFF coeff1, + TYPE_COEFF coeff2, + F&& operand, + uint32_t frames) +{ + static_assert( + std::is_same::type>::value, + "function must return the same type as used by matrix_coeff"); + for (uint32_t i = 0; i < frames; i++) { + *out = operand(coeff1 * *in1 + coeff2 * *in2); + out += stride_out; + in1 += stride_in; + in2 += stride_in; + } +} + +template +void +copy(TYPE_SAMPLE * out, + uint32_t stride_out, + const TYPE_SAMPLE * in, + uint32_t stride_in, + TYPE_COEFF coeff, + F&& operand, + uint32_t frames) +{ + static_assert( + std::is_same::type>::value, + "function must return the same type as used by matrix_coeff"); + for (uint32_t i = 0; i < frames; i++) { + *out = operand(coeff * *in); + out += stride_out; + in += stride_in; + } +} + +template +static int rematrix(const MixerContext * s, TYPE * aOut, const TYPE * aIn, + const TYPE_COEFF (&matrix_coeff)[COLS][COLS], + F&& aF, uint32_t frames) +{ + static_assert( + std::is_same::type>::value, + "function must return the same type as used by matrix_coeff"); + + for (uint32_t out_i = 0; out_i < s->_out_ch_count; out_i++) { + TYPE* out = aOut + out_i; + switch (s->_matrix_ch[out_i][0]) { + case 0: + for (uint32_t i = 0; i < frames; i++) { + out[i * s->_out_ch_count] = 0; + } + break; + case 1: { + int in_i = s->_matrix_ch[out_i][1]; + copy(out, + s->_out_ch_count, + aIn + in_i, + s->_in_ch_count, + matrix_coeff[out_i][in_i], + aF, + frames); + } break; + case 2: + sum2(out, + s->_out_ch_count, + aIn + s->_matrix_ch[out_i][1], + aIn + s->_matrix_ch[out_i][2], + s->_in_ch_count, + matrix_coeff[out_i][s->_matrix_ch[out_i][1]], + matrix_coeff[out_i][s->_matrix_ch[out_i][2]], + aF, + frames); + break; + default: + for (uint32_t i = 0; i < frames; i++) { + TYPE_COEFF v = 0; + for (uint32_t j = 0; j < s->_matrix_ch[out_i][0]; j++) { + uint32_t in_i = s->_matrix_ch[out_i][1 + j]; + v += + *(aIn + in_i + i * s->_in_ch_count) * matrix_coeff[out_i][in_i]; + } + out[i * s->_out_ch_count] = aF(v); + } + break; + } + } + return 0; +} + +struct cubeb_mixer +{ + cubeb_mixer(cubeb_sample_format format, + uint32_t in_channels, + cubeb_channel_layout in_layout, + uint32_t out_channels, + cubeb_channel_layout out_layout) + : _context(format, in_channels, in_layout, out_channels, out_layout) + { + } + + template + void copy_and_trunc(size_t frames, + const T * input_buffer, + T * output_buffer) const + { + if (_context._in_ch_count <= _context._out_ch_count) { + // Not enough channels to copy, fill the gaps with silence. + if (_context._in_ch_count == 1 && _context._out_ch_count >= 2) { + // Special case for upmixing mono input to stereo and more. We will + // duplicate the mono channel to the first two channels. On most system, + // the first two channels are for left and right. It is commonly + // expected that mono will on both left+right channels + for (uint32_t i = 0; i < frames; i++) { + output_buffer[0] = output_buffer[1] = *input_buffer; + PodZero(output_buffer + 2, _context._out_ch_count - 2); + output_buffer += _context._out_ch_count; + input_buffer++; + } + return; + } + for (uint32_t i = 0; i < frames; i++) { + PodCopy(output_buffer, input_buffer, _context._in_ch_count); + output_buffer += _context._in_ch_count; + input_buffer += _context._in_ch_count; + PodZero(output_buffer, _context._out_ch_count - _context._in_ch_count); + output_buffer += _context._out_ch_count - _context._in_ch_count; + } + } else { + for (uint32_t i = 0; i < frames; i++) { + PodCopy(output_buffer, input_buffer, _context._out_ch_count); + output_buffer += _context._out_ch_count; + input_buffer += _context._in_ch_count; + } + } + } + + int mix(size_t frames, + const void * input_buffer, + size_t input_buffer_size, + void * output_buffer, + size_t output_buffer_size) const + { + if (frames <= 0 || _context._out_ch_count == 0) { + return 0; + } + + // Check if output buffer is of sufficient size. + size_t size_read_needed = + frames * _context._in_ch_count * cubeb_sample_size(_context._format); + if (input_buffer_size < size_read_needed) { + // We don't have enough data to read! + return -1; + } + if (output_buffer_size * _context._in_ch_count < + size_read_needed * _context._out_ch_count) { + return -1; + } + + if (!valid()) { + // The channel layouts were invalid or unsupported, instead we will simply + // either drop the extra channels, or fill with silence the missing ones + if (_context._format == CUBEB_SAMPLE_FLOAT32NE) { + copy_and_trunc(frames, + static_cast(input_buffer), + static_cast(output_buffer)); + } else { + assert(_context._format == CUBEB_SAMPLE_S16NE); + copy_and_trunc(frames, + static_cast(input_buffer), + reinterpret_cast(output_buffer)); + } + return 0; + } + + switch (_context._format) + { + case CUBEB_SAMPLE_FLOAT32NE: { + auto f = [](float x) { return x; }; + return rematrix(&_context, + static_cast(output_buffer), + static_cast(input_buffer), + _context._matrix_flt, + f, + frames); + } + case CUBEB_SAMPLE_S16NE: + if (_context._clipping) { + auto f = [](int x) { + int y = (x + 16384) >> 15; + // clip the signed integer value into the -32768,32767 range. + if ((y + 0x8000U) & ~0xFFFF) { + return (y >> 31) ^ 0x7FFF; + } + return y; + }; + return rematrix(&_context, + static_cast(output_buffer), + static_cast(input_buffer), + _context._matrix32, + f, + frames); + } else { + auto f = [](int x) { return (x + 16384) >> 15; }; + return rematrix(&_context, + static_cast(output_buffer), + static_cast(input_buffer), + _context._matrix32, + f, + frames); + } + break; + default: + assert(false); + break; + } + + return -1; + } + + // Return false if any of the input or ouput layout were invalid. + bool valid() const { return _context._valid; } + + virtual ~cubeb_mixer(){}; + + MixerContext _context; +}; + +cubeb_mixer* cubeb_mixer_create(cubeb_sample_format format, + uint32_t in_channels, + cubeb_channel_layout in_layout, + uint32_t out_channels, + cubeb_channel_layout out_layout) +{ + return new cubeb_mixer( + format, in_channels, in_layout, out_channels, out_layout); +} + +void cubeb_mixer_destroy(cubeb_mixer * mixer) +{ + delete mixer; +} + +int cubeb_mixer_mix(cubeb_mixer * mixer, + size_t frames, + const void * input_buffer, + size_t input_buffer_size, + void * output_buffer, + size_t output_buffer_size) +{ + return mixer->mix( + frames, input_buffer, input_buffer_size, output_buffer, output_buffer_size); +} diff --git a/dep/cubeb/src/cubeb_mixer.h b/dep/cubeb/src/cubeb_mixer.h new file mode 100644 index 000000000..d43a237f9 --- /dev/null +++ b/dep/cubeb/src/cubeb_mixer.h @@ -0,0 +1,37 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef CUBEB_MIXER +#define CUBEB_MIXER + +#include "cubeb/cubeb.h" // for cubeb_channel_layout and cubeb_stream_params. + +#if defined(__cplusplus) +extern "C" { +#endif + +typedef struct cubeb_mixer cubeb_mixer; +cubeb_mixer * cubeb_mixer_create(cubeb_sample_format format, + uint32_t in_channels, + cubeb_channel_layout in_layout, + uint32_t out_channels, + cubeb_channel_layout out_layout); +void cubeb_mixer_destroy(cubeb_mixer * mixer); +int cubeb_mixer_mix(cubeb_mixer * mixer, + size_t frames, + const void * input_buffer, + size_t input_buffer_size, + void * output_buffer, + size_t output_buffer_size); + +unsigned int cubeb_channel_layout_nb_channels(cubeb_channel_layout channel_layout); + +#if defined(__cplusplus) +} +#endif + +#endif // CUBEB_MIXER diff --git a/dep/cubeb/src/cubeb_opensl.c b/dep/cubeb/src/cubeb_opensl.c new file mode 100644 index 000000000..021390e66 --- /dev/null +++ b/dep/cubeb/src/cubeb_opensl.c @@ -0,0 +1,1760 @@ +/* + * Copyright © 2012 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#undef NDEBUG +#include +#include +#include +#include +#include +#include +#include +#include +#if defined(__ANDROID__) +#include +#include +#include "android/sles_definitions.h" +#include +#include +#include +#endif +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" +#include "cubeb_resampler.h" +#include "cubeb-sles.h" +#include "cubeb_array_queue.h" +#include "android/cubeb-output-latency.h" + +#if defined(__ANDROID__) +#ifdef LOG +#undef LOG +#endif +//#define LOGGING_ENABLED +#ifdef LOGGING_ENABLED +#define LOG(args...) __android_log_print(ANDROID_LOG_INFO, "Cubeb_OpenSL" , ## args) +#else +#define LOG(...) +#endif + +//#define TIMESTAMP_ENABLED +#ifdef TIMESTAMP_ENABLED +#define FILENAME (strrchr(__FILE__, '/') ? strrchr(__FILE__, '/') + 1 : __FILE__) +#define LOG_TS(args...) __android_log_print(ANDROID_LOG_INFO, "Cubeb_OpenSL ES: Timestamp(usec)" , ## args) +#define TIMESTAMP(msg) do { \ + struct timeval timestamp; \ + int ts_ret = gettimeofday(×tamp, NULL); \ + if (ts_ret == 0) { \ + LOG_TS("%lld: %s (%s %s:%d)", timestamp.tv_sec * 1000000LL + timestamp.tv_usec, msg, __FUNCTION__, FILENAME, __LINE__);\ + } else { \ + LOG_TS("Error: %s (%s %s:%d) - %s", msg, __FUNCTION__, FILENAME, __LINE__);\ + } \ +} while(0) +#else +#define TIMESTAMP(...) +#endif + +#define ANDROID_VERSION_GINGERBREAD_MR1 10 +#define ANDROID_VERSION_JELLY_BEAN 18 +#define ANDROID_VERSION_LOLLIPOP 21 +#define ANDROID_VERSION_MARSHMALLOW 23 +#define ANDROID_VERSION_N_MR1 25 +#endif + +#define DEFAULT_SAMPLE_RATE 48000 +#define DEFAULT_NUM_OF_FRAMES 480 +// If the latency requested is above this threshold, this stream is considered +// intended for playback (vs. real-time). Tell Android it should favor saving +// power over performance or latency. +// This is around 100ms at 44100 or 48000 +#define POWERSAVE_LATENCY_FRAMES_THRESHOLD 4000 + +static struct cubeb_ops const opensl_ops; + +struct cubeb { + struct cubeb_ops const * ops; + void * lib; + SLInterfaceID SL_IID_BUFFERQUEUE; + SLInterfaceID SL_IID_PLAY; +#if defined(__ANDROID__) + SLInterfaceID SL_IID_ANDROIDCONFIGURATION; + SLInterfaceID SL_IID_ANDROIDSIMPLEBUFFERQUEUE; +#endif + SLInterfaceID SL_IID_VOLUME; + SLInterfaceID SL_IID_RECORD; + SLObjectItf engObj; + SLEngineItf eng; + SLObjectItf outmixObj; + output_latency_function * p_output_latency_function; +}; + +#define NELEMS(A) (sizeof(A) / sizeof A[0]) +#define NBUFS 2 + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr; + /**/ + pthread_mutex_t mutex; + SLObjectItf playerObj; + SLPlayItf play; + SLBufferQueueItf bufq; + SLVolumeItf volume; + void ** queuebuf; + uint32_t queuebuf_capacity; + int queuebuf_idx; + long queuebuf_len; + long bytespersec; + long framesize; + /* Total number of played frames. + * Synchronized by stream::mutex lock. */ + long written; + /* Flag indicating draining. Synchronized + * by stream::mutex lock. */ + int draining; + /* Flags to determine in/out.*/ + uint32_t input_enabled; + uint32_t output_enabled; + /* Recorder abstract object. */ + SLObjectItf recorderObj; + /* Recorder Itf for input capture. */ + SLRecordItf recorderItf; + /* Buffer queue for input capture. */ + SLAndroidSimpleBufferQueueItf recorderBufferQueueItf; + /* Store input buffers. */ + void ** input_buffer_array; + /* The capacity of the array. + * On capture only can be small (4). + * On full duplex is calculated to + * store 1 sec of data buffers. */ + uint32_t input_array_capacity; + /* Current filled index of input buffer array. + * It is initiated to -1 indicating buffering + * have not started yet. */ + int input_buffer_index; + /* Length of input buffer.*/ + uint32_t input_buffer_length; + /* Input frame size */ + uint32_t input_frame_size; + /* Device sampling rate. If user rate is not + * accepted an compatible rate is set. If it is + * accepted this is equal to params.rate. */ + uint32_t input_device_rate; + /* Exchange input buffers between input + * and full duplex threads. */ + array_queue * input_queue; + /* Silent input buffer used on full duplex. */ + void * input_silent_buffer; + /* Number of input frames from the start of the stream*/ + uint32_t input_total_frames; + /* Flag to stop the execution of user callback and + * close all working threads. Synchronized by + * stream::mutex lock. */ + uint32_t shutdown; + /* Store user callback. */ + cubeb_data_callback data_callback; + /* Store state callback. */ + cubeb_state_callback state_callback; + + cubeb_resampler * resampler; + unsigned int user_output_rate; + unsigned int output_configured_rate; + unsigned int buffer_size_frames; + // Audio output latency used in cubeb_stream_get_position(). + unsigned int output_latency_ms; + int64_t lastPosition; + int64_t lastPositionTimeStamp; + int64_t lastCompensativePosition; + int voice; +}; + +/* Forward declaration. */ +static int opensl_stop_player(cubeb_stream * stm); +static int opensl_stop_recorder(cubeb_stream * stm); + +static int +opensl_get_draining(cubeb_stream * stm) +{ +#ifdef DEBUG + int r = pthread_mutex_trylock(&stm->mutex); + assert((r == EDEADLK || r == EBUSY) && "get_draining: mutex should be locked but it's not."); +#endif + return stm->draining; +} + +static void +opensl_set_draining(cubeb_stream * stm, int value) +{ +#ifdef DEBUG + int r = pthread_mutex_trylock(&stm->mutex); + LOG("set draining try r = %d", r); + assert((r == EDEADLK || r == EBUSY) && "set_draining: mutex should be locked but it's not."); +#endif + assert(value == 0 || value == 1); + stm->draining = value; +} + +static void +opensl_notify_drained(cubeb_stream * stm) +{ + assert(stm); + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + int draining = opensl_get_draining(stm); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + if (draining) { + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + if (stm->play) { + LOG("stop player in play_callback"); + r = opensl_stop_player(stm); + assert(r == CUBEB_OK); + } + if (stm->recorderItf) { + r = opensl_stop_recorder(stm); + assert(r == CUBEB_OK); + } + } +} + +static uint32_t +opensl_get_shutdown(cubeb_stream * stm) +{ +#ifdef DEBUG + int r = pthread_mutex_trylock(&stm->mutex); + assert((r == EDEADLK || r == EBUSY) && "get_shutdown: mutex should be locked but it's not."); +#endif + return stm->shutdown; +} + +static void +opensl_set_shutdown(cubeb_stream * stm, uint32_t value) +{ +#ifdef DEBUG + int r = pthread_mutex_trylock(&stm->mutex); + LOG("set shutdown try r = %d", r); + assert((r == EDEADLK || r == EBUSY) && "set_shutdown: mutex should be locked but it's not."); +#endif + assert(value == 0 || value == 1); + stm->shutdown = value; +} + +static void +play_callback(SLPlayItf caller, void * user_ptr, SLuint32 event) +{ + cubeb_stream * stm = user_ptr; + assert(stm); + switch (event) { + case SL_PLAYEVENT_HEADATMARKER: + opensl_notify_drained(stm); + break; + default: + break; + } +} + +static void +recorder_marker_callback (SLRecordItf caller, void * pContext, SLuint32 event) +{ + cubeb_stream * stm = pContext; + assert(stm); + + if (event == SL_RECORDEVENT_HEADATMARKER) { + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + int draining = opensl_get_draining(stm); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + if (draining) { + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + if (stm->recorderItf) { + r = opensl_stop_recorder(stm); + assert(r == CUBEB_OK); + } + if (stm->play) { + r = opensl_stop_player(stm); + assert(r == CUBEB_OK); + } + } + } +} + +static void +bufferqueue_callback(SLBufferQueueItf caller, void * user_ptr) +{ + cubeb_stream * stm = user_ptr; + assert(stm); + SLBufferQueueState state; + SLresult res; + long written = 0; + + res = (*stm->bufq)->GetState(stm->bufq, &state); + assert(res == SL_RESULT_SUCCESS); + + if (state.count > 1) { + return; + } + + uint8_t *buf = stm->queuebuf[stm->queuebuf_idx]; + written = 0; + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + int draining = opensl_get_draining(stm); + uint32_t shutdown = opensl_get_shutdown(stm); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + if (!draining && !shutdown) { + written = cubeb_resampler_fill(stm->resampler, + NULL, NULL, + buf, stm->queuebuf_len / stm->framesize); + LOG("bufferqueue_callback: resampler fill returned %ld frames", written); + if (written < 0 || written * stm->framesize > stm->queuebuf_len) { + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + opensl_set_shutdown(stm, 1); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + opensl_stop_player(stm); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + return; + } + } + + // Keep sending silent data even in draining mode to prevent the audio + // back-end from being stopped automatically by OpenSL/ES. + assert(stm->queuebuf_len >= written * stm->framesize); + memset(buf + written * stm->framesize, 0, stm->queuebuf_len - written * stm->framesize); + res = (*stm->bufq)->Enqueue(stm->bufq, buf, stm->queuebuf_len); + assert(res == SL_RESULT_SUCCESS); + stm->queuebuf_idx = (stm->queuebuf_idx + 1) % stm->queuebuf_capacity; + + if (written > 0) { + pthread_mutex_lock(&stm->mutex); + stm->written += written; + pthread_mutex_unlock(&stm->mutex); + } + + if (!draining && written * stm->framesize < stm->queuebuf_len) { + LOG("bufferqueue_callback draining"); + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + int64_t written_duration = INT64_C(1000) * stm->written * stm->framesize / stm->bytespersec; + opensl_set_draining(stm, 1); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + if (written_duration == 0) { + // since we didn't write any sample, it's not possible to reach the marker + // time and trigger the callback. We should initiative notify drained. + opensl_notify_drained(stm); + } else { + // Use SL_PLAYEVENT_HEADATMARKER event from slPlayCallback of SLPlayItf + // to make sure all the data has been processed. + (*stm->play)->SetMarkerPosition(stm->play, (SLmillisecond)written_duration); + } + return; + } +} + +static int +opensl_enqueue_recorder(cubeb_stream * stm, void ** last_filled_buffer) +{ + assert(stm); + + int current_index = stm->input_buffer_index; + void * last_buffer = NULL; + + if (current_index < 0) { + // This is the first enqueue + current_index = 0; + } else { + // The current index hold the last filled buffer get it before advance index. + last_buffer = stm->input_buffer_array[current_index]; + // Advance to get next available buffer + current_index = (current_index + 1) % stm->input_array_capacity; + } + // enqueue next empty buffer to be filled by the recorder + SLresult res = (*stm->recorderBufferQueueItf)->Enqueue(stm->recorderBufferQueueItf, + stm->input_buffer_array[current_index], + stm->input_buffer_length); + if (res != SL_RESULT_SUCCESS ) { + LOG("Enqueue recorder failed. Error code: %lu", res); + return CUBEB_ERROR; + } + // All good, update buffer and index. + stm->input_buffer_index = current_index; + if (last_filled_buffer) { + *last_filled_buffer = last_buffer; + } + return CUBEB_OK; +} + +// input data callback +void recorder_callback(SLAndroidSimpleBufferQueueItf bq, void * context) +{ + assert(context); + cubeb_stream * stm = context; + assert(stm->recorderBufferQueueItf); + + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + uint32_t shutdown = opensl_get_shutdown(stm); + int draining = opensl_get_draining(stm); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + if (shutdown || draining) { + // According to the OpenSL ES 1.1 Specification, 8.14 SLBufferQueueItf + // page 184, on transition to the SL_RECORDSTATE_STOPPED state, + // the application should continue to enqueue buffers onto the queue + // to retrieve the residual recorded data in the system. + r = opensl_enqueue_recorder(stm, NULL); + assert(r == CUBEB_OK); + return; + } + + // Enqueue next available buffer and get the last filled buffer. + void * input_buffer = NULL; + r = opensl_enqueue_recorder(stm, &input_buffer); + assert(r == CUBEB_OK); + assert(input_buffer); + // Fill resampler with last input + long input_frame_count = stm->input_buffer_length / stm->input_frame_size; + long got = cubeb_resampler_fill(stm->resampler, + input_buffer, + &input_frame_count, + NULL, + 0); + // Error case + if (got < 0 || got > input_frame_count) { + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + opensl_set_shutdown(stm, 1); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + r = opensl_stop_recorder(stm); + assert(r == CUBEB_OK); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + } + + // Advance total stream frames + stm->input_total_frames += got; + + if (got < input_frame_count) { + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + opensl_set_draining(stm, 1); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + int64_t duration = INT64_C(1000) * stm->input_total_frames / stm->input_device_rate; + (*stm->recorderItf)->SetMarkerPosition(stm->recorderItf, (SLmillisecond)duration); + return; + } +} + +void recorder_fullduplex_callback(SLAndroidSimpleBufferQueueItf bq, void * context) +{ + assert(context); + cubeb_stream * stm = context; + assert(stm->recorderBufferQueueItf); + + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + int draining = opensl_get_draining(stm); + uint32_t shutdown = opensl_get_shutdown(stm); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + if (shutdown || draining) { + /* On draining and shutdown the recorder should have been stoped from + * the one set the flags. Accordint to the doc, on transition to + * the SL_RECORDSTATE_STOPPED state, the application should + * continue to enqueue buffers onto the queue to retrieve the residual + * recorded data in the system. */ + LOG("Input shutdown %d or drain %d", shutdown, draining); + int r = opensl_enqueue_recorder(stm, NULL); + assert(r == CUBEB_OK); + return; + } + + // Enqueue next available buffer and get the last filled buffer. + void * input_buffer = NULL; + r = opensl_enqueue_recorder(stm, &input_buffer); + assert(r == CUBEB_OK); + assert(input_buffer); + + assert(stm->input_queue); + r = array_queue_push(stm->input_queue, input_buffer); + if (r == -1) { + LOG("Input queue is full, drop input ..."); + return; + } + + LOG("Input pushed in the queue, input array %zu", + array_queue_get_size(stm->input_queue)); +} + +static void +player_fullduplex_callback(SLBufferQueueItf caller, void * user_ptr) +{ + TIMESTAMP("ENTER"); + cubeb_stream * stm = user_ptr; + assert(stm); + SLresult res; + + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + int draining = opensl_get_draining(stm); + uint32_t shutdown = opensl_get_shutdown(stm); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + // Get output + void * output_buffer = NULL; + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + output_buffer = stm->queuebuf[stm->queuebuf_idx]; + // Advance the output buffer queue index + stm->queuebuf_idx = (stm->queuebuf_idx + 1) % stm->queuebuf_capacity; + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + if (shutdown || draining) { + LOG("Shutdown/draining, send silent"); + // Set silent on buffer + memset(output_buffer, 0, stm->queuebuf_len); + + // Enqueue data in player buffer queue + res = (*stm->bufq)->Enqueue(stm->bufq, + output_buffer, + stm->queuebuf_len); + assert(res == SL_RESULT_SUCCESS); + return; + } + + // Get input. + void * input_buffer = array_queue_pop(stm->input_queue); + long input_frame_count = stm->input_buffer_length / stm->input_frame_size; + long frames_needed = stm->queuebuf_len / stm->framesize; + if (!input_buffer) { + LOG("Input hole set silent input buffer"); + input_buffer = stm->input_silent_buffer; + } + + long written = 0; + // Trigger user callback through resampler + written = cubeb_resampler_fill(stm->resampler, + input_buffer, + &input_frame_count, + output_buffer, + frames_needed); + + LOG("Fill: written %ld, frames_needed %ld, input array size %zu", + written, frames_needed, array_queue_get_size(stm->input_queue)); + + if (written < 0 || written > frames_needed) { + // Error case + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + opensl_set_shutdown(stm, 1); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + opensl_stop_player(stm); + opensl_stop_recorder(stm); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + memset(output_buffer, 0, stm->queuebuf_len); + + // Enqueue data in player buffer queue + res = (*stm->bufq)->Enqueue(stm->bufq, + output_buffer, + stm->queuebuf_len); + assert(res == SL_RESULT_SUCCESS); + return; + } + + // Advance total out written frames counter + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + stm->written += written; + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + if ( written < frames_needed) { + r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + int64_t written_duration = INT64_C(1000) * stm->written * stm->framesize / stm->bytespersec; + opensl_set_draining(stm, 1); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + // Use SL_PLAYEVENT_HEADATMARKER event from slPlayCallback of SLPlayItf + // to make sure all the data has been processed. + (*stm->play)->SetMarkerPosition(stm->play, (SLmillisecond)written_duration); + } + + // Keep sending silent data even in draining mode to prevent the audio + // back-end from being stopped automatically by OpenSL/ES. + memset((uint8_t *)output_buffer + written * stm->framesize, 0, + stm->queuebuf_len - written * stm->framesize); + + // Enqueue data in player buffer queue + res = (*stm->bufq)->Enqueue(stm->bufq, + output_buffer, + stm->queuebuf_len); + assert(res == SL_RESULT_SUCCESS); + TIMESTAMP("EXIT"); +} + +static void opensl_destroy(cubeb * ctx); + +#if defined(__ANDROID__) +#if (__ANDROID_API__ >= ANDROID_VERSION_LOLLIPOP) +typedef int (system_property_get)(const char*, char*); + +static int +wrap_system_property_get(const char* name, char* value) +{ + void* libc = dlopen("libc.so", RTLD_LAZY); + if (!libc) { + LOG("Failed to open libc.so"); + return -1; + } + system_property_get* func = (system_property_get*) + dlsym(libc, "__system_property_get"); + int ret = -1; + if (func) { + ret = func(name, value); + } + dlclose(libc); + return ret; +} +#endif + +static int +get_android_version(void) +{ + char version_string[PROP_VALUE_MAX]; + + memset(version_string, 0, PROP_VALUE_MAX); + +#if (__ANDROID_API__ >= ANDROID_VERSION_LOLLIPOP) + int len = wrap_system_property_get("ro.build.version.sdk", version_string); +#else + int len = __system_property_get("ro.build.version.sdk", version_string); +#endif + if (len <= 0) { + LOG("Failed to get Android version!\n"); + return len; + } + + int version = (int)strtol(version_string, NULL, 10); + LOG("Android version %d", version); + return version; +} +#endif + +/*static*/ int +opensl_init(cubeb ** context, char const * context_name) +{ + cubeb * ctx; + +#if defined(__ANDROID__) + int android_version = get_android_version(); + if (android_version > 0 && android_version <= ANDROID_VERSION_GINGERBREAD_MR1) { + // Don't even attempt to run on Gingerbread and lower + return CUBEB_ERROR; + } +#endif + + *context = NULL; + + ctx = calloc(1, sizeof(*ctx)); + assert(ctx); + + ctx->ops = &opensl_ops; + + ctx->lib = dlopen("libOpenSLES.so", RTLD_LAZY); + if (!ctx->lib) { + free(ctx); + return CUBEB_ERROR; + } + + typedef SLresult (*slCreateEngine_t)(SLObjectItf *, + SLuint32, + const SLEngineOption *, + SLuint32, + const SLInterfaceID *, + const SLboolean *); + slCreateEngine_t f_slCreateEngine = + (slCreateEngine_t)dlsym(ctx->lib, "slCreateEngine"); + SLInterfaceID SL_IID_ENGINE = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_ENGINE"); + SLInterfaceID SL_IID_OUTPUTMIX = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_OUTPUTMIX"); + ctx->SL_IID_VOLUME = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_VOLUME"); + ctx->SL_IID_BUFFERQUEUE = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_BUFFERQUEUE"); +#if defined(__ANDROID__) + ctx->SL_IID_ANDROIDCONFIGURATION = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_ANDROIDCONFIGURATION"); + ctx->SL_IID_ANDROIDSIMPLEBUFFERQUEUE = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE"); +#endif + ctx->SL_IID_PLAY = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_PLAY"); + ctx->SL_IID_RECORD = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_RECORD"); + + if (!f_slCreateEngine || + !SL_IID_ENGINE || + !SL_IID_OUTPUTMIX || + !ctx->SL_IID_BUFFERQUEUE || +#if defined(__ANDROID__) + !ctx->SL_IID_ANDROIDCONFIGURATION || + !ctx->SL_IID_ANDROIDSIMPLEBUFFERQUEUE || +#endif + !ctx->SL_IID_PLAY || + !ctx->SL_IID_RECORD) { + opensl_destroy(ctx); + return CUBEB_ERROR; + } + + const SLEngineOption opt[] = {{SL_ENGINEOPTION_THREADSAFE, SL_BOOLEAN_TRUE}}; + + SLresult res; + res = cubeb_get_sles_engine(&ctx->engObj, 1, opt, 0, NULL, NULL); + + if (res != SL_RESULT_SUCCESS) { + opensl_destroy(ctx); + return CUBEB_ERROR; + } + + res = cubeb_realize_sles_engine(ctx->engObj); + if (res != SL_RESULT_SUCCESS) { + opensl_destroy(ctx); + return CUBEB_ERROR; + } + + res = (*ctx->engObj)->GetInterface(ctx->engObj, SL_IID_ENGINE, &ctx->eng); + if (res != SL_RESULT_SUCCESS) { + opensl_destroy(ctx); + return CUBEB_ERROR; + } + + const SLInterfaceID idsom[] = {SL_IID_OUTPUTMIX}; + const SLboolean reqom[] = {SL_BOOLEAN_TRUE}; + res = (*ctx->eng)->CreateOutputMix(ctx->eng, &ctx->outmixObj, 1, idsom, reqom); + if (res != SL_RESULT_SUCCESS) { + opensl_destroy(ctx); + return CUBEB_ERROR; + } + + res = (*ctx->outmixObj)->Realize(ctx->outmixObj, SL_BOOLEAN_FALSE); + if (res != SL_RESULT_SUCCESS) { + opensl_destroy(ctx); + return CUBEB_ERROR; + } + + ctx->p_output_latency_function = cubeb_output_latency_load_method(android_version); + if (!cubeb_output_latency_method_is_loaded(ctx->p_output_latency_function)) { + LOG("Warning: output latency is not available, cubeb_stream_get_position() is not supported"); + } + + *context = ctx; + + LOG("Cubeb init (%p) success", ctx); + return CUBEB_OK; +} + +static char const * +opensl_get_backend_id(cubeb * ctx) +{ + return "opensl"; +} + +static int +opensl_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + assert(ctx && max_channels); + /* The android mixer handles up to two channels, see + http://androidxref.com/4.2.2_r1/xref/frameworks/av/services/audioflinger/AudioFlinger.h#67 */ + *max_channels = 2; + + return CUBEB_OK; +} + +static void +opensl_destroy(cubeb * ctx) +{ + if (ctx->outmixObj) + (*ctx->outmixObj)->Destroy(ctx->outmixObj); + if (ctx->engObj) + cubeb_destroy_sles_engine(&ctx->engObj); + dlclose(ctx->lib); + if (ctx->p_output_latency_function) + cubeb_output_latency_unload_method(ctx->p_output_latency_function); + free(ctx); +} + +static void opensl_stream_destroy(cubeb_stream * stm); + +#if defined(__ANDROID__) && (__ANDROID_API__ >= ANDROID_VERSION_LOLLIPOP) +static int +opensl_set_format_ext(SLAndroidDataFormat_PCM_EX * format, cubeb_stream_params * params) +{ + assert(format); + assert(params); + + format->formatType = SL_ANDROID_DATAFORMAT_PCM_EX; + format->numChannels = params->channels; + // sampleRate is in milliHertz + format->sampleRate = params->rate * 1000; + format->channelMask = params->channels == 1 ? + SL_SPEAKER_FRONT_CENTER : + SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; + + switch (params->format) { + case CUBEB_SAMPLE_S16LE: + format->bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; + format->containerSize = SL_PCMSAMPLEFORMAT_FIXED_16; + format->representation = SL_ANDROID_PCM_REPRESENTATION_SIGNED_INT; + format->endianness = SL_BYTEORDER_LITTLEENDIAN; + break; + case CUBEB_SAMPLE_S16BE: + format->bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; + format->containerSize = SL_PCMSAMPLEFORMAT_FIXED_16; + format->representation = SL_ANDROID_PCM_REPRESENTATION_SIGNED_INT; + format->endianness = SL_BYTEORDER_BIGENDIAN; + break; + case CUBEB_SAMPLE_FLOAT32LE: + format->bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_32; + format->containerSize = SL_PCMSAMPLEFORMAT_FIXED_32; + format->representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT; + format->endianness = SL_BYTEORDER_LITTLEENDIAN; + break; + case CUBEB_SAMPLE_FLOAT32BE: + format->bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_32; + format->containerSize = SL_PCMSAMPLEFORMAT_FIXED_32; + format->representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT; + format->endianness = SL_BYTEORDER_BIGENDIAN; + break; + default: + return CUBEB_ERROR_INVALID_FORMAT; + } + return CUBEB_OK; +} +#endif + +static int +opensl_set_format(SLDataFormat_PCM * format, cubeb_stream_params * params) +{ + assert(format); + assert(params); + + format->formatType = SL_DATAFORMAT_PCM; + format->numChannels = params->channels; + // samplesPerSec is in milliHertz + format->samplesPerSec = params->rate * 1000; + format->bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; + format->containerSize = SL_PCMSAMPLEFORMAT_FIXED_16; + format->channelMask = params->channels == 1 ? + SL_SPEAKER_FRONT_CENTER : + SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; + + switch (params->format) { + case CUBEB_SAMPLE_S16LE: + format->endianness = SL_BYTEORDER_LITTLEENDIAN; + break; + case CUBEB_SAMPLE_S16BE: + format->endianness = SL_BYTEORDER_BIGENDIAN; + break; + default: + return CUBEB_ERROR_INVALID_FORMAT; + } + return CUBEB_OK; +} + +static int +opensl_configure_capture(cubeb_stream * stm, cubeb_stream_params * params) +{ + assert(stm); + assert(params); + + SLDataLocator_AndroidSimpleBufferQueue lDataLocatorOut; + lDataLocatorOut.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE; + lDataLocatorOut.numBuffers = NBUFS; + + SLDataFormat_PCM lDataFormat; + int r = opensl_set_format(&lDataFormat, params); + if (r != CUBEB_OK) { + return CUBEB_ERROR_INVALID_FORMAT; + } + + /* For now set device rate to params rate. */ + stm->input_device_rate = params->rate; + + SLDataSink lDataSink; + lDataSink.pLocator = &lDataLocatorOut; + lDataSink.pFormat = &lDataFormat; + + SLDataLocator_IODevice lDataLocatorIn; + lDataLocatorIn.locatorType = SL_DATALOCATOR_IODEVICE; + lDataLocatorIn.deviceType = SL_IODEVICE_AUDIOINPUT; + lDataLocatorIn.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT; + lDataLocatorIn.device = NULL; + + SLDataSource lDataSource; + lDataSource.pLocator = &lDataLocatorIn; + lDataSource.pFormat = NULL; + + const SLInterfaceID lSoundRecorderIIDs[] = { stm->context->SL_IID_RECORD, + stm->context->SL_IID_ANDROIDSIMPLEBUFFERQUEUE, + stm->context->SL_IID_ANDROIDCONFIGURATION }; + + const SLboolean lSoundRecorderReqs[] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE }; + // create the audio recorder abstract object + SLresult res = (*stm->context->eng)->CreateAudioRecorder(stm->context->eng, + &stm->recorderObj, + &lDataSource, + &lDataSink, + NELEMS(lSoundRecorderIIDs), + lSoundRecorderIIDs, + lSoundRecorderReqs); + // Sample rate not supported. Try again with default sample rate! + if (res == SL_RESULT_CONTENT_UNSUPPORTED) { + if (stm->output_enabled && stm->output_configured_rate != 0) { + // Set the same with the player. Since there is no + // api for input device this is a safe choice. + stm->input_device_rate = stm->output_configured_rate; + } else { + // The output preferred rate is used for an input only scenario. + // The default rate expected to be supported from all android devices. + stm->input_device_rate = DEFAULT_SAMPLE_RATE; + } + lDataFormat.samplesPerSec = stm->input_device_rate * 1000; + res = (*stm->context->eng)->CreateAudioRecorder(stm->context->eng, + &stm->recorderObj, + &lDataSource, + &lDataSink, + NELEMS(lSoundRecorderIIDs), + lSoundRecorderIIDs, + lSoundRecorderReqs); + + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to create recorder. Error code: %lu", res); + return CUBEB_ERROR; + } + } + + + if (get_android_version() > ANDROID_VERSION_JELLY_BEAN) { + SLAndroidConfigurationItf recorderConfig; + res = (*stm->recorderObj) + ->GetInterface(stm->recorderObj, + stm->context->SL_IID_ANDROIDCONFIGURATION, + &recorderConfig); + + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to get the android configuration interface for recorder. Error " + "code: %lu", + res); + return CUBEB_ERROR; + } + + // Voice recognition is the lowest latency, according to the docs. Camcorder + // uses a microphone that is in the same direction as the camera. + SLint32 streamType = stm->voice ? SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION + : SL_ANDROID_RECORDING_PRESET_CAMCORDER; + + res = (*recorderConfig) + ->SetConfiguration(recorderConfig, SL_ANDROID_KEY_RECORDING_PRESET, + &streamType, sizeof(SLint32)); + + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to set the android configuration to VOICE for the recorder. " + "Error code: %lu", res); + return CUBEB_ERROR; + } + } + // realize the audio recorder + res = (*stm->recorderObj)->Realize(stm->recorderObj, SL_BOOLEAN_FALSE); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to realize recorder. Error code: %lu", res); + return CUBEB_ERROR; + } + // get the record interface + res = (*stm->recorderObj)->GetInterface(stm->recorderObj, + stm->context->SL_IID_RECORD, + &stm->recorderItf); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to get recorder interface. Error code: %lu", res); + return CUBEB_ERROR; + } + + res = (*stm->recorderItf)->RegisterCallback(stm->recorderItf, recorder_marker_callback, stm); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to register recorder marker callback. Error code: %lu", res); + return CUBEB_ERROR; + } + + (*stm->recorderItf)->SetMarkerPosition(stm->recorderItf, (SLmillisecond)0); + + res = (*stm->recorderItf)->SetCallbackEventsMask(stm->recorderItf, (SLuint32)SL_RECORDEVENT_HEADATMARKER); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to set headatmarker event mask. Error code: %lu", res); + return CUBEB_ERROR; + } + // get the simple android buffer queue interface + res = (*stm->recorderObj)->GetInterface(stm->recorderObj, + stm->context->SL_IID_ANDROIDSIMPLEBUFFERQUEUE, + &stm->recorderBufferQueueItf); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to get recorder (android) buffer queue interface. Error code: %lu", res); + return CUBEB_ERROR; + } + + // register callback on record (input) buffer queue + slAndroidSimpleBufferQueueCallback rec_callback = recorder_callback; + if (stm->output_enabled) { + // Register full duplex callback instead. + rec_callback = recorder_fullduplex_callback; + } + res = (*stm->recorderBufferQueueItf)->RegisterCallback(stm->recorderBufferQueueItf, + rec_callback, + stm); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to register recorder buffer queue callback. Error code: %lu", res); + return CUBEB_ERROR; + } + + // Calculate length of input buffer according to requested latency + stm->input_frame_size = params->channels * sizeof(int16_t); + stm->input_buffer_length = (stm->input_frame_size * stm->buffer_size_frames); + + // Calculate the capacity of input array + stm->input_array_capacity = NBUFS; + if (stm->output_enabled) { + // Full duplex, update capacity to hold 1 sec of data + stm->input_array_capacity = 1 * stm->input_device_rate / stm->input_buffer_length; + } + // Allocate input array + stm->input_buffer_array = (void**)calloc(1, sizeof(void*)*stm->input_array_capacity); + // Buffering has not started yet. + stm->input_buffer_index = -1; + // Prepare input buffers + for(uint32_t i = 0; i < stm->input_array_capacity; ++i) { + stm->input_buffer_array[i] = calloc(1, stm->input_buffer_length); + } + + // On full duplex allocate input queue and silent buffer + if (stm->output_enabled) { + stm->input_queue = array_queue_create(stm->input_array_capacity); + assert(stm->input_queue); + stm->input_silent_buffer = calloc(1, stm->input_buffer_length); + assert(stm->input_silent_buffer); + } + + // Enqueue buffer to start rolling once recorder started + r = opensl_enqueue_recorder(stm, NULL); + if (r != CUBEB_OK) { + return r; + } + + LOG("Cubeb stream init recorder success"); + + return CUBEB_OK; +} + +static int +opensl_configure_playback(cubeb_stream * stm, cubeb_stream_params * params) { + assert(stm); + assert(params); + + stm->user_output_rate = params->rate; + if(params->format == CUBEB_SAMPLE_S16NE || params->format == CUBEB_SAMPLE_S16BE) { + stm->framesize = params->channels * sizeof(int16_t); + } else if(params->format == CUBEB_SAMPLE_FLOAT32NE || params->format == CUBEB_SAMPLE_FLOAT32BE) { + stm->framesize = params->channels * sizeof(float); + } + stm->lastPosition = -1; + stm->lastPositionTimeStamp = 0; + stm->lastCompensativePosition = -1; + + void* format = NULL; + SLuint32* format_sample_rate = NULL; + +#if defined(__ANDROID__) && (__ANDROID_API__ >= ANDROID_VERSION_LOLLIPOP) + SLAndroidDataFormat_PCM_EX pcm_ext_format; + if (get_android_version() >= ANDROID_VERSION_LOLLIPOP) { + if (opensl_set_format_ext(&pcm_ext_format, params) != CUBEB_OK) { + return CUBEB_ERROR_INVALID_FORMAT; + } + format = &pcm_ext_format; + format_sample_rate = &pcm_ext_format.sampleRate; + } +#endif + + SLDataFormat_PCM pcm_format; + if(!format) { + if(opensl_set_format(&pcm_format, params) != CUBEB_OK) { + return CUBEB_ERROR_INVALID_FORMAT; + } + format = &pcm_format; + format_sample_rate = &pcm_format.samplesPerSec; + } + + SLDataLocator_BufferQueue loc_bufq; + loc_bufq.locatorType = SL_DATALOCATOR_BUFFERQUEUE; + loc_bufq.numBuffers = NBUFS; + SLDataSource source; + source.pLocator = &loc_bufq; + source.pFormat = format; + + SLDataLocator_OutputMix loc_outmix; + loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX; + loc_outmix.outputMix = stm->context->outmixObj; + SLDataSink sink; + sink.pLocator = &loc_outmix; + sink.pFormat = NULL; + +#if defined(__ANDROID__) + const SLInterfaceID ids[] = {stm->context->SL_IID_BUFFERQUEUE, + stm->context->SL_IID_VOLUME, + stm->context->SL_IID_ANDROIDCONFIGURATION}; + const SLboolean req[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE}; +#else + const SLInterfaceID ids[] = {ctx->SL_IID_BUFFERQUEUE, ctx->SL_IID_VOLUME}; + const SLboolean req[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE}; +#endif + assert(NELEMS(ids) == NELEMS(req)); + + uint32_t preferred_sampling_rate = stm->user_output_rate; + SLresult res = SL_RESULT_CONTENT_UNSUPPORTED; + if (preferred_sampling_rate) { + res = (*stm->context->eng)->CreateAudioPlayer(stm->context->eng, + &stm->playerObj, + &source, + &sink, + NELEMS(ids), + ids, + req); + } + + // Sample rate not supported? Try again with primary sample rate! + if (res == SL_RESULT_CONTENT_UNSUPPORTED && + preferred_sampling_rate != DEFAULT_SAMPLE_RATE) { + preferred_sampling_rate = DEFAULT_SAMPLE_RATE; + *format_sample_rate = preferred_sampling_rate * 1000; + res = (*stm->context->eng)->CreateAudioPlayer(stm->context->eng, + &stm->playerObj, + &source, + &sink, + NELEMS(ids), + ids, + req); + } + + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to create audio player. Error code: %lu", res); + return CUBEB_ERROR; + } + + stm->output_configured_rate = preferred_sampling_rate; + stm->bytespersec = stm->output_configured_rate * stm->framesize; + stm->queuebuf_len = stm->framesize * stm->buffer_size_frames; + + // Calculate the capacity of input array + stm->queuebuf_capacity = NBUFS; + if (stm->output_enabled) { + // Full duplex, update capacity to hold 1 sec of data + stm->queuebuf_capacity = 1 * stm->output_configured_rate / stm->queuebuf_len; + } + // Allocate input array + stm->queuebuf = (void**)calloc(1, sizeof(void*) * stm->queuebuf_capacity); + for (uint32_t i = 0; i < stm->queuebuf_capacity; ++i) { + stm->queuebuf[i] = calloc(1, stm->queuebuf_len); + assert(stm->queuebuf[i]); + } + + SLAndroidConfigurationItf playerConfig = NULL; + + if (get_android_version() >= ANDROID_VERSION_N_MR1) { + res = (*stm->playerObj) + ->GetInterface(stm->playerObj, + stm->context->SL_IID_ANDROIDCONFIGURATION, + &playerConfig); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to get Android configuration interface. Error code: %lu", res); + return CUBEB_ERROR; + } + + SLint32 streamType = SL_ANDROID_STREAM_MEDIA; + if (stm->voice) { + streamType = SL_ANDROID_STREAM_VOICE; + } + res = (*playerConfig)->SetConfiguration(playerConfig, + SL_ANDROID_KEY_STREAM_TYPE, + &streamType, + sizeof(streamType)); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to set Android configuration to %d Error code: %lu", + streamType, res); + } + + SLuint32 performanceMode = SL_ANDROID_PERFORMANCE_LATENCY; + if (stm->buffer_size_frames > POWERSAVE_LATENCY_FRAMES_THRESHOLD) { + performanceMode = SL_ANDROID_PERFORMANCE_POWER_SAVING; + } + + res = (*playerConfig)->SetConfiguration(playerConfig, + SL_ANDROID_KEY_PERFORMANCE_MODE, + &performanceMode, + sizeof(performanceMode)); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to set Android performance mode to %d Error code: %lu. This is" + " not fatal", performanceMode, res); + } + } + + res = (*stm->playerObj)->Realize(stm->playerObj, SL_BOOLEAN_FALSE); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to realize player object. Error code: %lu", res); + return CUBEB_ERROR; + } + + // There are two ways of getting the audio output latency: + // - a configuration value, only available on some devices (notably devices + // running FireOS) + // - A Java method, that we call using JNI. + // + // The first method is prefered, if available, because it can account for more + // latency causes, and is more precise. + + // Latency has to be queried after the realization of the interface, when + // using SL_IID_ANDROIDCONFIGURATION. + SLuint32 audioLatency = 0; + SLuint32 paramSize = sizeof(SLuint32); + // The reported latency is in milliseconds. + if (playerConfig) { + res = (*playerConfig)->GetConfiguration(playerConfig, + (const SLchar *)"androidGetAudioLatency", + ¶mSize, + &audioLatency); + if (res == SL_RESULT_SUCCESS) { + LOG("Got playback latency using android configuration extension"); + stm->output_latency_ms = audioLatency; + } + } + // `playerConfig` is available, but the above failed, or `playerConfig` is not + // available. In both cases, we need to acquire the output latency by an other + // mean. + if ((playerConfig && res != SL_RESULT_SUCCESS) || + !playerConfig) { + if (cubeb_output_latency_method_is_loaded(stm->context->p_output_latency_function)) { + LOG("Got playback latency using JNI"); + stm->output_latency_ms = cubeb_get_output_latency(stm->context->p_output_latency_function); + } else { + LOG("No alternate latency querying method loaded, A/V sync will be off."); + stm->output_latency_ms = 0; + } + } + + LOG("Audio output latency: %dms", stm->output_latency_ms); + + res = (*stm->playerObj)->GetInterface(stm->playerObj, + stm->context->SL_IID_PLAY, + &stm->play); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to get play interface. Error code: %lu", res); + return CUBEB_ERROR; + } + + res = (*stm->playerObj)->GetInterface(stm->playerObj, + stm->context->SL_IID_BUFFERQUEUE, + &stm->bufq); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to get bufferqueue interface. Error code: %lu", res); + return CUBEB_ERROR; + } + + res = (*stm->playerObj)->GetInterface(stm->playerObj, + stm->context->SL_IID_VOLUME, + &stm->volume); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to get volume interface. Error code: %lu", res); + return CUBEB_ERROR; + } + + res = (*stm->play)->RegisterCallback(stm->play, play_callback, stm); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to register play callback. Error code: %lu", res); + return CUBEB_ERROR; + } + + // Work around wilhelm/AudioTrack badness, bug 1221228 + (*stm->play)->SetMarkerPosition(stm->play, (SLmillisecond)0); + + res = (*stm->play)->SetCallbackEventsMask(stm->play, (SLuint32)SL_PLAYEVENT_HEADATMARKER); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to set headatmarker event mask. Error code: %lu", res); + return CUBEB_ERROR; + } + + slBufferQueueCallback player_callback = bufferqueue_callback; + if (stm->input_enabled) { + player_callback = player_fullduplex_callback; + } + res = (*stm->bufq)->RegisterCallback(stm->bufq, player_callback, stm); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to register bufferqueue callback. Error code: %lu", res); + return CUBEB_ERROR; + } + + { + // Enqueue a silent frame so once the player becomes playing, the frame + // will be consumed and kick off the buffer queue callback. + // Note the duration of a single frame is less than 1ms. We don't bother + // adjusting the playback position. + uint8_t *buf = stm->queuebuf[stm->queuebuf_idx++]; + memset(buf, 0, stm->framesize); + res = (*stm->bufq)->Enqueue(stm->bufq, buf, stm->framesize); + assert(res == SL_RESULT_SUCCESS); + } + + LOG("Cubeb stream init playback success"); + return CUBEB_OK; +} + +static int +opensl_validate_stream_param(cubeb_stream_params * stream_params) +{ + if ((stream_params && + (stream_params->channels < 1 || stream_params->channels > 32))) { + return CUBEB_ERROR_INVALID_FORMAT; + } + if ((stream_params && + (stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK))) { + LOG("Loopback is not supported"); + return CUBEB_ERROR_NOT_SUPPORTED; + } + return CUBEB_OK; +} + +int has_pref_set(cubeb_stream_params* input_params, + cubeb_stream_params* output_params, + cubeb_stream_prefs pref) +{ + return (input_params && input_params->prefs & pref) || + (output_params && output_params->prefs & pref); +} + +static int +opensl_stream_init(cubeb * ctx, cubeb_stream ** stream, char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, cubeb_state_callback state_callback, + void * user_ptr) +{ + cubeb_stream * stm; + + assert(ctx); + if (input_device || output_device) { + LOG("Device selection is not supported in Android. The default will be used"); + } + + *stream = NULL; + + int r = opensl_validate_stream_param(output_stream_params); + if(r != CUBEB_OK) { + LOG("Output stream params not valid"); + return r; + } + r = opensl_validate_stream_param(input_stream_params); + if(r != CUBEB_OK) { + LOG("Input stream params not valid"); + return r; + } + + stm = calloc(1, sizeof(*stm)); + assert(stm); + + stm->context = ctx; + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + stm->buffer_size_frames = latency_frames ? latency_frames : DEFAULT_NUM_OF_FRAMES; + stm->input_enabled = (input_stream_params) ? 1 : 0; + stm->output_enabled = (output_stream_params) ? 1 : 0; + stm->shutdown = 1; + stm->voice = has_pref_set(input_stream_params, output_stream_params, CUBEB_STREAM_PREF_VOICE); + + LOG("cubeb stream prefs: voice: %s", stm->voice ? "true" : "false"); + + +#ifdef DEBUG + pthread_mutexattr_t attr; + pthread_mutexattr_init(&attr); + pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_ERRORCHECK); + r = pthread_mutex_init(&stm->mutex, &attr); +#else + r = pthread_mutex_init(&stm->mutex, NULL); +#endif + assert(r == 0); + + if (output_stream_params) { + LOG("Playback params: Rate %d, channels %d, format %d, latency in frames %d.", + output_stream_params->rate, output_stream_params->channels, + output_stream_params->format, stm->buffer_size_frames); + r = opensl_configure_playback(stm, output_stream_params); + if (r != CUBEB_OK) { + opensl_stream_destroy(stm); + return r; + } + } + + if (input_stream_params) { + LOG("Capture params: Rate %d, channels %d, format %d, latency in frames %d.", + input_stream_params->rate, input_stream_params->channels, + input_stream_params->format, stm->buffer_size_frames); + r = opensl_configure_capture(stm, input_stream_params); + if (r != CUBEB_OK) { + opensl_stream_destroy(stm); + return r; + } + } + + /* Configure resampler*/ + uint32_t target_sample_rate; + if (input_stream_params) { + target_sample_rate = input_stream_params->rate; + } else { + assert(output_stream_params); + target_sample_rate = output_stream_params->rate; + } + + // Use the actual configured rates for input + // and output. + cubeb_stream_params input_params; + if (input_stream_params) { + input_params = *input_stream_params; + input_params.rate = stm->input_device_rate; + } + cubeb_stream_params output_params; + if (output_stream_params) { + output_params = *output_stream_params; + output_params.rate = stm->output_configured_rate; + } + + stm->resampler = cubeb_resampler_create(stm, + input_stream_params ? &input_params : NULL, + output_stream_params ? &output_params : NULL, + target_sample_rate, + data_callback, + user_ptr, + CUBEB_RESAMPLER_QUALITY_DEFAULT); + if (!stm->resampler) { + LOG("Failed to create resampler"); + opensl_stream_destroy(stm); + return CUBEB_ERROR; + } + + *stream = stm; + LOG("Cubeb stream (%p) init success", stm); + return CUBEB_OK; +} + +static int +opensl_start_player(cubeb_stream * stm) +{ + assert(stm->playerObj); + SLuint32 playerState; + (*stm->playerObj)->GetState(stm->playerObj, &playerState); + if (playerState == SL_OBJECT_STATE_REALIZED) { + SLresult res = (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PLAYING); + if(res != SL_RESULT_SUCCESS) { + LOG("Failed to start player. Error code: %lu", res); + return CUBEB_ERROR; + } + } + return CUBEB_OK; +} + +static int +opensl_start_recorder(cubeb_stream * stm) +{ + assert(stm->recorderObj); + SLuint32 recorderState; + (*stm->recorderObj)->GetState(stm->recorderObj, &recorderState); + if (recorderState == SL_OBJECT_STATE_REALIZED) { + SLresult res = (*stm->recorderItf)->SetRecordState(stm->recorderItf, SL_RECORDSTATE_RECORDING); + if(res != SL_RESULT_SUCCESS) { + LOG("Failed to start recorder. Error code: %lu", res); + return CUBEB_ERROR; + } + } + return CUBEB_OK; +} + +static int +opensl_stream_start(cubeb_stream * stm) +{ + assert(stm); + + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + opensl_set_shutdown(stm, 0); + opensl_set_draining(stm, 0); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + if (stm->playerObj) { + r = opensl_start_player(stm); + if (r != CUBEB_OK) { + return r; + } + } + + if (stm->recorderObj) { + int r = opensl_start_recorder(stm); + if (r != CUBEB_OK) { + return r; + } + } + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED); + LOG("Cubeb stream (%p) started", stm); + return CUBEB_OK; +} + +static int +opensl_stop_player(cubeb_stream * stm) +{ + assert(stm->playerObj); + assert(stm->shutdown || stm->draining); + + SLresult res = (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PAUSED); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to stop player. Error code: %lu", res); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +static int +opensl_stop_recorder(cubeb_stream * stm) +{ + assert(stm->recorderObj); + assert(stm->shutdown || stm->draining); + + SLresult res = (*stm->recorderItf)->SetRecordState(stm->recorderItf, SL_RECORDSTATE_PAUSED); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to stop recorder. Error code: %lu", res); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +static int +opensl_stream_stop(cubeb_stream * stm) +{ + assert(stm); + + int r = pthread_mutex_lock(&stm->mutex); + assert(r == 0); + opensl_set_shutdown(stm, 1); + r = pthread_mutex_unlock(&stm->mutex); + assert(r == 0); + + if (stm->playerObj) { + r = opensl_stop_player(stm); + if (r != CUBEB_OK) { + return r; + } + } + + if (stm->recorderObj) { + int r = opensl_stop_recorder(stm); + if (r != CUBEB_OK) { + return r; + } + } + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED); + LOG("Cubeb stream (%p) stopped", stm); + return CUBEB_OK; +} + +static int +opensl_destroy_recorder(cubeb_stream * stm) +{ + assert(stm); + assert(stm->recorderObj); + + if (stm->recorderBufferQueueItf) { + SLresult res = (*stm->recorderBufferQueueItf)->Clear(stm->recorderBufferQueueItf); + if (res != SL_RESULT_SUCCESS) { + LOG("Failed to clear recorder buffer queue. Error code: %lu", res); + return CUBEB_ERROR; + } + stm->recorderBufferQueueItf = NULL; + for (uint32_t i = 0; i < stm->input_array_capacity; ++i) { + free(stm->input_buffer_array[i]); + } + } + + (*stm->recorderObj)->Destroy(stm->recorderObj); + stm->recorderObj = NULL; + stm->recorderItf = NULL; + + if (stm->input_queue) { + array_queue_destroy(stm->input_queue); + } + free(stm->input_silent_buffer); + + return CUBEB_OK; +} + +static void +opensl_stream_destroy(cubeb_stream * stm) +{ + assert(stm->draining || stm->shutdown); + + if (stm->playerObj) { + (*stm->playerObj)->Destroy(stm->playerObj); + stm->playerObj = NULL; + stm->play = NULL; + stm->bufq = NULL; + for (uint32_t i = 0; i < stm->queuebuf_capacity; ++i) { + free(stm->queuebuf[i]); + } + } + + if (stm->recorderObj) { + int r = opensl_destroy_recorder(stm); + assert(r == CUBEB_OK); + } + + if (stm->resampler) { + cubeb_resampler_destroy(stm->resampler); + } + + pthread_mutex_destroy(&stm->mutex); + + LOG("Cubeb stream (%p) destroyed", stm); + free(stm); +} + +static int +opensl_stream_get_position(cubeb_stream * stm, uint64_t * position) +{ + SLmillisecond msec; + uint32_t compensation_msec = 0; + SLresult res; + + res = (*stm->play)->GetPosition(stm->play, &msec); + if (res != SL_RESULT_SUCCESS) + return CUBEB_ERROR; + + struct timespec t; + clock_gettime(CLOCK_MONOTONIC, &t); + if(stm->lastPosition == msec) { + compensation_msec = + (t.tv_sec*1000000000LL + t.tv_nsec - stm->lastPositionTimeStamp) / 1000000; + } else { + stm->lastPositionTimeStamp = t.tv_sec*1000000000LL + t.tv_nsec; + stm->lastPosition = msec; + } + + uint64_t samplerate = stm->user_output_rate; + uint32_t output_latency = stm->output_latency_ms; + + pthread_mutex_lock(&stm->mutex); + int64_t maximum_position = stm->written * (int64_t)stm->user_output_rate / stm->output_configured_rate; + pthread_mutex_unlock(&stm->mutex); + assert(maximum_position >= 0); + + if (msec > output_latency) { + int64_t unadjusted_position; + if (stm->lastCompensativePosition > msec + compensation_msec) { + // Over compensation, use lastCompensativePosition. + unadjusted_position = + samplerate * (stm->lastCompensativePosition - output_latency) / 1000; + } else { + unadjusted_position = + samplerate * (msec - output_latency + compensation_msec) / 1000; + stm->lastCompensativePosition = msec + compensation_msec; + } + *position = unadjusted_position < maximum_position ? + unadjusted_position : maximum_position; + } else { + *position = 0; + } + return CUBEB_OK; +} + +static int +opensl_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ + assert(stm); + assert(latency); + + uint32_t stream_latency_frames = + stm->user_output_rate * (stm->output_latency_ms / 1000); + + return stream_latency_frames + cubeb_resampler_latency(stm->resampler); +} + +int +opensl_stream_set_volume(cubeb_stream * stm, float volume) +{ + SLresult res; + SLmillibel max_level, millibels; + float unclamped_millibels; + + res = (*stm->volume)->GetMaxVolumeLevel(stm->volume, &max_level); + + if (res != SL_RESULT_SUCCESS) { + return CUBEB_ERROR; + } + + /* millibels are 100*dB, so the conversion from the volume's linear amplitude + * is 100 * 20 * log(volume). However we clamp the resulting value before + * passing it to lroundf() in order to prevent it from silently returning an + * erroneous value when the unclamped value exceeds the size of a long. */ + unclamped_millibels = 100.0f * 20.0f * log10f(fmaxf(volume, 0.0f)); + unclamped_millibels = fmaxf(unclamped_millibels, SL_MILLIBEL_MIN); + unclamped_millibels = fminf(unclamped_millibels, max_level); + + millibels = lroundf(unclamped_millibels); + + res = (*stm->volume)->SetVolumeLevel(stm->volume, millibels); + + if (res != SL_RESULT_SUCCESS) { + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +static struct cubeb_ops const opensl_ops = { + .init = opensl_init, + .get_backend_id = opensl_get_backend_id, + .get_max_channel_count = opensl_get_max_channel_count, + .get_min_latency = NULL, + .get_preferred_sample_rate = NULL, + .enumerate_devices = NULL, + .device_collection_destroy = NULL, + .destroy = opensl_destroy, + .stream_init = opensl_stream_init, + .stream_destroy = opensl_stream_destroy, + .stream_start = opensl_stream_start, + .stream_stop = opensl_stream_stop, + .stream_reset_default_device = NULL, + .stream_get_position = opensl_stream_get_position, + .stream_get_latency = opensl_stream_get_latency, + .stream_set_volume = opensl_stream_set_volume, + .stream_get_current_device = NULL, + .stream_device_destroy = NULL, + .stream_register_device_changed_callback = NULL, + .register_device_collection_changed = NULL +}; diff --git a/dep/cubeb/src/cubeb_osx_run_loop.cpp b/dep/cubeb/src/cubeb_osx_run_loop.cpp new file mode 100644 index 000000000..de4e43970 --- /dev/null +++ b/dep/cubeb/src/cubeb_osx_run_loop.cpp @@ -0,0 +1,36 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#include +#include "cubeb_osx_run_loop.h" +#include "cubeb_log.h" +#include +#include +#include +#include + +void cubeb_set_coreaudio_notification_runloop() +{ + /* This is needed so that AudioUnit listeners get called on this thread, and + * not the main thread. If we don't do that, they are not called, or a crash + * occur, depending on the OSX version. */ + AudioObjectPropertyAddress runloop_address = { + kAudioHardwarePropertyRunLoop, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster + }; + + CFRunLoopRef run_loop = nullptr; + + OSStatus r; + r = AudioObjectSetPropertyData(kAudioObjectSystemObject, + &runloop_address, + 0, NULL, sizeof(CFRunLoopRef), &run_loop); + if (r != noErr) { + LOG("Could not make global CoreAudio notifications use their own thread."); + } +} diff --git a/dep/cubeb/src/cubeb_osx_run_loop.h b/dep/cubeb/src/cubeb_osx_run_loop.h new file mode 100644 index 000000000..78cd68d09 --- /dev/null +++ b/dep/cubeb/src/cubeb_osx_run_loop.h @@ -0,0 +1,22 @@ +/* + * Copyright © 2014 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +/* On OSX 10.6 and after, the notification callbacks from the audio hardware are + * called on the main thread. Setting the kAudioHardwarePropertyRunLoop property + * to null tells the OSX to use a separate thread for that. + * + * This has to be called only once per process, so it is in a separate header + * for easy integration in other code bases. */ +#if defined(__cplusplus) +extern "C" { +#endif + +void cubeb_set_coreaudio_notification_runloop(); + +#if defined(__cplusplus) +} +#endif diff --git a/dep/cubeb/src/cubeb_pulse.c b/dep/cubeb/src/cubeb_pulse.c new file mode 100644 index 000000000..94ead3a9c --- /dev/null +++ b/dep/cubeb/src/cubeb_pulse.c @@ -0,0 +1,1605 @@ +/* + * Copyright © 2011 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#undef NDEBUG +#include +#include +#include +#include +#include +#include +#include "cubeb-internal.h" +#include "cubeb/cubeb.h" +#include "cubeb_mixer.h" +#include "cubeb_strings.h" + +#ifdef DISABLE_LIBPULSE_DLOPEN +#define WRAP(x) x +#else +#define WRAP(x) cubeb_##x +#define LIBPULSE_API_VISIT(X) \ + X(pa_channel_map_can_balance) \ + X(pa_channel_map_init) \ + X(pa_context_connect) \ + X(pa_context_disconnect) \ + X(pa_context_drain) \ + X(pa_context_get_server_info) \ + X(pa_context_get_sink_info_by_name) \ + X(pa_context_get_sink_info_list) \ + X(pa_context_get_sink_input_info) \ + X(pa_context_get_source_info_list) \ + X(pa_context_get_state) \ + X(pa_context_new) \ + X(pa_context_rttime_new) \ + X(pa_context_set_sink_input_volume) \ + X(pa_context_set_state_callback) \ + X(pa_context_unref) \ + X(pa_cvolume_set) \ + X(pa_cvolume_set_balance) \ + X(pa_frame_size) \ + X(pa_operation_get_state) \ + X(pa_operation_unref) \ + X(pa_proplist_gets) \ + X(pa_rtclock_now) \ + X(pa_stream_begin_write) \ + X(pa_stream_cancel_write) \ + X(pa_stream_connect_playback) \ + X(pa_stream_cork) \ + X(pa_stream_disconnect) \ + X(pa_stream_get_channel_map) \ + X(pa_stream_get_index) \ + X(pa_stream_get_latency) \ + X(pa_stream_get_sample_spec) \ + X(pa_stream_get_state) \ + X(pa_stream_get_time) \ + X(pa_stream_new) \ + X(pa_stream_set_state_callback) \ + X(pa_stream_set_write_callback) \ + X(pa_stream_unref) \ + X(pa_stream_update_timing_info) \ + X(pa_stream_write) \ + X(pa_sw_volume_from_linear) \ + X(pa_threaded_mainloop_free) \ + X(pa_threaded_mainloop_get_api) \ + X(pa_threaded_mainloop_in_thread) \ + X(pa_threaded_mainloop_lock) \ + X(pa_threaded_mainloop_new) \ + X(pa_threaded_mainloop_signal) \ + X(pa_threaded_mainloop_start) \ + X(pa_threaded_mainloop_stop) \ + X(pa_threaded_mainloop_unlock) \ + X(pa_threaded_mainloop_wait) \ + X(pa_usec_to_bytes) \ + X(pa_stream_set_read_callback) \ + X(pa_stream_connect_record) \ + X(pa_stream_readable_size) \ + X(pa_stream_writable_size) \ + X(pa_stream_peek) \ + X(pa_stream_drop) \ + X(pa_stream_get_buffer_attr) \ + X(pa_stream_get_device_name) \ + X(pa_context_set_subscribe_callback) \ + X(pa_context_subscribe) \ + X(pa_mainloop_api_once) \ + X(pa_get_library_version) \ + X(pa_channel_map_init_auto) \ + +#define MAKE_TYPEDEF(x) static typeof(x) * cubeb_##x; +LIBPULSE_API_VISIT(MAKE_TYPEDEF); +#undef MAKE_TYPEDEF +#endif + +#if PA_CHECK_VERSION(2, 0, 0) +static int has_pulse_v2 = 0; +#endif + +static struct cubeb_ops const pulse_ops; + +struct cubeb_default_sink_info { + pa_channel_map channel_map; + uint32_t sample_spec_rate; + pa_sink_flags_t flags; +}; + +struct cubeb { + struct cubeb_ops const * ops; + void * libpulse; + pa_threaded_mainloop * mainloop; + pa_context * context; + struct cubeb_default_sink_info * default_sink_info; + char * context_name; + int error; + cubeb_device_collection_changed_callback output_collection_changed_callback; + void * output_collection_changed_user_ptr; + cubeb_device_collection_changed_callback input_collection_changed_callback; + void * input_collection_changed_user_ptr; + cubeb_strings * device_ids; +}; + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr; + /**/ + pa_stream * output_stream; + pa_stream * input_stream; + cubeb_data_callback data_callback; + cubeb_state_callback state_callback; + pa_time_event * drain_timer; + pa_sample_spec output_sample_spec; + pa_sample_spec input_sample_spec; + int shutdown; + float volume; + cubeb_state state; +}; + +static const float PULSE_NO_GAIN = -1.0; + +enum cork_state { + UNCORK = 0, + CORK = 1 << 0, + NOTIFY = 1 << 1 +}; + +static int +intern_device_id(cubeb * ctx, char const ** id) +{ + char const * interned; + + assert(ctx); + assert(id); + + interned = cubeb_strings_intern(ctx->device_ids, *id); + if (!interned) { + return CUBEB_ERROR; + } + + *id = interned; + + return CUBEB_OK; +} + +static void +sink_info_callback(pa_context * context, const pa_sink_info * info, int eol, void * u) +{ + (void)context; + cubeb * ctx = u; + if (!eol) { + free(ctx->default_sink_info); + ctx->default_sink_info = malloc(sizeof(struct cubeb_default_sink_info)); + memcpy(&ctx->default_sink_info->channel_map, &info->channel_map, sizeof(pa_channel_map)); + ctx->default_sink_info->sample_spec_rate = info->sample_spec.rate; + ctx->default_sink_info->flags = info->flags; + } + WRAP(pa_threaded_mainloop_signal)(ctx->mainloop, 0); +} + +static void +server_info_callback(pa_context * context, const pa_server_info * info, void * u) +{ + pa_operation * o; + o = WRAP(pa_context_get_sink_info_by_name)(context, info->default_sink_name, sink_info_callback, u); + if (o) { + WRAP(pa_operation_unref)(o); + } +} + +static void +context_state_callback(pa_context * c, void * u) +{ + cubeb * ctx = u; + if (!PA_CONTEXT_IS_GOOD(WRAP(pa_context_get_state)(c))) { + ctx->error = 1; + } + WRAP(pa_threaded_mainloop_signal)(ctx->mainloop, 0); +} + +static void +context_notify_callback(pa_context * c, void * u) +{ + (void)c; + cubeb * ctx = u; + WRAP(pa_threaded_mainloop_signal)(ctx->mainloop, 0); +} + +static void +stream_success_callback(pa_stream * s, int success, void * u) +{ + (void)s; + (void)success; + cubeb_stream * stm = u; + WRAP(pa_threaded_mainloop_signal)(stm->context->mainloop, 0); +} + +static void +stream_state_change_callback(cubeb_stream * stm, cubeb_state s) +{ + stm->state = s; + stm->state_callback(stm, stm->user_ptr, s); +} + +static void +stream_drain_callback(pa_mainloop_api * a, pa_time_event * e, struct timeval const * tv, void * u) +{ + (void)a; + (void)tv; + cubeb_stream * stm = u; + assert(stm->drain_timer == e); + stream_state_change_callback(stm, CUBEB_STATE_DRAINED); + /* there's no pa_rttime_free, so use this instead. */ + a->time_free(stm->drain_timer); + stm->drain_timer = NULL; + WRAP(pa_threaded_mainloop_signal)(stm->context->mainloop, 0); +} + +static void +stream_state_callback(pa_stream * s, void * u) +{ + cubeb_stream * stm = u; + if (!PA_STREAM_IS_GOOD(WRAP(pa_stream_get_state)(s))) { + stream_state_change_callback(stm, CUBEB_STATE_ERROR); + } + WRAP(pa_threaded_mainloop_signal)(stm->context->mainloop, 0); +} + +static void +trigger_user_callback(pa_stream * s, void const * input_data, size_t nbytes, cubeb_stream * stm) +{ + void * buffer; + size_t size; + int r; + long got; + size_t towrite, read_offset; + size_t frame_size; + + frame_size = WRAP(pa_frame_size)(&stm->output_sample_spec); + assert(nbytes % frame_size == 0); + + towrite = nbytes; + read_offset = 0; + while (towrite) { + size = towrite; + r = WRAP(pa_stream_begin_write)(s, &buffer, &size); + // Note: this has failed running under rr on occassion - needs investigation. + assert(r == 0); + assert(size > 0); + assert(size % frame_size == 0); + + LOGV("Trigger user callback with output buffer size=%zd, read_offset=%zd", size, read_offset); + got = stm->data_callback(stm, stm->user_ptr, (uint8_t const *)input_data + read_offset, buffer, size / frame_size); + if (got < 0) { + WRAP(pa_stream_cancel_write)(s); + stm->shutdown = 1; + return; + } + // If more iterations move offset of read buffer + if (input_data) { + size_t in_frame_size = WRAP(pa_frame_size)(&stm->input_sample_spec); + read_offset += (size / frame_size) * in_frame_size; + } + + if (stm->volume != PULSE_NO_GAIN) { + uint32_t samples = size * stm->output_sample_spec.channels / frame_size ; + + if (stm->output_sample_spec.format == PA_SAMPLE_S16BE || + stm->output_sample_spec.format == PA_SAMPLE_S16LE) { + short * b = buffer; + for (uint32_t i = 0; i < samples; i++) { + b[i] *= stm->volume; + } + } else { + float * b = buffer; + for (uint32_t i = 0; i < samples; i++) { + b[i] *= stm->volume; + } + } + } + + r = WRAP(pa_stream_write)(s, buffer, got * frame_size, NULL, 0, PA_SEEK_RELATIVE); + assert(r == 0); + + if ((size_t) got < size / frame_size) { + pa_usec_t latency = 0; + r = WRAP(pa_stream_get_latency)(s, &latency, NULL); + if (r == -PA_ERR_NODATA) { + /* this needs a better guess. */ + latency = 100 * PA_USEC_PER_MSEC; + } + assert(r == 0 || r == -PA_ERR_NODATA); + /* pa_stream_drain is useless, see PA bug# 866. this is a workaround. */ + /* arbitrary safety margin: double the current latency. */ + assert(!stm->drain_timer); + stm->drain_timer = WRAP(pa_context_rttime_new)(stm->context->context, WRAP(pa_rtclock_now)() + 2 * latency, stream_drain_callback, stm); + stm->shutdown = 1; + return; + } + + towrite -= size; + } + + assert(towrite == 0); +} + +static int +read_from_input(pa_stream * s, void const ** buffer, size_t * size) +{ + size_t readable_size = WRAP(pa_stream_readable_size)(s); + if (readable_size > 0) { + if (WRAP(pa_stream_peek)(s, buffer, size) < 0) { + return -1; + } + } + return readable_size; +} + +static void +stream_write_callback(pa_stream * s, size_t nbytes, void * u) +{ + LOGV("Output callback to be written buffer size %zd", nbytes); + cubeb_stream * stm = u; + if (stm->shutdown || + stm->state != CUBEB_STATE_STARTED) { + return; + } + + if (!stm->input_stream){ + // Output/playback only operation. + // Write directly to output + assert(!stm->input_stream && stm->output_stream); + trigger_user_callback(s, NULL, nbytes, stm); + } +} + +static void +stream_read_callback(pa_stream * s, size_t nbytes, void * u) +{ + LOGV("Input callback buffer size %zd", nbytes); + cubeb_stream * stm = u; + if (stm->shutdown) { + return; + } + + void const * read_data = NULL; + size_t read_size; + while (read_from_input(s, &read_data, &read_size) > 0) { + /* read_data can be NULL in case of a hole. */ + if (read_data) { + size_t in_frame_size = WRAP(pa_frame_size)(&stm->input_sample_spec); + size_t read_frames = read_size / in_frame_size; + + if (stm->output_stream) { + // input/capture + output/playback operation + size_t out_frame_size = WRAP(pa_frame_size)(&stm->output_sample_spec); + size_t write_size = read_frames * out_frame_size; + // Offer full duplex data for writing + trigger_user_callback(stm->output_stream, read_data, write_size, stm); + } else { + // input/capture only operation. Call callback directly + long got = stm->data_callback(stm, stm->user_ptr, read_data, NULL, read_frames); + if (got < 0 || (size_t) got != read_frames) { + WRAP(pa_stream_cancel_write)(s); + stm->shutdown = 1; + break; + } + } + } + if (read_size > 0) { + WRAP(pa_stream_drop)(s); + } + + if (stm->shutdown) { + return; + } + } +} + +static int +wait_until_context_ready(cubeb * ctx) +{ + for (;;) { + pa_context_state_t state = WRAP(pa_context_get_state)(ctx->context); + if (!PA_CONTEXT_IS_GOOD(state)) + return -1; + if (state == PA_CONTEXT_READY) + break; + WRAP(pa_threaded_mainloop_wait)(ctx->mainloop); + } + return 0; +} + +static int +wait_until_io_stream_ready(pa_stream * stream, pa_threaded_mainloop * mainloop) +{ + if (!stream || !mainloop){ + return -1; + } + for (;;) { + pa_stream_state_t state = WRAP(pa_stream_get_state)(stream); + if (!PA_STREAM_IS_GOOD(state)) + return -1; + if (state == PA_STREAM_READY) + break; + WRAP(pa_threaded_mainloop_wait)(mainloop); + } + return 0; +} + +static int +wait_until_stream_ready(cubeb_stream * stm) +{ + if (stm->output_stream && + wait_until_io_stream_ready(stm->output_stream, stm->context->mainloop) == -1) { + return -1; + } + if(stm->input_stream && + wait_until_io_stream_ready(stm->input_stream, stm->context->mainloop) == -1) { + return -1; + } + return 0; +} + +static int +operation_wait(cubeb * ctx, pa_stream * stream, pa_operation * o) +{ + while (WRAP(pa_operation_get_state)(o) == PA_OPERATION_RUNNING) { + WRAP(pa_threaded_mainloop_wait)(ctx->mainloop); + if (!PA_CONTEXT_IS_GOOD(WRAP(pa_context_get_state)(ctx->context))) { + return -1; + } + if (stream && !PA_STREAM_IS_GOOD(WRAP(pa_stream_get_state)(stream))) { + return -1; + } + } + return 0; +} + +static void +cork_io_stream(cubeb_stream * stm, pa_stream * io_stream, enum cork_state state) +{ + pa_operation * o; + if (!io_stream) { + return; + } + o = WRAP(pa_stream_cork)(io_stream, state & CORK, stream_success_callback, stm); + if (o) { + operation_wait(stm->context, io_stream, o); + WRAP(pa_operation_unref)(o); + } +} + +static void +stream_cork(cubeb_stream * stm, enum cork_state state) +{ + WRAP(pa_threaded_mainloop_lock)(stm->context->mainloop); + cork_io_stream(stm, stm->output_stream, state); + cork_io_stream(stm, stm->input_stream, state); + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + + if (state & NOTIFY) { + stream_state_change_callback(stm, state & CORK ? CUBEB_STATE_STOPPED + : CUBEB_STATE_STARTED); + } +} + +static int +stream_update_timing_info(cubeb_stream * stm) +{ + int r = -1; + pa_operation * o = NULL; + if (stm->output_stream) { + o = WRAP(pa_stream_update_timing_info)(stm->output_stream, stream_success_callback, stm); + if (o) { + r = operation_wait(stm->context, stm->output_stream, o); + WRAP(pa_operation_unref)(o); + } + if (r != 0) { + return r; + } + } + + if (stm->input_stream) { + o = WRAP(pa_stream_update_timing_info)(stm->input_stream, stream_success_callback, stm); + if (o) { + r = operation_wait(stm->context, stm->input_stream, o); + WRAP(pa_operation_unref)(o); + } + } + + return r; +} + +static pa_channel_position_t +cubeb_channel_to_pa_channel(cubeb_channel channel) +{ + switch (channel) { + case CHANNEL_FRONT_LEFT: + return PA_CHANNEL_POSITION_FRONT_LEFT; + case CHANNEL_FRONT_RIGHT: + return PA_CHANNEL_POSITION_FRONT_RIGHT; + case CHANNEL_FRONT_CENTER: + return PA_CHANNEL_POSITION_FRONT_CENTER; + case CHANNEL_LOW_FREQUENCY: + return PA_CHANNEL_POSITION_LFE; + case CHANNEL_BACK_LEFT: + return PA_CHANNEL_POSITION_REAR_LEFT; + case CHANNEL_BACK_RIGHT: + return PA_CHANNEL_POSITION_REAR_RIGHT; + case CHANNEL_FRONT_LEFT_OF_CENTER: + return PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER; + case CHANNEL_FRONT_RIGHT_OF_CENTER: + return PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER; + case CHANNEL_BACK_CENTER: + return PA_CHANNEL_POSITION_REAR_CENTER; + case CHANNEL_SIDE_LEFT: + return PA_CHANNEL_POSITION_SIDE_LEFT; + case CHANNEL_SIDE_RIGHT: + return PA_CHANNEL_POSITION_SIDE_RIGHT; + case CHANNEL_TOP_CENTER: + return PA_CHANNEL_POSITION_TOP_CENTER; + case CHANNEL_TOP_FRONT_LEFT: + return PA_CHANNEL_POSITION_TOP_FRONT_LEFT; + case CHANNEL_TOP_FRONT_CENTER: + return PA_CHANNEL_POSITION_TOP_FRONT_CENTER; + case CHANNEL_TOP_FRONT_RIGHT: + return PA_CHANNEL_POSITION_TOP_FRONT_RIGHT; + case CHANNEL_TOP_BACK_LEFT: + return PA_CHANNEL_POSITION_TOP_REAR_LEFT; + case CHANNEL_TOP_BACK_CENTER: + return PA_CHANNEL_POSITION_TOP_REAR_CENTER; + case CHANNEL_TOP_BACK_RIGHT: + return PA_CHANNEL_POSITION_TOP_REAR_RIGHT; + default: + return PA_CHANNEL_POSITION_INVALID; + } +} + +static void +layout_to_channel_map(cubeb_channel_layout layout, pa_channel_map * cm) +{ + assert(cm && layout != CUBEB_LAYOUT_UNDEFINED); + + WRAP(pa_channel_map_init)(cm); + + uint32_t channels = 0; + cubeb_channel_layout channelMap = layout; + for (uint32_t i = 0 ; channelMap != 0; ++i) { + uint32_t channel = (channelMap & 1) << i; + if (channel != 0) { + cm->map[channels] = cubeb_channel_to_pa_channel(channel); + channels++; + } + channelMap = channelMap >> 1; + } + unsigned int channels_from_layout = cubeb_channel_layout_nb_channels(layout); + assert(channels_from_layout <= UINT8_MAX); + cm->channels = (uint8_t) channels_from_layout; + + // Special case single channel center mapping as mono. + if (cm->channels == 1 && cm->map[0] == PA_CHANNEL_POSITION_FRONT_CENTER) { + cm->map[0] = PA_CHANNEL_POSITION_MONO; + } +} + +static void pulse_context_destroy(cubeb * ctx); +static void pulse_destroy(cubeb * ctx); + +static int +pulse_context_init(cubeb * ctx) +{ + int r; + + if (ctx->context) { + assert(ctx->error == 1); + pulse_context_destroy(ctx); + } + + ctx->context = WRAP(pa_context_new)(WRAP(pa_threaded_mainloop_get_api)(ctx->mainloop), + ctx->context_name); + if (!ctx->context) { + return -1; + } + WRAP(pa_context_set_state_callback)(ctx->context, context_state_callback, ctx); + + WRAP(pa_threaded_mainloop_lock)(ctx->mainloop); + r = WRAP(pa_context_connect)(ctx->context, NULL, 0, NULL); + + if (r < 0 || wait_until_context_ready(ctx) != 0) { + WRAP(pa_threaded_mainloop_unlock)(ctx->mainloop); + pulse_context_destroy(ctx); + ctx->context = NULL; + return -1; + } + + WRAP(pa_threaded_mainloop_unlock)(ctx->mainloop); + + ctx->error = 0; + + return 0; +} + +static int pulse_subscribe_notifications(cubeb * context, + pa_subscription_mask_t mask); + +/*static*/ int +pulse_init(cubeb ** context, char const * context_name) +{ + void * libpulse = NULL; + cubeb * ctx; + pa_operation * o; + + *context = NULL; + +#ifndef DISABLE_LIBPULSE_DLOPEN + libpulse = dlopen("libpulse.so.0", RTLD_LAZY); + if (!libpulse) { + libpulse = dlopen("libpulse.so", RTLD_LAZY); + if (!libpulse) { + return CUBEB_ERROR; + } + } + +#define LOAD(x) { \ + cubeb_##x = dlsym(libpulse, #x); \ + if (!cubeb_##x) { \ + dlclose(libpulse); \ + return CUBEB_ERROR; \ + } \ + } + + LIBPULSE_API_VISIT(LOAD); +#undef LOAD +#endif + +#if PA_CHECK_VERSION(2, 0, 0) + const char* version = WRAP(pa_get_library_version)(); + has_pulse_v2 = strtol(version, NULL, 10) >= 2; +#endif + + ctx = calloc(1, sizeof(*ctx)); + assert(ctx); + + ctx->ops = &pulse_ops; + ctx->libpulse = libpulse; + if (cubeb_strings_init(&ctx->device_ids) != CUBEB_OK) { + pulse_destroy(ctx); + return CUBEB_ERROR; + } + + ctx->mainloop = WRAP(pa_threaded_mainloop_new)(); + ctx->default_sink_info = NULL; + + WRAP(pa_threaded_mainloop_start)(ctx->mainloop); + + ctx->context_name = context_name ? strdup(context_name) : NULL; + if (pulse_context_init(ctx) != 0) { + pulse_destroy(ctx); + return CUBEB_ERROR; + } + + /* server_info_callback performs a second async query, which is + responsible for initializing default_sink_info and signalling the + mainloop to end the wait. */ + WRAP(pa_threaded_mainloop_lock)(ctx->mainloop); + o = WRAP(pa_context_get_server_info)(ctx->context, server_info_callback, ctx); + if (o) { + operation_wait(ctx, NULL, o); + WRAP(pa_operation_unref)(o); + } + WRAP(pa_threaded_mainloop_unlock)(ctx->mainloop); + + /* Update `default_sink_info` when the default device changes. */ + pulse_subscribe_notifications(ctx, PA_SUBSCRIPTION_MASK_SERVER); + + *context = ctx; + + return CUBEB_OK; +} + +static char const * +pulse_get_backend_id(cubeb * ctx) +{ + (void)ctx; + return "pulse"; +} + +static int +pulse_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + (void)ctx; + assert(ctx && max_channels); + + if (!ctx->default_sink_info) + return CUBEB_ERROR; + + *max_channels = ctx->default_sink_info->channel_map.channels; + + return CUBEB_OK; +} + +static int +pulse_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) +{ + assert(ctx && rate); + (void)ctx; + + if (!ctx->default_sink_info) + return CUBEB_ERROR; + + *rate = ctx->default_sink_info->sample_spec_rate; + + return CUBEB_OK; +} + +static int +pulse_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames) +{ + (void)ctx; + // According to PulseAudio developers, this is a safe minimum. + *latency_frames = 25 * params.rate / 1000; + + return CUBEB_OK; +} + +static void +pulse_context_destroy(cubeb * ctx) +{ + pa_operation * o; + + WRAP(pa_threaded_mainloop_lock)(ctx->mainloop); + o = WRAP(pa_context_drain)(ctx->context, context_notify_callback, ctx); + if (o) { + operation_wait(ctx, NULL, o); + WRAP(pa_operation_unref)(o); + } + WRAP(pa_context_set_state_callback)(ctx->context, NULL, NULL); + WRAP(pa_context_disconnect)(ctx->context); + WRAP(pa_context_unref)(ctx->context); + WRAP(pa_threaded_mainloop_unlock)(ctx->mainloop); +} + +static void +pulse_destroy(cubeb * ctx) +{ + free(ctx->context_name); + if (ctx->context) { + pulse_context_destroy(ctx); + } + + if (ctx->mainloop) { + WRAP(pa_threaded_mainloop_stop)(ctx->mainloop); + WRAP(pa_threaded_mainloop_free)(ctx->mainloop); + } + + if (ctx->device_ids) { + cubeb_strings_destroy(ctx->device_ids); + } + + if (ctx->libpulse) { + dlclose(ctx->libpulse); + } + free(ctx->default_sink_info); + free(ctx); +} + +static void pulse_stream_destroy(cubeb_stream * stm); + +static pa_sample_format_t +to_pulse_format(cubeb_sample_format format) +{ + switch (format) { + case CUBEB_SAMPLE_S16LE: + return PA_SAMPLE_S16LE; + case CUBEB_SAMPLE_S16BE: + return PA_SAMPLE_S16BE; + case CUBEB_SAMPLE_FLOAT32LE: + return PA_SAMPLE_FLOAT32LE; + case CUBEB_SAMPLE_FLOAT32BE: + return PA_SAMPLE_FLOAT32BE; + default: + return PA_SAMPLE_INVALID; + } +} + +static cubeb_channel_layout +pulse_default_layout_for_channels(uint32_t ch) +{ + assert (ch > 0 && ch <= 8); + switch (ch) { + case 1: return CUBEB_LAYOUT_MONO; + case 2: return CUBEB_LAYOUT_STEREO; + case 3: return CUBEB_LAYOUT_3F; + case 4: return CUBEB_LAYOUT_QUAD; + case 5: return CUBEB_LAYOUT_3F2; + case 6: return CUBEB_LAYOUT_3F_LFE | + CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT; + case 7: return CUBEB_LAYOUT_3F3R_LFE; + case 8: return CUBEB_LAYOUT_3F4_LFE; + } + // Never get here! + return CUBEB_LAYOUT_UNDEFINED; +} + +static int +create_pa_stream(cubeb_stream * stm, + pa_stream ** pa_stm, + cubeb_stream_params * stream_params, + char const * stream_name) +{ + assert(stm && stream_params); + assert(&stm->input_stream == pa_stm || (&stm->output_stream == pa_stm && + (stream_params->layout == CUBEB_LAYOUT_UNDEFINED || + (stream_params->layout != CUBEB_LAYOUT_UNDEFINED && + cubeb_channel_layout_nb_channels(stream_params->layout) == stream_params->channels)))); + if (stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + *pa_stm = NULL; + pa_sample_spec ss; + ss.format = to_pulse_format(stream_params->format); + if (ss.format == PA_SAMPLE_INVALID) + return CUBEB_ERROR_INVALID_FORMAT; + ss.rate = stream_params->rate; + if (stream_params->channels > UINT8_MAX) + return CUBEB_ERROR_INVALID_FORMAT; + ss.channels = (uint8_t) stream_params->channels; + + if (stream_params->layout == CUBEB_LAYOUT_UNDEFINED) { + pa_channel_map cm; + if (stream_params->channels <= 8 && + !WRAP(pa_channel_map_init_auto)(&cm, stream_params->channels, PA_CHANNEL_MAP_DEFAULT)) { + LOG("Layout undefined and PulseAudio's default layout has not been configured, guess one."); + layout_to_channel_map(pulse_default_layout_for_channels(stream_params->channels), &cm); + *pa_stm = WRAP(pa_stream_new)(stm->context->context, stream_name, &ss, &cm); + } else { + LOG("Layout undefined, PulseAudio will use its default."); + *pa_stm = WRAP(pa_stream_new)(stm->context->context, stream_name, &ss, NULL); + } + } else { + pa_channel_map cm; + layout_to_channel_map(stream_params->layout, &cm); + *pa_stm = WRAP(pa_stream_new)(stm->context->context, stream_name, &ss, &cm); + } + return (*pa_stm == NULL) ? CUBEB_ERROR : CUBEB_OK; +} + +static pa_buffer_attr +set_buffering_attribute(unsigned int latency_frames, pa_sample_spec * sample_spec) +{ + pa_buffer_attr battr; + battr.maxlength = -1; + battr.prebuf = -1; + battr.tlength = latency_frames * WRAP(pa_frame_size)(sample_spec); + battr.minreq = battr.tlength / 4; + battr.fragsize = battr.minreq; + + LOG("Requested buffer attributes maxlength %u, tlength %u, prebuf %u, minreq %u, fragsize %u", + battr.maxlength, battr.tlength, battr.prebuf, battr.minreq, battr.fragsize); + + return battr; +} + +static int +pulse_stream_init(cubeb * context, + cubeb_stream ** stream, + char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + cubeb_stream * stm; + pa_buffer_attr battr; + int r; + + assert(context); + + // If the connection failed for some reason, try to reconnect + if (context->error == 1 && pulse_context_init(context) != 0) { + return CUBEB_ERROR; + } + + *stream = NULL; + + stm = calloc(1, sizeof(*stm)); + assert(stm); + + stm->context = context; + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + stm->volume = PULSE_NO_GAIN; + stm->state = -1; + assert(stm->shutdown == 0); + + WRAP(pa_threaded_mainloop_lock)(stm->context->mainloop); + if (output_stream_params) { + r = create_pa_stream(stm, &stm->output_stream, output_stream_params, stream_name); + if (r != CUBEB_OK) { + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + pulse_stream_destroy(stm); + return r; + } + + stm->output_sample_spec = *(WRAP(pa_stream_get_sample_spec)(stm->output_stream)); + + WRAP(pa_stream_set_state_callback)(stm->output_stream, stream_state_callback, stm); + WRAP(pa_stream_set_write_callback)(stm->output_stream, stream_write_callback, stm); + + battr = set_buffering_attribute(latency_frames, &stm->output_sample_spec); + WRAP(pa_stream_connect_playback)(stm->output_stream, + (char const *) output_device, + &battr, + PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING | + PA_STREAM_START_CORKED | PA_STREAM_ADJUST_LATENCY, + NULL, NULL); + } + + // Set up input stream + if (input_stream_params) { + r = create_pa_stream(stm, &stm->input_stream, input_stream_params, stream_name); + if (r != CUBEB_OK) { + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + pulse_stream_destroy(stm); + return r; + } + + stm->input_sample_spec = *(WRAP(pa_stream_get_sample_spec)(stm->input_stream)); + + WRAP(pa_stream_set_state_callback)(stm->input_stream, stream_state_callback, stm); + WRAP(pa_stream_set_read_callback)(stm->input_stream, stream_read_callback, stm); + + battr = set_buffering_attribute(latency_frames, &stm->input_sample_spec); + WRAP(pa_stream_connect_record)(stm->input_stream, + (char const *) input_device, + &battr, + PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING | + PA_STREAM_START_CORKED | PA_STREAM_ADJUST_LATENCY); + } + + r = wait_until_stream_ready(stm); + if (r == 0) { + /* force a timing update now, otherwise timing info does not become valid + until some point after initialization has completed. */ + r = stream_update_timing_info(stm); + } + + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + + if (r != 0) { + pulse_stream_destroy(stm); + return CUBEB_ERROR; + } + + if (g_cubeb_log_level) { + if (output_stream_params){ + const pa_buffer_attr * output_att; + output_att = WRAP(pa_stream_get_buffer_attr)(stm->output_stream); + LOG("Output buffer attributes maxlength %u, tlength %u, prebuf %u, minreq %u, fragsize %u",output_att->maxlength, output_att->tlength, + output_att->prebuf, output_att->minreq, output_att->fragsize); + } + + if (input_stream_params){ + const pa_buffer_attr * input_att; + input_att = WRAP(pa_stream_get_buffer_attr)(stm->input_stream); + LOG("Input buffer attributes maxlength %u, tlength %u, prebuf %u, minreq %u, fragsize %u",input_att->maxlength, input_att->tlength, + input_att->prebuf, input_att->minreq, input_att->fragsize); + } + } + + *stream = stm; + LOG("Cubeb stream (%p) init successful.", *stream); + + return CUBEB_OK; +} + +static void +pulse_stream_destroy(cubeb_stream * stm) +{ + stream_cork(stm, CORK); + + WRAP(pa_threaded_mainloop_lock)(stm->context->mainloop); + if (stm->output_stream) { + + if (stm->drain_timer) { + /* there's no pa_rttime_free, so use this instead. */ + WRAP(pa_threaded_mainloop_get_api)(stm->context->mainloop)->time_free(stm->drain_timer); + } + + WRAP(pa_stream_set_state_callback)(stm->output_stream, NULL, NULL); + WRAP(pa_stream_set_write_callback)(stm->output_stream, NULL, NULL); + WRAP(pa_stream_disconnect)(stm->output_stream); + WRAP(pa_stream_unref)(stm->output_stream); + } + + if (stm->input_stream) { + WRAP(pa_stream_set_state_callback)(stm->input_stream, NULL, NULL); + WRAP(pa_stream_set_read_callback)(stm->input_stream, NULL, NULL); + WRAP(pa_stream_disconnect)(stm->input_stream); + WRAP(pa_stream_unref)(stm->input_stream); + } + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + + LOG("Cubeb stream (%p) destroyed successfully.", stm); + free(stm); +} + +static void +pulse_defer_event_cb(pa_mainloop_api * a, void * userdata) +{ + (void)a; + cubeb_stream * stm = userdata; + if (stm->shutdown) { + return; + } + size_t writable_size = WRAP(pa_stream_writable_size)(stm->output_stream); + trigger_user_callback(stm->output_stream, NULL, writable_size, stm); +} + +static int +pulse_stream_start(cubeb_stream * stm) +{ + stm->shutdown = 0; + stream_cork(stm, UNCORK | NOTIFY); + + if (stm->output_stream && !stm->input_stream) { + /* On output only case need to manually call user cb once in order to make + * things roll. This is done via a defer event in order to execute it + * from PA server thread. */ + WRAP(pa_threaded_mainloop_lock)(stm->context->mainloop); + WRAP(pa_mainloop_api_once)(WRAP(pa_threaded_mainloop_get_api)(stm->context->mainloop), + pulse_defer_event_cb, stm); + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + } + + LOG("Cubeb stream (%p) started successfully.", stm); + return CUBEB_OK; +} + +static int +pulse_stream_stop(cubeb_stream * stm) +{ + WRAP(pa_threaded_mainloop_lock)(stm->context->mainloop); + stm->shutdown = 1; + // If draining is taking place wait to finish + while (stm->drain_timer) { + WRAP(pa_threaded_mainloop_wait)(stm->context->mainloop); + } + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + + stream_cork(stm, CORK | NOTIFY); + LOG("Cubeb stream (%p) stopped successfully.", stm); + return CUBEB_OK; +} + +static int +pulse_stream_get_position(cubeb_stream * stm, uint64_t * position) +{ + int r, in_thread; + pa_usec_t r_usec; + uint64_t bytes; + + if (!stm || !stm->output_stream) { + return CUBEB_ERROR; + } + + in_thread = WRAP(pa_threaded_mainloop_in_thread)(stm->context->mainloop); + + if (!in_thread) { + WRAP(pa_threaded_mainloop_lock)(stm->context->mainloop); + } + r = WRAP(pa_stream_get_time)(stm->output_stream, &r_usec); + if (!in_thread) { + WRAP(pa_threaded_mainloop_unlock)(stm->context->mainloop); + } + + if (r != 0) { + return CUBEB_ERROR; + } + + bytes = WRAP(pa_usec_to_bytes)(r_usec, &stm->output_sample_spec); + *position = bytes / WRAP(pa_frame_size)(&stm->output_sample_spec); + + return CUBEB_OK; +} + +static int +pulse_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ + pa_usec_t r_usec; + int negative, r; + + if (!stm || !stm->output_stream) { + return CUBEB_ERROR; + } + + r = WRAP(pa_stream_get_latency)(stm->output_stream, &r_usec, &negative); + assert(!negative); + if (r) { + return CUBEB_ERROR; + } + + *latency = r_usec * stm->output_sample_spec.rate / PA_USEC_PER_SEC; + return CUBEB_OK; +} + +static void +volume_success(pa_context *c, int success, void *userdata) +{ + (void)success; + (void)c; + cubeb_stream * stream = userdata; + assert(success); + WRAP(pa_threaded_mainloop_signal)(stream->context->mainloop, 0); +} + +static int +pulse_stream_set_volume(cubeb_stream * stm, float volume) +{ + uint32_t index; + pa_operation * op; + pa_volume_t vol; + pa_cvolume cvol; + const pa_sample_spec * ss; + cubeb * ctx; + + if (!stm->output_stream) { + return CUBEB_ERROR; + } + + WRAP(pa_threaded_mainloop_lock)(stm->context->mainloop); + + /* if the pulse daemon is configured to use flat volumes, + * apply our own gain instead of changing the input volume on the sink. */ + ctx = stm->context; + if (ctx->default_sink_info && + (ctx->default_sink_info->flags & PA_SINK_FLAT_VOLUME)) { + stm->volume = volume; + } else { + ss = WRAP(pa_stream_get_sample_spec)(stm->output_stream); + + vol = WRAP(pa_sw_volume_from_linear)(volume); + WRAP(pa_cvolume_set)(&cvol, ss->channels, vol); + + index = WRAP(pa_stream_get_index)(stm->output_stream); + + op = WRAP(pa_context_set_sink_input_volume)(ctx->context, + index, &cvol, volume_success, + stm); + if (op) { + operation_wait(ctx, stm->output_stream, op); + WRAP(pa_operation_unref)(op); + } + } + + WRAP(pa_threaded_mainloop_unlock)(ctx->mainloop); + + return CUBEB_OK; +} + +typedef struct { + char * default_sink_name; + char * default_source_name; + + cubeb_device_info * devinfo; + uint32_t max; + uint32_t count; + cubeb * context; +} pulse_dev_list_data; + +static cubeb_device_fmt +pulse_format_to_cubeb_format(pa_sample_format_t format) +{ + switch (format) { + case PA_SAMPLE_S16LE: + return CUBEB_DEVICE_FMT_S16LE; + case PA_SAMPLE_S16BE: + return CUBEB_DEVICE_FMT_S16BE; + case PA_SAMPLE_FLOAT32LE: + return CUBEB_DEVICE_FMT_F32LE; + case PA_SAMPLE_FLOAT32BE: + return CUBEB_DEVICE_FMT_F32BE; + default: + return CUBEB_DEVICE_FMT_F32NE; + } +} + +static void +pulse_ensure_dev_list_data_list_size (pulse_dev_list_data * list_data) +{ + if (list_data->count == list_data->max) { + list_data->max += 8; + list_data->devinfo = realloc(list_data->devinfo, + sizeof(cubeb_device_info) * list_data->max); + } +} + +static cubeb_device_state +pulse_get_state_from_sink_port(pa_sink_port_info * info) +{ + if (info != NULL) { +#if PA_CHECK_VERSION(2, 0, 0) + if (has_pulse_v2 && info->available == PA_PORT_AVAILABLE_NO) + return CUBEB_DEVICE_STATE_UNPLUGGED; + else /*if (info->available == PA_PORT_AVAILABLE_YES) + UNKNOWN */ +#endif + return CUBEB_DEVICE_STATE_ENABLED; + } + + return CUBEB_DEVICE_STATE_ENABLED; +} + +static void +pulse_sink_info_cb(pa_context * context, const pa_sink_info * info, + int eol, void * user_data) +{ + pulse_dev_list_data * list_data = user_data; + cubeb_device_info * devinfo; + char const * prop = NULL; + char const * device_id = NULL; + + (void)context; + + if (eol) { + WRAP(pa_threaded_mainloop_signal)(list_data->context->mainloop, 0); + return; + } + + if (info == NULL) + return; + + device_id = info->name; + if (intern_device_id(list_data->context, &device_id) != CUBEB_OK) { + assert(NULL); + return; + } + + pulse_ensure_dev_list_data_list_size(list_data); + devinfo = &list_data->devinfo[list_data->count]; + memset(devinfo, 0, sizeof(cubeb_device_info)); + + devinfo->device_id = device_id; + devinfo->devid = (cubeb_devid) devinfo->device_id; + devinfo->friendly_name = strdup(info->description); + prop = WRAP(pa_proplist_gets)(info->proplist, "sysfs.path"); + if (prop) + devinfo->group_id = strdup(prop); + prop = WRAP(pa_proplist_gets)(info->proplist, "device.vendor.name"); + if (prop) + devinfo->vendor_name = strdup(prop); + + devinfo->type = CUBEB_DEVICE_TYPE_OUTPUT; + devinfo->state = pulse_get_state_from_sink_port(info->active_port); + devinfo->preferred = (strcmp(info->name, list_data->default_sink_name) == 0) ? + CUBEB_DEVICE_PREF_ALL : CUBEB_DEVICE_PREF_NONE; + + devinfo->format = CUBEB_DEVICE_FMT_ALL; + devinfo->default_format = pulse_format_to_cubeb_format(info->sample_spec.format); + devinfo->max_channels = info->channel_map.channels; + devinfo->min_rate = 1; + devinfo->max_rate = PA_RATE_MAX; + devinfo->default_rate = info->sample_spec.rate; + + devinfo->latency_lo = 0; + devinfo->latency_hi = 0; + + list_data->count += 1; +} + +static cubeb_device_state +pulse_get_state_from_source_port(pa_source_port_info * info) +{ + if (info != NULL) { +#if PA_CHECK_VERSION(2, 0, 0) + if (has_pulse_v2 && info->available == PA_PORT_AVAILABLE_NO) + return CUBEB_DEVICE_STATE_UNPLUGGED; + else /*if (info->available == PA_PORT_AVAILABLE_YES) + UNKNOWN */ +#endif + return CUBEB_DEVICE_STATE_ENABLED; + } + + return CUBEB_DEVICE_STATE_ENABLED; +} + +static void +pulse_source_info_cb(pa_context * context, const pa_source_info * info, + int eol, void * user_data) +{ + pulse_dev_list_data * list_data = user_data; + cubeb_device_info * devinfo; + char const * prop = NULL; + char const * device_id = NULL; + + (void)context; + + if (eol) { + WRAP(pa_threaded_mainloop_signal)(list_data->context->mainloop, 0); + return; + } + + device_id = info->name; + if (intern_device_id(list_data->context, &device_id) != CUBEB_OK) { + assert(NULL); + return; + } + + pulse_ensure_dev_list_data_list_size(list_data); + devinfo = &list_data->devinfo[list_data->count]; + memset(devinfo, 0, sizeof(cubeb_device_info)); + + devinfo->device_id = device_id; + devinfo->devid = (cubeb_devid) devinfo->device_id; + devinfo->friendly_name = strdup(info->description); + prop = WRAP(pa_proplist_gets)(info->proplist, "sysfs.path"); + if (prop) + devinfo->group_id = strdup(prop); + prop = WRAP(pa_proplist_gets)(info->proplist, "device.vendor.name"); + if (prop) + devinfo->vendor_name = strdup(prop); + + devinfo->type = CUBEB_DEVICE_TYPE_INPUT; + devinfo->state = pulse_get_state_from_source_port(info->active_port); + devinfo->preferred = (strcmp(info->name, list_data->default_source_name) == 0) ? + CUBEB_DEVICE_PREF_ALL : CUBEB_DEVICE_PREF_NONE; + + devinfo->format = CUBEB_DEVICE_FMT_ALL; + devinfo->default_format = pulse_format_to_cubeb_format(info->sample_spec.format); + devinfo->max_channels = info->channel_map.channels; + devinfo->min_rate = 1; + devinfo->max_rate = PA_RATE_MAX; + devinfo->default_rate = info->sample_spec.rate; + + devinfo->latency_lo = 0; + devinfo->latency_hi = 0; + + list_data->count += 1; +} + +static void +pulse_server_info_cb(pa_context * c, const pa_server_info * i, void * userdata) +{ + pulse_dev_list_data * list_data = userdata; + + (void)c; + + free(list_data->default_sink_name); + free(list_data->default_source_name); + list_data->default_sink_name = + i->default_sink_name ? strdup(i->default_sink_name) : NULL; + list_data->default_source_name = + i->default_source_name ? strdup(i->default_source_name) : NULL; + + WRAP(pa_threaded_mainloop_signal)(list_data->context->mainloop, 0); +} + +static int +pulse_enumerate_devices(cubeb * context, cubeb_device_type type, + cubeb_device_collection * collection) +{ + pulse_dev_list_data user_data = { NULL, NULL, NULL, 0, 0, context }; + pa_operation * o; + + WRAP(pa_threaded_mainloop_lock)(context->mainloop); + + o = WRAP(pa_context_get_server_info)(context->context, + pulse_server_info_cb, &user_data); + if (o) { + operation_wait(context, NULL, o); + WRAP(pa_operation_unref)(o); + } + + if (type & CUBEB_DEVICE_TYPE_OUTPUT) { + o = WRAP(pa_context_get_sink_info_list)(context->context, + pulse_sink_info_cb, &user_data); + if (o) { + operation_wait(context, NULL, o); + WRAP(pa_operation_unref)(o); + } + } + + if (type & CUBEB_DEVICE_TYPE_INPUT) { + o = WRAP(pa_context_get_source_info_list)(context->context, + pulse_source_info_cb, &user_data); + if (o) { + operation_wait(context, NULL, o); + WRAP(pa_operation_unref)(o); + } + } + + WRAP(pa_threaded_mainloop_unlock)(context->mainloop); + + collection->device = user_data.devinfo; + collection->count = user_data.count; + + free(user_data.default_sink_name); + free(user_data.default_source_name); + return CUBEB_OK; +} + +static int +pulse_device_collection_destroy(cubeb * ctx, cubeb_device_collection * collection) +{ + size_t n; + + for (n = 0; n < collection->count; n++) { + free((void *) collection->device[n].friendly_name); + free((void *) collection->device[n].vendor_name); + free((void *) collection->device[n].group_id); + } + + free(collection->device); + return CUBEB_OK; +} + +static int +pulse_stream_get_current_device(cubeb_stream * stm, cubeb_device ** const device) +{ +#if PA_CHECK_VERSION(0, 9, 8) + *device = calloc(1, sizeof(cubeb_device)); + if (*device == NULL) + return CUBEB_ERROR; + + if (stm->input_stream) { + const char * name = WRAP(pa_stream_get_device_name)(stm->input_stream); + (*device)->input_name = (name == NULL) ? NULL : strdup(name); + } + + if (stm->output_stream) { + const char * name = WRAP(pa_stream_get_device_name)(stm->output_stream); + (*device)->output_name = (name == NULL) ? NULL : strdup(name); + } + + return CUBEB_OK; +#else + return CUBEB_ERROR_NOT_SUPPORTED; +#endif +} + +static int +pulse_stream_device_destroy(cubeb_stream * stream, + cubeb_device * device) +{ + (void)stream; + free(device->input_name); + free(device->output_name); + free(device); + return CUBEB_OK; +} + +static void +pulse_subscribe_callback(pa_context * ctx, + pa_subscription_event_type_t t, + uint32_t index, void * userdata) +{ + (void)ctx; + cubeb * context = userdata; + + switch (t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) { + case PA_SUBSCRIPTION_EVENT_SERVER: + if ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_CHANGE) { + LOG("Server changed %d", index); + WRAP(pa_context_get_server_info)(context->context, server_info_callback, context); + } + break; + case PA_SUBSCRIPTION_EVENT_SOURCE: + case PA_SUBSCRIPTION_EVENT_SINK: + + if (g_cubeb_log_level) { + if ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE && + (t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE) { + LOG("Removing source index %d", index); + } else if ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE && + (t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW) { + LOG("Adding source index %d", index); + } + if ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK && + (t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE) { + LOG("Removing sink index %d", index); + } else if ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK && + (t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW) { + LOG("Adding sink index %d", index); + } + } + + if ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE || + (t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW) { + if ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE) { + context->input_collection_changed_callback(context, context->input_collection_changed_user_ptr); + } + if ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK) { + context->output_collection_changed_callback(context, context->output_collection_changed_user_ptr); + } + } + break; + } +} + +static void +subscribe_success(pa_context *c, int success, void *userdata) +{ + (void)c; + cubeb * context = userdata; + assert(success); + WRAP(pa_threaded_mainloop_signal)(context->mainloop, 0); +} + +static int +pulse_subscribe_notifications(cubeb * context, pa_subscription_mask_t mask) { + WRAP(pa_threaded_mainloop_lock)(context->mainloop); + + WRAP(pa_context_set_subscribe_callback)(context->context, pulse_subscribe_callback, context); + + pa_operation * o; + o = WRAP(pa_context_subscribe)(context->context, mask, subscribe_success, context); + if (o == NULL) { + WRAP(pa_threaded_mainloop_unlock)(context->mainloop); + LOG("Context subscribe failed"); + return CUBEB_ERROR; + } + operation_wait(context, NULL, o); + WRAP(pa_operation_unref)(o); + + WRAP(pa_threaded_mainloop_unlock)(context->mainloop); + + return CUBEB_OK; +} + +static int +pulse_register_device_collection_changed(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection_changed_callback collection_changed_callback, + void * user_ptr) +{ + if (devtype & CUBEB_DEVICE_TYPE_INPUT) { + context->input_collection_changed_callback = collection_changed_callback; + context->input_collection_changed_user_ptr = user_ptr; + } + if (devtype & CUBEB_DEVICE_TYPE_OUTPUT) { + context->output_collection_changed_callback = collection_changed_callback; + context->output_collection_changed_user_ptr = user_ptr; + } + + pa_subscription_mask_t mask = PA_SUBSCRIPTION_MASK_NULL; + if (context->input_collection_changed_callback) { + /* Input added or removed */ + mask |= PA_SUBSCRIPTION_MASK_SOURCE; + } + if (context->output_collection_changed_callback) { + /* Output added or removed */ + mask |= PA_SUBSCRIPTION_MASK_SINK; + } + /* Default device changed, this is always registered in order to update the + * `default_sink_info` when the default device changes. */ + mask |= PA_SUBSCRIPTION_MASK_SERVER; + + return pulse_subscribe_notifications(context, mask); +} + +static struct cubeb_ops const pulse_ops = { + .init = pulse_init, + .get_backend_id = pulse_get_backend_id, + .get_max_channel_count = pulse_get_max_channel_count, + .get_min_latency = pulse_get_min_latency, + .get_preferred_sample_rate = pulse_get_preferred_sample_rate, + .enumerate_devices = pulse_enumerate_devices, + .device_collection_destroy = pulse_device_collection_destroy, + .destroy = pulse_destroy, + .stream_init = pulse_stream_init, + .stream_destroy = pulse_stream_destroy, + .stream_start = pulse_stream_start, + .stream_stop = pulse_stream_stop, + .stream_reset_default_device = NULL, + .stream_get_position = pulse_stream_get_position, + .stream_get_latency = pulse_stream_get_latency, + .stream_set_volume = pulse_stream_set_volume, + .stream_get_current_device = pulse_stream_get_current_device, + .stream_device_destroy = pulse_stream_device_destroy, + .stream_register_device_changed_callback = NULL, + .register_device_collection_changed = pulse_register_device_collection_changed +}; diff --git a/dep/cubeb/src/cubeb_resampler.cpp b/dep/cubeb/src/cubeb_resampler.cpp new file mode 100644 index 000000000..3fc09d294 --- /dev/null +++ b/dep/cubeb/src/cubeb_resampler.cpp @@ -0,0 +1,372 @@ +/* + * Copyright © 2014 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#ifndef NOMINMAX +#define NOMINMAX +#endif // NOMINMAX + +#include +#include +#include +#include +#include +#include +#include "cubeb_resampler.h" +#include "cubeb-speex-resampler.h" +#include "cubeb_resampler_internal.h" +#include "cubeb_utils.h" + +int +to_speex_quality(cubeb_resampler_quality q) +{ + switch(q) { + case CUBEB_RESAMPLER_QUALITY_VOIP: + return SPEEX_RESAMPLER_QUALITY_VOIP; + case CUBEB_RESAMPLER_QUALITY_DEFAULT: + return SPEEX_RESAMPLER_QUALITY_DEFAULT; + case CUBEB_RESAMPLER_QUALITY_DESKTOP: + return SPEEX_RESAMPLER_QUALITY_DESKTOP; + default: + assert(false); + return 0XFFFFFFFF; + } +} + +uint32_t min_buffered_audio_frame(uint32_t sample_rate) +{ + return sample_rate / 20; +} + +template +passthrough_resampler::passthrough_resampler(cubeb_stream * s, + cubeb_data_callback cb, + void * ptr, + uint32_t input_channels, + uint32_t sample_rate) + : processor(input_channels) + , stream(s) + , data_callback(cb) + , user_ptr(ptr) + , sample_rate(sample_rate) +{ +} + +template +long passthrough_resampler::fill(void * input_buffer, long * input_frames_count, + void * output_buffer, long output_frames) +{ + if (input_buffer) { + assert(input_frames_count); + } + assert((input_buffer && output_buffer && + *input_frames_count + static_cast(samples_to_frames(internal_input_buffer.length())) >= output_frames) || + (output_buffer && !input_buffer && (!input_frames_count || *input_frames_count == 0)) || + (input_buffer && !output_buffer && output_frames == 0)); + + // When we have no pending input data and exactly as much input + // as output data, we don't need to copy it into the internal buffer + // and can directly forward it to the callback. + void * in_buf = input_buffer; + unsigned long pop_input_count = 0u; + if (input_buffer && !output_buffer) { + output_frames = *input_frames_count; + } else if(input_buffer) { + if (internal_input_buffer.length() != 0) { + // In this case we have pending input data left and have + // to first append the input so we can pass it as one pointer + // to the callback + internal_input_buffer.push(static_cast(input_buffer), + frames_to_samples(*input_frames_count)); + in_buf = internal_input_buffer.data(); + pop_input_count = frames_to_samples(output_frames); + } else if(*input_frames_count > output_frames) { + // In this case we have more input that we need output and + // fill the overflowing input into internal_input_buffer + // Since we have no other pending data, we can nonetheless + // pass the current input data directly to the callback + assert(pop_input_count == 0); + unsigned long samples_off = frames_to_samples(output_frames); + internal_input_buffer.push(static_cast(input_buffer) + samples_off, + frames_to_samples(*input_frames_count - output_frames)); + } + } + + long rv = data_callback(stream, user_ptr, in_buf, output_buffer, output_frames); + + if (input_buffer) { + if (pop_input_count) { + internal_input_buffer.pop(nullptr, pop_input_count); + } + + *input_frames_count = output_frames; + drop_audio_if_needed(); + } + + return rv; +} + +template +cubeb_resampler_speex + ::cubeb_resampler_speex(InputProcessor * input_processor, + OutputProcessor * output_processor, + cubeb_stream * s, + cubeb_data_callback cb, + void * ptr) + : input_processor(input_processor) + , output_processor(output_processor) + , stream(s) + , data_callback(cb) + , user_ptr(ptr) +{ + if (input_processor && output_processor) { + // Add some delay on the processor that has the lowest delay so that the + // streams are synchronized. + uint32_t in_latency = input_processor->latency(); + uint32_t out_latency = output_processor->latency(); + if (in_latency > out_latency) { + uint32_t latency_diff = in_latency - out_latency; + output_processor->add_latency(latency_diff); + } else if (in_latency < out_latency) { + uint32_t latency_diff = out_latency - in_latency; + input_processor->add_latency(latency_diff); + } + fill_internal = &cubeb_resampler_speex::fill_internal_duplex; + } else if (input_processor) { + fill_internal = &cubeb_resampler_speex::fill_internal_input; + } else if (output_processor) { + fill_internal = &cubeb_resampler_speex::fill_internal_output; + } +} + +template +cubeb_resampler_speex + ::~cubeb_resampler_speex() +{ } + +template +long +cubeb_resampler_speex +::fill(void * input_buffer, long * input_frames_count, + void * output_buffer, long output_frames_needed) +{ + /* Input and output buffers, typed */ + T * in_buffer = reinterpret_cast(input_buffer); + T * out_buffer = reinterpret_cast(output_buffer); + return (this->*fill_internal)(in_buffer, input_frames_count, + out_buffer, output_frames_needed); +} + +template +long +cubeb_resampler_speex +::fill_internal_output(T * input_buffer, long * input_frames_count, + T * output_buffer, long output_frames_needed) +{ + assert(!input_buffer && (!input_frames_count || *input_frames_count == 0) && + output_buffer && output_frames_needed); + + if (!draining) { + long got = 0; + T * out_unprocessed = nullptr; + long output_frames_before_processing = 0; + + /* fill directly the input buffer of the output processor to save a copy */ + output_frames_before_processing = + output_processor->input_needed_for_output(output_frames_needed); + + out_unprocessed = + output_processor->input_buffer(output_frames_before_processing); + + got = data_callback(stream, user_ptr, + nullptr, out_unprocessed, + output_frames_before_processing); + + if (got < output_frames_before_processing) { + draining = true; + + if (got < 0) { + return got; + } + } + + output_processor->written(got); + } + + /* Process the output. If not enough frames have been returned from the + * callback, drain the processors. */ + return output_processor->output(output_buffer, output_frames_needed); +} + +template +long +cubeb_resampler_speex +::fill_internal_input(T * input_buffer, long * input_frames_count, + T * output_buffer, long /*output_frames_needed*/) +{ + assert(input_buffer && input_frames_count && *input_frames_count && + !output_buffer); + + /* The input data, after eventual resampling. This is passed to the callback. */ + T * resampled_input = nullptr; + uint32_t resampled_frame_count = input_processor->output_for_input(*input_frames_count); + + /* process the input, and present exactly `output_frames_needed` in the + * callback. */ + input_processor->input(input_buffer, *input_frames_count); + + size_t frames_resampled = 0; + resampled_input = input_processor->output(resampled_frame_count, &frames_resampled); + *input_frames_count = frames_resampled; + + long got = data_callback(stream, user_ptr, + resampled_input, nullptr, resampled_frame_count); + + /* Return the number of initial input frames or part of it. + * Since output_frames_needed == 0 in input scenario, the only + * available number outside resampler is the initial number of frames. */ + return (*input_frames_count) * (got / resampled_frame_count); +} + +template +long +cubeb_resampler_speex +::fill_internal_duplex(T * in_buffer, long * input_frames_count, + T * out_buffer, long output_frames_needed) +{ + if (draining) { + // discard input and drain any signal remaining in the resampler. + return output_processor->output(out_buffer, output_frames_needed); + } + + /* The input data, after eventual resampling. This is passed to the callback. */ + T * resampled_input = nullptr; + /* The output buffer passed down in the callback, that might be resampled. */ + T * out_unprocessed = nullptr; + long output_frames_before_processing = 0; + /* The number of frames returned from the callback. */ + long got = 0; + + /* We need to determine how much frames to present to the consumer. + * - If we have a two way stream, but we're only resampling input, we resample + * the input to the number of output frames. + * - If we have a two way stream, but we're only resampling the output, we + * resize the input buffer of the output resampler to the number of input + * frames, and we resample it afterwards. + * - If we resample both ways, we resample the input to the number of frames + * we would need to pass down to the consumer (before resampling the output), + * get the output data, and resample it to the number of frames needed by the + * caller. */ + + output_frames_before_processing = + output_processor->input_needed_for_output(output_frames_needed); + /* fill directly the input buffer of the output processor to save a copy */ + out_unprocessed = + output_processor->input_buffer(output_frames_before_processing); + + if (in_buffer) { + /* process the input, and present exactly `output_frames_needed` in the + * callback. */ + input_processor->input(in_buffer, *input_frames_count); + + size_t frames_resampled = 0; + resampled_input = + input_processor->output(output_frames_before_processing, &frames_resampled); + *input_frames_count = frames_resampled; + } else { + resampled_input = nullptr; + } + + got = data_callback(stream, user_ptr, + resampled_input, out_unprocessed, + output_frames_before_processing); + + if (got < output_frames_before_processing) { + draining = true; + + if (got < 0) { + return got; + } + } + + output_processor->written(got); + + input_processor->drop_audio_if_needed(); + + /* Process the output. If not enough frames have been returned from the + * callback, drain the processors. */ + got = output_processor->output(out_buffer, output_frames_needed); + + output_processor->drop_audio_if_needed(); + + return got; +} + +/* Resampler C API */ + +cubeb_resampler * +cubeb_resampler_create(cubeb_stream * stream, + cubeb_stream_params * input_params, + cubeb_stream_params * output_params, + unsigned int target_rate, + cubeb_data_callback callback, + void * user_ptr, + cubeb_resampler_quality quality) +{ + cubeb_sample_format format; + + assert(input_params || output_params); + + if (input_params) { + format = input_params->format; + } else { + format = output_params->format; + } + + switch(format) { + case CUBEB_SAMPLE_S16NE: + return cubeb_resampler_create_internal(stream, + input_params, + output_params, + target_rate, + callback, + user_ptr, + quality); + case CUBEB_SAMPLE_FLOAT32NE: + return cubeb_resampler_create_internal(stream, + input_params, + output_params, + target_rate, + callback, + user_ptr, + quality); + default: + assert(false); + return nullptr; + } +} + +long +cubeb_resampler_fill(cubeb_resampler * resampler, + void * input_buffer, + long * input_frames_count, + void * output_buffer, + long output_frames_needed) +{ + return resampler->fill(input_buffer, input_frames_count, + output_buffer, output_frames_needed); +} + +void +cubeb_resampler_destroy(cubeb_resampler * resampler) +{ + delete resampler; +} + +long +cubeb_resampler_latency(cubeb_resampler * resampler) +{ + return resampler->latency(); +} diff --git a/dep/cubeb/src/cubeb_resampler.h b/dep/cubeb/src/cubeb_resampler.h new file mode 100644 index 000000000..f6b551373 --- /dev/null +++ b/dep/cubeb/src/cubeb_resampler.h @@ -0,0 +1,85 @@ +/* + * Copyright © 2014 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#ifndef CUBEB_RESAMPLER_H +#define CUBEB_RESAMPLER_H + +#include "cubeb/cubeb.h" + +#if defined(__cplusplus) +extern "C" { +#endif + +typedef struct cubeb_resampler cubeb_resampler; + +typedef enum { + CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_QUALITY_DEFAULT, + CUBEB_RESAMPLER_QUALITY_DESKTOP +} cubeb_resampler_quality; + +/** + * Create a resampler to adapt the requested sample rate into something that + * is accepted by the audio backend. + * @param stream A cubeb_stream instance supplied to the data callback. + * @param input_params Used to calculate bytes per frame and buffer size for + * resampling of the input side of the stream. NULL if input should not be + * resampled. + * @param output_params Used to calculate bytes per frame and buffer size for + * resampling of the output side of the stream. NULL if output should not be + * resampled. + * @param target_rate The sampling rate after resampling for the input side of + * the stream, and/or the sampling rate prior to resampling of the output side + * of the stream. + * @param callback A callback to request data for resampling. + * @param user_ptr User data supplied to the data callback. + * @param quality Quality of the resampler. + * @retval A non-null pointer if success. + */ +cubeb_resampler * cubeb_resampler_create(cubeb_stream * stream, + cubeb_stream_params * input_params, + cubeb_stream_params * output_params, + unsigned int target_rate, + cubeb_data_callback callback, + void * user_ptr, + cubeb_resampler_quality quality); + +/** + * Fill the buffer with frames acquired using the data callback. Resampling will + * happen if necessary. + * @param resampler A cubeb_resampler instance. + * @param input_buffer A buffer of input samples + * @param input_frame_count The size of the buffer. Returns the number of frames + * consumed. + * @param output_buffer The buffer to be filled. + * @param output_frames_needed Number of frames that should be produced. + * @retval Number of frames that are actually produced. + * @retval CUBEB_ERROR on error. + */ +long cubeb_resampler_fill(cubeb_resampler * resampler, + void * input_buffer, + long * input_frame_count, + void * output_buffer, + long output_frames_needed); + +/** + * Destroy a cubeb_resampler. + * @param resampler A cubeb_resampler instance. + */ +void cubeb_resampler_destroy(cubeb_resampler * resampler); + +/** + * Returns the latency, in frames, of the resampler. + * @param resampler A cubeb resampler instance. + * @retval The latency, in frames, induced by the resampler. + */ +long cubeb_resampler_latency(cubeb_resampler * resampler); + +#if defined(__cplusplus) +} +#endif + +#endif /* CUBEB_RESAMPLER_H */ diff --git a/dep/cubeb/src/cubeb_resampler_internal.h b/dep/cubeb/src/cubeb_resampler_internal.h new file mode 100644 index 000000000..fb69992ff --- /dev/null +++ b/dep/cubeb/src/cubeb_resampler_internal.h @@ -0,0 +1,602 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#if !defined(CUBEB_RESAMPLER_INTERNAL) +#define CUBEB_RESAMPLER_INTERNAL + +#include +#include +#include +#include +#ifdef CUBEB_GECKO_BUILD +#include "mozilla/UniquePtr.h" +// In libc++, symbols such as std::unique_ptr may be defined in std::__1. +// The _LIBCPP_BEGIN_NAMESPACE_STD and _LIBCPP_END_NAMESPACE_STD macros +// will expand to the correct namespace. +#ifdef _LIBCPP_BEGIN_NAMESPACE_STD +#define MOZ_BEGIN_STD_NAMESPACE _LIBCPP_BEGIN_NAMESPACE_STD +#define MOZ_END_STD_NAMESPACE _LIBCPP_END_NAMESPACE_STD +#else +#define MOZ_BEGIN_STD_NAMESPACE namespace std { +#define MOZ_END_STD_NAMESPACE } +#endif +MOZ_BEGIN_STD_NAMESPACE + using mozilla::DefaultDelete; + using mozilla::UniquePtr; + #define default_delete DefaultDelete + #define unique_ptr UniquePtr +MOZ_END_STD_NAMESPACE +#endif +#include "cubeb/cubeb.h" +#include "cubeb_utils.h" +#include "cubeb-speex-resampler.h" +#include "cubeb_resampler.h" +#include + +/* This header file contains the internal C++ API of the resamplers, for testing. */ + +// When dropping audio input frames to prevent building +// an input delay, this function returns the number of frames +// to keep in the buffer. +// @parameter sample_rate The sample rate of the stream. +// @return A number of frames to keep. +uint32_t min_buffered_audio_frame(uint32_t sample_rate); + +int to_speex_quality(cubeb_resampler_quality q); + +struct cubeb_resampler { + virtual long fill(void * input_buffer, long * input_frames_count, + void * output_buffer, long frames_needed) = 0; + virtual long latency() = 0; + virtual ~cubeb_resampler() {} +}; + +/** Base class for processors. This is just used to share methods for now. */ +class processor { +public: + explicit processor(uint32_t channels) + : channels(channels) + {} +protected: + size_t frames_to_samples(size_t frames) const + { + return frames * channels; + } + size_t samples_to_frames(size_t samples) const + { + assert(!(samples % channels)); + return samples / channels; + } + /** The number of channel of the audio buffers to be resampled. */ + const uint32_t channels; +}; + +template +class passthrough_resampler : public cubeb_resampler + , public processor { +public: + passthrough_resampler(cubeb_stream * s, + cubeb_data_callback cb, + void * ptr, + uint32_t input_channels, + uint32_t sample_rate); + + virtual long fill(void * input_buffer, long * input_frames_count, + void * output_buffer, long output_frames); + + virtual long latency() + { + return 0; + } + + void drop_audio_if_needed() + { + uint32_t to_keep = min_buffered_audio_frame(sample_rate); + uint32_t available = samples_to_frames(internal_input_buffer.length()); + if (available > to_keep) { + internal_input_buffer.pop(nullptr, frames_to_samples(available - to_keep)); + } + } + +private: + cubeb_stream * const stream; + const cubeb_data_callback data_callback; + void * const user_ptr; + /* This allows to buffer some input to account for the fact that we buffer + * some inputs. */ + auto_array internal_input_buffer; + uint32_t sample_rate; +}; + +/** Bidirectional resampler, can resample an input and an output stream, or just + * an input stream or output stream. In this case a delay is inserted in the + * opposite direction to keep the streams synchronized. */ +template +class cubeb_resampler_speex : public cubeb_resampler { +public: + cubeb_resampler_speex(InputProcessing * input_processor, + OutputProcessing * output_processor, + cubeb_stream * s, + cubeb_data_callback cb, + void * ptr); + + virtual ~cubeb_resampler_speex(); + + virtual long fill(void * input_buffer, long * input_frames_count, + void * output_buffer, long output_frames_needed); + + virtual long latency() + { + if (input_processor && output_processor) { + assert(input_processor->latency() == output_processor->latency()); + return input_processor->latency(); + } else if (input_processor) { + return input_processor->latency(); + } else { + return output_processor->latency(); + } + } + +private: + typedef long(cubeb_resampler_speex::*processing_callback)(T * input_buffer, long * input_frames_count, T * output_buffer, long output_frames_needed); + + long fill_internal_duplex(T * input_buffer, long * input_frames_count, + T * output_buffer, long output_frames_needed); + long fill_internal_input(T * input_buffer, long * input_frames_count, + T * output_buffer, long output_frames_needed); + long fill_internal_output(T * input_buffer, long * input_frames_count, + T * output_buffer, long output_frames_needed); + + std::unique_ptr input_processor; + std::unique_ptr output_processor; + processing_callback fill_internal; + cubeb_stream * const stream; + const cubeb_data_callback data_callback; + void * const user_ptr; + bool draining = false; +}; + +/** Handles one way of a (possibly) duplex resampler, working on interleaved + * audio buffers of type T. This class is designed so that the number of frames + * coming out of the resampler can be precisely controled. It manages its own + * input buffer, and can use the caller's output buffer, or allocate its own. */ +template +class cubeb_resampler_speex_one_way : public processor { +public: + /** The sample type of this resampler, either 16-bit integers or 32-bit + * floats. */ + typedef T sample_type; + /** Construct a resampler resampling from #source_rate to #target_rate, that + * can be arbitrary, strictly positive number. + * @parameter channels The number of channels this resampler will resample. + * @parameter source_rate The sample-rate of the audio input. + * @parameter target_rate The sample-rate of the audio output. + * @parameter quality A number between 0 (fast, low quality) and 10 (slow, + * high quality). */ + cubeb_resampler_speex_one_way(uint32_t channels, + uint32_t source_rate, + uint32_t target_rate, + int quality) + : processor(channels) + , resampling_ratio(static_cast(source_rate) / target_rate) + , source_rate(source_rate) + , additional_latency(0) + , leftover_samples(0) + { + int r; + speex_resampler = speex_resampler_init(channels, source_rate, + target_rate, quality, &r); + assert(r == RESAMPLER_ERR_SUCCESS && "resampler allocation failure"); + } + + /** Destructor, deallocate the resampler */ + virtual ~cubeb_resampler_speex_one_way() + { + speex_resampler_destroy(speex_resampler); + } + + /** Sometimes, it is necessary to add latency on one way of a two-way + * resampler so that the stream are synchronized. This must be called only on + * a fresh resampler, otherwise, silent samples will be inserted in the + * stream. + * @param frames the number of frames of latency to add. */ + void add_latency(size_t frames) + { + additional_latency += frames; + resampling_in_buffer.push_silence(frames_to_samples(frames)); + } + + /* Fill the resampler with `input_frame_count` frames. */ + void input(T * input_buffer, size_t input_frame_count) + { + resampling_in_buffer.push(input_buffer, + frames_to_samples(input_frame_count)); + } + + /** Outputs exactly `output_frame_count` into `output_buffer`. + * `output_buffer` has to be at least `output_frame_count` long. */ + size_t output(T * output_buffer, size_t output_frame_count) + { + uint32_t in_len = samples_to_frames(resampling_in_buffer.length()); + uint32_t out_len = output_frame_count; + + speex_resample(resampling_in_buffer.data(), &in_len, + output_buffer, &out_len); + + /* This shifts back any unresampled samples to the beginning of the input + buffer. */ + resampling_in_buffer.pop(nullptr, frames_to_samples(in_len)); + + return out_len; + } + + size_t output_for_input(uint32_t input_frames) + { + return (size_t)floorf((input_frames + samples_to_frames(resampling_in_buffer.length())) + / resampling_ratio); + } + + /** Returns a buffer containing exactly `output_frame_count` resampled frames. + * The consumer should not hold onto the pointer. */ + T * output(size_t output_frame_count, size_t * input_frames_used) + { + if (resampling_out_buffer.capacity() < frames_to_samples(output_frame_count)) { + resampling_out_buffer.reserve(frames_to_samples(output_frame_count)); + } + + uint32_t in_len = samples_to_frames(resampling_in_buffer.length()); + uint32_t out_len = output_frame_count; + + speex_resample(resampling_in_buffer.data(), &in_len, + resampling_out_buffer.data(), &out_len); + + assert(out_len == output_frame_count); + + /* This shifts back any unresampled samples to the beginning of the input + buffer. */ + resampling_in_buffer.pop(nullptr, frames_to_samples(in_len)); + *input_frames_used = in_len; + + return resampling_out_buffer.data(); + } + + /** Get the latency of the resampler, in output frames. */ + uint32_t latency() const + { + /* The documentation of the resampler talks about "samples" here, but it + * only consider a single channel here so it's the same number of frames. */ + int latency = 0; + + latency = + speex_resampler_get_output_latency(speex_resampler) + additional_latency; + + assert(latency >= 0); + + return latency; + } + + /** Returns the number of frames to pass in the input of the resampler to have + * exactly `output_frame_count` resampled frames. This can return a number + * slightly bigger than what is strictly necessary, but it guaranteed that the + * number of output frames will be exactly equal. */ + uint32_t input_needed_for_output(uint32_t output_frame_count) const + { + int32_t unresampled_frames_left = samples_to_frames(resampling_in_buffer.length()); + int32_t resampled_frames_left = samples_to_frames(resampling_out_buffer.length()); + float input_frames_needed = + (output_frame_count - unresampled_frames_left) * resampling_ratio + - resampled_frames_left; + if (input_frames_needed < 0) { + return 0; + } + return (uint32_t)ceilf(input_frames_needed); + } + + /** Returns a pointer to the input buffer, that contains empty space for at + * least `frame_count` elements. This is useful so that consumer can directly + * write into the input buffer of the resampler. The pointer returned is + * adjusted so that leftover data are not overwritten. + */ + T * input_buffer(size_t frame_count) + { + leftover_samples = resampling_in_buffer.length(); + resampling_in_buffer.reserve(leftover_samples + + frames_to_samples(frame_count)); + return resampling_in_buffer.data() + leftover_samples; + } + + /** This method works with `input_buffer`, and allows to inform the processor + how much frames have been written in the provided buffer. */ + void written(size_t written_frames) + { + resampling_in_buffer.set_length(leftover_samples + + frames_to_samples(written_frames)); + } + + void drop_audio_if_needed() + { + // Keep at most 100ms buffered. + uint32_t available = samples_to_frames(resampling_in_buffer.length()); + uint32_t to_keep = min_buffered_audio_frame(source_rate); + if (available > to_keep) { + resampling_in_buffer.pop(nullptr, frames_to_samples(available - to_keep)); + } + } +private: + /** Wrapper for the speex resampling functions to have a typed + * interface. */ + void speex_resample(float * input_buffer, uint32_t * input_frame_count, + float * output_buffer, uint32_t * output_frame_count) + { +#ifndef NDEBUG + int rv; + rv = +#endif + speex_resampler_process_interleaved_float(speex_resampler, + input_buffer, + input_frame_count, + output_buffer, + output_frame_count); + assert(rv == RESAMPLER_ERR_SUCCESS); + } + + void speex_resample(short * input_buffer, uint32_t * input_frame_count, + short * output_buffer, uint32_t * output_frame_count) + { +#ifndef NDEBUG + int rv; + rv = +#endif + speex_resampler_process_interleaved_int(speex_resampler, + input_buffer, + input_frame_count, + output_buffer, + output_frame_count); + assert(rv == RESAMPLER_ERR_SUCCESS); + } + /** The state for the speex resampler used internaly. */ + SpeexResamplerState * speex_resampler; + /** Source rate / target rate. */ + const float resampling_ratio; + const uint32_t source_rate; + /** Storage for the input frames, to be resampled. Also contains + * any unresampled frames after resampling. */ + auto_array resampling_in_buffer; + /* Storage for the resampled frames, to be passed back to the caller. */ + auto_array resampling_out_buffer; + /** Additional latency inserted into the pipeline for synchronisation. */ + uint32_t additional_latency; + /** When `input_buffer` is called, this allows tracking the number of samples + that were in the buffer. */ + uint32_t leftover_samples; +}; + +/** This class allows delaying an audio stream by `frames` frames. */ +template +class delay_line : public processor { +public: + /** Constructor + * @parameter frames the number of frames of delay. + * @parameter channels the number of channels of this delay line. + * @parameter sample_rate sample-rate of the audio going through this delay line */ + delay_line(uint32_t frames, uint32_t channels, uint32_t sample_rate) + : processor(channels) + , length(frames) + , leftover_samples(0) + , sample_rate(sample_rate) + { + /* Fill the delay line with some silent frames to add latency. */ + delay_input_buffer.push_silence(frames * channels); + } + /* Add some latency to the delay line. + * @param frames the number of frames of latency to add. */ + void add_latency(size_t frames) + { + length += frames; + delay_input_buffer.push_silence(frames_to_samples(frames)); + } + /** Push some frames into the delay line. + * @parameter buffer the frames to push. + * @parameter frame_count the number of frames in #buffer. */ + void input(T * buffer, uint32_t frame_count) + { + delay_input_buffer.push(buffer, frames_to_samples(frame_count)); + } + /** Pop some frames from the internal buffer, into a internal output buffer. + * @parameter frames_needed the number of frames to be returned. + * @return a buffer containing the delayed frames. The consumer should not + * hold onto the pointer. */ + T * output(uint32_t frames_needed, size_t * input_frames_used) + { + if (delay_output_buffer.capacity() < frames_to_samples(frames_needed)) { + delay_output_buffer.reserve(frames_to_samples(frames_needed)); + } + + delay_output_buffer.clear(); + delay_output_buffer.push(delay_input_buffer.data(), + frames_to_samples(frames_needed)); + delay_input_buffer.pop(nullptr, frames_to_samples(frames_needed)); + *input_frames_used = frames_needed; + + return delay_output_buffer.data(); + } + /** Get a pointer to the first writable location in the input buffer> + * @parameter frames_needed the number of frames the user needs to write into + * the buffer. + * @returns a pointer to a location in the input buffer where #frames_needed + * can be writen. */ + T * input_buffer(uint32_t frames_needed) + { + leftover_samples = delay_input_buffer.length(); + delay_input_buffer.reserve(leftover_samples + frames_to_samples(frames_needed)); + return delay_input_buffer.data() + leftover_samples; + } + /** This method works with `input_buffer`, and allows to inform the processor + how much frames have been written in the provided buffer. */ + void written(size_t frames_written) + { + delay_input_buffer.set_length(leftover_samples + + frames_to_samples(frames_written)); + } + /** Drains the delay line, emptying the buffer. + * @parameter output_buffer the buffer in which the frames are written. + * @parameter frames_needed the maximum number of frames to write. + * @return the actual number of frames written. */ + size_t output(T * output_buffer, uint32_t frames_needed) + { + uint32_t in_len = samples_to_frames(delay_input_buffer.length()); + uint32_t out_len = frames_needed; + + uint32_t to_pop = std::min(in_len, out_len); + + delay_input_buffer.pop(output_buffer, frames_to_samples(to_pop)); + + return to_pop; + } + /** Returns the number of frames one needs to input into the delay line to get + * #frames_needed frames back. + * @parameter frames_needed the number of frames one want to write into the + * delay_line + * @returns the number of frames one will get. */ + size_t input_needed_for_output(uint32_t frames_needed) const + { + return frames_needed; + } + /** Returns the number of frames produces for `input_frames` frames in input */ + size_t output_for_input(uint32_t input_frames) + { + return input_frames; + } + /** The number of frames this delay line delays the stream by. + * @returns The number of frames of delay. */ + size_t latency() + { + return length; + } + + void drop_audio_if_needed() + { + size_t available = samples_to_frames(delay_input_buffer.length()); + uint32_t to_keep = min_buffered_audio_frame(sample_rate); + if (available > to_keep) { + delay_input_buffer.pop(nullptr, frames_to_samples(available - to_keep)); + } + } +private: + /** The length, in frames, of this delay line */ + uint32_t length; + /** When `input_buffer` is called, this allows tracking the number of samples + that where in the buffer. */ + uint32_t leftover_samples; + /** The input buffer, where the delay is applied. */ + auto_array delay_input_buffer; + /** The output buffer. This is only ever used if using the ::output with a + * single argument. */ + auto_array delay_output_buffer; + uint32_t sample_rate; +}; + +/** This sits behind the C API and is more typed. */ +template +cubeb_resampler * +cubeb_resampler_create_internal(cubeb_stream * stream, + cubeb_stream_params * input_params, + cubeb_stream_params * output_params, + unsigned int target_rate, + cubeb_data_callback callback, + void * user_ptr, + cubeb_resampler_quality quality) +{ + std::unique_ptr> input_resampler = nullptr; + std::unique_ptr> output_resampler = nullptr; + std::unique_ptr> input_delay = nullptr; + std::unique_ptr> output_delay = nullptr; + + assert((input_params || output_params) && + "need at least one valid parameter pointer."); + + /* All the streams we have have a sample rate that matches the target + sample rate, use a no-op resampler, that simply forwards the buffers to the + callback. */ + if (((input_params && input_params->rate == target_rate) && + (output_params && output_params->rate == target_rate)) || + (input_params && !output_params && (input_params->rate == target_rate)) || + (output_params && !input_params && (output_params->rate == target_rate))) { + return new passthrough_resampler(stream, callback, + user_ptr, + input_params ? input_params->channels : 0, + target_rate); + } + + /* Determine if we need to resampler one or both directions, and create the + resamplers. */ + if (output_params && (output_params->rate != target_rate)) { + output_resampler.reset( + new cubeb_resampler_speex_one_way(output_params->channels, + target_rate, + output_params->rate, + to_speex_quality(quality))); + if (!output_resampler) { + return NULL; + } + } + + if (input_params && (input_params->rate != target_rate)) { + input_resampler.reset( + new cubeb_resampler_speex_one_way(input_params->channels, + input_params->rate, + target_rate, + to_speex_quality(quality))); + if (!input_resampler) { + return NULL; + } + } + + /* If we resample only one direction but we have a duplex stream, insert a + * delay line with a length equal to the resampler latency of the + * other direction so that the streams are synchronized. */ + if (input_resampler && !output_resampler && input_params && output_params) { + output_delay.reset(new delay_line(input_resampler->latency(), + output_params->channels, + output_params->rate)); + if (!output_delay) { + return NULL; + } + } else if (output_resampler && !input_resampler && input_params && output_params) { + input_delay.reset(new delay_line(output_resampler->latency(), + input_params->channels, + output_params->rate)); + if (!input_delay) { + return NULL; + } + } + + if (input_resampler && output_resampler) { + return new cubeb_resampler_speex, + cubeb_resampler_speex_one_way> + (input_resampler.release(), + output_resampler.release(), + stream, callback, user_ptr); + } else if (input_resampler) { + return new cubeb_resampler_speex, + delay_line> + (input_resampler.release(), + output_delay.release(), + stream, callback, user_ptr); + } else { + return new cubeb_resampler_speex, + cubeb_resampler_speex_one_way> + (input_delay.release(), + output_resampler.release(), + stream, callback, user_ptr); + } +} + +#endif /* CUBEB_RESAMPLER_INTERNAL */ diff --git a/dep/cubeb/src/cubeb_ring_array.h b/dep/cubeb/src/cubeb_ring_array.h new file mode 100644 index 000000000..51b3b321a --- /dev/null +++ b/dep/cubeb/src/cubeb_ring_array.h @@ -0,0 +1,159 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef CUBEB_RING_ARRAY_H +#define CUBEB_RING_ARRAY_H + +#include "cubeb_utils.h" + +/** Ring array of pointers is used to hold buffers. In case that + asynchronous producer/consumer callbacks do not arrive in a + repeated order the ring array stores the buffers and fetch + them in the correct order. */ + +typedef struct { + AudioBuffer * buffer_array; /**< Array that hold pointers of the allocated space for the buffers. */ + unsigned int tail; /**< Index of the last element (first to deliver). */ + unsigned int count; /**< Number of elements in the array. */ + unsigned int capacity; /**< Total length of the array. */ +} ring_array; + +static int +single_audiobuffer_init(AudioBuffer * buffer, + uint32_t bytesPerFrame, + uint32_t channelsPerFrame, + uint32_t frames) +{ + assert(buffer); + assert(bytesPerFrame > 0 && channelsPerFrame && frames > 0); + + size_t size = bytesPerFrame * frames; + buffer->mData = operator new(size); + if (buffer->mData == NULL) { + return CUBEB_ERROR; + } + PodZero(static_cast(buffer->mData), size); + + buffer->mNumberChannels = channelsPerFrame; + buffer->mDataByteSize = size; + + return CUBEB_OK; +} + +/** Initialize the ring array. + @param ra The ring_array pointer of allocated structure. + @retval 0 on success. */ +int +ring_array_init(ring_array * ra, + uint32_t capacity, + uint32_t bytesPerFrame, + uint32_t channelsPerFrame, + uint32_t framesPerBuffer) +{ + assert(ra); + if (capacity == 0 || bytesPerFrame == 0 || + channelsPerFrame == 0 || framesPerBuffer == 0) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + ra->capacity = capacity; + ra->tail = 0; + ra->count = 0; + + ra->buffer_array = new AudioBuffer[ra->capacity]; + PodZero(ra->buffer_array, ra->capacity); + if (ra->buffer_array == NULL) { + return CUBEB_ERROR; + } + + for (unsigned int i = 0; i < ra->capacity; ++i) { + if (single_audiobuffer_init(&ra->buffer_array[i], + bytesPerFrame, + channelsPerFrame, + framesPerBuffer) != CUBEB_OK) { + return CUBEB_ERROR; + } + } + + return CUBEB_OK; +} + +/** Destroy the ring array. + @param ra The ring_array pointer.*/ +void +ring_array_destroy(ring_array * ra) +{ + assert(ra); + if (ra->buffer_array == NULL){ + return; + } + for (unsigned int i = 0; i < ra->capacity; ++i) { + if (ra->buffer_array[i].mData) { + operator delete(ra->buffer_array[i].mData); + } + } + delete [] ra->buffer_array; +} + +/** Get the allocated buffer to be stored with fresh data. + @param ra The ring_array pointer. + @retval Pointer of the allocated space to be stored with fresh data or NULL if full. */ +AudioBuffer * +ring_array_get_free_buffer(ring_array * ra) +{ + assert(ra && ra->buffer_array); + assert(ra->buffer_array[0].mData != NULL); + if (ra->count == ra->capacity) { + return NULL; + } + + assert(ra->count == 0 || (ra->tail + ra->count) % ra->capacity != ra->tail); + AudioBuffer * ret = &ra->buffer_array[(ra->tail + ra->count) % ra->capacity]; + + ++ra->count; + assert(ra->count <= ra->capacity); + + return ret; +} + +/** Get the next available buffer with data. + @param ra The ring_array pointer. + @retval Pointer of the next in order data buffer or NULL if empty. */ +AudioBuffer * +ring_array_get_data_buffer(ring_array * ra) +{ + assert(ra && ra->buffer_array); + assert(ra->buffer_array[0].mData != NULL); + + if (ra->count == 0) { + return NULL; + } + AudioBuffer * ret = &ra->buffer_array[ra->tail]; + + ra->tail = (ra->tail + 1) % ra->capacity; + assert(ra->tail < ra->capacity); + + assert(ra->count > 0); + --ra->count; + + return ret; +} + +/** When array is empty get the first allocated buffer in the array. + @param ra The ring_array pointer. + @retval If arrays is empty, pointer of the allocated space else NULL. */ +AudioBuffer * +ring_array_get_dummy_buffer(ring_array * ra) +{ + assert(ra && ra->buffer_array); + assert(ra->capacity > 0); + if (ra->count > 0) { + return NULL; + } + return &ra->buffer_array[0]; +} + +#endif //CUBEB_RING_ARRAY_H diff --git a/dep/cubeb/src/cubeb_ringbuffer.h b/dep/cubeb/src/cubeb_ringbuffer.h new file mode 100644 index 000000000..b6696e886 --- /dev/null +++ b/dep/cubeb/src/cubeb_ringbuffer.h @@ -0,0 +1,495 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef CUBEB_RING_BUFFER_H +#define CUBEB_RING_BUFFER_H + +#include "cubeb_utils.h" +#include +#include +#include +#include +#include + +/** + * Single producer single consumer lock-free and wait-free ring buffer. + * + * This data structure allows producing data from one thread, and consuming it on + * another thread, safely and without explicit synchronization. If used on two + * threads, this data structure uses atomics for thread safety. It is possible + * to disable the use of atomics at compile time and only use this data + * structure on one thread. + * + * The role for the producer and the consumer must be constant, i.e., the + * producer should always be on one thread and the consumer should always be on + * another thread. + * + * Some words about the inner workings of this class: + * - Capacity is fixed. Only one allocation is performed, in the constructor. + * When reading and writing, the return value of the method allows checking if + * the ring buffer is empty or full. + * - We always keep the read index at least one element ahead of the write + * index, so we can distinguish between an empty and a full ring buffer: an + * empty ring buffer is when the write index is at the same position as the + * read index. A full buffer is when the write index is exactly one position + * before the read index. + * - We synchronize updates to the read index after having read the data, and + * the write index after having written the data. This means that the each + * thread can only touch a portion of the buffer that is not touched by the + * other thread. + * - Callers are expected to provide buffers. When writing to the queue, + * elements are copied into the internal storage from the buffer passed in. + * When reading from the queue, the user is expected to provide a buffer. + * Because this is a ring buffer, data might not be contiguous in memory, + * providing an external buffer to copy into is an easy way to have linear + * data for further processing. + */ +template +class ring_buffer_base +{ +public: + /** + * Constructor for a ring buffer. + * + * This performs an allocation, but is the only allocation that will happen + * for the life time of a `ring_buffer_base`. + * + * @param capacity The maximum number of element this ring buffer will hold. + */ + ring_buffer_base(int capacity) + /* One more element to distinguish from empty and full buffer. */ + : capacity_(capacity + 1) + { + assert(storage_capacity() < + std::numeric_limits::max() / 2 && + "buffer too large for the type of index used."); + assert(capacity_ > 0); + + data_.reset(new T[storage_capacity()]); + /* If this queue is using atomics, initializing those members as the last + * action in the constructor acts as a full barrier, and allow capacity() to + * be thread-safe. */ + write_index_ = 0; + read_index_ = 0; + } + /** + * Push `count` zero or default constructed elements in the array. + * + * Only safely called on the producer thread. + * + * @param count The number of elements to enqueue. + * @return The number of element enqueued. + */ + int enqueue_default(int count) + { + return enqueue(nullptr, count); + } + /** + * @brief Put an element in the queue + * + * Only safely called on the producer thread. + * + * @param element The element to put in the queue. + * + * @return 1 if the element was inserted, 0 otherwise. + */ + int enqueue(T& element) + { + return enqueue(&element, 1); + } + /** + * Push `count` elements in the ring buffer. + * + * Only safely called on the producer thread. + * + * @param elements a pointer to a buffer containing at least `count` elements. + * If `elements` is nullptr, zero or default constructed elements are enqueued. + * @param count The number of elements to read from `elements` + * @return The number of elements successfully coped from `elements` and inserted + * into the ring buffer. + */ + int enqueue(T * elements, int count) + { +#ifndef NDEBUG + assert_correct_thread(producer_id); +#endif + + int rd_idx = read_index_.load(std::memory_order::memory_order_relaxed); + int wr_idx = write_index_.load(std::memory_order::memory_order_relaxed); + + if (full_internal(rd_idx, wr_idx)) { + return 0; + } + + int to_write = + std::min(available_write_internal(rd_idx, wr_idx), count); + + /* First part, from the write index to the end of the array. */ + int first_part = std::min(storage_capacity() - wr_idx, + to_write); + /* Second part, from the beginning of the array */ + int second_part = to_write - first_part; + + if (elements) { + Copy(data_.get() + wr_idx, elements, first_part); + Copy(data_.get(), elements + first_part, second_part); + } else { + ConstructDefault(data_.get() + wr_idx, first_part); + ConstructDefault(data_.get(), second_part); + } + + write_index_.store(increment_index(wr_idx, to_write), std::memory_order::memory_order_release); + + return to_write; + } + /** + * Retrieve at most `count` elements from the ring buffer, and copy them to + * `elements`, if non-null. + * + * Only safely called on the consumer side. + * + * @param elements A pointer to a buffer with space for at least `count` + * elements. If `elements` is `nullptr`, `count` element will be discarded. + * @param count The maximum number of elements to dequeue. + * @return The number of elements written to `elements`. + */ + int dequeue(T * elements, int count) + { +#ifndef NDEBUG + assert_correct_thread(consumer_id); +#endif + + int wr_idx = write_index_.load(std::memory_order::memory_order_acquire); + int rd_idx = read_index_.load(std::memory_order::memory_order_relaxed); + + if (empty_internal(rd_idx, wr_idx)) { + return 0; + } + + int to_read = + std::min(available_read_internal(rd_idx, wr_idx), count); + + int first_part = std::min(storage_capacity() - rd_idx, to_read); + int second_part = to_read - first_part; + + if (elements) { + Copy(elements, data_.get() + rd_idx, first_part); + Copy(elements + first_part, data_.get(), second_part); + } + + read_index_.store(increment_index(rd_idx, to_read), std::memory_order::memory_order_relaxed); + + return to_read; + } + /** + * Get the number of available element for consuming. + * + * Only safely called on the consumer thread. + * + * @return The number of available elements for reading. + */ + int available_read() const + { +#ifndef NDEBUG + assert_correct_thread(consumer_id); +#endif + return available_read_internal(read_index_.load(std::memory_order::memory_order_relaxed), + write_index_.load(std::memory_order::memory_order_relaxed)); + } + /** + * Get the number of available elements for consuming. + * + * Only safely called on the producer thread. + * + * @return The number of empty slots in the buffer, available for writing. + */ + int available_write() const + { +#ifndef NDEBUG + assert_correct_thread(producer_id); +#endif + return available_write_internal(read_index_.load(std::memory_order::memory_order_relaxed), + write_index_.load(std::memory_order::memory_order_relaxed)); + } + /** + * Get the total capacity, for this ring buffer. + * + * Can be called safely on any thread. + * + * @return The maximum capacity of this ring buffer. + */ + int capacity() const + { + return storage_capacity() - 1; + } + /** + * Reset the consumer and producer thread identifier, in case the thread are + * being changed. This has to be externally synchronized. This is no-op when + * asserts are disabled. + */ + void reset_thread_ids() + { +#ifndef NDEBUG + consumer_id = producer_id = std::thread::id(); +#endif + } +private: + /** Return true if the ring buffer is empty. + * + * @param read_index the read index to consider + * @param write_index the write index to consider + * @return true if the ring buffer is empty, false otherwise. + **/ + bool empty_internal(int read_index, + int write_index) const + { + return write_index == read_index; + } + /** Return true if the ring buffer is full. + * + * This happens if the write index is exactly one element behind the read + * index. + * + * @param read_index the read index to consider + * @param write_index the write index to consider + * @return true if the ring buffer is full, false otherwise. + **/ + bool full_internal(int read_index, + int write_index) const + { + return (write_index + 1) % storage_capacity() == read_index; + } + /** + * Return the size of the storage. It is one more than the number of elements + * that can be stored in the buffer. + * + * @return the number of elements that can be stored in the buffer. + */ + int storage_capacity() const + { + return capacity_; + } + /** + * Returns the number of elements available for reading. + * + * @return the number of available elements for reading. + */ + int + available_read_internal(int read_index, + int write_index) const + { + if (write_index >= read_index) { + return write_index - read_index; + } else { + return write_index + storage_capacity() - read_index; + } + } + /** + * Returns the number of empty elements, available for writing. + * + * @return the number of elements that can be written into the array. + */ + int + available_write_internal(int read_index, + int write_index) const + { + /* We substract one element here to always keep at least one sample + * free in the buffer, to distinguish between full and empty array. */ + int rv = read_index - write_index - 1; + if (write_index >= read_index) { + rv += storage_capacity(); + } + return rv; + } + /** + * Increments an index, wrapping it around the storage. + * + * @param index a reference to the index to increment. + * @param increment the number by which `index` is incremented. + * @return the new index. + */ + int + increment_index(int index, int increment) const + { + assert(increment >= 0); + return (index + increment) % storage_capacity(); + } + /** + * @brief This allows checking that enqueue (resp. dequeue) are always called + * by the right thread. + * + * @param id the id of the thread that has called the calling method first. + */ +#ifndef NDEBUG + static void assert_correct_thread(std::thread::id& id) + { + if (id == std::thread::id()) { + id = std::this_thread::get_id(); + return; + } + assert(id == std::this_thread::get_id()); + } +#endif + /** Index at which the oldest element is at, in samples. */ + std::atomic read_index_; + /** Index at which to write new elements. `write_index` is always at + * least one element ahead of `read_index_`. */ + std::atomic write_index_; + /** Maximum number of elements that can be stored in the ring buffer. */ + const int capacity_; + /** Data storage */ + std::unique_ptr data_; +#ifndef NDEBUG + /** The id of the only thread that is allowed to read from the queue. */ + mutable std::thread::id consumer_id; + /** The id of the only thread that is allowed to write from the queue. */ + mutable std::thread::id producer_id; +#endif +}; + +/** + * Adapter for `ring_buffer_base` that exposes an interface in frames. + */ +template +class audio_ring_buffer_base +{ +public: + /** + * @brief Constructor. + * + * @param channel_count Number of channels. + * @param capacity_in_frames The capacity in frames. + */ + audio_ring_buffer_base(int channel_count, int capacity_in_frames) + : channel_count(channel_count) + , ring_buffer(frames_to_samples(capacity_in_frames)) + { + assert(channel_count > 0); + } + /** + * @brief Enqueue silence. + * + * Only safely called on the producer thread. + * + * @param frame_count The number of frames of silence to enqueue. + * @return The number of frames of silence actually written to the queue. + */ + int enqueue_default(int frame_count) + { + return samples_to_frames(ring_buffer.enqueue(nullptr, frames_to_samples(frame_count))); + } + /** + * @brief Enqueue `frames_count` frames of audio. + * + * Only safely called from the producer thread. + * + * @param [in] frames If non-null, the frames to enqueue. + * Otherwise, silent frames are enqueued. + * @param frame_count The number of frames to enqueue. + * + * @return The number of frames enqueued + */ + + int enqueue(T * frames, int frame_count) + { + return samples_to_frames(ring_buffer.enqueue(frames, frames_to_samples(frame_count))); + } + + /** + * @brief Removes `frame_count` frames from the buffer, and + * write them to `frames` if it is non-null. + * + * Only safely called on the consumer thread. + * + * @param frames If non-null, the frames are copied to `frames`. + * Otherwise, they are dropped. + * @param frame_count The number of frames to remove. + * + * @return The number of frames actually dequeud. + */ + int dequeue(T * frames, int frame_count) + { + return samples_to_frames(ring_buffer.dequeue(frames, frames_to_samples(frame_count))); + } + /** + * Get the number of available frames of audio for consuming. + * + * Only safely called on the consumer thread. + * + * @return The number of available frames of audio for reading. + */ + int available_read() const + { + return samples_to_frames(ring_buffer.available_read()); + } + /** + * Get the number of available frames of audio for consuming. + * + * Only safely called on the producer thread. + * + * @return The number of empty slots in the buffer, available for writing. + */ + int available_write() const + { + return samples_to_frames(ring_buffer.available_write()); + } + /** + * Get the total capacity, for this ring buffer. + * + * Can be called safely on any thread. + * + * @return The maximum capacity of this ring buffer. + */ + int capacity() const + { + return samples_to_frames(ring_buffer.capacity()); + } +private: + /** + * @brief Frames to samples conversion. + * + * @param frames The number of frames. + * + * @return A number of samples. + */ + int frames_to_samples(int frames) const + { + return frames * channel_count; + } + /** + * @brief Samples to frames conversion. + * + * @param samples The number of samples. + * + * @return A number of frames. + */ + int samples_to_frames(int samples) const + { + return samples / channel_count; + } + /** Number of channels of audio that will stream through this ring buffer. */ + int channel_count; + /** The underlying ring buffer that is used to store the data. */ + ring_buffer_base ring_buffer; +}; + +/** + * Lock-free instantiation of the `ring_buffer_base` type. This is safe to use + * from two threads, one producer, one consumer (that never change role), + * without explicit synchronization. + */ +template +using lock_free_queue = ring_buffer_base; +/** + * Lock-free instantiation of the `audio_ring_buffer` type. This is safe to use + * from two threads, one producer, one consumer (that never change role), + * without explicit synchronization. + */ +template +using lock_free_audio_ring_buffer = audio_ring_buffer_base; + +#endif // CUBEB_RING_BUFFER_H diff --git a/dep/cubeb/src/cubeb_sndio.c b/dep/cubeb/src/cubeb_sndio.c new file mode 100644 index 000000000..34b3513d5 --- /dev/null +++ b/dep/cubeb/src/cubeb_sndio.c @@ -0,0 +1,669 @@ +/* + * Copyright (c) 2011 Alexandre Ratchov + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" + +#if defined(CUBEB_SNDIO_DEBUG) +#define DPR(...) fprintf(stderr, __VA_ARGS__); +#else +#define DPR(...) do {} while(0) +#endif + +#ifdef DISABLE_LIBSNDIO_DLOPEN +#define WRAP(x) x +#else +#define WRAP(x) cubeb_##x +#define LIBSNDIO_API_VISIT(X) \ + X(sio_close) \ + X(sio_eof) \ + X(sio_getpar) \ + X(sio_initpar) \ + X(sio_nfds) \ + X(sio_onmove) \ + X(sio_open) \ + X(sio_pollfd) \ + X(sio_read) \ + X(sio_revents) \ + X(sio_setpar) \ + X(sio_start) \ + X(sio_stop) \ + X(sio_write) \ + +#define MAKE_TYPEDEF(x) static typeof(x) * cubeb_##x; +LIBSNDIO_API_VISIT(MAKE_TYPEDEF); +#undef MAKE_TYPEDEF +#endif + +static struct cubeb_ops const sndio_ops; + +struct cubeb { + struct cubeb_ops const * ops; + void * libsndio; +}; + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * arg; /* user arg to {data,state}_cb */ + /**/ + pthread_t th; /* to run real-time audio i/o */ + pthread_mutex_t mtx; /* protects hdl and pos */ + struct sio_hdl *hdl; /* link us to sndio */ + int mode; /* bitmap of SIO_{PLAY,REC} */ + int active; /* cubec_start() called */ + int conv; /* need float->s16 conversion */ + unsigned char *rbuf; /* rec data consumed from here */ + unsigned char *pbuf; /* play data is prepared here */ + unsigned int nfr; /* number of frames in ibuf and obuf */ + unsigned int rbpf; /* rec bytes per frame */ + unsigned int pbpf; /* play bytes per frame */ + unsigned int rchan; /* number of rec channels */ + unsigned int pchan; /* number of play channels */ + unsigned int nblks; /* number of blocks in the buffer */ + uint64_t hwpos; /* frame number Joe hears right now */ + uint64_t swpos; /* number of frames produced/consumed */ + cubeb_data_callback data_cb; /* cb to preapare data */ + cubeb_state_callback state_cb; /* cb to notify about state changes */ + float volume; /* current volume */ +}; + +static void +s16_setvol(void *ptr, long nsamp, float volume) +{ + int16_t *dst = ptr; + int32_t mult = volume * 32768; + int32_t s; + + while (nsamp-- > 0) { + s = *dst; + s = (s * mult) >> 15; + *(dst++) = s; + } +} + +static void +float_to_s16(void *ptr, long nsamp, float volume) +{ + int16_t *dst = ptr; + float *src = ptr; + float mult = volume * 32768; + int s; + + while (nsamp-- > 0) { + s = lrintf(*(src++) * mult); + if (s < -32768) + s = -32768; + else if (s > 32767) + s = 32767; + *(dst++) = s; + } +} + +static void +s16_to_float(void *ptr, long nsamp) +{ + int16_t *src = ptr; + float *dst = ptr; + + src += nsamp; + dst += nsamp; + while (nsamp-- > 0) + *(--dst) = (1. / 32768) * *(--src); +} + +static const char * +sndio_get_device() +{ +#ifndef __OpenBSD__ + /* + * On other platforms default to sndio devices, + * so cubebs other backends can be used instead. + */ + const char *dev = getenv("AUDIODEVICE"); + if (dev == NULL || *dev == '\0') + return "snd/0"; + return dev; +#else + return SIO_DEVANY; +#endif +} + +static void +sndio_onmove(void *arg, int delta) +{ + cubeb_stream *s = (cubeb_stream *)arg; + + s->hwpos += delta; +} + +static void * +sndio_mainloop(void *arg) +{ + struct pollfd *pfds; + cubeb_stream *s = arg; + int n, eof = 0, prime, nfds, events, revents, state = CUBEB_STATE_STARTED; + size_t pstart = 0, pend = 0, rstart = 0, rend = 0; + long nfr; + + nfds = WRAP(sio_nfds)(s->hdl); + pfds = calloc(nfds, sizeof (struct pollfd)); + if (pfds == NULL) + return NULL; + + DPR("sndio_mainloop()\n"); + s->state_cb(s, s->arg, CUBEB_STATE_STARTED); + pthread_mutex_lock(&s->mtx); + if (!WRAP(sio_start)(s->hdl)) { + pthread_mutex_unlock(&s->mtx); + free(pfds); + return NULL; + } + DPR("sndio_mainloop(), started\n"); + + if (s->mode & SIO_PLAY) { + pstart = pend = s->nfr * s->pbpf; + prime = s->nblks; + if (s->mode & SIO_REC) { + memset(s->rbuf, 0, s->nfr * s->rbpf); + rstart = rend = s->nfr * s->rbpf; + } + } else { + prime = 0; + rstart = 0; + rend = s->nfr * s->rbpf; + } + + for (;;) { + if (!s->active) { + DPR("sndio_mainloop() stopped\n"); + state = CUBEB_STATE_STOPPED; + break; + } + + /* do we have a complete block? */ + if ((!(s->mode & SIO_PLAY) || pstart == pend) && + (!(s->mode & SIO_REC) || rstart == rend)) { + + if (eof) { + DPR("sndio_mainloop() drained\n"); + state = CUBEB_STATE_DRAINED; + break; + } + + if ((s->mode & SIO_REC) && s->conv) + s16_to_float(s->rbuf, s->nfr * s->rchan); + + /* invoke call-back, it returns less that s->nfr if done */ + pthread_mutex_unlock(&s->mtx); + nfr = s->data_cb(s, s->arg, s->rbuf, s->pbuf, s->nfr); + pthread_mutex_lock(&s->mtx); + if (nfr < 0) { + DPR("sndio_mainloop() cb err\n"); + state = CUBEB_STATE_ERROR; + break; + } + s->swpos += nfr; + + /* was this last call-back invocation (aka end-of-stream) ? */ + if (nfr < s->nfr) { + + if (!(s->mode & SIO_PLAY) || nfr == 0) { + state = CUBEB_STATE_DRAINED; + break; + } + + /* need to write (aka drain) the partial play block we got */ + pend = nfr * s->pbpf; + eof = 1; + } + + if (prime > 0) + prime--; + + if (s->mode & SIO_PLAY) { + if (s->conv) + float_to_s16(s->pbuf, nfr * s->pchan, s->volume); + else + s16_setvol(s->pbuf, nfr * s->pchan, s->volume); + } + + if (s->mode & SIO_REC) + rstart = 0; + if (s->mode & SIO_PLAY) + pstart = 0; + } + + events = 0; + if ((s->mode & SIO_REC) && rstart < rend && prime == 0) + events |= POLLIN; + if ((s->mode & SIO_PLAY) && pstart < pend) + events |= POLLOUT; + nfds = WRAP(sio_pollfd)(s->hdl, pfds, events); + + if (nfds > 0) { + pthread_mutex_unlock(&s->mtx); + n = poll(pfds, nfds, -1); + pthread_mutex_lock(&s->mtx); + if (n < 0) + continue; + } + + revents = WRAP(sio_revents)(s->hdl, pfds); + + if (revents & POLLHUP) { + state = CUBEB_STATE_ERROR; + break; + } + + if (revents & POLLOUT) { + n = WRAP(sio_write)(s->hdl, s->pbuf + pstart, pend - pstart); + if (n == 0 && WRAP(sio_eof)(s->hdl)) { + DPR("sndio_mainloop() werr\n"); + state = CUBEB_STATE_ERROR; + break; + } + pstart += n; + } + + if (revents & POLLIN) { + n = WRAP(sio_read)(s->hdl, s->rbuf + rstart, rend - rstart); + if (n == 0 && WRAP(sio_eof)(s->hdl)) { + DPR("sndio_mainloop() rerr\n"); + state = CUBEB_STATE_ERROR; + break; + } + rstart += n; + } + + /* skip rec block, if not recording (yet) */ + if (prime > 0 && (s->mode & SIO_REC)) + rstart = rend; + } + WRAP(sio_stop)(s->hdl); + s->hwpos = s->swpos; + pthread_mutex_unlock(&s->mtx); + s->state_cb(s, s->arg, state); + free(pfds); + return NULL; +} + +/*static*/ int +sndio_init(cubeb **context, char const *context_name) +{ + void * libsndio = NULL; + struct sio_hdl *hdl; + + assert(context); + +#ifndef DISABLE_LIBSNDIO_DLOPEN + libsndio = dlopen("libsndio.so.7.0", RTLD_LAZY); + if (!libsndio) { + libsndio = dlopen("libsndio.so", RTLD_LAZY); + if (!libsndio) { + DPR("sndio_init(%s) failed dlopen(libsndio.so)\n", context_name); + return CUBEB_ERROR; + } + } + +#define LOAD(x) { \ + cubeb_##x = dlsym(libsndio, #x); \ + if (!cubeb_##x) { \ + DPR("sndio_init(%s) failed dlsym(%s)\n", context_name, #x); \ + dlclose(libsndio); \ + return CUBEB_ERROR; \ + } \ + } + + LIBSNDIO_API_VISIT(LOAD); +#undef LOAD +#endif + + /* test if sndio works */ + hdl = WRAP(sio_open)(sndio_get_device(), SIO_PLAY, 1); + if (hdl == NULL) { + return CUBEB_ERROR; + } + WRAP(sio_close)(hdl); + + DPR("sndio_init(%s)\n", context_name); + *context = malloc(sizeof(**context)); + if (*context == NULL) + return CUBEB_ERROR; + (*context)->libsndio = libsndio; + (*context)->ops = &sndio_ops; + (void)context_name; + return CUBEB_OK; +} + +static char const * +sndio_get_backend_id(cubeb *context) +{ + return "sndio"; +} + +static void +sndio_destroy(cubeb *context) +{ + DPR("sndio_destroy()\n"); + if (context->libsndio) + dlclose(context->libsndio); + free(context); +} + +static int +sndio_stream_init(cubeb * context, + cubeb_stream ** stream, + char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void *user_ptr) +{ + cubeb_stream *s; + struct sio_par wpar, rpar; + cubeb_sample_format format; + int rate; + size_t bps; + + DPR("sndio_stream_init(%s)\n", stream_name); + + s = malloc(sizeof(cubeb_stream)); + if (s == NULL) + return CUBEB_ERROR; + memset(s, 0, sizeof(cubeb_stream)); + s->mode = 0; + if (input_stream_params) { + if (input_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + DPR("sndio_stream_init(), loopback not supported\n"); + goto err; + } + s->mode |= SIO_REC; + format = input_stream_params->format; + rate = input_stream_params->rate; + } + if (output_stream_params) { + if (output_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + DPR("sndio_stream_init(), loopback not supported\n"); + goto err; + } + s->mode |= SIO_PLAY; + format = output_stream_params->format; + rate = output_stream_params->rate; + } + if (s->mode == 0) { + DPR("sndio_stream_init(), neither playing nor recording\n"); + goto err; + } + s->context = context; + s->hdl = WRAP(sio_open)(sndio_get_device(), s->mode, 1); + if (s->hdl == NULL) { + DPR("sndio_stream_init(), sio_open() failed\n"); + goto err; + } + WRAP(sio_initpar)(&wpar); + wpar.sig = 1; + wpar.bits = 16; + switch (format) { + case CUBEB_SAMPLE_S16LE: + wpar.le = 1; + break; + case CUBEB_SAMPLE_S16BE: + wpar.le = 0; + break; + case CUBEB_SAMPLE_FLOAT32NE: + wpar.le = SIO_LE_NATIVE; + break; + default: + DPR("sndio_stream_init() unsupported format\n"); + goto err; + } + wpar.rate = rate; + if (s->mode & SIO_REC) + wpar.rchan = input_stream_params->channels; + if (s->mode & SIO_PLAY) + wpar.pchan = output_stream_params->channels; + wpar.appbufsz = latency_frames; + if (!WRAP(sio_setpar)(s->hdl, &wpar) || !WRAP(sio_getpar)(s->hdl, &rpar)) { + DPR("sndio_stream_init(), sio_setpar() failed\n"); + goto err; + } + if (rpar.bits != wpar.bits || rpar.le != wpar.le || + rpar.sig != wpar.sig || rpar.rate != wpar.rate || + ((s->mode & SIO_REC) && rpar.rchan != wpar.rchan) || + ((s->mode & SIO_PLAY) && rpar.pchan != wpar.pchan)) { + DPR("sndio_stream_init() unsupported params\n"); + goto err; + } + WRAP(sio_onmove)(s->hdl, sndio_onmove, s); + s->active = 0; + s->nfr = rpar.round; + s->rbpf = rpar.bps * rpar.rchan; + s->pbpf = rpar.bps * rpar.pchan; + s->rchan = rpar.rchan; + s->pchan = rpar.pchan; + s->nblks = rpar.bufsz / rpar.round; + s->data_cb = data_callback; + s->state_cb = state_callback; + s->arg = user_ptr; + s->mtx = (pthread_mutex_t)PTHREAD_MUTEX_INITIALIZER; + s->hwpos = s->swpos = 0; + if (format == CUBEB_SAMPLE_FLOAT32LE) { + s->conv = 1; + bps = sizeof(float); + } else { + s->conv = 0; + bps = rpar.bps; + } + if (s->mode & SIO_PLAY) { + s->pbuf = malloc(bps * rpar.pchan * rpar.round); + if (s->pbuf == NULL) + goto err; + } + if (s->mode & SIO_REC) { + s->rbuf = malloc(bps * rpar.rchan * rpar.round); + if (s->rbuf == NULL) + goto err; + } + s->volume = 1.; + *stream = s; + DPR("sndio_stream_init() end, ok\n"); + (void)context; + (void)stream_name; + return CUBEB_OK; +err: + if (s->hdl) + WRAP(sio_close)(s->hdl); + if (s->pbuf) + free(s->pbuf); + if (s->rbuf) + free(s->pbuf); + free(s); + return CUBEB_ERROR; +} + +static int +sndio_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + assert(ctx && max_channels); + + *max_channels = 8; + + return CUBEB_OK; +} + +static int +sndio_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) +{ + /* + * We've no device-independent prefered rate; any rate will work if + * sndiod is running. If it isn't, 48kHz is what is most likely to + * work as most (but not all) devices support it. + */ + *rate = 48000; + return CUBEB_OK; +} + +static int +sndio_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames) +{ + /* + * We've no device-independent minimum latency. + */ + *latency_frames = 2048; + + return CUBEB_OK; +} + +static void +sndio_stream_destroy(cubeb_stream *s) +{ + DPR("sndio_stream_destroy()\n"); + WRAP(sio_close)(s->hdl); + if (s->mode & SIO_PLAY) + free(s->pbuf); + if (s->mode & SIO_REC) + free(s->rbuf); + free(s); +} + +static int +sndio_stream_start(cubeb_stream *s) +{ + int err; + + DPR("sndio_stream_start()\n"); + s->active = 1; + err = pthread_create(&s->th, NULL, sndio_mainloop, s); + if (err) { + s->active = 0; + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +static int +sndio_stream_stop(cubeb_stream *s) +{ + void *dummy; + + DPR("sndio_stream_stop()\n"); + if (s->active) { + s->active = 0; + pthread_join(s->th, &dummy); + } + return CUBEB_OK; +} + +static int +sndio_stream_get_position(cubeb_stream *s, uint64_t *p) +{ + pthread_mutex_lock(&s->mtx); + DPR("sndio_stream_get_position() %" PRId64 "\n", s->hwpos); + *p = s->hwpos; + pthread_mutex_unlock(&s->mtx); + return CUBEB_OK; +} + +static int +sndio_stream_set_volume(cubeb_stream *s, float volume) +{ + DPR("sndio_stream_set_volume(%f)\n", volume); + pthread_mutex_lock(&s->mtx); + if (volume < 0.) + volume = 0.; + else if (volume > 1.0) + volume = 1.; + s->volume = volume; + pthread_mutex_unlock(&s->mtx); + return CUBEB_OK; +} + +int +sndio_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ + // http://www.openbsd.org/cgi-bin/man.cgi?query=sio_open + // in the "Measuring the latency and buffers usage" paragraph. + *latency = stm->swpos - stm->hwpos; + return CUBEB_OK; +} + +static int +sndio_enumerate_devices(cubeb *context, cubeb_device_type type, + cubeb_device_collection *collection) +{ + static char dev[] = SIO_DEVANY; + cubeb_device_info *device; + + device = malloc(sizeof(cubeb_device_info)); + if (device == NULL) + return CUBEB_ERROR; + + device->devid = dev; /* passed to stream_init() */ + device->device_id = dev; /* printable in UI */ + device->friendly_name = dev; /* same, but friendly */ + device->group_id = dev; /* actual device if full-duplex */ + device->vendor_name = NULL; /* may be NULL */ + device->type = type; /* Input/Output */ + device->state = CUBEB_DEVICE_STATE_ENABLED; + device->preferred = CUBEB_DEVICE_PREF_ALL; + device->format = CUBEB_DEVICE_FMT_S16NE; + device->default_format = CUBEB_DEVICE_FMT_S16NE; + device->max_channels = 16; + device->default_rate = 48000; + device->min_rate = 4000; + device->max_rate = 192000; + device->latency_lo = 480; + device->latency_hi = 9600; + collection->device = device; + collection->count = 1; + return CUBEB_OK; +} + +static int +sndio_device_collection_destroy(cubeb * context, + cubeb_device_collection * collection) +{ + free(collection->device); + return CUBEB_OK; +} + +static struct cubeb_ops const sndio_ops = { + .init = sndio_init, + .get_backend_id = sndio_get_backend_id, + .get_max_channel_count = sndio_get_max_channel_count, + .get_min_latency = sndio_get_min_latency, + .get_preferred_sample_rate = sndio_get_preferred_sample_rate, + .enumerate_devices = sndio_enumerate_devices, + .device_collection_destroy = sndio_device_collection_destroy, + .destroy = sndio_destroy, + .stream_init = sndio_stream_init, + .stream_destroy = sndio_stream_destroy, + .stream_start = sndio_stream_start, + .stream_stop = sndio_stream_stop, + .stream_reset_default_device = NULL, + .stream_get_position = sndio_stream_get_position, + .stream_get_latency = sndio_stream_get_latency, + .stream_set_volume = sndio_stream_set_volume, + .stream_get_current_device = NULL, + .stream_device_destroy = NULL, + .stream_register_device_changed_callback = NULL, + .register_device_collection_changed = NULL +}; diff --git a/dep/cubeb/src/cubeb_strings.c b/dep/cubeb/src/cubeb_strings.c new file mode 100644 index 000000000..79d7d21b3 --- /dev/null +++ b/dep/cubeb/src/cubeb_strings.c @@ -0,0 +1,155 @@ +/* + * Copyright © 2011 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#include "cubeb_strings.h" + +#include +#include +#include + +#define CUBEB_STRINGS_INLINE_COUNT 4 + +struct cubeb_strings { + uint32_t size; + uint32_t count; + char ** data; + char * small_store[CUBEB_STRINGS_INLINE_COUNT]; +}; + +int +cubeb_strings_init(cubeb_strings ** strings) +{ + cubeb_strings* strs = NULL; + + if (!strings) { + return CUBEB_ERROR; + } + + strs = calloc(1, sizeof(cubeb_strings)); + assert(strs); + + if (!strs) { + return CUBEB_ERROR; + } + + strs->size = sizeof(strs->small_store) / sizeof(strs->small_store[0]); + strs->count = 0; + strs->data = strs->small_store; + + *strings = strs; + + return CUBEB_OK; +} + +void +cubeb_strings_destroy(cubeb_strings * strings) +{ + char ** sp = NULL; + char ** se = NULL; + + if (!strings) { + return; + } + + sp = strings->data; + se = sp + strings->count; + + for ( ; sp != se; sp++) { + if (*sp) { + free(*sp); + } + } + + if (strings->data != strings->small_store) { + free(strings->data); + } + + free(strings); +} + +/** Look for string in string storage. + @param strings Opaque pointer to interned string storage. + @param s String to look up. + @retval Read-only string or NULL if not found. */ +static char const * +cubeb_strings_lookup(cubeb_strings * strings, char const * s) +{ + char ** sp = NULL; + char ** se = NULL; + + if (!strings || !s) { + return NULL; + } + + sp = strings->data; + se = sp + strings->count; + + for ( ; sp != se; sp++) { + if (*sp && strcmp(*sp, s) == 0) { + return *sp; + } + } + + return NULL; +} + +static char const * +cubeb_strings_push(cubeb_strings * strings, char const * s) +{ + char * is = NULL; + + if (strings->count == strings->size) { + char ** new_data; + uint32_t value_size = sizeof(char const *); + uint32_t new_size = strings->size * 2; + if (!new_size || value_size > (uint32_t)-1 / new_size) { + // overflow + return NULL; + } + + if (strings->small_store == strings->data) { + // First time heap allocation. + new_data = malloc(new_size * value_size); + if (new_data) { + memcpy(new_data, strings->small_store, sizeof(strings->small_store)); + } + } else { + new_data = realloc(strings->data, new_size * value_size); + } + + if (!new_data) { + // out of memory + return NULL; + } + + strings->size = new_size; + strings->data = new_data; + } + + is = strdup(s); + strings->data[strings->count++] = is; + + return is; +} + +char const * +cubeb_strings_intern(cubeb_strings * strings, char const * s) +{ + char const * is = NULL; + + if (!strings || !s) { + return NULL; + } + + is = cubeb_strings_lookup(strings, s); + if (is) { + return is; + } + + return cubeb_strings_push(strings, s); +} + diff --git a/dep/cubeb/src/cubeb_strings.h b/dep/cubeb/src/cubeb_strings.h new file mode 100644 index 000000000..a918a01c5 --- /dev/null +++ b/dep/cubeb/src/cubeb_strings.h @@ -0,0 +1,44 @@ +/* + * Copyright © 2011 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#ifndef CUBEB_STRINGS_H +#define CUBEB_STRINGS_H + +#include "cubeb/cubeb.h" + +#if defined(__cplusplus) +extern "C" { +#endif + +/** Opaque handle referencing interned string storage. */ +typedef struct cubeb_strings cubeb_strings; + +/** Initialize an interned string structure. + @param strings An out param where an opaque pointer to the + interned string storage will be returned. + @retval CUBEB_OK in case of success. + @retval CUBEB_ERROR in case of error. */ +CUBEB_EXPORT int cubeb_strings_init(cubeb_strings ** strings); + +/** Destroy an interned string structure freeing all associated memory. + @param strings An opaque pointer to the interned string storage to + destroy. */ +CUBEB_EXPORT void cubeb_strings_destroy(cubeb_strings * strings); + +/** Add string to internal storage. + @param strings Opaque pointer to interned string storage. + @param s String to add to storage. + @retval CUBEB_OK + @retval CUBEB_ERROR + */ +CUBEB_EXPORT char const * cubeb_strings_intern(cubeb_strings * strings, char const * s); + +#if defined(__cplusplus) +} +#endif + +#endif // !CUBEB_STRINGS_H diff --git a/dep/cubeb/src/cubeb_sun.c b/dep/cubeb/src/cubeb_sun.c new file mode 100644 index 000000000..64ab0b5b1 --- /dev/null +++ b/dep/cubeb/src/cubeb_sun.c @@ -0,0 +1,752 @@ +/* + * Copyright © 2019 Nia Alarie + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" + +#define BYTES_TO_FRAMES(bytes, channels) \ + (bytes / (channels * sizeof(int16_t))) + +#define FRAMES_TO_BYTES(frames, channels) \ + (frames * (channels * sizeof(int16_t))) + +/* Default to 4 + 1 for the default device. */ +#ifndef SUN_DEVICE_COUNT +#define SUN_DEVICE_COUNT (5) +#endif + +/* Supported well by most hardware. */ +#ifndef SUN_PREFER_RATE +#define SUN_PREFER_RATE (48000) +#endif + +/* Standard acceptable minimum. */ +#ifndef SUN_LATENCY_MS +#define SUN_LATENCY_MS (40) +#endif + +#ifndef SUN_DEFAULT_DEVICE +#define SUN_DEFAULT_DEVICE "/dev/audio" +#endif + +#ifndef SUN_POLL_TIMEOUT +#define SUN_POLL_TIMEOUT (1000) +#endif + +#ifndef SUN_BUFFER_FRAMES +#define SUN_BUFFER_FRAMES (32) +#endif + +/* + * Supported on NetBSD regardless of hardware. + */ + +#ifndef SUN_MAX_CHANNELS +# ifdef __NetBSD__ +# define SUN_MAX_CHANNELS (12) +# else +# define SUN_MAX_CHANNELS (2) +# endif +#endif + +#ifndef SUN_MIN_RATE +#define SUN_MIN_RATE (1000) +#endif + +#ifndef SUN_MAX_RATE +#define SUN_MAX_RATE (192000) +#endif + +static struct cubeb_ops const sun_ops; + +struct cubeb { + struct cubeb_ops const * ops; +}; + +struct cubeb_stream { + struct cubeb * context; + void * user_ptr; + pthread_t thread; + pthread_mutex_t mutex; /* protects running, volume, frames_written */ + int floating; + int running; + int play_fd; + int record_fd; + float volume; + struct audio_info p_info; /* info for the play fd */ + struct audio_info r_info; /* info for the record fd */ + cubeb_data_callback data_cb; + cubeb_state_callback state_cb; + int16_t * play_buf; + int16_t * record_buf; + float * f_play_buf; + float * f_record_buf; + char input_name[32]; + char output_name[32]; + uint64_t frames_written; + uint64_t blocks_written; +}; + +int +sun_init(cubeb ** context, char const * context_name) +{ + cubeb * c; + + (void)context_name; + if ((c = calloc(1, sizeof(cubeb))) == NULL) { + return CUBEB_ERROR; + } + c->ops = &sun_ops; + *context = c; + return CUBEB_OK; +} + +static void +sun_destroy(cubeb * context) +{ + free(context); +} + +static char const * +sun_get_backend_id(cubeb * context) +{ + return "sun"; +} + +static int +sun_get_preferred_sample_rate(cubeb * context, uint32_t * rate) +{ + (void)context; + + *rate = SUN_PREFER_RATE; + return CUBEB_OK; +} + +static int +sun_get_max_channel_count(cubeb * context, uint32_t * max_channels) +{ + (void)context; + + *max_channels = SUN_MAX_CHANNELS; + return CUBEB_OK; +} + +static int +sun_get_min_latency(cubeb * context, cubeb_stream_params params, + uint32_t * latency_frames) +{ + (void)context; + + *latency_frames = SUN_LATENCY_MS * params.rate / 1000; + return CUBEB_OK; +} + +static int +sun_get_hwinfo(const char * device, struct audio_info * format, + int * props, struct audio_device * dev) +{ + int fd = -1; + + if ((fd = open(device, O_RDONLY)) == -1) { + goto error; + } +#ifdef AUDIO_GETFORMAT + if (ioctl(fd, AUDIO_GETFORMAT, format) != 0) { + goto error; + } +#endif +#ifdef AUDIO_GETPROPS + if (ioctl(fd, AUDIO_GETPROPS, props) != 0) { + goto error; + } +#endif + if (ioctl(fd, AUDIO_GETDEV, dev) != 0) { + goto error; + } + close(fd); + return CUBEB_OK; +error: + if (fd != -1) { + close(fd); + } + return CUBEB_ERROR; +} + +/* + * XXX: PR kern/54264 + */ +static int +sun_prinfo_verify_sanity(struct audio_prinfo * prinfo) +{ + return prinfo->precision >= 8 && prinfo->precision <= 32 && + prinfo->channels >= 1 && prinfo->channels < SUN_MAX_CHANNELS && + prinfo->sample_rate < SUN_MAX_RATE && prinfo->sample_rate > SUN_MIN_RATE; +} + +static int +sun_enumerate_devices(cubeb * context, cubeb_device_type type, + cubeb_device_collection * collection) +{ + unsigned i; + cubeb_device_info device = {0}; + char dev[16] = SUN_DEFAULT_DEVICE; + char dev_friendly[64]; + struct audio_info hwfmt; + struct audio_device hwname; + struct audio_prinfo *prinfo = NULL; + int hwprops; + + collection->device = calloc(SUN_DEVICE_COUNT, sizeof(cubeb_device_info)); + if (collection->device == NULL) { + return CUBEB_ERROR; + } + collection->count = 0; + + for (i = 0; i < SUN_DEVICE_COUNT; ++i) { + if (i > 0) { + (void)snprintf(dev, sizeof(dev), "/dev/audio%u", i - 1); + } + if (sun_get_hwinfo(dev, &hwfmt, &hwprops, &hwname) != CUBEB_OK) { + continue; + } +#ifdef AUDIO_GETPROPS + device.type = 0; + if ((hwprops & AUDIO_PROP_CAPTURE) != 0 && + sun_prinfo_verify_sanity(&hwfmt.record)) { + /* the device supports recording, probably */ + device.type |= CUBEB_DEVICE_TYPE_INPUT; + } + if ((hwprops & AUDIO_PROP_PLAYBACK) != 0 && + sun_prinfo_verify_sanity(&hwfmt.play)) { + /* the device supports playback, probably */ + device.type |= CUBEB_DEVICE_TYPE_OUTPUT; + } + switch (device.type) { + case 0: + /* device doesn't do input or output, aliens probably involved */ + continue; + case CUBEB_DEVICE_TYPE_INPUT: + if ((type & CUBEB_DEVICE_TYPE_INPUT) == 0) { + /* this device is input only, not scanning for those, skip it */ + continue; + } + break; + case CUBEB_DEVICE_TYPE_OUTPUT: + if ((type & CUBEB_DEVICE_TYPE_OUTPUT) == 0) { + /* this device is output only, not scanning for those, skip it */ + continue; + } + break; + } + if ((type & CUBEB_DEVICE_TYPE_INPUT) != 0) { + prinfo = &hwfmt.record; + } + if ((type & CUBEB_DEVICE_TYPE_OUTPUT) != 0) { + prinfo = &hwfmt.play; + } +#endif + if (i > 0) { + (void)snprintf(dev_friendly, sizeof(dev_friendly), "%s %s %s (%d)", + hwname.name, hwname.version, hwname.config, i - 1); + } else { + (void)snprintf(dev_friendly, sizeof(dev_friendly), "%s %s %s (default)", + hwname.name, hwname.version, hwname.config); + } + device.devid = (void *)(uintptr_t)i; + device.device_id = strdup(dev); + device.friendly_name = strdup(dev_friendly); + device.group_id = strdup(dev); + device.vendor_name = strdup(hwname.name); + device.type = type; + device.state = CUBEB_DEVICE_STATE_ENABLED; + device.preferred = (i == 0) ? CUBEB_DEVICE_PREF_ALL : CUBEB_DEVICE_PREF_NONE; +#ifdef AUDIO_GETFORMAT + device.max_channels = prinfo->channels; + device.default_rate = prinfo->sample_rate; +#else + device.max_channels = 2; + device.default_rate = SUN_PREFER_RATE; +#endif + device.default_format = CUBEB_DEVICE_FMT_S16NE; + device.format = CUBEB_DEVICE_FMT_S16NE; + device.min_rate = SUN_MIN_RATE; + device.max_rate = SUN_MAX_RATE; + device.latency_lo = SUN_LATENCY_MS * SUN_MIN_RATE / 1000; + device.latency_hi = SUN_LATENCY_MS * SUN_MAX_RATE / 1000; + collection->device[collection->count++] = device; + } + return CUBEB_OK; +} + +static int +sun_device_collection_destroy(cubeb * context, + cubeb_device_collection * collection) +{ + unsigned i; + + for (i = 0; i < collection->count; ++i) { + free((char *)collection->device[i].device_id); + free((char *)collection->device[i].friendly_name); + free((char *)collection->device[i].group_id); + free((char *)collection->device[i].vendor_name); + } + free(collection->device); + return CUBEB_OK; +} + +static int +sun_copy_params(int fd, cubeb_stream * stream, cubeb_stream_params * params, + struct audio_info * info, struct audio_prinfo * prinfo) +{ + prinfo->channels = params->channels; + prinfo->sample_rate = params->rate; + prinfo->precision = 16; +#ifdef AUDIO_ENCODING_SLINEAR_LE + switch (params->format) { + case CUBEB_SAMPLE_S16LE: + prinfo->encoding = AUDIO_ENCODING_SLINEAR_LE; + break; + case CUBEB_SAMPLE_S16BE: + prinfo->encoding = AUDIO_ENCODING_SLINEAR_BE; + break; + case CUBEB_SAMPLE_FLOAT32NE: + stream->floating = 1; + prinfo->encoding = AUDIO_ENCODING_SLINEAR; + break; + default: + LOG("Unsupported format"); + return CUBEB_ERROR_INVALID_FORMAT; + } +#else + switch (params->format) { + case CUBEB_SAMPLE_S16NE: + prinfo->encoding = AUDIO_ENCODING_LINEAR; + break; + case CUBEB_SAMPLE_FLOAT32NE: + stream->floating = 1; + prinfo->encoding = AUDIO_ENCODING_LINEAR; + break; + default: + LOG("Unsupported format"); + return CUBEB_ERROR_INVALID_FORMAT; + } +#endif + if (ioctl(fd, AUDIO_SETINFO, info) == -1) { + return CUBEB_ERROR; + } + if (ioctl(fd, AUDIO_GETINFO, info) == -1) { + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +static int +sun_stream_stop(cubeb_stream * s) +{ + pthread_mutex_lock(&s->mutex); + if (s->running) { + s->running = 0; + pthread_mutex_unlock(&s->mutex); + pthread_join(s->thread, NULL); + } else { + pthread_mutex_unlock(&s->mutex); + } + return CUBEB_OK; +} + +static void +sun_stream_destroy(cubeb_stream * s) +{ + pthread_mutex_destroy(&s->mutex); + sun_stream_stop(s); + if (s->play_fd != -1) { + close(s->play_fd); + } + if (s->record_fd != -1) { + close(s->record_fd); + } + free(s->f_play_buf); + free(s->f_record_buf); + free(s->play_buf); + free(s->record_buf); + free(s); +} + +static void +sun_float_to_linear(float * in, int16_t * out, + unsigned channels, long frames, float vol) +{ + unsigned i, sample_count = frames * channels; + float multiplier = vol * 0x8000; + + for (i = 0; i < sample_count; ++i) { + int32_t sample = lrintf(in[i] * multiplier); + if (sample < -0x8000) { + out[i] = -0x8000; + } else if (sample > 0x7fff) { + out[i] = 0x7fff; + } else { + out[i] = sample; + } + } +} + +static void +sun_linear_to_float(int16_t * in, float * out, + unsigned channels, long frames) +{ + unsigned i, sample_count = frames * channels; + + for (i = 0; i < sample_count; ++i) { + out[i] = (1.0 / 0x8000) * in[i]; + } +} + +static void +sun_linear_set_vol(int16_t * buf, unsigned channels, long frames, float vol) +{ + unsigned i, sample_count = frames * channels; + int32_t multiplier = vol * 0x8000; + + for (i = 0; i < sample_count; ++i) { + buf[i] = (buf[i] * multiplier) >> 15; + } +} + +static void * +sun_io_routine(void * arg) +{ + cubeb_stream *s = arg; + cubeb_state state = CUBEB_STATE_STARTED; + size_t to_read = 0; + long to_write = 0; + size_t write_ofs = 0; + size_t read_ofs = 0; + int drain = 0; + + s->state_cb(s, s->user_ptr, CUBEB_STATE_STARTED); + while (state != CUBEB_STATE_ERROR) { + pthread_mutex_lock(&s->mutex); + if (!s->running) { + pthread_mutex_unlock(&s->mutex); + state = CUBEB_STATE_STOPPED; + break; + } + pthread_mutex_unlock(&s->mutex); + if (s->floating) { + if (s->record_fd != -1) { + sun_linear_to_float(s->record_buf, s->f_record_buf, + s->r_info.record.channels, SUN_BUFFER_FRAMES); + } + to_write = s->data_cb(s, s->user_ptr, + s->f_record_buf, s->f_play_buf, SUN_BUFFER_FRAMES); + if (to_write == CUBEB_ERROR) { + state = CUBEB_STATE_ERROR; + break; + } + if (s->play_fd != -1) { + pthread_mutex_lock(&s->mutex); + sun_float_to_linear(s->f_play_buf, s->play_buf, + s->p_info.play.channels, to_write, s->volume); + pthread_mutex_unlock(&s->mutex); + } + } else { + to_write = s->data_cb(s, s->user_ptr, + s->record_buf, s->play_buf, SUN_BUFFER_FRAMES); + if (to_write == CUBEB_ERROR) { + state = CUBEB_STATE_ERROR; + break; + } + if (s->play_fd != -1) { + pthread_mutex_lock(&s->mutex); + sun_linear_set_vol(s->play_buf, s->p_info.play.channels, to_write, s->volume); + pthread_mutex_unlock(&s->mutex); + } + } + if (to_write < SUN_BUFFER_FRAMES) { + drain = 1; + } + to_write = s->play_fd != -1 ? to_write : 0; + to_read = s->record_fd != -1 ? SUN_BUFFER_FRAMES : 0; + write_ofs = 0; + read_ofs = 0; + while (to_write > 0 || to_read > 0) { + size_t bytes; + ssize_t n, frames; + + if (to_write > 0) { + bytes = FRAMES_TO_BYTES(to_write, s->p_info.play.channels); + if ((n = write(s->play_fd, s->play_buf + write_ofs, bytes)) < 0) { + state = CUBEB_STATE_ERROR; + break; + } + frames = BYTES_TO_FRAMES(n, s->p_info.play.channels); + pthread_mutex_lock(&s->mutex); + s->frames_written += frames; + pthread_mutex_unlock(&s->mutex); + to_write -= frames; + write_ofs += frames; + } + if (to_read > 0) { + bytes = FRAMES_TO_BYTES(to_read, s->r_info.record.channels); + if ((n = read(s->record_fd, s->record_buf + read_ofs, bytes)) < 0) { + state = CUBEB_STATE_ERROR; + break; + } + frames = BYTES_TO_FRAMES(n, s->r_info.record.channels); + to_read -= frames; + read_ofs += frames; + } + } + if (drain && state != CUBEB_STATE_ERROR) { + state = CUBEB_STATE_DRAINED; + break; + } + } + s->state_cb(s, s->user_ptr, state); + return NULL; +} + +static int +sun_stream_init(cubeb * context, + cubeb_stream ** stream, + char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + int ret = CUBEB_OK; + cubeb_stream *s = NULL; + + (void)stream_name; + (void)latency_frames; + if ((s = calloc(1, sizeof(cubeb_stream))) == NULL) { + ret = CUBEB_ERROR; + goto error; + } + s->record_fd = -1; + s->play_fd = -1; + if (input_device != 0) { + snprintf(s->input_name, sizeof(s->input_name), + "/dev/audio%zu", (uintptr_t)input_device - 1); + } else { + snprintf(s->input_name, sizeof(s->input_name), "%s", SUN_DEFAULT_DEVICE); + } + if (output_device != 0) { + snprintf(s->output_name, sizeof(s->output_name), + "/dev/audio%zu", (uintptr_t)output_device - 1); + } else { + snprintf(s->output_name, sizeof(s->output_name), "%s", SUN_DEFAULT_DEVICE); + } + if (input_stream_params != NULL) { + if (input_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + LOG("Loopback not supported"); + ret = CUBEB_ERROR_NOT_SUPPORTED; + goto error; + } + if (s->record_fd == -1) { + if ((s->record_fd = open(s->input_name, O_RDONLY)) == -1) { + LOG("Audio device cannot be opened as read-only"); + ret = CUBEB_ERROR_DEVICE_UNAVAILABLE; + goto error; + } + } + AUDIO_INITINFO(&s->r_info); +#ifdef AUMODE_RECORD + s->r_info.mode = AUMODE_RECORD; +#endif + if ((ret = sun_copy_params(s->record_fd, s, input_stream_params, + &s->r_info, &s->r_info.record)) != CUBEB_OK) { + LOG("Setting record params failed"); + goto error; + } + } + if (output_stream_params != NULL) { + if (output_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + LOG("Loopback not supported"); + ret = CUBEB_ERROR_NOT_SUPPORTED; + goto error; + } + if (s->play_fd == -1) { + if ((s->play_fd = open(s->output_name, O_WRONLY)) == -1) { + LOG("Audio device cannot be opened as write-only"); + ret = CUBEB_ERROR_DEVICE_UNAVAILABLE; + goto error; + } + } + AUDIO_INITINFO(&s->p_info); +#ifdef AUMODE_PLAY + s->p_info.mode = AUMODE_PLAY; +#endif + if ((ret = sun_copy_params(s->play_fd, s, output_stream_params, + &s->p_info, &s->p_info.play)) != CUBEB_OK) { + LOG("Setting play params failed"); + goto error; + } + } + s->context = context; + s->volume = 1.0; + s->state_cb = state_callback; + s->data_cb = data_callback; + s->user_ptr = user_ptr; + if (pthread_mutex_init(&s->mutex, NULL) != 0) { + LOG("Failed to create mutex"); + goto error; + } + if (s->play_fd != -1 && (s->play_buf = calloc(SUN_BUFFER_FRAMES, + s->p_info.play.channels * sizeof(int16_t))) == NULL) { + ret = CUBEB_ERROR; + goto error; + } + if (s->record_fd != -1 && (s->record_buf = calloc(SUN_BUFFER_FRAMES, + s->r_info.record.channels * sizeof(int16_t))) == NULL) { + ret = CUBEB_ERROR; + goto error; + } + if (s->floating) { + if (s->play_fd != -1 && (s->f_play_buf = calloc(SUN_BUFFER_FRAMES, + s->p_info.play.channels * sizeof(float))) == NULL) { + ret = CUBEB_ERROR; + goto error; + } + if (s->record_fd != -1 && (s->f_record_buf = calloc(SUN_BUFFER_FRAMES, + s->r_info.record.channels * sizeof(float))) == NULL) { + ret = CUBEB_ERROR; + goto error; + } + } + *stream = s; + return CUBEB_OK; +error: + if (s != NULL) { + sun_stream_destroy(s); + } + return ret; +} + +static int +sun_stream_start(cubeb_stream * s) +{ + s->running = 1; + if (pthread_create(&s->thread, NULL, sun_io_routine, s) != 0) { + LOG("Couldn't create thread"); + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +static int +sun_stream_get_position(cubeb_stream * s, uint64_t * position) +{ +#ifdef AUDIO_GETOOFFS + struct audio_offset offset; + + if (ioctl(s->play_fd, AUDIO_GETOOFFS, &offset) == -1) { + return CUBEB_ERROR; + } + s->blocks_written += offset.deltablks; + *position = BYTES_TO_FRAMES(s->blocks_written * s->p_info.blocksize, + s->p_info.play.channels); + return CUBEB_OK; +#else + pthread_mutex_lock(&s->mutex); + *position = s->frames_written; + pthread_mutex_unlock(&s->mutex); + return CUBEB_OK; +#endif +} + +static int +sun_stream_get_latency(cubeb_stream * stream, uint32_t * latency) +{ +#ifdef AUDIO_GETBUFINFO + struct audio_info info; + + if (ioctl(stream->play_fd, AUDIO_GETBUFINFO, &info) == -1) { + return CUBEB_ERROR; + } + + *latency = BYTES_TO_FRAMES(info.play.seek + info.blocksize, + info.play.channels); + return CUBEB_OK; +#else + cubeb_stream_params params; + + params.rate = stream->p_info.play.sample_rate; + + return sun_get_min_latency(NULL, params, latency); +#endif +} + +static int +sun_stream_set_volume(cubeb_stream * stream, float volume) +{ + pthread_mutex_lock(&stream->mutex); + stream->volume = volume; + pthread_mutex_unlock(&stream->mutex); + return CUBEB_OK; +} + +static int +sun_get_current_device(cubeb_stream * stream, cubeb_device ** const device) +{ + *device = calloc(1, sizeof(cubeb_device)); + if (*device == NULL) { + return CUBEB_ERROR; + } + (*device)->input_name = stream->record_fd != -1 ? + strdup(stream->input_name) : NULL; + (*device)->output_name = stream->play_fd != -1 ? + strdup(stream->output_name) : NULL; + return CUBEB_OK; +} + +static int +sun_stream_device_destroy(cubeb_stream * stream, cubeb_device * device) +{ + (void)stream; + free(device->input_name); + free(device->output_name); + free(device); + return CUBEB_OK; +} + +static struct cubeb_ops const sun_ops = { + .init = sun_init, + .get_backend_id = sun_get_backend_id, + .get_max_channel_count = sun_get_max_channel_count, + .get_min_latency = sun_get_min_latency, + .get_preferred_sample_rate = sun_get_preferred_sample_rate, + .enumerate_devices = sun_enumerate_devices, + .device_collection_destroy = sun_device_collection_destroy, + .destroy = sun_destroy, + .stream_init = sun_stream_init, + .stream_destroy = sun_stream_destroy, + .stream_start = sun_stream_start, + .stream_stop = sun_stream_stop, + .stream_reset_default_device = NULL, + .stream_get_position = sun_stream_get_position, + .stream_get_latency = sun_stream_get_latency, + .stream_set_volume = sun_stream_set_volume, + .stream_get_current_device = sun_get_current_device, + .stream_device_destroy = sun_stream_device_destroy, + .stream_register_device_changed_callback = NULL, + .register_device_collection_changed = NULL +}; diff --git a/dep/cubeb/src/cubeb_utils.cpp b/dep/cubeb/src/cubeb_utils.cpp new file mode 100644 index 000000000..85572a9fe --- /dev/null +++ b/dep/cubeb/src/cubeb_utils.cpp @@ -0,0 +1,23 @@ +/* + * Copyright © 2018 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#include "cubeb_utils.h" + +size_t cubeb_sample_size(cubeb_sample_format format) +{ + switch (format) { + case CUBEB_SAMPLE_S16LE: + case CUBEB_SAMPLE_S16BE: + return sizeof(int16_t); + case CUBEB_SAMPLE_FLOAT32LE: + case CUBEB_SAMPLE_FLOAT32BE: + return sizeof(float); + default: + // should never happen as all cases are handled above. + assert(false); + } +} diff --git a/dep/cubeb/src/cubeb_utils.h b/dep/cubeb/src/cubeb_utils.h new file mode 100644 index 000000000..df6751155 --- /dev/null +++ b/dep/cubeb/src/cubeb_utils.h @@ -0,0 +1,343 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#if !defined(CUBEB_UTILS) +#define CUBEB_UTILS + +#include "cubeb/cubeb.h" + +#ifdef __cplusplus + +#include +#include +#include +#include +#include +#if defined(_WIN32) +#include "cubeb_utils_win.h" +#else +#include "cubeb_utils_unix.h" +#endif + +/** Similar to memcpy, but accounts for the size of an element. */ +template +void PodCopy(T * destination, const T * source, size_t count) +{ + static_assert(std::is_trivial::value, "Requires trivial type"); + assert(destination && source); + memcpy(destination, source, count * sizeof(T)); +} + +/** Similar to memmove, but accounts for the size of an element. */ +template +void PodMove(T * destination, const T * source, size_t count) +{ + static_assert(std::is_trivial::value, "Requires trivial type"); + assert(destination && source); + memmove(destination, source, count * sizeof(T)); +} + +/** Similar to a memset to zero, but accounts for the size of an element. */ +template +void PodZero(T * destination, size_t count) +{ + static_assert(std::is_trivial::value, "Requires trivial type"); + assert(destination); + memset(destination, 0, count * sizeof(T)); +} + +namespace { +template +void Copy(T * destination, const T * source, size_t count, Trait) +{ + for (size_t i = 0; i < count; i++) { + destination[i] = source[i]; + } +} + +template +void Copy(T * destination, const T * source, size_t count, std::true_type) +{ + PodCopy(destination, source, count); +} +} + +/** + * This allows copying a number of elements from a `source` pointer to a + * `destination` pointer, using `memcpy` if it is safe to do so, or a loop that + * calls the constructors and destructors otherwise. + */ +template +void Copy(T * destination, const T * source, size_t count) +{ + assert(destination && source); + Copy(destination, source, count, typename std::is_trivial::type()); +} + +namespace { +template +void ConstructDefault(T * destination, size_t count, Trait) +{ + for (size_t i = 0; i < count; i++) { + destination[i] = T(); + } +} + +template +void ConstructDefault(T * destination, + size_t count, std::true_type) +{ + PodZero(destination, count); +} +} + +/** + * This allows zeroing (using memset) or default-constructing a number of + * elements calling the constructors and destructors if necessary. + */ +template +void ConstructDefault(T * destination, size_t count) +{ + assert(destination); + ConstructDefault(destination, count, + typename std::is_arithmetic::type()); +} + +template +class auto_array +{ +public: + explicit auto_array(uint32_t capacity = 0) + : data_(capacity ? new T[capacity] : nullptr) + , capacity_(capacity) + , length_(0) + {} + + ~auto_array() + { + delete [] data_; + } + + /** Get a constant pointer to the underlying data. */ + T * data() const + { + return data_; + } + + T * end() const + { + return data_ + length_; + } + + const T& at(size_t index) const + { + assert(index < length_ && "out of range"); + return data_[index]; + } + + T& at(size_t index) + { + assert(index < length_ && "out of range"); + return data_[index]; + } + + /** Get how much underlying storage this auto_array has. */ + size_t capacity() const + { + return capacity_; + } + + /** Get how much elements this auto_array contains. */ + size_t length() const + { + return length_; + } + + /** Keeps the storage, but removes all the elements from the array. */ + void clear() + { + length_ = 0; + } + + /** Change the storage of this auto array, copying the elements to the new + * storage. + * @returns true in case of success + * @returns false if the new capacity is not big enough to accomodate for the + * elements in the array. + */ + bool reserve(size_t new_capacity) + { + if (new_capacity < length_) { + return false; + } + T * new_data = new T[new_capacity]; + if (data_ && length_) { + PodCopy(new_data, data_, length_); + } + capacity_ = new_capacity; + delete [] data_; + data_ = new_data; + + return true; + } + + /** Append `length` elements to the end of the array, resizing the array if + * needed. + * @parameter elements the elements to append to the array. + * @parameter length the number of elements to append to the array. + */ + void push(const T * elements, size_t length) + { + if (length_ + length > capacity_) { + reserve(length_ + length); + } + PodCopy(data_ + length_, elements, length); + length_ += length; + } + + /** Append `length` zero-ed elements to the end of the array, resizing the + * array if needed. + * @parameter length the number of elements to append to the array. + */ + void push_silence(size_t length) + { + if (length_ + length > capacity_) { + reserve(length + length_); + } + PodZero(data_ + length_, length); + length_ += length; + } + + /** Prepend `length` zero-ed elements to the end of the array, resizing the + * array if needed. + * @parameter length the number of elements to prepend to the array. + */ + void push_front_silence(size_t length) + { + if (length_ + length > capacity_) { + reserve(length + length_); + } + PodMove(data_ + length, data_, length_); + PodZero(data_, length); + length_ += length; + } + + /** Return the number of free elements in the array. */ + size_t available() const + { + return capacity_ - length_; + } + + /** Copies `length` elements to `elements` if it is not null, and shift + * the remaining elements of the `auto_array` to the beginning. + * @parameter elements a buffer to copy the elements to, or nullptr. + * @parameter length the number of elements to copy. + * @returns true in case of success. + * @returns false if the auto_array contains less than `length` elements. */ + bool pop(T * elements, size_t length) + { + if (length > length_) { + return false; + } + if (elements) { + PodCopy(elements, data_, length); + } + PodMove(data_, data_ + length, length_ - length); + + length_ -= length; + + return true; + } + + void set_length(size_t length) + { + assert(length <= capacity_); + length_ = length; + } + +private: + /** The underlying storage */ + T * data_; + /** The size, in number of elements, of the storage. */ + size_t capacity_; + /** The number of elements the array contains. */ + size_t length_; +}; + +struct auto_array_wrapper { + virtual void push(void * elements, size_t length) = 0; + virtual size_t length() = 0; + virtual void push_silence(size_t length) = 0; + virtual bool pop(size_t length) = 0; + virtual void * data() = 0; + virtual void * end() = 0; + virtual void clear() = 0; + virtual bool reserve(size_t capacity) = 0; + virtual void set_length(size_t length) = 0; + virtual ~auto_array_wrapper() {} +}; + +template +struct auto_array_wrapper_impl : public auto_array_wrapper { + auto_array_wrapper_impl() {} + + explicit auto_array_wrapper_impl(uint32_t size) + : ar(size) + {} + + void push(void * elements, size_t length) override { + ar.push(static_cast(elements), length); + } + + size_t length() override { + return ar.length(); + } + + void push_silence(size_t length) override { + ar.push_silence(length); + } + + bool pop(size_t length) override { + return ar.pop(nullptr, length); + } + + void * data() override { + return ar.data(); + } + + void * end() override { + return ar.end(); + } + + void clear() override { + ar.clear(); + } + + bool reserve(size_t capacity) override { + return ar.reserve(capacity); + } + + void set_length(size_t length) override { + ar.set_length(length); + } + + ~auto_array_wrapper_impl() { + ar.clear(); + } + +private: + auto_array ar; +}; + +extern "C" { + size_t cubeb_sample_size(cubeb_sample_format format); +} + +using auto_lock = std::lock_guard; +#endif // __cplusplus + +#endif /* CUBEB_UTILS */ diff --git a/dep/cubeb/src/cubeb_utils_unix.h b/dep/cubeb/src/cubeb_utils_unix.h new file mode 100644 index 000000000..4876d015f --- /dev/null +++ b/dep/cubeb/src/cubeb_utils_unix.h @@ -0,0 +1,89 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#if !defined(CUBEB_UTILS_UNIX) +#define CUBEB_UTILS_UNIX + +#include +#include +#include + +/* This wraps a critical section to track the owner in debug mode. */ +class owned_critical_section +{ +public: + owned_critical_section() + { + pthread_mutexattr_t attr; + pthread_mutexattr_init(&attr); +#ifndef NDEBUG + pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_ERRORCHECK); +#else + pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_NORMAL); +#endif + +#ifndef NDEBUG + int r = +#endif + pthread_mutex_init(&mutex, &attr); +#ifndef NDEBUG + assert(r == 0); +#endif + + pthread_mutexattr_destroy(&attr); + } + + ~owned_critical_section() + { +#ifndef NDEBUG + int r = +#endif + pthread_mutex_destroy(&mutex); +#ifndef NDEBUG + assert(r == 0); +#endif + } + + void lock() + { +#ifndef NDEBUG + int r = +#endif + pthread_mutex_lock(&mutex); +#ifndef NDEBUG + assert(r == 0 && "Deadlock"); +#endif + } + + void unlock() + { +#ifndef NDEBUG + int r = +#endif + pthread_mutex_unlock(&mutex); +#ifndef NDEBUG + assert(r == 0 && "Unlocking unlocked mutex"); +#endif + } + + void assert_current_thread_owns() + { +#ifndef NDEBUG + int r = pthread_mutex_lock(&mutex); + assert(r == EDEADLK); +#endif + } + +private: + pthread_mutex_t mutex; + + // Disallow copy and assignment because pthread_mutex_t cannot be copied. + owned_critical_section(const owned_critical_section&); + owned_critical_section& operator=(const owned_critical_section&); +}; + +#endif /* CUBEB_UTILS_UNIX */ diff --git a/dep/cubeb/src/cubeb_utils_win.h b/dep/cubeb/src/cubeb_utils_win.h new file mode 100644 index 000000000..0112ad6d3 --- /dev/null +++ b/dep/cubeb/src/cubeb_utils_win.h @@ -0,0 +1,71 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ + +#if !defined(CUBEB_UTILS_WIN) +#define CUBEB_UTILS_WIN + +#include +#include "cubeb-internal.h" + +/* This wraps a critical section to track the owner in debug mode, adapted from + NSPR and http://blogs.msdn.com/b/oldnewthing/archive/2013/07/12/10433554.aspx */ +class owned_critical_section +{ +public: + owned_critical_section() +#ifndef NDEBUG + : owner(0) +#endif + { + InitializeCriticalSection(&critical_section); + } + + ~owned_critical_section() + { + DeleteCriticalSection(&critical_section); + } + + void lock() + { + EnterCriticalSection(&critical_section); +#ifndef NDEBUG + XASSERT(owner != GetCurrentThreadId() && "recursive locking"); + owner = GetCurrentThreadId(); +#endif + } + + void unlock() + { +#ifndef NDEBUG + /* GetCurrentThreadId cannot return 0: it is not a the valid thread id */ + owner = 0; +#endif + LeaveCriticalSection(&critical_section); + } + + /* This is guaranteed to have the good behaviour if it succeeds. The behaviour + is undefined otherwise. */ + void assert_current_thread_owns() + { +#ifndef NDEBUG + /* This implies owner != 0, because GetCurrentThreadId cannot return 0. */ + XASSERT(owner == GetCurrentThreadId()); +#endif + } + +private: + CRITICAL_SECTION critical_section; +#ifndef NDEBUG + DWORD owner; +#endif + + // Disallow copy and assignment because CRICICAL_SECTION cannot be copied. + owned_critical_section(const owned_critical_section&); + owned_critical_section& operator=(const owned_critical_section&); +}; + +#endif /* CUBEB_UTILS_WIN */ diff --git a/dep/cubeb/src/cubeb_wasapi.cpp b/dep/cubeb/src/cubeb_wasapi.cpp new file mode 100644 index 000000000..e85462f12 --- /dev/null +++ b/dep/cubeb/src/cubeb_wasapi.cpp @@ -0,0 +1,2965 @@ +/* + * Copyright © 2013 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#define _WIN32_WINNT 0x0600 +#define NOMINMAX + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" +#include "cubeb_mixer.h" +#include "cubeb_resampler.h" +#include "cubeb_strings.h" +#include "cubeb_utils.h" + +// Windows 10 exposes the IAudioClient3 interface to create low-latency streams. +// Copy the interface definition from audioclient.h here to make the code simpler +// and so that we can still access IAudioClient3 via COM if cubeb was compiled +// against an older SDK. +#ifndef __IAudioClient3_INTERFACE_DEFINED__ +#define __IAudioClient3_INTERFACE_DEFINED__ +MIDL_INTERFACE("7ED4EE07-8E67-4CD4-8C1A-2B7A5987AD42") +IAudioClient3 : public IAudioClient +{ +public: + virtual HRESULT STDMETHODCALLTYPE GetSharedModeEnginePeriod( + /* [annotation][in] */ + _In_ const WAVEFORMATEX *pFormat, + /* [annotation][out] */ + _Out_ UINT32 *pDefaultPeriodInFrames, + /* [annotation][out] */ + _Out_ UINT32 *pFundamentalPeriodInFrames, + /* [annotation][out] */ + _Out_ UINT32 *pMinPeriodInFrames, + /* [annotation][out] */ + _Out_ UINT32 *pMaxPeriodInFrames) = 0; + + virtual HRESULT STDMETHODCALLTYPE GetCurrentSharedModeEnginePeriod( + /* [unique][annotation][out] */ + _Out_ WAVEFORMATEX **ppFormat, + /* [annotation][out] */ + _Out_ UINT32 *pCurrentPeriodInFrames) = 0; + + virtual HRESULT STDMETHODCALLTYPE InitializeSharedAudioStream( + /* [annotation][in] */ + _In_ DWORD StreamFlags, + /* [annotation][in] */ + _In_ UINT32 PeriodInFrames, + /* [annotation][in] */ + _In_ const WAVEFORMATEX *pFormat, + /* [annotation][in] */ + _In_opt_ LPCGUID AudioSessionGuid) = 0; +}; +#ifdef __CRT_UUID_DECL +// Required for MinGW +__CRT_UUID_DECL(IAudioClient3, 0x7ED4EE07, 0x8E67, 0x4CD4, 0x8C, 0x1A, 0x2B, 0x7A, 0x59, 0x87, 0xAD, 0x42) +#endif +#endif +// Copied from audioclient.h in the Windows 10 SDK +#ifndef AUDCLNT_E_ENGINE_PERIODICITY_LOCKED +#define AUDCLNT_E_ENGINE_PERIODICITY_LOCKED AUDCLNT_ERR(0x028) +#endif + +#ifndef PKEY_Device_FriendlyName +DEFINE_PROPERTYKEY(PKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 14); // DEVPROP_TYPE_STRING +#endif +#ifndef PKEY_Device_InstanceId +DEFINE_PROPERTYKEY(PKEY_Device_InstanceId, 0x78c34fc8, 0x104a, 0x4aca, 0x9e, 0xa4, 0x52, 0x4d, 0x52, 0x99, 0x6e, 0x57, 0x00000100); // VT_LPWSTR +#endif + +namespace { +struct com_heap_ptr_deleter { + void operator()(void * ptr) const noexcept { + CoTaskMemFree(ptr); + } +}; + +template +using com_heap_ptr = std::unique_ptr; + +template +constexpr size_t +ARRAY_LENGTH(T(&)[N]) +{ + return N; +} + +template +class no_addref_release : public T { + ULONG STDMETHODCALLTYPE AddRef() = 0; + ULONG STDMETHODCALLTYPE Release() = 0; +}; + +template +class com_ptr { +public: + com_ptr() noexcept = default; + + com_ptr(com_ptr const & other) noexcept = delete; + com_ptr & operator=(com_ptr const & other) noexcept = delete; + T ** operator&() const noexcept = delete; + + ~com_ptr() noexcept { + release(); + } + + com_ptr(com_ptr && other) noexcept + : ptr(other.ptr) + { + other.ptr = nullptr; + } + + com_ptr & operator=(com_ptr && other) noexcept { + if (ptr != other.ptr) { + release(); + ptr = other.ptr; + other.ptr = nullptr; + } + return *this; + } + + explicit operator bool() const noexcept { + return nullptr != ptr; + } + + no_addref_release * operator->() const noexcept { + return static_cast *>(ptr); + } + + T * get() const noexcept { + return ptr; + } + + T ** receive() noexcept { + XASSERT(ptr == nullptr); + return &ptr; + } + + void ** receive_vpp() noexcept { + return reinterpret_cast(receive()); + } + + com_ptr & operator=(std::nullptr_t) noexcept { + release(); + return *this; + } + + void reset(T * p = nullptr) noexcept { + release(); + ptr = p; + } + +private: + void release() noexcept { + T * temp = ptr; + + if (temp) { + ptr = nullptr; + temp->Release(); + } + } + + T * ptr = nullptr; +}; + +extern cubeb_ops const wasapi_ops; + +int wasapi_stream_stop(cubeb_stream * stm); +int wasapi_stream_start(cubeb_stream * stm); +void close_wasapi_stream(cubeb_stream * stm); +int setup_wasapi_stream(cubeb_stream * stm); +ERole pref_to_role(cubeb_stream_prefs param); +static char const * wstr_to_utf8(wchar_t const * str); +static std::unique_ptr utf8_to_wstr(char const * str); + +} + +class wasapi_collection_notification_client; +class monitor_device_notifications; + +struct cubeb { + cubeb_ops const * ops = &wasapi_ops; + cubeb_strings * device_ids; + /* Device enumerator to get notifications when the + device collection change. */ + com_ptr device_collection_enumerator; + com_ptr collection_notification_client; + /* Collection changed for input (capture) devices. */ + cubeb_device_collection_changed_callback input_collection_changed_callback = nullptr; + void * input_collection_changed_user_ptr = nullptr; + /* Collection changed for output (render) devices. */ + cubeb_device_collection_changed_callback output_collection_changed_callback = nullptr; + void * output_collection_changed_user_ptr = nullptr; +}; + +class wasapi_endpoint_notification_client; + +/* We have three possible callbacks we can use with a stream: + * - input only + * - output only + * - synchronized input and output + * + * Returns true when we should continue to play, false otherwise. + */ +typedef bool (*wasapi_refill_callback)(cubeb_stream * stm); + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context = nullptr; + void * user_ptr = nullptr; + /**/ + + /* Mixer pameters. We need to convert the input stream to this + samplerate/channel layout, as WASAPI does not resample nor upmix + itself. */ + cubeb_stream_params input_mix_params = { CUBEB_SAMPLE_FLOAT32NE, 0, 0, CUBEB_LAYOUT_UNDEFINED, CUBEB_STREAM_PREF_NONE }; + cubeb_stream_params output_mix_params = { CUBEB_SAMPLE_FLOAT32NE, 0, 0, CUBEB_LAYOUT_UNDEFINED, CUBEB_STREAM_PREF_NONE }; + /* Stream parameters. This is what the client requested, + * and what will be presented in the callback. */ + cubeb_stream_params input_stream_params = { CUBEB_SAMPLE_FLOAT32NE, 0, 0, CUBEB_LAYOUT_UNDEFINED, CUBEB_STREAM_PREF_NONE }; + cubeb_stream_params output_stream_params = { CUBEB_SAMPLE_FLOAT32NE, 0, 0, CUBEB_LAYOUT_UNDEFINED, CUBEB_STREAM_PREF_NONE }; + /* A MMDevice role for this stream: either communication or console here. */ + ERole role; + /* The input and output device, or NULL for default. */ + std::unique_ptr input_device; + std::unique_ptr output_device; + /* The latency initially requested for this stream, in frames. */ + unsigned latency = 0; + cubeb_state_callback state_callback = nullptr; + cubeb_data_callback data_callback = nullptr; + wasapi_refill_callback refill_callback = nullptr; + /* True when a loopback device is requested with no output device. In this + case a dummy output device is opened to drive the loopback, but should not + be exposed. */ + bool has_dummy_output = false; + /* Lifetime considerations: + - client, render_client, audio_clock and audio_stream_volume are interface + pointer to the IAudioClient. + - The lifetime for device_enumerator and notification_client, resampler, + mix_buffer are the same as the cubeb_stream instance. */ + + /* Main handle on the WASAPI stream. */ + com_ptr output_client; + /* Interface pointer to use the event-driven interface. */ + com_ptr render_client; + /* Interface pointer to use the volume facilities. */ + com_ptr audio_stream_volume; + /* Interface pointer to use the stream audio clock. */ + com_ptr audio_clock; + /* Frames written to the stream since it was opened. Reset on device + change. Uses mix_params.rate. */ + UINT64 frames_written = 0; + /* Frames written to the (logical) stream since it was first + created. Updated on device change. Uses stream_params.rate. */ + UINT64 total_frames_written = 0; + /* Last valid reported stream position. Used to ensure the position + reported by stream_get_position increases monotonically. */ + UINT64 prev_position = 0; + /* Device enumerator to be able to be notified when the default + device change. */ + com_ptr device_enumerator; + /* Device notification client, to be able to be notified when the default + audio device changes and route the audio to the new default audio output + device */ + com_ptr notification_client; + /* Main andle to the WASAPI capture stream. */ + com_ptr input_client; + /* Interface to use the event driven capture interface */ + com_ptr capture_client; + /* This event is set by the stream_stop and stream_destroy + function, so the render loop can exit properly. */ + HANDLE shutdown_event = 0; + /* Set by OnDefaultDeviceChanged when a stream reconfiguration is required. + The reconfiguration is handled by the render loop thread. */ + HANDLE reconfigure_event = 0; + /* This is set by WASAPI when we should refill the stream. */ + HANDLE refill_event = 0; + /* This is set by WASAPI when we should read from the input stream. In + * practice, we read from the input stream in the output callback, so + * this is not used, but it is necessary to start getting input data. */ + HANDLE input_available_event = 0; + /* Each cubeb_stream has its own thread. */ + HANDLE thread = 0; + /* The lock protects all members that are touched by the render thread or + change during a device reset, including: audio_clock, audio_stream_volume, + client, frames_written, mix_params, total_frames_written, prev_position. */ + owned_critical_section stream_reset_lock; + /* Maximum number of frames that can be passed down in a callback. */ + uint32_t input_buffer_frame_count = 0; + /* Maximum number of frames that can be requested in a callback. */ + uint32_t output_buffer_frame_count = 0; + /* Resampler instance. Resampling will only happen if necessary. */ + std::unique_ptr resampler = { nullptr, cubeb_resampler_destroy }; + /* Mixer interfaces */ + std::unique_ptr output_mixer = { nullptr, cubeb_mixer_destroy }; + std::unique_ptr input_mixer = { nullptr, cubeb_mixer_destroy }; + /* A buffer for up/down mixing multi-channel audio output. */ + std::vector mix_buffer; + /* WASAPI input works in "packets". We re-linearize the audio packets + * into this buffer before handing it to the resampler. */ + std::unique_ptr linear_input_buffer; + /* Bytes per sample. This multiplied by the number of channels is the number + * of bytes per frame. */ + size_t bytes_per_sample = 0; + /* WAVEFORMATEXTENSIBLE sub-format: either PCM or float. */ + GUID waveformatextensible_sub_format = GUID_NULL; + /* Stream volume. Set via stream_set_volume and used to reset volume on + device changes. */ + float volume = 1.0; + /* True if the stream is draining. */ + bool draining = false; + /* True when we've destroyed the stream. This pointer is leaked on stream + * destruction if we could not join the thread. */ + std::atomic*> emergency_bailout { nullptr }; + /* Synchronizes render thread start to ensure safe access to emergency_bailout. */ + HANDLE thread_ready_event = 0; +}; + +class monitor_device_notifications { +public: + monitor_device_notifications(cubeb * context) + : cubeb_context(context) + { + create_thread(); + } + + ~monitor_device_notifications() + { + SetEvent(shutdown); + WaitForSingleObject(thread, INFINITE); + CloseHandle(thread); + + CloseHandle(input_changed); + CloseHandle(output_changed); + CloseHandle(shutdown); + } + + void notify(EDataFlow flow) + { + XASSERT(cubeb_context); + if (flow == eCapture && cubeb_context->input_collection_changed_callback) { + bool res = SetEvent(input_changed); + if (!res) { + LOG("Failed to set input changed event"); + } + return; + } + if (flow == eRender && cubeb_context->output_collection_changed_callback) { + bool res = SetEvent(output_changed); + if (!res) { + LOG("Failed to set output changed event"); + } + } + } +private: + static unsigned int __stdcall + thread_proc(LPVOID args) + { + XASSERT(args); + static_cast(args) + ->notification_thread_loop(); + return 0; + } + + void notification_thread_loop() + { + struct auto_com { + auto_com() { + HRESULT hr = CoInitializeEx(nullptr, COINIT_MULTITHREADED); + XASSERT(SUCCEEDED(hr)); + } + ~auto_com() { + CoUninitialize(); + } + } com; + + HANDLE wait_array[3] = { + input_changed, + output_changed, + shutdown, + }; + + while (true) { + Sleep(200); + + DWORD wait_result = WaitForMultipleObjects(ARRAY_LENGTH(wait_array), + wait_array, + FALSE, + INFINITE); + if (wait_result == WAIT_OBJECT_0) { // input changed + cubeb_context->input_collection_changed_callback(cubeb_context, + cubeb_context->input_collection_changed_user_ptr); + } else if (wait_result == WAIT_OBJECT_0 + 1) { // output changed + cubeb_context->output_collection_changed_callback(cubeb_context, + cubeb_context->output_collection_changed_user_ptr); + } else if (wait_result == WAIT_OBJECT_0 + 2) { // shutdown + break; + } else { + LOG("Unexpected result %lu", wait_result); + } + } // loop + } + + void create_thread() + { + output_changed = CreateEvent(nullptr, 0, 0, nullptr); + if (!output_changed) { + LOG("Failed to create output changed event."); + return; + } + + input_changed = CreateEvent(nullptr, 0, 0, nullptr); + if (!input_changed) { + LOG("Failed to create input changed event."); + return; + } + + shutdown = CreateEvent(nullptr, 0, 0, nullptr); + if (!shutdown) { + LOG("Failed to create shutdown event."); + return; + } + + thread = (HANDLE) _beginthreadex(nullptr, + 256 * 1024, + thread_proc, + this, + STACK_SIZE_PARAM_IS_A_RESERVATION, + nullptr); + if (!thread) { + LOG("Failed to create thread."); + return; + } + } + + HANDLE thread = INVALID_HANDLE_VALUE; + HANDLE output_changed = INVALID_HANDLE_VALUE; + HANDLE input_changed = INVALID_HANDLE_VALUE; + HANDLE shutdown = INVALID_HANDLE_VALUE; + + cubeb * cubeb_context = nullptr; +}; + +class wasapi_collection_notification_client : public IMMNotificationClient +{ +public: + /* The implementation of MSCOM was copied from MSDN. */ + ULONG STDMETHODCALLTYPE + AddRef() + { + return InterlockedIncrement(&ref_count); + } + + ULONG STDMETHODCALLTYPE + Release() + { + ULONG ulRef = InterlockedDecrement(&ref_count); + if (0 == ulRef) { + delete this; + } + return ulRef; + } + + HRESULT STDMETHODCALLTYPE + QueryInterface(REFIID riid, VOID **ppvInterface) + { + if (__uuidof(IUnknown) == riid) { + AddRef(); + *ppvInterface = (IUnknown*)this; + } else if (__uuidof(IMMNotificationClient) == riid) { + AddRef(); + *ppvInterface = (IMMNotificationClient*)this; + } else { + *ppvInterface = NULL; + return E_NOINTERFACE; + } + return S_OK; + } + + wasapi_collection_notification_client(cubeb * context) + : ref_count(1) + , cubeb_context(context) + , monitor_notifications(context) + { + XASSERT(cubeb_context); + } + + virtual ~wasapi_collection_notification_client() + { } + + HRESULT STDMETHODCALLTYPE + OnDefaultDeviceChanged(EDataFlow flow, ERole role, LPCWSTR device_id) + { + LOG("collection: Audio device default changed, id = %S.", device_id); + return S_OK; + } + + /* The remaining methods are not implemented, they simply log when called (if + log is enabled), for debugging. */ + HRESULT STDMETHODCALLTYPE OnDeviceAdded(LPCWSTR device_id) + { + LOG("collection: Audio device added."); + return S_OK; + }; + + HRESULT STDMETHODCALLTYPE OnDeviceRemoved(LPCWSTR device_id) + { + LOG("collection: Audio device removed."); + return S_OK; + } + + HRESULT STDMETHODCALLTYPE + OnDeviceStateChanged(LPCWSTR device_id, DWORD new_state) + { + XASSERT(cubeb_context->output_collection_changed_callback || + cubeb_context->input_collection_changed_callback); + LOG("collection: Audio device state changed, id = %S, state = %lu.", device_id, new_state); + EDataFlow flow; + HRESULT hr = GetDataFlow(device_id, &flow); + if (FAILED(hr)) { + return hr; + } + monitor_notifications.notify(flow); + return S_OK; + } + + HRESULT STDMETHODCALLTYPE + OnPropertyValueChanged(LPCWSTR device_id, const PROPERTYKEY key) + { + //Audio device property value changed. + return S_OK; + } + +private: + HRESULT GetDataFlow(LPCWSTR device_id, EDataFlow * flow) + { + com_ptr device; + com_ptr endpoint; + + HRESULT hr = cubeb_context->device_collection_enumerator + ->GetDevice(device_id, device.receive()); + if (FAILED(hr)) { + LOG("collection: Could not get device: %lx", hr); + return hr; + } + + hr = device->QueryInterface(IID_PPV_ARGS(endpoint.receive())); + if (FAILED(hr)) { + LOG("collection: Could not get endpoint: %lx", hr); + return hr; + } + + return endpoint->GetDataFlow(flow); + } + + /* refcount for this instance, necessary to implement MSCOM semantics. */ + LONG ref_count; + + cubeb * cubeb_context = nullptr; + monitor_device_notifications monitor_notifications; +}; + +class wasapi_endpoint_notification_client : public IMMNotificationClient +{ +public: + /* The implementation of MSCOM was copied from MSDN. */ + ULONG STDMETHODCALLTYPE + AddRef() + { + return InterlockedIncrement(&ref_count); + } + + ULONG STDMETHODCALLTYPE + Release() + { + ULONG ulRef = InterlockedDecrement(&ref_count); + if (0 == ulRef) { + delete this; + } + return ulRef; + } + + HRESULT STDMETHODCALLTYPE + QueryInterface(REFIID riid, VOID **ppvInterface) + { + if (__uuidof(IUnknown) == riid) { + AddRef(); + *ppvInterface = (IUnknown*)this; + } else if (__uuidof(IMMNotificationClient) == riid) { + AddRef(); + *ppvInterface = (IMMNotificationClient*)this; + } else { + *ppvInterface = NULL; + return E_NOINTERFACE; + } + return S_OK; + } + + wasapi_endpoint_notification_client(HANDLE event, ERole role) + : ref_count(1) + , reconfigure_event(event) + , role(role) + { } + + virtual ~wasapi_endpoint_notification_client() + { } + + HRESULT STDMETHODCALLTYPE + OnDefaultDeviceChanged(EDataFlow flow, ERole role, LPCWSTR device_id) + { + LOG("endpoint: Audio device default changed."); + + /* we only support a single stream type for now. */ + if (flow != eRender && role != this->role) { + return S_OK; + } + + BOOL ok = SetEvent(reconfigure_event); + if (!ok) { + LOG("endpoint: SetEvent on reconfigure_event failed: %lx", GetLastError()); + } + + return S_OK; + } + + /* The remaining methods are not implemented, they simply log when called (if + log is enabled), for debugging. */ + HRESULT STDMETHODCALLTYPE OnDeviceAdded(LPCWSTR device_id) + { + LOG("endpoint: Audio device added."); + return S_OK; + }; + + HRESULT STDMETHODCALLTYPE OnDeviceRemoved(LPCWSTR device_id) + { + LOG("endpoint: Audio device removed."); + return S_OK; + } + + HRESULT STDMETHODCALLTYPE + OnDeviceStateChanged(LPCWSTR device_id, DWORD new_state) + { + LOG("endpoint: Audio device state changed."); + return S_OK; + } + + HRESULT STDMETHODCALLTYPE + OnPropertyValueChanged(LPCWSTR device_id, const PROPERTYKEY key) + { + //Audio device property value changed. + return S_OK; + } +private: + /* refcount for this instance, necessary to implement MSCOM semantics. */ + LONG ref_count; + HANDLE reconfigure_event; + ERole role; +}; + +namespace { + +char const * +intern_device_id(cubeb * ctx, wchar_t const * id) +{ + XASSERT(id); + + char const * tmp = wstr_to_utf8(id); + if (!tmp) + return nullptr; + + char const * interned = cubeb_strings_intern(ctx->device_ids, tmp); + + free((void *) tmp); + + return interned; +} + +bool has_input(cubeb_stream * stm) +{ + return stm->input_stream_params.rate != 0; +} + +bool has_output(cubeb_stream * stm) +{ + return stm->output_stream_params.rate != 0; +} + +double stream_to_mix_samplerate_ratio(cubeb_stream_params & stream, cubeb_stream_params & mixer) +{ + return double(stream.rate) / mixer.rate; +} + +/* Convert the channel layout into the corresponding KSAUDIO_CHANNEL_CONFIG. + See more: https://msdn.microsoft.com/en-us/library/windows/hardware/ff537083(v=vs.85).aspx */ + +cubeb_channel_layout +mask_to_channel_layout(WAVEFORMATEX const * fmt) +{ + cubeb_channel_layout mask = 0; + + if (fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { + WAVEFORMATEXTENSIBLE const * ext = reinterpret_cast(fmt); + mask = ext->dwChannelMask; + } else if (fmt->wFormatTag == WAVE_FORMAT_PCM || + fmt->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) { + if (fmt->nChannels == 1) { + mask = CHANNEL_FRONT_CENTER; + } else if (fmt->nChannels == 2) { + mask = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT; + } + } + return mask; +} + +uint32_t +get_rate(cubeb_stream * stm) +{ + return has_input(stm) ? stm->input_stream_params.rate + : stm->output_stream_params.rate; +} + +uint32_t +hns_to_frames(uint32_t rate, REFERENCE_TIME hns) +{ + return std::ceil((hns - 1) / 10000000.0 * rate); +} + +uint32_t +hns_to_frames(cubeb_stream * stm, REFERENCE_TIME hns) +{ + return hns_to_frames(get_rate(stm), hns); +} + +REFERENCE_TIME +frames_to_hns(cubeb_stream * stm, uint32_t frames) +{ + return std::ceil(frames * 10000000.0 / get_rate(stm)); +} + +/* This returns the size of a frame in the stream, before the eventual upmix + occurs. */ +static size_t +frames_to_bytes_before_mix(cubeb_stream * stm, size_t frames) +{ + // This is called only when we has a output client. + XASSERT(has_output(stm)); + return stm->output_stream_params.channels * stm->bytes_per_sample * frames; +} + +/* This function handles the processing of the input and output audio, + * converting it to rate and channel layout specified at initialization. + * It then calls the data callback, via the resampler. */ +long +refill(cubeb_stream * stm, void * input_buffer, long input_frames_count, + void * output_buffer, long output_frames_needed) +{ + XASSERT(!stm->draining); + /* If we need to upmix after resampling, resample into the mix buffer to + avoid a copy. Avoid exposing output if it is a dummy stream. */ + void * dest = nullptr; + if (has_output(stm) && !stm->has_dummy_output) { + if (stm->output_mixer) { + dest = stm->mix_buffer.data(); + } else { + dest = output_buffer; + } + } + + long out_frames = cubeb_resampler_fill(stm->resampler.get(), + input_buffer, + &input_frames_count, + dest, + output_frames_needed); + /* TODO: Report out_frames < 0 as an error via the API. */ + XASSERT(out_frames >= 0); + + { + auto_lock lock(stm->stream_reset_lock); + stm->frames_written += out_frames; + } + + /* Go in draining mode if we got fewer frames than requested. If the stream + has no output we still expect the callback to return number of frames read + from input, otherwise we stop. */ + if ((out_frames < output_frames_needed) || + (!has_output(stm) && out_frames < input_frames_count)) { + LOG("start draining."); + stm->draining = true; + } + + /* If this is not true, there will be glitches. + It is alright to have produced less frames if we are draining, though. */ + XASSERT(out_frames == output_frames_needed || stm->draining || !has_output(stm) || stm->has_dummy_output); + + // We don't bother mixing dummy output as it will be silenced, otherwise mix output if needed + if (!stm->has_dummy_output && has_output(stm) && stm->output_mixer) { + XASSERT(dest == stm->mix_buffer.data()); + size_t dest_size = + out_frames * stm->output_stream_params.channels * stm->bytes_per_sample; + XASSERT(dest_size <= stm->mix_buffer.size()); + size_t output_buffer_size = + out_frames * stm->output_mix_params.channels * stm->bytes_per_sample; + int ret = cubeb_mixer_mix(stm->output_mixer.get(), + out_frames, + dest, + dest_size, + output_buffer, + output_buffer_size); + if (ret < 0) { + LOG("Error remixing content (%d)", ret); + } + } + + return out_frames; +} + +int wasapi_stream_reset_default_device(cubeb_stream * stm); + +/* This helper grabs all the frames available from a capture client, put them in + * linear_input_buffer. linear_input_buffer should be cleared before the + * callback exits. This helper does not work with exclusive mode streams. */ +bool get_input_buffer(cubeb_stream * stm) +{ + XASSERT(has_input(stm)); + + HRESULT hr; + BYTE * input_packet = NULL; + DWORD flags; + UINT64 dev_pos; + UINT32 next; + /* Get input packets until we have captured enough frames, and put them in a + * contiguous buffer. */ + uint32_t offset = 0; + // If the input stream is event driven we should only ever expect to read a + // single packet each time. However, if we're pulling from the stream we may + // need to grab multiple packets worth of frames that have accumulated (so + // need a loop). + for (hr = stm->capture_client->GetNextPacketSize(&next); + next > 0; + hr = stm->capture_client->GetNextPacketSize(&next)) { + if (hr == AUDCLNT_E_DEVICE_INVALIDATED) { + // Application can recover from this error. More info + // https://msdn.microsoft.com/en-us/library/windows/desktop/dd316605(v=vs.85).aspx + LOG("Device invalidated error, reset default device"); + wasapi_stream_reset_default_device(stm); + return true; + } + + if (FAILED(hr)) { + LOG("cannot get next packet size: %lx", hr); + return false; + } + + UINT32 frames; + hr = stm->capture_client->GetBuffer(&input_packet, + &frames, + &flags, + &dev_pos, + NULL); + if (FAILED(hr)) { + LOG("GetBuffer failed for capture: %lx", hr); + return false; + } + XASSERT(frames == next); + + UINT32 input_stream_samples = frames * stm->input_stream_params.channels; + // We do not explicitly handle the AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY + // flag. There a two primary (non exhaustive) scenarios we anticipate this + // flag being set in: + // - The first GetBuffer after Start has this flag undefined. In this + // case the flag may be set but is meaningless and can be ignored. + // - If a glitch is introduced into the input. This should not happen + // for event based inputs, and should be mitigated by using a dummy + // stream to drive input in the case of input only loopback. Without + // a dummy output, input only loopback would glitch on silence. However, + // the dummy input should push silence to the loopback and prevent + // discontinuities. See https://blogs.msdn.microsoft.com/matthew_van_eerde/2008/12/16/sample-wasapi-loopback-capture-record-what-you-hear/ + // As the first scenario can be ignored, and we anticipate the second + // scenario is mitigated, we ignore the flag. + // For more info: https://msdn.microsoft.com/en-us/library/windows/desktop/dd370859(v=vs.85).aspx, + // https://msdn.microsoft.com/en-us/library/windows/desktop/dd371458(v=vs.85).aspx + if (flags & AUDCLNT_BUFFERFLAGS_SILENT) { + LOG("insert silence: ps=%u", frames); + stm->linear_input_buffer->push_silence(input_stream_samples); + } else { + if (stm->input_mixer) { + bool ok = stm->linear_input_buffer->reserve( + stm->linear_input_buffer->length() + input_stream_samples); + XASSERT(ok); + size_t input_packet_size = + frames * stm->input_mix_params.channels * + cubeb_sample_size(stm->input_mix_params.format); + size_t linear_input_buffer_size = + input_stream_samples * + cubeb_sample_size(stm->input_stream_params.format); + cubeb_mixer_mix(stm->input_mixer.get(), + frames, + input_packet, + input_packet_size, + stm->linear_input_buffer->end(), + linear_input_buffer_size); + stm->linear_input_buffer->set_length( + stm->linear_input_buffer->length() + input_stream_samples); + } else { + stm->linear_input_buffer->push( + input_packet, input_stream_samples); + } + } + hr = stm->capture_client->ReleaseBuffer(frames); + if (FAILED(hr)) { + LOG("FAILED to release intput buffer"); + return false; + } + offset += input_stream_samples; + } + + XASSERT(stm->linear_input_buffer->length() >= offset); + + return true; +} + +/* Get an output buffer from the render_client. It has to be released before + * exiting the callback. */ +bool get_output_buffer(cubeb_stream * stm, void *& buffer, size_t & frame_count) +{ + UINT32 padding_out; + HRESULT hr; + + XASSERT(has_output(stm)); + + hr = stm->output_client->GetCurrentPadding(&padding_out); + if (hr == AUDCLNT_E_DEVICE_INVALIDATED) { + // Application can recover from this error. More info + // https://msdn.microsoft.com/en-us/library/windows/desktop/dd316605(v=vs.85).aspx + LOG("Device invalidated error, reset default device"); + wasapi_stream_reset_default_device(stm); + return true; + } + + if (FAILED(hr)) { + LOG("Failed to get padding: %lx", hr); + return false; + } + + XASSERT(padding_out <= stm->output_buffer_frame_count); + + if (stm->draining) { + if (padding_out == 0) { + LOG("Draining finished."); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + return false; + } + LOG("Draining."); + return true; + } + + frame_count = stm->output_buffer_frame_count - padding_out; + BYTE * output_buffer; + + hr = stm->render_client->GetBuffer(frame_count, &output_buffer); + if (FAILED(hr)) { + LOG("cannot get render buffer"); + return false; + } + + buffer = output_buffer; + + return true; +} + +/** + * This function gets input data from a input device, and pass it along with an + * output buffer to the resamplers. */ +bool +refill_callback_duplex(cubeb_stream * stm) +{ + HRESULT hr; + void * output_buffer = nullptr; + size_t output_frames = 0; + size_t input_frames; + bool rv; + + XASSERT(has_input(stm) && has_output(stm)); + + rv = get_input_buffer(stm); + if (!rv) { + return rv; + } + + input_frames = stm->linear_input_buffer->length() / stm->input_stream_params.channels; + if (!input_frames) { + return true; + } + + rv = get_output_buffer(stm, output_buffer, output_frames); + if (!rv) { + hr = stm->render_client->ReleaseBuffer(output_frames, 0); + return rv; + } + + /* This can only happen when debugging, and having breakpoints set in the + * callback in a way that it makes the stream underrun. */ + if (output_frames == 0) { + return true; + } + + /* Wait for draining is not important on duplex. */ + if (stm->draining) { + return false; + } + + if (stm->has_dummy_output) { + ALOGV("Duplex callback (dummy output): input frames: %Iu, output frames: %Iu", + input_frames, output_frames); + + // We don't want to expose the dummy output to the callback so don't pass + // the output buffer (it will be released later with silence in it) + refill(stm, + stm->linear_input_buffer->data(), + input_frames, + nullptr, + 0); + } else { + ALOGV("Duplex callback: input frames: %Iu, output frames: %Iu", + input_frames, output_frames); + + refill(stm, + stm->linear_input_buffer->data(), + input_frames, + output_buffer, + output_frames); + } + + stm->linear_input_buffer->clear(); + + if (stm->has_dummy_output) { + // If output is a dummy output, make sure it's silent + hr = stm->render_client->ReleaseBuffer(output_frames, AUDCLNT_BUFFERFLAGS_SILENT); + } else { + hr = stm->render_client->ReleaseBuffer(output_frames, 0); + } + if (FAILED(hr)) { + LOG("failed to release buffer: %lx", hr); + return false; + } + return true; +} + +bool +refill_callback_input(cubeb_stream * stm) +{ + bool rv; + size_t input_frames; + + XASSERT(has_input(stm) && !has_output(stm)); + + rv = get_input_buffer(stm); + if (!rv) { + return rv; + } + + input_frames = stm->linear_input_buffer->length() / stm->input_stream_params.channels; + if (!input_frames) { + return true; + } + + ALOGV("Input callback: input frames: %Iu", input_frames); + + long read = refill(stm, + stm->linear_input_buffer->data(), + input_frames, + nullptr, + 0); + + XASSERT(read >= 0); + + stm->linear_input_buffer->clear(); + + return !stm->draining; +} + +bool +refill_callback_output(cubeb_stream * stm) +{ + bool rv; + HRESULT hr; + void * output_buffer = nullptr; + size_t output_frames = 0; + + XASSERT(!has_input(stm) && has_output(stm)); + + rv = get_output_buffer(stm, output_buffer, output_frames); + if (!rv) { + return rv; + } + + if (stm->draining || output_frames == 0) { + return true; + } + + long got = refill(stm, + nullptr, + 0, + output_buffer, + output_frames); + + ALOGV("Output callback: output frames requested: %Iu, got %ld", + output_frames, got); + + XASSERT(got >= 0); + XASSERT(size_t(got) == output_frames || stm->draining); + + hr = stm->render_client->ReleaseBuffer(got, 0); + if (FAILED(hr)) { + LOG("failed to release buffer: %lx", hr); + return false; + } + + return size_t(got) == output_frames || stm->draining; +} + +static unsigned int __stdcall +wasapi_stream_render_loop(LPVOID stream) +{ + cubeb_stream * stm = static_cast(stream); + std::atomic * emergency_bailout = stm->emergency_bailout; + + // Signal wasapi_stream_start that we've copied emergency_bailout. + BOOL ok = SetEvent(stm->thread_ready_event); + if (!ok) { + LOG("thread_ready SetEvent failed: %lx", GetLastError()); + return 0; + } + + bool is_playing = true; + HANDLE wait_array[4] = { + stm->shutdown_event, + stm->reconfigure_event, + stm->refill_event, + stm->input_available_event + }; + HANDLE mmcss_handle = NULL; + HRESULT hr = 0; + DWORD mmcss_task_index = 0; + struct auto_com { + auto_com() { + HRESULT hr = CoInitializeEx(nullptr, COINIT_MULTITHREADED); + XASSERT(SUCCEEDED(hr)); + } + ~auto_com() { + CoUninitialize(); + } + } com; + + /* We could consider using "Pro Audio" here for WebAudio and + maybe WebRTC. */ + mmcss_handle = AvSetMmThreadCharacteristicsA("Audio", &mmcss_task_index); + if (!mmcss_handle) { + /* This is not fatal, but we might glitch under heavy load. */ + LOG("Unable to use mmcss to bump the render thread priority: %lx", GetLastError()); + } + + /* WaitForMultipleObjects timeout can trigger in cases where we don't want to + treat it as a timeout, such as across a system sleep/wake cycle. Trigger + the timeout error handling only when the timeout_limit is reached, which is + reset on each successful loop. */ + unsigned timeout_count = 0; + const unsigned timeout_limit = 3; + while (is_playing) { + // We want to check the emergency bailout variable before a + // and after the WaitForMultipleObject, because the handles WaitForMultipleObjects + // is going to wait on might have been closed already. + if (*emergency_bailout) { + delete emergency_bailout; + return 0; + } + DWORD waitResult = WaitForMultipleObjects(ARRAY_LENGTH(wait_array), + wait_array, + FALSE, + 1000); + if (*emergency_bailout) { + delete emergency_bailout; + return 0; + } + if (waitResult != WAIT_TIMEOUT) { + timeout_count = 0; + } + switch (waitResult) { + case WAIT_OBJECT_0: { /* shutdown */ + is_playing = false; + /* We don't check if the drain is actually finished here, we just want to + shutdown. */ + if (stm->draining) { + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + } + continue; + } + case WAIT_OBJECT_0 + 1: { /* reconfigure */ + XASSERT(stm->output_client || stm->input_client); + LOG("Reconfiguring the stream"); + /* Close the stream */ + if (stm->output_client) { + stm->output_client->Stop(); + LOG("Output stopped."); + } + if (stm->input_client) { + stm->input_client->Stop(); + LOG("Input stopped."); + } + { + auto_lock lock(stm->stream_reset_lock); + close_wasapi_stream(stm); + LOG("Stream closed."); + /* Reopen a stream and start it immediately. This will automatically pick the + new default device for this role. */ + int r = setup_wasapi_stream(stm); + if (r != CUBEB_OK) { + LOG("Error setting up the stream during reconfigure."); + /* Don't destroy the stream here, since we expect the caller to do + so after the error has propagated via the state callback. */ + is_playing = false; + hr = E_FAIL; + continue; + } + LOG("Stream setup successfuly."); + } + XASSERT(stm->output_client || stm->input_client); + if (stm->output_client) { + hr = stm->output_client->Start(); + if (FAILED(hr)) { + LOG("Error starting output after reconfigure, error: %lx", hr); + is_playing = false; + continue; + } + LOG("Output started after reconfigure."); + } + if (stm->input_client) { + hr = stm->input_client->Start(); + if (FAILED(hr)) { + LOG("Error starting input after reconfiguring, error: %lx", hr); + is_playing = false; + continue; + } + LOG("Input started after reconfigure."); + } + break; + } + case WAIT_OBJECT_0 + 2: /* refill */ + XASSERT((has_input(stm) && has_output(stm)) || + (!has_input(stm) && has_output(stm))); + is_playing = stm->refill_callback(stm); + break; + case WAIT_OBJECT_0 + 3: /* input available */ + if (has_input(stm) && has_output(stm)) { continue; } + is_playing = stm->refill_callback(stm); + break; + case WAIT_TIMEOUT: + XASSERT(stm->shutdown_event == wait_array[0]); + if (++timeout_count >= timeout_limit) { + LOG("Render loop reached the timeout limit."); + is_playing = false; + hr = E_FAIL; + } + break; + default: + LOG("case %lu not handled in render loop.", waitResult); + abort(); + } + } + + if (FAILED(hr)) { + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + } + + if (mmcss_handle) { + AvRevertMmThreadCharacteristics(mmcss_handle); + } + + return 0; +} + +void wasapi_destroy(cubeb * context); + +HRESULT register_notification_client(cubeb_stream * stm) +{ + HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), + NULL, CLSCTX_INPROC_SERVER, + IID_PPV_ARGS(stm->device_enumerator.receive())); + if (FAILED(hr)) { + LOG("Could not get device enumerator: %lx", hr); + return hr; + } + + stm->notification_client.reset(new wasapi_endpoint_notification_client(stm->reconfigure_event, stm->role)); + + hr = stm->device_enumerator->RegisterEndpointNotificationCallback(stm->notification_client.get()); + if (FAILED(hr)) { + LOG("Could not register endpoint notification callback: %lx", hr); + stm->notification_client = nullptr; + stm->device_enumerator = nullptr; + } + + return hr; +} + +HRESULT unregister_notification_client(cubeb_stream * stm) +{ + XASSERT(stm); + HRESULT hr; + + if (!stm->device_enumerator) { + return S_OK; + } + + hr = stm->device_enumerator->UnregisterEndpointNotificationCallback(stm->notification_client.get()); + if (FAILED(hr)) { + // We can't really do anything here, we'll probably leak the + // notification client, but we can at least release the enumerator. + stm->device_enumerator = nullptr; + return S_OK; + } + + stm->notification_client = nullptr; + stm->device_enumerator = nullptr; + + return S_OK; +} + +HRESULT get_endpoint(com_ptr & device, LPCWSTR devid) +{ + com_ptr enumerator; + HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), + NULL, CLSCTX_INPROC_SERVER, + IID_PPV_ARGS(enumerator.receive())); + if (FAILED(hr)) { + LOG("Could not get device enumerator: %lx", hr); + return hr; + } + + hr = enumerator->GetDevice(devid, device.receive()); + if (FAILED(hr)) { + LOG("Could not get device: %lx", hr); + return hr; + } + + return S_OK; +} + +HRESULT register_collection_notification_client(cubeb * context) +{ + HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), + NULL, CLSCTX_INPROC_SERVER, + IID_PPV_ARGS(context->device_collection_enumerator.receive())); + if (FAILED(hr)) { + LOG("Could not get device enumerator: %lx", hr); + return hr; + } + + context->collection_notification_client.reset(new wasapi_collection_notification_client(context)); + + hr = context->device_collection_enumerator->RegisterEndpointNotificationCallback( + context->collection_notification_client.get()); + if (FAILED(hr)) { + LOG("Could not register endpoint notification callback: %lx", hr); + context->collection_notification_client.reset(); + context->device_collection_enumerator.reset(); + } + + return hr; +} + +HRESULT unregister_collection_notification_client(cubeb * context) +{ + HRESULT hr = context->device_collection_enumerator-> + UnregisterEndpointNotificationCallback(context->collection_notification_client.get()); + if (FAILED(hr)) { + return hr; + } + + context->collection_notification_client = nullptr; + context->device_collection_enumerator = nullptr; + + return hr; +} + +HRESULT get_default_endpoint(com_ptr & device, EDataFlow direction, ERole role) +{ + com_ptr enumerator; + HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), + NULL, CLSCTX_INPROC_SERVER, + IID_PPV_ARGS(enumerator.receive())); + if (FAILED(hr)) { + LOG("Could not get device enumerator: %lx", hr); + return hr; + } + hr = enumerator->GetDefaultAudioEndpoint(direction, role, device.receive()); + if (FAILED(hr)) { + LOG("Could not get default audio endpoint: %lx", hr); + return hr; + } + + return ERROR_SUCCESS; +} + +double +current_stream_delay(cubeb_stream * stm) +{ + stm->stream_reset_lock.assert_current_thread_owns(); + + /* If the default audio endpoint went away during playback and we weren't + able to configure a new one, it's possible the caller may call this + before the error callback has propogated back. */ + if (!stm->audio_clock) { + return 0; + } + + UINT64 freq; + HRESULT hr = stm->audio_clock->GetFrequency(&freq); + if (FAILED(hr)) { + LOG("GetFrequency failed: %lx", hr); + return 0; + } + + UINT64 pos; + hr = stm->audio_clock->GetPosition(&pos, NULL); + if (FAILED(hr)) { + LOG("GetPosition failed: %lx", hr); + return 0; + } + + double cur_pos = static_cast(pos) / freq; + double max_pos = static_cast(stm->frames_written) / stm->output_mix_params.rate; + double delay = max_pos - cur_pos; + XASSERT(delay >= 0); + + return delay; +} + +int +stream_set_volume(cubeb_stream * stm, float volume) +{ + stm->stream_reset_lock.assert_current_thread_owns(); + + if (!stm->audio_stream_volume) { + return CUBEB_ERROR; + } + + uint32_t channels; + HRESULT hr = stm->audio_stream_volume->GetChannelCount(&channels); + if (FAILED(hr)) { + LOG("could not get the channel count: %lx", hr); + return CUBEB_ERROR; + } + + /* up to 9.1 for now */ + if (channels > 10) { + return CUBEB_ERROR_NOT_SUPPORTED; + } + + float volumes[10]; + for (uint32_t i = 0; i < channels; i++) { + volumes[i] = volume; + } + + hr = stm->audio_stream_volume->SetAllVolumes(channels, volumes); + if (FAILED(hr)) { + LOG("could not set the channels volume: %lx", hr); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} +} // namespace anonymous + +extern "C" { +int wasapi_init(cubeb ** context, char const * context_name) +{ + /* We don't use the device yet, but need to make sure we can initialize one + so that this backend is not incorrectly enabled on platforms that don't + support WASAPI. */ + com_ptr device; + HRESULT hr = get_default_endpoint(device, eRender, eConsole); + if (FAILED(hr)) { + XASSERT(hr != CO_E_NOTINITIALIZED); + LOG("It wasn't able to find a default rendering device: %lx", hr); + hr = get_default_endpoint(device, eCapture, eConsole); + if (FAILED(hr)) { + LOG("It wasn't able to find a default capture device: %lx", hr); + return CUBEB_ERROR; + } + } + + cubeb * ctx = new cubeb(); + + ctx->ops = &wasapi_ops; + if (cubeb_strings_init(&ctx->device_ids) != CUBEB_OK) { + delete ctx; + return CUBEB_ERROR; + } + + *context = ctx; + + return CUBEB_OK; +} +} + +namespace { +bool stop_and_join_render_thread(cubeb_stream * stm) +{ + bool rv = true; + LOG("Stop and join render thread."); + if (!stm->thread) { + LOG("No thread present."); + return true; + } + + // If we've already leaked the thread, just return, + // there is not much we can do. + if (!stm->emergency_bailout.load()) { + return false; + } + + BOOL ok = SetEvent(stm->shutdown_event); + if (!ok) { + LOG("Destroy SetEvent failed: %lx", GetLastError()); + } + + /* Wait five seconds for the rendering thread to return. It's supposed to + * check its event loop very often, five seconds is rather conservative. */ + DWORD r = WaitForSingleObject(stm->thread, 5000); + if (r != WAIT_OBJECT_0) { + /* Something weird happened, leak the thread and continue the shutdown + * process. */ + *(stm->emergency_bailout) = true; + // We give the ownership to the rendering thread. + stm->emergency_bailout = nullptr; + LOG("Destroy WaitForSingleObject on thread failed: %lx, %lx", r, GetLastError()); + rv = false; + } + + // Only attempts to close and null out the thread and event if the + // WaitForSingleObject above succeeded, so that calling this function again + // attemps to clean up the thread and event each time. + if (rv) { + LOG("Closing thread."); + CloseHandle(stm->thread); + stm->thread = NULL; + + CloseHandle(stm->shutdown_event); + stm->shutdown_event = 0; + } + + return rv; +} + +void wasapi_destroy(cubeb * context) +{ + if (context->device_ids) { + cubeb_strings_destroy(context->device_ids); + } + + delete context; +} + +char const * wasapi_get_backend_id(cubeb * context) +{ + return "wasapi"; +} + +int +wasapi_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + XASSERT(ctx && max_channels); + + com_ptr device; + HRESULT hr = get_default_endpoint(device, eRender, eConsole); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + + com_ptr client; + hr = device->Activate(__uuidof(IAudioClient), + CLSCTX_INPROC_SERVER, + NULL, client.receive_vpp()); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + + WAVEFORMATEX * tmp = nullptr; + hr = client->GetMixFormat(&tmp); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + com_heap_ptr mix_format(tmp); + + *max_channels = mix_format->nChannels; + + return CUBEB_OK; +} + +int +wasapi_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames) +{ + if (params.format != CUBEB_SAMPLE_FLOAT32NE && params.format != CUBEB_SAMPLE_S16NE) { + return CUBEB_ERROR_INVALID_FORMAT; + } + + ERole role = pref_to_role(params.prefs); + + com_ptr device; + HRESULT hr = get_default_endpoint(device, eRender, role); + if (FAILED(hr)) { + LOG("Could not get default endpoint: %lx", hr); + return CUBEB_ERROR; + } + + com_ptr client; + hr = device->Activate(__uuidof(IAudioClient), + CLSCTX_INPROC_SERVER, + NULL, client.receive_vpp()); + if (FAILED(hr)) { + LOG("Could not activate device for latency: %lx", hr); + return CUBEB_ERROR; + } + + REFERENCE_TIME minimum_period; + REFERENCE_TIME default_period; + hr = client->GetDevicePeriod(&default_period, &minimum_period); + if (FAILED(hr)) { + LOG("Could not get device period: %lx", hr); + return CUBEB_ERROR; + } + + LOG("default device period: %I64d, minimum device period: %I64d", default_period, minimum_period); + + /* If we're on Windows 10, we can use IAudioClient3 to get minimal latency. + Otherwise, according to the docs, the best latency we can achieve is by + synchronizing the stream and the engine. + http://msdn.microsoft.com/en-us/library/windows/desktop/dd370871%28v=vs.85%29.aspx */ + + #ifdef _WIN32_WINNT_WIN10 + *latency_frames = hns_to_frames(params.rate, minimum_period); + #else + *latency_frames = hns_to_frames(params.rate, default_period); + #endif + + LOG("Minimum latency in frames: %u", *latency_frames); + + return CUBEB_OK; +} + +int +wasapi_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) +{ + com_ptr device; + HRESULT hr = get_default_endpoint(device, eRender, eConsole); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + + com_ptr client; + hr = device->Activate(__uuidof(IAudioClient), + CLSCTX_INPROC_SERVER, + NULL, client.receive_vpp()); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + + WAVEFORMATEX * tmp = nullptr; + hr = client->GetMixFormat(&tmp); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + com_heap_ptr mix_format(tmp); + + *rate = mix_format->nSamplesPerSec; + + LOG("Preferred sample rate for output: %u", *rate); + + return CUBEB_OK; +} + +void wasapi_stream_destroy(cubeb_stream * stm); + +static void +waveformatex_update_derived_properties(WAVEFORMATEX * format) +{ + format->nBlockAlign = format->wBitsPerSample * format->nChannels / 8; + format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign; + if (format->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { + WAVEFORMATEXTENSIBLE * format_pcm = reinterpret_cast(format); + format_pcm->Samples.wValidBitsPerSample = format->wBitsPerSample; + } +} + +/* Based on the mix format and the stream format, try to find a way to play + what the user requested. */ +static void +handle_channel_layout(cubeb_stream * stm, EDataFlow direction, com_heap_ptr & mix_format, const cubeb_stream_params * stream_params) +{ + com_ptr & audio_client = (direction == eRender) ? stm->output_client : stm->input_client; + XASSERT(audio_client); + /* The docs say that GetMixFormat is always of type WAVEFORMATEXTENSIBLE [1], + so the reinterpret_cast below should be safe. In practice, this is not + true, and we just want to bail out and let the rest of the code find a good + conversion path instead of trying to make WASAPI do it by itself. + [1]: http://msdn.microsoft.com/en-us/library/windows/desktop/dd370811%28v=vs.85%29.aspx*/ + if (mix_format->wFormatTag != WAVE_FORMAT_EXTENSIBLE) { + return; + } + + WAVEFORMATEXTENSIBLE * format_pcm = reinterpret_cast(mix_format.get()); + + /* Stash a copy of the original mix format in case we need to restore it later. */ + WAVEFORMATEXTENSIBLE hw_mix_format = *format_pcm; + + /* Get the channel mask by the channel layout. + If the layout is not supported, we will get a closest settings below. */ + format_pcm->dwChannelMask = stream_params->layout; + mix_format->nChannels = stream_params->channels; + waveformatex_update_derived_properties(mix_format.get()); + + /* Check if wasapi will accept our channel layout request. */ + WAVEFORMATEX * closest; + HRESULT hr = audio_client->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, + mix_format.get(), + &closest); + if (hr == S_FALSE) { + /* Channel layout not supported, but WASAPI gives us a suggestion. Use it, + and handle the eventual upmix/downmix ourselves. Ignore the subformat of + the suggestion, since it seems to always be IEEE_FLOAT. */ + LOG("Using WASAPI suggested format: channels: %d", closest->nChannels); + XASSERT(closest->wFormatTag == WAVE_FORMAT_EXTENSIBLE); + WAVEFORMATEXTENSIBLE * closest_pcm = reinterpret_cast(closest); + format_pcm->dwChannelMask = closest_pcm->dwChannelMask; + mix_format->nChannels = closest->nChannels; + waveformatex_update_derived_properties(mix_format.get()); + } else if (hr == AUDCLNT_E_UNSUPPORTED_FORMAT) { + /* Not supported, no suggestion. This should not happen, but it does in the + field with some sound cards. We restore the mix format, and let the rest + of the code figure out the right conversion path. */ + XASSERT(mix_format->wFormatTag == WAVE_FORMAT_EXTENSIBLE); + *reinterpret_cast(mix_format.get()) = hw_mix_format; + } else if (hr == S_OK) { + LOG("Requested format accepted by WASAPI."); + } else { + LOG("IsFormatSupported unhandled error: %lx", hr); + } +} + +static bool +initialize_iaudioclient3(com_ptr & audio_client, + cubeb_stream * stm, + const com_heap_ptr & mix_format, + DWORD flags, + EDataFlow direction) +{ + com_ptr audio_client3; + audio_client->QueryInterface(audio_client3.receive()); + if (!audio_client3) { + LOG("Could not get IAudioClient3 interface"); + return false; + } + + if (flags & AUDCLNT_STREAMFLAGS_LOOPBACK) { + // IAudioClient3 doesn't work with loopback streams, and will return error + // 88890021: AUDCLNT_E_INVALID_STREAM_FLAG + LOG("Audio stream is loopback, not using IAudioClient3"); + return false; + } + + // IAudioClient3 doesn't support AUDCLNT_STREAMFLAGS_NOPERSIST, and will return + // AUDCLNT_E_INVALID_STREAM_FLAG. This is undocumented. + flags = flags ^ AUDCLNT_STREAMFLAGS_NOPERSIST; + + // Some people have reported glitches with capture streams: + // http://blog.nirbheek.in/2018/03/low-latency-audio-on-windows-with.html + if (direction == eCapture) { + LOG("Audio stream is capture, not using IAudioClient3"); + return false; + } + + // Possibly initialize a shared-mode stream using IAudioClient3. Initializing + // a stream this way lets you request lower latencies, but also locks the global + // WASAPI engine at that latency. + // - If we request a shared-mode stream, streams created with IAudioClient will + // have their latency adjusted to match. When the shared-mode stream is + // closed, they'll go back to normal. + // - If there's already a shared-mode stream running, then we cannot request + // the engine change to a different latency - we have to match it. + // - It's antisocial to lock the WASAPI engine at its default latency. If we + // would do this, then stop and use IAudioClient instead. + + HRESULT hr; + uint32_t default_period = 0, fundamental_period = 0, min_period = 0, max_period = 0; + hr = audio_client3->GetSharedModeEnginePeriod(mix_format.get(), &default_period, &fundamental_period, &min_period, &max_period); + if (FAILED(hr)) { + LOG("Could not get shared mode engine period: error: %lx", hr); + return false; + } + uint32_t requested_latency = stm->latency; + if (requested_latency >= default_period) { + LOG("Requested latency %i greater than default latency %i, not using IAudioClient3", requested_latency, default_period); + return false; + } + LOG("Got shared mode engine period: default=%i fundamental=%i min=%i max=%i", default_period, fundamental_period, min_period, max_period); + // Snap requested latency to a valid value + uint32_t old_requested_latency = requested_latency; + if (requested_latency < min_period) { + requested_latency = min_period; + } + requested_latency -= (requested_latency - min_period) % fundamental_period; + if (requested_latency != old_requested_latency) { + LOG("Requested latency %i was adjusted to %i", old_requested_latency, requested_latency); + } + + hr = audio_client3->InitializeSharedAudioStream(flags, requested_latency, mix_format.get(), NULL); + if (SUCCEEDED(hr)) { + return true; + } + else if (hr == AUDCLNT_E_ENGINE_PERIODICITY_LOCKED) { + LOG("Got AUDCLNT_E_ENGINE_PERIODICITY_LOCKED, adjusting latency request"); + } else { + LOG("Could not initialize shared stream with IAudioClient3: error: %lx", hr); + return false; + } + + uint32_t current_period = 0; + WAVEFORMATEX* current_format = nullptr; + // We have to pass a valid WAVEFORMATEX** and not nullptr, otherwise + // GetCurrentSharedModeEnginePeriod will return E_POINTER + hr = audio_client3->GetCurrentSharedModeEnginePeriod(¤t_format, ¤t_period); + CoTaskMemFree(current_format); + if (FAILED(hr)) { + LOG("Could not get current shared mode engine period: error: %lx", hr); + return false; + } + + if (current_period >= default_period) { + LOG("Current shared mode engine period %i too high, not using IAudioClient", current_period); + return false; + } + + hr = audio_client3->InitializeSharedAudioStream(flags, current_period, mix_format.get(), NULL); + if (SUCCEEDED(hr)) { + LOG("Current shared mode engine period is %i instead of requested %i", current_period, requested_latency); + return true; + } + + LOG("Could not initialize shared stream with IAudioClient3: error: %lx", hr); + return false; +} + +#define DIRECTION_NAME (direction == eCapture ? "capture" : "render") + +template +int setup_wasapi_stream_one_side(cubeb_stream * stm, + cubeb_stream_params * stream_params, + wchar_t const * devid, + EDataFlow direction, + REFIID riid, + com_ptr & audio_client, + uint32_t * buffer_frame_count, + HANDLE & event, + T & render_or_capture_client, + cubeb_stream_params * mix_params) +{ + com_ptr device; + HRESULT hr; + bool is_loopback = stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK; + if (is_loopback && direction != eCapture) { + LOG("Loopback pref can only be used with capture streams!\n"); + return CUBEB_ERROR; + } + + stm->stream_reset_lock.assert_current_thread_owns(); + bool try_again = false; + // This loops until we find a device that works, or we've exhausted all + // possibilities. + do { + if (devid) { + hr = get_endpoint(device, devid); + if (FAILED(hr)) { + LOG("Could not get %s endpoint, error: %lx\n", DIRECTION_NAME, hr); + return CUBEB_ERROR; + } + } else { + // If caller has requested loopback but not specified a device, look for + // the default render device. Otherwise look for the default device + // appropriate to the direction. + hr = get_default_endpoint(device, is_loopback ? eRender : direction, pref_to_role(stream_params->prefs)); + if (FAILED(hr)) { + if (is_loopback) { + LOG("Could not get default render endpoint for loopback, error: %lx\n", hr); + } else { + LOG("Could not get default %s endpoint, error: %lx\n", DIRECTION_NAME, hr); + } + return CUBEB_ERROR; + } + } + + /* Get a client. We will get all other interfaces we need from + * this pointer. */ + hr = device->Activate(__uuidof(IAudioClient3), + CLSCTX_INPROC_SERVER, + NULL, audio_client.receive_vpp()); + if (hr == E_NOINTERFACE) { + hr = device->Activate(__uuidof(IAudioClient), + CLSCTX_INPROC_SERVER, + NULL, audio_client.receive_vpp()); + } + + if (FAILED(hr)) { + LOG("Could not activate the device to get an audio" + " client for %s: error: %lx\n", DIRECTION_NAME, hr); + // A particular device can't be activated because it has been + // unplugged, try fall back to the default audio device. + if (devid && hr == AUDCLNT_E_DEVICE_INVALIDATED) { + LOG("Trying again with the default %s audio device.", DIRECTION_NAME); + devid = nullptr; + device = nullptr; + try_again = true; + } else { + return CUBEB_ERROR; + } + } else { + try_again = false; + } + } while (try_again); + + /* We have to distinguish between the format the mixer uses, + * and the format the stream we want to play uses. */ + WAVEFORMATEX * tmp = nullptr; + hr = audio_client->GetMixFormat(&tmp); + if (FAILED(hr)) { + LOG("Could not fetch current mix format from the audio" + " client for %s: error: %lx", DIRECTION_NAME, hr); + return CUBEB_ERROR; + } + com_heap_ptr mix_format(tmp); + + mix_format->wBitsPerSample = stm->bytes_per_sample * 8; + if (mix_format->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { + WAVEFORMATEXTENSIBLE * format_pcm = reinterpret_cast(mix_format.get()); + format_pcm->SubFormat = stm->waveformatextensible_sub_format; + } + waveformatex_update_derived_properties(mix_format.get()); + + /* Set channel layout only when there're more than two channels. Otherwise, + * use the default setting retrieved from the stream format of the audio + * engine's internal processing by GetMixFormat. */ + if (mix_format->nChannels > 2) { + handle_channel_layout(stm, direction, mix_format, stream_params); + } + + mix_params->format = stream_params->format; + mix_params->rate = mix_format->nSamplesPerSec; + mix_params->channels = mix_format->nChannels; + mix_params->layout = mask_to_channel_layout(mix_format.get()); + + LOG("Setup requested=[f=%d r=%u c=%u l=%u] mix=[f=%d r=%u c=%u l=%u]", + stream_params->format, stream_params->rate, stream_params->channels, + stream_params->layout, + mix_params->format, mix_params->rate, mix_params->channels, + mix_params->layout); + + DWORD flags = AUDCLNT_STREAMFLAGS_NOPERSIST; + + // Check if a loopback device should be requested. Note that event callbacks + // do not work with loopback devices, so only request these if not looping. + if (is_loopback) { + flags |= AUDCLNT_STREAMFLAGS_LOOPBACK; + } else { + flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK; + } + + if (initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction)) { + LOG("Initialized with IAudioClient3"); + } else { + hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, + flags, + frames_to_hns(stm, stm->latency), + 0, + mix_format.get(), + NULL); + } + if (FAILED(hr)) { + LOG("Unable to initialize audio client for %s: %lx.", DIRECTION_NAME, hr); + return CUBEB_ERROR; + } + + hr = audio_client->GetBufferSize(buffer_frame_count); + if (FAILED(hr)) { + LOG("Could not get the buffer size from the client" + " for %s %lx.", DIRECTION_NAME, hr); + return CUBEB_ERROR; + } + + // Events are used if not looping back + if (!is_loopback) { + hr = audio_client->SetEventHandle(event); + if (FAILED(hr)) { + LOG("Could set the event handle for the %s client %lx.", + DIRECTION_NAME, hr); + return CUBEB_ERROR; + } + } + + hr = audio_client->GetService(riid, render_or_capture_client.receive_vpp()); + if (FAILED(hr)) { + LOG("Could not get the %s client %lx.", DIRECTION_NAME, hr); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +#undef DIRECTION_NAME + +int setup_wasapi_stream(cubeb_stream * stm) +{ + int rv; + + stm->stream_reset_lock.assert_current_thread_owns(); + + XASSERT((!stm->output_client || !stm->input_client) && "WASAPI stream already setup, close it first."); + + if (has_input(stm)) { + LOG("(%p) Setup capture: device=%p", stm, stm->input_device.get()); + rv = setup_wasapi_stream_one_side(stm, + &stm->input_stream_params, + stm->input_device.get(), + eCapture, + __uuidof(IAudioCaptureClient), + stm->input_client, + &stm->input_buffer_frame_count, + stm->input_available_event, + stm->capture_client, + &stm->input_mix_params); + if (rv != CUBEB_OK) { + LOG("Failure to open the input side."); + return rv; + } + + // We initializing an input stream, buffer ahead two buffers worth of silence. + // This delays the input side slightly, but allow to not glitch when no input + // is available when calling into the resampler to call the callback: the input + // refill event will be set shortly after to compensate for this lack of data. + // In debug, four buffers are used, to avoid tripping up assertions down the line. +#if !defined(DEBUG) + const int silent_buffer_count = 2; +#else + const int silent_buffer_count = 6; +#endif + stm->linear_input_buffer->push_silence(stm->input_buffer_frame_count * + stm->input_stream_params.channels * + silent_buffer_count); + } + + // If we don't have an output device but are requesting a loopback device, + // we attempt to open that same device in output mode in order to drive the + // loopback via the output events. + stm->has_dummy_output = false; + if (!has_output(stm) && stm->input_stream_params.prefs & CUBEB_STREAM_PREF_LOOPBACK) { + stm->output_stream_params.rate = stm->input_stream_params.rate; + stm->output_stream_params.channels = stm->input_stream_params.channels; + stm->output_stream_params.layout = stm->input_stream_params.layout; + if (stm->input_device) { + size_t len = wcslen(stm->input_device.get()); + std::unique_ptr tmp(new wchar_t[len + 1]); + if (wcsncpy_s(tmp.get(), len + 1, stm->input_device.get(), len) != 0) { + LOG("Failed to copy device identifier while copying input stream" + " configuration to output stream configuration to drive loopback."); + return CUBEB_ERROR; + } + stm->output_device = move(tmp); + } + stm->has_dummy_output = true; + } + + if (has_output(stm)) { + LOG("(%p) Setup render: device=%p", stm, stm->output_device.get()); + rv = setup_wasapi_stream_one_side(stm, + &stm->output_stream_params, + stm->output_device.get(), + eRender, + __uuidof(IAudioRenderClient), + stm->output_client, + &stm->output_buffer_frame_count, + stm->refill_event, + stm->render_client, + &stm->output_mix_params); + if (rv != CUBEB_OK) { + LOG("Failure to open the output side."); + return rv; + } + + HRESULT hr = stm->output_client->GetService(__uuidof(IAudioStreamVolume), + stm->audio_stream_volume.receive_vpp()); + if (FAILED(hr)) { + LOG("Could not get the IAudioStreamVolume: %lx", hr); + return CUBEB_ERROR; + } + + XASSERT(stm->frames_written == 0); + hr = stm->output_client->GetService(__uuidof(IAudioClock), + stm->audio_clock.receive_vpp()); + if (FAILED(hr)) { + LOG("Could not get the IAudioClock: %lx", hr); + return CUBEB_ERROR; + } + + /* Restore the stream volume over a device change. */ + if (stream_set_volume(stm, stm->volume) != CUBEB_OK) { + LOG("Could not set the volume."); + return CUBEB_ERROR; + } + } + + /* If we have both input and output, we resample to + * the highest sample rate available. */ + int32_t target_sample_rate; + if (has_input(stm) && has_output(stm)) { + XASSERT(stm->input_stream_params.rate == stm->output_stream_params.rate); + target_sample_rate = stm->input_stream_params.rate; + } else if (has_input(stm)) { + target_sample_rate = stm->input_stream_params.rate; + } else { + XASSERT(has_output(stm)); + target_sample_rate = stm->output_stream_params.rate; + } + + LOG("Target sample rate: %d", target_sample_rate); + + /* If we are playing/capturing a mono stream, we only resample one channel, + and copy it over, so we are always resampling the number + of channels of the stream, not the number of channels + that WASAPI wants. */ + cubeb_stream_params input_params = stm->input_mix_params; + input_params.channels = stm->input_stream_params.channels; + cubeb_stream_params output_params = stm->output_mix_params; + output_params.channels = stm->output_stream_params.channels; + + stm->resampler.reset( + cubeb_resampler_create(stm, + has_input(stm) ? &input_params : nullptr, + has_output(stm) ? &output_params : nullptr, + target_sample_rate, + stm->data_callback, + stm->user_ptr, + CUBEB_RESAMPLER_QUALITY_DESKTOP)); + if (!stm->resampler) { + LOG("Could not get a resampler"); + return CUBEB_ERROR; + } + + XASSERT(has_input(stm) || has_output(stm)); + + if (has_input(stm) && has_output(stm)) { + stm->refill_callback = refill_callback_duplex; + } else if (has_input(stm)) { + stm->refill_callback = refill_callback_input; + } else if (has_output(stm)) { + stm->refill_callback = refill_callback_output; + } + + // Create input mixer. + if (has_input(stm) && + ((stm->input_mix_params.layout != CUBEB_LAYOUT_UNDEFINED && + stm->input_mix_params.layout != stm->input_stream_params.layout) || + (stm->input_mix_params.channels != stm->input_stream_params.channels))) { + if (stm->input_mix_params.layout == CUBEB_LAYOUT_UNDEFINED) { + LOG("Input stream using undefined layout! Any mixing may be " + "unpredictable!\n"); + } + stm->input_mixer.reset(cubeb_mixer_create(stm->input_stream_params.format, + stm->input_mix_params.channels, + stm->input_mix_params.layout, + stm->input_stream_params.channels, + stm->input_stream_params.layout)); + assert(stm->input_mixer); + } + + // Create output mixer. + if (has_output(stm) && stm->output_mix_params.layout != stm->output_stream_params.layout) { + if (stm->output_mix_params.layout == CUBEB_LAYOUT_UNDEFINED) { + LOG("Output stream using undefined layout! Any mixing may be unpredictable!\n"); + } + stm->output_mixer.reset(cubeb_mixer_create(stm->output_stream_params.format, + stm->output_stream_params.channels, + stm->output_stream_params.layout, + stm->output_mix_params.channels, + stm->output_mix_params.layout)); + assert(stm->output_mixer); + // Input is up/down mixed when depacketized in get_input_buffer. + stm->mix_buffer.resize( + frames_to_bytes_before_mix(stm, stm->output_buffer_frame_count)); + } + + return CUBEB_OK; +} + +ERole +pref_to_role(cubeb_stream_prefs prefs) +{ + if (prefs & CUBEB_STREAM_PREF_VOICE) { + return eCommunications; + } + + return eConsole; +} + +int +wasapi_stream_init(cubeb * context, cubeb_stream ** stream, + char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, cubeb_data_callback data_callback, + cubeb_state_callback state_callback, void * user_ptr) +{ + int rv; + + XASSERT(context && stream && (input_stream_params || output_stream_params)); + + if (output_stream_params && input_stream_params && + output_stream_params->format != input_stream_params->format) { + return CUBEB_ERROR_INVALID_FORMAT; + } + + std::unique_ptr stm(new cubeb_stream(), wasapi_stream_destroy); + + stm->context = context; + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + + if (stm->output_stream_params.prefs & CUBEB_STREAM_PREF_VOICE || + stm->input_stream_params.prefs & CUBEB_STREAM_PREF_VOICE) { + stm->role = eCommunications; + } else { + stm->role = eConsole; + } + + if (input_stream_params) { + stm->input_stream_params = *input_stream_params; + stm->input_device = utf8_to_wstr(reinterpret_cast(input_device)); + } + if (output_stream_params) { + stm->output_stream_params = *output_stream_params; + stm->output_device = utf8_to_wstr(reinterpret_cast(output_device)); + } + + switch (output_stream_params ? output_stream_params->format : input_stream_params->format) { + case CUBEB_SAMPLE_S16NE: + stm->bytes_per_sample = sizeof(short); + stm->waveformatextensible_sub_format = KSDATAFORMAT_SUBTYPE_PCM; + stm->linear_input_buffer.reset(new auto_array_wrapper_impl); + break; + case CUBEB_SAMPLE_FLOAT32NE: + stm->bytes_per_sample = sizeof(float); + stm->waveformatextensible_sub_format = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; + stm->linear_input_buffer.reset(new auto_array_wrapper_impl); + break; + default: + return CUBEB_ERROR_INVALID_FORMAT; + } + + stm->latency = latency_frames; + + stm->reconfigure_event = CreateEvent(NULL, 0, 0, NULL); + if (!stm->reconfigure_event) { + LOG("Can't create the reconfigure event, error: %lx", GetLastError()); + return CUBEB_ERROR; + } + + /* Unconditionally create the two events so that the wait logic is simpler. */ + stm->refill_event = CreateEvent(NULL, 0, 0, NULL); + if (!stm->refill_event) { + LOG("Can't create the refill event, error: %lx", GetLastError()); + return CUBEB_ERROR; + } + + stm->input_available_event = CreateEvent(NULL, 0, 0, NULL); + if (!stm->input_available_event) { + LOG("Can't create the input available event , error: %lx", GetLastError()); + return CUBEB_ERROR; + } + + { + /* Locking here is not strictly necessary, because we don't have a + notification client that can reset the stream yet, but it lets us + assert that the lock is held in the function. */ + auto_lock lock(stm->stream_reset_lock); + rv = setup_wasapi_stream(stm.get()); + } + if (rv != CUBEB_OK) { + return rv; + } + + if (!((input_stream_params ? + (input_stream_params->prefs & CUBEB_STREAM_PREF_DISABLE_DEVICE_SWITCHING) : 0) || + (output_stream_params ? + (output_stream_params->prefs & CUBEB_STREAM_PREF_DISABLE_DEVICE_SWITCHING) : 0))) { + HRESULT hr = register_notification_client(stm.get()); + if (FAILED(hr)) { + /* this is not fatal, we can still play audio, but we won't be able + to keep using the default audio endpoint if it changes. */ + LOG("failed to register notification client, %lx", hr); + } + } + + *stream = stm.release(); + + LOG("Stream init succesfull (%p)", *stream); + return CUBEB_OK; +} + +void close_wasapi_stream(cubeb_stream * stm) +{ + XASSERT(stm); + + stm->stream_reset_lock.assert_current_thread_owns(); + + stm->output_client = nullptr; + stm->render_client = nullptr; + + stm->input_client = nullptr; + stm->capture_client = nullptr; + + stm->audio_stream_volume = nullptr; + + stm->audio_clock = nullptr; + stm->total_frames_written += static_cast(round(stm->frames_written * stream_to_mix_samplerate_ratio(stm->output_stream_params, stm->output_mix_params))); + stm->frames_written = 0; + + stm->resampler.reset(); + stm->output_mixer.reset(); + stm->input_mixer.reset(); + stm->mix_buffer.clear(); +} + +void wasapi_stream_destroy(cubeb_stream * stm) +{ + XASSERT(stm); + LOG("Stream destroy (%p)", stm); + + // Only free stm->emergency_bailout if we could join the thread. + // If we could not join the thread, stm->emergency_bailout is true + // and is still alive until the thread wakes up and exits cleanly. + if (stop_and_join_render_thread(stm)) { + delete stm->emergency_bailout.load(); + stm->emergency_bailout = nullptr; + } + + if (stm->notification_client) { + unregister_notification_client(stm); + } + + CloseHandle(stm->reconfigure_event); + CloseHandle(stm->refill_event); + CloseHandle(stm->input_available_event); + + // The variables intialized in wasapi_stream_init, + // must be destroyed in wasapi_stream_destroy. + stm->linear_input_buffer.reset(); + + { + auto_lock lock(stm->stream_reset_lock); + close_wasapi_stream(stm); + } + + delete stm; +} + +enum StreamDirection { + OUTPUT, + INPUT +}; + +int stream_start_one_side(cubeb_stream * stm, StreamDirection dir) +{ + XASSERT((dir == OUTPUT && stm->output_client) || + (dir == INPUT && stm->input_client)); + + HRESULT hr = dir == OUTPUT ? stm->output_client->Start() : stm->input_client->Start(); + if (hr == AUDCLNT_E_DEVICE_INVALIDATED) { + LOG("audioclient invalidated for %s device, reconfiguring", + dir == OUTPUT ? "output" : "input"); + + BOOL ok = ResetEvent(stm->reconfigure_event); + if (!ok) { + LOG("resetting reconfig event failed for %s stream: %lx", + dir == OUTPUT ? "output" : "input", GetLastError()); + } + + close_wasapi_stream(stm); + int r = setup_wasapi_stream(stm); + if (r != CUBEB_OK) { + LOG("reconfigure failed"); + return r; + } + + HRESULT hr2 = dir == OUTPUT ? stm->output_client->Start() : stm->input_client->Start(); + if (FAILED(hr2)) { + LOG("could not start the %s stream after reconfig: %lx", + dir == OUTPUT ? "output" : "input", hr); + return CUBEB_ERROR; + } + } else if (FAILED(hr)) { + LOG("could not start the %s stream: %lx.", + dir == OUTPUT ? "output" : "input", hr); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +int wasapi_stream_start(cubeb_stream * stm) +{ + auto_lock lock(stm->stream_reset_lock); + + XASSERT(stm && !stm->thread && !stm->shutdown_event); + XASSERT(stm->output_client || stm->input_client); + + stm->emergency_bailout = new std::atomic(false); + + if (stm->output_client) { + int rv = stream_start_one_side(stm, OUTPUT); + if (rv != CUBEB_OK) { + return rv; + } + } + + if (stm->input_client) { + int rv = stream_start_one_side(stm, INPUT); + if (rv != CUBEB_OK) { + return rv; + } + } + + stm->shutdown_event = CreateEvent(NULL, 0, 0, NULL); + if (!stm->shutdown_event) { + LOG("Can't create the shutdown event, error: %lx", GetLastError()); + return CUBEB_ERROR; + } + + stm->thread_ready_event = CreateEvent(NULL, 0, 0, NULL); + if (!stm->thread_ready_event) { + LOG("Can't create the thread_ready event, error: %lx", GetLastError()); + return CUBEB_ERROR; + } + + cubeb_async_log_reset_threads(); + stm->thread = (HANDLE) _beginthreadex(NULL, 512 * 1024, wasapi_stream_render_loop, stm, STACK_SIZE_PARAM_IS_A_RESERVATION, NULL); + if (stm->thread == NULL) { + LOG("could not create WASAPI render thread."); + return CUBEB_ERROR; + } + + // Wait for wasapi_stream_render_loop to signal that emergency_bailout has + // been read, avoiding a bailout situation where we could free `stm` + // before wasapi_stream_render_loop had a chance to run. + HRESULT hr = WaitForSingleObject(stm->thread_ready_event, INFINITE); + XASSERT(hr == WAIT_OBJECT_0); + CloseHandle(stm->thread_ready_event); + stm->thread_ready_event = 0; + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED); + + return CUBEB_OK; +} + +int wasapi_stream_stop(cubeb_stream * stm) +{ + XASSERT(stm); + HRESULT hr; + + { + auto_lock lock(stm->stream_reset_lock); + + if (stm->output_client) { + hr = stm->output_client->Stop(); + if (FAILED(hr)) { + LOG("could not stop AudioClient (output)"); + return CUBEB_ERROR; + } + } + + if (stm->input_client) { + hr = stm->input_client->Stop(); + if (FAILED(hr)) { + LOG("could not stop AudioClient (input)"); + return CUBEB_ERROR; + } + } + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED); + } + + if (stop_and_join_render_thread(stm)) { + delete stm->emergency_bailout.load(); + stm->emergency_bailout = nullptr; + } else { + // If we could not join the thread, put the stream in error. + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + return CUBEB_ERROR; + } + + return CUBEB_OK; +} + +int wasapi_stream_reset_default_device(cubeb_stream * stm) +{ + XASSERT(stm && stm->reconfigure_event); + BOOL ok = SetEvent(stm->reconfigure_event); + if (!ok) { + LOG("SetEvent on reconfigure_event failed: %lx", GetLastError()); + return CUBEB_ERROR; + } + return CUBEB_OK; +} + +int wasapi_stream_get_position(cubeb_stream * stm, uint64_t * position) +{ + XASSERT(stm && position); + auto_lock lock(stm->stream_reset_lock); + + if (!has_output(stm)) { + return CUBEB_ERROR; + } + + /* Calculate how far behind the current stream head the playback cursor is. */ + uint64_t stream_delay = static_cast(current_stream_delay(stm) * stm->output_stream_params.rate); + + /* Calculate the logical stream head in frames at the stream sample rate. */ + uint64_t max_pos = stm->total_frames_written + + static_cast(round(stm->frames_written * stream_to_mix_samplerate_ratio(stm->output_stream_params, stm->output_mix_params))); + + *position = max_pos; + if (stream_delay <= *position) { + *position -= stream_delay; + } + + if (*position < stm->prev_position) { + *position = stm->prev_position; + } + stm->prev_position = *position; + + return CUBEB_OK; +} + +int wasapi_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ + XASSERT(stm && latency); + + if (!has_output(stm)) { + return CUBEB_ERROR; + } + + auto_lock lock(stm->stream_reset_lock); + + /* The GetStreamLatency method only works if the + AudioClient has been initialized. */ + if (!stm->output_client) { + return CUBEB_ERROR; + } + + REFERENCE_TIME latency_hns; + HRESULT hr = stm->output_client->GetStreamLatency(&latency_hns); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + *latency = hns_to_frames(stm, latency_hns); + + return CUBEB_OK; +} + +int wasapi_stream_set_volume(cubeb_stream * stm, float volume) +{ + auto_lock lock(stm->stream_reset_lock); + + if (!has_output(stm)) { + return CUBEB_ERROR; + } + + if (stream_set_volume(stm, volume) != CUBEB_OK) { + return CUBEB_ERROR; + } + + stm->volume = volume; + + return CUBEB_OK; +} + +static char const * +wstr_to_utf8(LPCWSTR str) +{ + int size = ::WideCharToMultiByte(CP_UTF8, 0, str, -1, nullptr, 0, NULL, NULL); + if (size <= 0) { + return nullptr; + } + + char * ret = static_cast(malloc(size)); + ::WideCharToMultiByte(CP_UTF8, 0, str, -1, ret, size, NULL, NULL); + return ret; +} + +static std::unique_ptr +utf8_to_wstr(char const * str) +{ + int size = ::MultiByteToWideChar(CP_UTF8, 0, str, -1, nullptr, 0); + if (size <= 0) { + return nullptr; + } + + std::unique_ptr ret(new wchar_t[size]); + ::MultiByteToWideChar(CP_UTF8, 0, str, -1, ret.get(), size); + return ret; +} + +static com_ptr +wasapi_get_device_node(IMMDeviceEnumerator * enumerator, IMMDevice * dev) +{ + com_ptr ret; + com_ptr devtopo; + com_ptr connector; + + if (SUCCEEDED(dev->Activate(__uuidof(IDeviceTopology), CLSCTX_ALL, NULL, devtopo.receive_vpp())) && + SUCCEEDED(devtopo->GetConnector(0, connector.receive()))) { + wchar_t * tmp = nullptr; + if (SUCCEEDED(connector->GetDeviceIdConnectedTo(&tmp))) { + com_heap_ptr filterid(tmp); + if (FAILED(enumerator->GetDevice(filterid.get(), ret.receive()))) + ret = NULL; + } + } + + return ret; +} + +static BOOL +wasapi_is_default_device(EDataFlow flow, ERole role, LPCWSTR device_id, + IMMDeviceEnumerator * enumerator) +{ + BOOL ret = FALSE; + com_ptr dev; + HRESULT hr; + + hr = enumerator->GetDefaultAudioEndpoint(flow, role, dev.receive()); + if (SUCCEEDED(hr)) { + wchar_t * tmp = nullptr; + if (SUCCEEDED(dev->GetId(&tmp))) { + com_heap_ptr defdevid(tmp); + ret = (wcscmp(defdevid.get(), device_id) == 0); + } + } + + return ret; +} + +int +wasapi_create_device(cubeb * ctx, cubeb_device_info& ret, IMMDeviceEnumerator * enumerator, IMMDevice * dev) +{ + com_ptr endpoint; + com_ptr devnode; + com_ptr client; + EDataFlow flow; + DWORD state = DEVICE_STATE_NOTPRESENT; + com_ptr propstore; + REFERENCE_TIME def_period, min_period; + HRESULT hr; + + struct prop_variant : public PROPVARIANT { + prop_variant() { PropVariantInit(this); } + ~prop_variant() { PropVariantClear(this); } + prop_variant(prop_variant const &) = delete; + prop_variant & operator=(prop_variant const &) = delete; + }; + + hr = dev->QueryInterface(IID_PPV_ARGS(endpoint.receive())); + if (FAILED(hr)) return CUBEB_ERROR; + + hr = endpoint->GetDataFlow(&flow); + if (FAILED(hr)) return CUBEB_ERROR; + + wchar_t * tmp = nullptr; + hr = dev->GetId(&tmp); + if (FAILED(hr)) return CUBEB_ERROR; + com_heap_ptr device_id(tmp); + + char const * device_id_intern = intern_device_id(ctx, device_id.get()); + if (!device_id_intern) { + return CUBEB_ERROR; + } + + hr = dev->OpenPropertyStore(STGM_READ, propstore.receive()); + if (FAILED(hr)) return CUBEB_ERROR; + + hr = dev->GetState(&state); + if (FAILED(hr)) return CUBEB_ERROR; + + ret.device_id = device_id_intern; + ret.devid = reinterpret_cast(ret.device_id); + prop_variant namevar; + hr = propstore->GetValue(PKEY_Device_FriendlyName, &namevar); + if (SUCCEEDED(hr)) + ret.friendly_name = wstr_to_utf8(namevar.pwszVal); + + devnode = wasapi_get_device_node(enumerator, dev); + if (devnode) { + com_ptr ps; + hr = devnode->OpenPropertyStore(STGM_READ, ps.receive()); + if (FAILED(hr)) return CUBEB_ERROR; + + prop_variant instancevar; + hr = ps->GetValue(PKEY_Device_InstanceId, &instancevar); + if (SUCCEEDED(hr)) { + ret.group_id = wstr_to_utf8(instancevar.pwszVal); + } + } + + ret.preferred = CUBEB_DEVICE_PREF_NONE; + if (wasapi_is_default_device(flow, eConsole, device_id.get(), enumerator)) + ret.preferred = (cubeb_device_pref)(ret.preferred | CUBEB_DEVICE_PREF_MULTIMEDIA); + if (wasapi_is_default_device(flow, eCommunications, device_id.get(), enumerator)) + ret.preferred = (cubeb_device_pref)(ret.preferred | CUBEB_DEVICE_PREF_VOICE); + if (wasapi_is_default_device(flow, eConsole, device_id.get(), enumerator)) + ret.preferred = (cubeb_device_pref)(ret.preferred | CUBEB_DEVICE_PREF_NOTIFICATION); + + if (flow == eRender) ret.type = CUBEB_DEVICE_TYPE_OUTPUT; + else if (flow == eCapture) ret.type = CUBEB_DEVICE_TYPE_INPUT; + switch (state) { + case DEVICE_STATE_ACTIVE: + ret.state = CUBEB_DEVICE_STATE_ENABLED; + break; + case DEVICE_STATE_UNPLUGGED: + ret.state = CUBEB_DEVICE_STATE_UNPLUGGED; + break; + default: + ret.state = CUBEB_DEVICE_STATE_DISABLED; + break; + }; + + ret.format = static_cast(CUBEB_DEVICE_FMT_F32NE | CUBEB_DEVICE_FMT_S16NE); + ret.default_format = CUBEB_DEVICE_FMT_F32NE; + prop_variant fmtvar; + hr = propstore->GetValue(PKEY_AudioEngine_DeviceFormat, &fmtvar); + if (SUCCEEDED(hr) && fmtvar.vt == VT_BLOB) { + if (fmtvar.blob.cbSize == sizeof(PCMWAVEFORMAT)) { + const PCMWAVEFORMAT * pcm = reinterpret_cast(fmtvar.blob.pBlobData); + + ret.max_rate = ret.min_rate = ret.default_rate = pcm->wf.nSamplesPerSec; + ret.max_channels = pcm->wf.nChannels; + } else if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX)) { + WAVEFORMATEX* wfx = reinterpret_cast(fmtvar.blob.pBlobData); + + if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX) + wfx->cbSize || + wfx->wFormatTag == WAVE_FORMAT_PCM) { + ret.max_rate = ret.min_rate = ret.default_rate = wfx->nSamplesPerSec; + ret.max_channels = wfx->nChannels; + } + } + } + + if (SUCCEEDED(dev->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL, client.receive_vpp())) && + SUCCEEDED(client->GetDevicePeriod(&def_period, &min_period))) { + ret.latency_lo = hns_to_frames(ret.default_rate, min_period); + ret.latency_hi = hns_to_frames(ret.default_rate, def_period); + } else { + ret.latency_lo = 0; + ret.latency_hi = 0; + } + + return CUBEB_OK; +} + +static int +wasapi_enumerate_devices(cubeb * context, cubeb_device_type type, + cubeb_device_collection * out) +{ + com_ptr enumerator; + com_ptr collection; + HRESULT hr; + UINT cc, i; + EDataFlow flow; + + hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, + CLSCTX_INPROC_SERVER, IID_PPV_ARGS(enumerator.receive())); + if (FAILED(hr)) { + LOG("Could not get device enumerator: %lx", hr); + return CUBEB_ERROR; + } + + if (type == CUBEB_DEVICE_TYPE_OUTPUT) flow = eRender; + else if (type == CUBEB_DEVICE_TYPE_INPUT) flow = eCapture; + else if (type & (CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_OUTPUT)) flow = eAll; + else return CUBEB_ERROR; + + hr = enumerator->EnumAudioEndpoints(flow, DEVICE_STATEMASK_ALL, collection.receive()); + if (FAILED(hr)) { + LOG("Could not enumerate audio endpoints: %lx", hr); + return CUBEB_ERROR; + } + + hr = collection->GetCount(&cc); + if (FAILED(hr)) { + LOG("IMMDeviceCollection::GetCount() failed: %lx", hr); + return CUBEB_ERROR; + } + cubeb_device_info * devices = new cubeb_device_info[cc]; + if (!devices) + return CUBEB_ERROR; + + PodZero(devices, cc); + out->count = 0; + for (i = 0; i < cc; i++) { + com_ptr dev; + hr = collection->Item(i, dev.receive()); + if (FAILED(hr)) { + LOG("IMMDeviceCollection::Item(%u) failed: %lx", i-1, hr); + continue; + } + if (wasapi_create_device(context, devices[out->count], + enumerator.get(), dev.get()) == CUBEB_OK) { + out->count += 1; + } + } + + out->device = devices; + return CUBEB_OK; +} + +static int +wasapi_device_collection_destroy(cubeb * /*ctx*/, cubeb_device_collection * collection) +{ + XASSERT(collection); + + for (size_t n = 0; n < collection->count; n++) { + cubeb_device_info& dev = collection->device[n]; + delete [] dev.friendly_name; + delete [] dev.group_id; + } + + delete [] collection->device; + return CUBEB_OK; +} + +static int +wasapi_register_device_collection_changed(cubeb * context, + cubeb_device_type devtype, + cubeb_device_collection_changed_callback collection_changed_callback, + void * user_ptr) +{ + if (devtype == CUBEB_DEVICE_TYPE_UNKNOWN) { + return CUBEB_ERROR_INVALID_PARAMETER; + } + + if (collection_changed_callback) { + // Make sure it has been unregistered first. + XASSERT(((devtype & CUBEB_DEVICE_TYPE_INPUT) && + !context->input_collection_changed_callback) || + ((devtype & CUBEB_DEVICE_TYPE_OUTPUT) && + !context->output_collection_changed_callback)); + + // Stop the notification client. Notifications arrive on + // a separate thread. We stop them here to avoid + // synchronization issues during the update. + if (context->device_collection_enumerator.get()) { + HRESULT hr = unregister_collection_notification_client(context); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + } + + if (devtype & CUBEB_DEVICE_TYPE_INPUT) { + context->input_collection_changed_callback = collection_changed_callback; + context->input_collection_changed_user_ptr = user_ptr; + } + if (devtype & CUBEB_DEVICE_TYPE_OUTPUT) { + context->output_collection_changed_callback = collection_changed_callback; + context->output_collection_changed_user_ptr = user_ptr; + } + + HRESULT hr = register_collection_notification_client(context); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + } else { + if (!context->device_collection_enumerator.get()) { + // Already unregistered, ignore it. + return CUBEB_OK; + } + + HRESULT hr = unregister_collection_notification_client(context); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + if (devtype & CUBEB_DEVICE_TYPE_INPUT) { + context->input_collection_changed_callback = nullptr; + context->input_collection_changed_user_ptr = nullptr; + } + if (devtype & CUBEB_DEVICE_TYPE_OUTPUT) { + context->output_collection_changed_callback = nullptr; + context->output_collection_changed_user_ptr = nullptr; + } + + // If after the updates we still have registered + // callbacks restart the notification client. + if (context->input_collection_changed_callback || + context->output_collection_changed_callback) { + hr = register_collection_notification_client(context); + if (FAILED(hr)) { + return CUBEB_ERROR; + } + } + } + + return CUBEB_OK; +} + +cubeb_ops const wasapi_ops = { + /*.init =*/ wasapi_init, + /*.get_backend_id =*/ wasapi_get_backend_id, + /*.get_max_channel_count =*/ wasapi_get_max_channel_count, + /*.get_min_latency =*/ wasapi_get_min_latency, + /*.get_preferred_sample_rate =*/ wasapi_get_preferred_sample_rate, + /*.enumerate_devices =*/ wasapi_enumerate_devices, + /*.device_collection_destroy =*/ wasapi_device_collection_destroy, + /*.destroy =*/ wasapi_destroy, + /*.stream_init =*/ wasapi_stream_init, + /*.stream_destroy =*/ wasapi_stream_destroy, + /*.stream_start =*/ wasapi_stream_start, + /*.stream_stop =*/ wasapi_stream_stop, + /*.stream_reset_default_device =*/ wasapi_stream_reset_default_device, + /*.stream_get_position =*/ wasapi_stream_get_position, + /*.stream_get_latency =*/ wasapi_stream_get_latency, + /*.stream_set_volume =*/ wasapi_stream_set_volume, + /*.stream_get_current_device =*/ NULL, + /*.stream_device_destroy =*/ NULL, + /*.stream_register_device_changed_callback =*/ NULL, + /*.register_device_collection_changed =*/ wasapi_register_device_collection_changed, +}; +} // namespace anonymous diff --git a/dep/cubeb/src/cubeb_winmm.c b/dep/cubeb/src/cubeb_winmm.c new file mode 100644 index 000000000..e064ca079 --- /dev/null +++ b/dep/cubeb/src/cubeb_winmm.c @@ -0,0 +1,1067 @@ +/* + * Copyright © 2011 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#undef WINVER +#define WINVER 0x0501 +#undef WIN32_LEAN_AND_MEAN + +#include +#include +#include +#include +#include +#include +#include +#include +#include "cubeb/cubeb.h" +#include "cubeb-internal.h" + +/* This is missing from the MinGW headers. Use a safe fallback. */ +#if !defined(MEMORY_ALLOCATION_ALIGNMENT) +#define MEMORY_ALLOCATION_ALIGNMENT 16 +#endif + +/**This is also missing from the MinGW headers. It also appears to be undocumented by Microsoft.*/ +#ifndef WAVE_FORMAT_48M08 +#define WAVE_FORMAT_48M08 0x00001000 /* 48 kHz, Mono, 8-bit */ +#endif +#ifndef WAVE_FORMAT_48M16 +#define WAVE_FORMAT_48M16 0x00002000 /* 48 kHz, Mono, 16-bit */ +#endif +#ifndef WAVE_FORMAT_48S08 +#define WAVE_FORMAT_48S08 0x00004000 /* 48 kHz, Stereo, 8-bit */ +#endif +#ifndef WAVE_FORMAT_48S16 +#define WAVE_FORMAT_48S16 0x00008000 /* 48 kHz, Stereo, 16-bit */ +#endif +#ifndef WAVE_FORMAT_96M08 +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#endif +#ifndef WAVE_FORMAT_96M16 +#define WAVE_FORMAT_96M16 0x00020000 /* 96 kHz, Mono, 16-bit */ +#endif +#ifndef WAVE_FORMAT_96S08 +#define WAVE_FORMAT_96S08 0x00040000 /* 96 kHz, Stereo, 8-bit */ +#endif +#ifndef WAVE_FORMAT_96S16 +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +/**Taken from winbase.h, also not in MinGW.*/ +#ifndef STACK_SIZE_PARAM_IS_A_RESERVATION +#define STACK_SIZE_PARAM_IS_A_RESERVATION 0x00010000 // Threads only +#endif + +#ifndef DRVM_MAPPER +#define DRVM_MAPPER (0x2000) +#endif +#ifndef DRVM_MAPPER_PREFERRED_GET +#define DRVM_MAPPER_PREFERRED_GET (DRVM_MAPPER+21) +#endif +#ifndef DRVM_MAPPER_CONSOLEVOICECOM_GET +#define DRVM_MAPPER_CONSOLEVOICECOM_GET (DRVM_MAPPER+23) +#endif + +#define CUBEB_STREAM_MAX 32 +#define NBUFS 4 + +const GUID KSDATAFORMAT_SUBTYPE_PCM = +{ 0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } }; +const GUID KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = +{ 0x00000003, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } }; + +struct cubeb_stream_item { + SLIST_ENTRY head; + cubeb_stream * stream; +}; + +static struct cubeb_ops const winmm_ops; + +struct cubeb { + struct cubeb_ops const * ops; + HANDLE event; + HANDLE thread; + int shutdown; + PSLIST_HEADER work; + CRITICAL_SECTION lock; + unsigned int active_streams; + unsigned int minimum_latency_ms; +}; + +struct cubeb_stream { + /* Note: Must match cubeb_stream layout in cubeb.c. */ + cubeb * context; + void * user_ptr; + /**/ + cubeb_stream_params params; + cubeb_data_callback data_callback; + cubeb_state_callback state_callback; + WAVEHDR buffers[NBUFS]; + size_t buffer_size; + int next_buffer; + int free_buffers; + int shutdown; + int draining; + HANDLE event; + HWAVEOUT waveout; + CRITICAL_SECTION lock; + uint64_t written; + float soft_volume; +}; + +static size_t +bytes_per_frame(cubeb_stream_params params) +{ + size_t bytes; + + switch (params.format) { + case CUBEB_SAMPLE_S16LE: + bytes = sizeof(signed short); + break; + case CUBEB_SAMPLE_FLOAT32LE: + bytes = sizeof(float); + break; + default: + XASSERT(0); + } + + return bytes * params.channels; +} + +static WAVEHDR * +winmm_get_next_buffer(cubeb_stream * stm) +{ + WAVEHDR * hdr = NULL; + + XASSERT(stm->free_buffers > 0 && stm->free_buffers <= NBUFS); + hdr = &stm->buffers[stm->next_buffer]; + XASSERT(hdr->dwFlags & WHDR_PREPARED || + (hdr->dwFlags & WHDR_DONE && !(hdr->dwFlags & WHDR_INQUEUE))); + stm->next_buffer = (stm->next_buffer + 1) % NBUFS; + stm->free_buffers -= 1; + + return hdr; +} + +static void +winmm_refill_stream(cubeb_stream * stm) +{ + WAVEHDR * hdr; + long got; + long wanted; + MMRESULT r; + + EnterCriticalSection(&stm->lock); + stm->free_buffers += 1; + XASSERT(stm->free_buffers > 0 && stm->free_buffers <= NBUFS); + + if (stm->draining) { + LeaveCriticalSection(&stm->lock); + if (stm->free_buffers == NBUFS) { + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); + } + SetEvent(stm->event); + return; + } + + if (stm->shutdown) { + LeaveCriticalSection(&stm->lock); + SetEvent(stm->event); + return; + } + + hdr = winmm_get_next_buffer(stm); + + wanted = (DWORD) stm->buffer_size / bytes_per_frame(stm->params); + + /* It is assumed that the caller is holding this lock. It must be dropped + during the callback to avoid deadlocks. */ + LeaveCriticalSection(&stm->lock); + got = stm->data_callback(stm, stm->user_ptr, NULL, hdr->lpData, wanted); + EnterCriticalSection(&stm->lock); + if (got < 0) { + LeaveCriticalSection(&stm->lock); + /* XXX handle this case */ + XASSERT(0); + return; + } else if (got < wanted) { + stm->draining = 1; + } + stm->written += got; + + XASSERT(hdr->dwFlags & WHDR_PREPARED); + + hdr->dwBufferLength = got * bytes_per_frame(stm->params); + XASSERT(hdr->dwBufferLength <= stm->buffer_size); + + if (stm->soft_volume != -1.0) { + if (stm->params.format == CUBEB_SAMPLE_FLOAT32NE) { + float * b = (float *) hdr->lpData; + uint32_t i; + for (i = 0; i < got * stm->params.channels; i++) { + b[i] *= stm->soft_volume; + } + } else { + short * b = (short *) hdr->lpData; + uint32_t i; + for (i = 0; i < got * stm->params.channels; i++) { + b[i] = (short) (b[i] * stm->soft_volume); + } + } + } + + r = waveOutWrite(stm->waveout, hdr, sizeof(*hdr)); + if (r != MMSYSERR_NOERROR) { + LeaveCriticalSection(&stm->lock); + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); + return; + } + + LeaveCriticalSection(&stm->lock); +} + +static unsigned __stdcall +winmm_buffer_thread(void * user_ptr) +{ + cubeb * ctx = (cubeb *) user_ptr; + XASSERT(ctx); + + for (;;) { + DWORD r; + PSLIST_ENTRY item; + + r = WaitForSingleObject(ctx->event, INFINITE); + XASSERT(r == WAIT_OBJECT_0); + + /* Process work items in batches so that a single stream can't + starve the others by continuously adding new work to the top of + the work item stack. */ + item = InterlockedFlushSList(ctx->work); + while (item != NULL) { + PSLIST_ENTRY tmp = item; + winmm_refill_stream(((struct cubeb_stream_item *) tmp)->stream); + item = item->Next; + _aligned_free(tmp); + } + + if (ctx->shutdown) { + break; + } + } + + return 0; +} + +static void CALLBACK +winmm_buffer_callback(HWAVEOUT waveout, UINT msg, DWORD_PTR user_ptr, DWORD_PTR p1, DWORD_PTR p2) +{ + cubeb_stream * stm = (cubeb_stream *) user_ptr; + struct cubeb_stream_item * item; + + if (msg != WOM_DONE) { + return; + } + + item = _aligned_malloc(sizeof(struct cubeb_stream_item), MEMORY_ALLOCATION_ALIGNMENT); + XASSERT(item); + item->stream = stm; + InterlockedPushEntrySList(stm->context->work, &item->head); + + SetEvent(stm->context->event); +} + +static unsigned int +calculate_minimum_latency(void) +{ + OSVERSIONINFOEX osvi; + DWORDLONG mask; + + /* Running under Terminal Services results in underruns with low latency. */ + if (GetSystemMetrics(SM_REMOTESESSION) == TRUE) { + return 500; + } + + /* Vista's WinMM implementation underruns when less than 200ms of audio is buffered. */ + memset(&osvi, 0, sizeof(OSVERSIONINFOEX)); + osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX); + osvi.dwMajorVersion = 6; + osvi.dwMinorVersion = 0; + + mask = 0; + VER_SET_CONDITION(mask, VER_MAJORVERSION, VER_EQUAL); + VER_SET_CONDITION(mask, VER_MINORVERSION, VER_EQUAL); + + if (VerifyVersionInfo(&osvi, VER_MAJORVERSION | VER_MINORVERSION, mask) != 0) { + return 200; + } + + return 100; +} + +static void winmm_destroy(cubeb * ctx); + +/*static*/ int +winmm_init(cubeb ** context, char const * context_name) +{ + cubeb * ctx; + + XASSERT(context); + *context = NULL; + + /* Don't initialize a context if there are no devices available. */ + if (waveOutGetNumDevs() == 0) { + return CUBEB_ERROR; + } + + ctx = calloc(1, sizeof(*ctx)); + XASSERT(ctx); + + ctx->ops = &winmm_ops; + + ctx->work = _aligned_malloc(sizeof(*ctx->work), MEMORY_ALLOCATION_ALIGNMENT); + XASSERT(ctx->work); + InitializeSListHead(ctx->work); + + ctx->event = CreateEvent(NULL, FALSE, FALSE, NULL); + if (!ctx->event) { + winmm_destroy(ctx); + return CUBEB_ERROR; + } + + ctx->thread = (HANDLE) _beginthreadex(NULL, 256 * 1024, winmm_buffer_thread, ctx, STACK_SIZE_PARAM_IS_A_RESERVATION, NULL); + if (!ctx->thread) { + winmm_destroy(ctx); + return CUBEB_ERROR; + } + + SetThreadPriority(ctx->thread, THREAD_PRIORITY_TIME_CRITICAL); + + InitializeCriticalSection(&ctx->lock); + ctx->active_streams = 0; + + ctx->minimum_latency_ms = calculate_minimum_latency(); + + *context = ctx; + + return CUBEB_OK; +} + +static char const * +winmm_get_backend_id(cubeb * ctx) +{ + return "winmm"; +} + +static void +winmm_destroy(cubeb * ctx) +{ + DWORD r; + + XASSERT(ctx->active_streams == 0); + XASSERT(!InterlockedPopEntrySList(ctx->work)); + + DeleteCriticalSection(&ctx->lock); + + if (ctx->thread) { + ctx->shutdown = 1; + SetEvent(ctx->event); + r = WaitForSingleObject(ctx->thread, INFINITE); + XASSERT(r == WAIT_OBJECT_0); + CloseHandle(ctx->thread); + } + + if (ctx->event) { + CloseHandle(ctx->event); + } + + _aligned_free(ctx->work); + + free(ctx); +} + +static void winmm_stream_destroy(cubeb_stream * stm); + +static int +winmm_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name, + cubeb_devid input_device, + cubeb_stream_params * input_stream_params, + cubeb_devid output_device, + cubeb_stream_params * output_stream_params, + unsigned int latency_frames, + cubeb_data_callback data_callback, + cubeb_state_callback state_callback, + void * user_ptr) +{ + MMRESULT r; + WAVEFORMATEXTENSIBLE wfx; + cubeb_stream * stm; + int i; + size_t bufsz; + + XASSERT(context); + XASSERT(stream); + XASSERT(output_stream_params); + + if (input_stream_params) { + /* Capture support not yet implemented. */ + return CUBEB_ERROR_NOT_SUPPORTED; + } + + if (input_device || output_device) { + /* Device selection not yet implemented. */ + return CUBEB_ERROR_DEVICE_UNAVAILABLE; + } + + if (output_stream_params->prefs & CUBEB_STREAM_PREF_LOOPBACK) { + /* Loopback is not supported */ + return CUBEB_ERROR_NOT_SUPPORTED; + } + + *stream = NULL; + + memset(&wfx, 0, sizeof(wfx)); + if (output_stream_params->channels > 2) { + wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; + wfx.Format.cbSize = sizeof(wfx) - sizeof(wfx.Format); + } else { + wfx.Format.wFormatTag = WAVE_FORMAT_PCM; + if (output_stream_params->format == CUBEB_SAMPLE_FLOAT32LE) { + wfx.Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT; + } + wfx.Format.cbSize = 0; + } + wfx.Format.nChannels = output_stream_params->channels; + wfx.Format.nSamplesPerSec = output_stream_params->rate; + + /* XXX fix channel mappings */ + wfx.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; + + switch (output_stream_params->format) { + case CUBEB_SAMPLE_S16LE: + wfx.Format.wBitsPerSample = 16; + wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; + break; + case CUBEB_SAMPLE_FLOAT32LE: + wfx.Format.wBitsPerSample = 32; + wfx.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; + break; + default: + return CUBEB_ERROR_INVALID_FORMAT; + } + + wfx.Format.nBlockAlign = (wfx.Format.wBitsPerSample * wfx.Format.nChannels) / 8; + wfx.Format.nAvgBytesPerSec = wfx.Format.nSamplesPerSec * wfx.Format.nBlockAlign; + wfx.Samples.wValidBitsPerSample = wfx.Format.wBitsPerSample; + + EnterCriticalSection(&context->lock); + /* CUBEB_STREAM_MAX is a horrible hack to avoid a situation where, when + many streams are active at once, a subset of them will not consume (via + playback) or release (via waveOutReset) their buffers. */ + if (context->active_streams >= CUBEB_STREAM_MAX) { + LeaveCriticalSection(&context->lock); + return CUBEB_ERROR; + } + context->active_streams += 1; + LeaveCriticalSection(&context->lock); + + stm = calloc(1, sizeof(*stm)); + XASSERT(stm); + + stm->context = context; + + stm->params = *output_stream_params; + + stm->data_callback = data_callback; + stm->state_callback = state_callback; + stm->user_ptr = user_ptr; + stm->written = 0; + + uint32_t latency_ms = latency_frames * 1000 / output_stream_params->rate; + + if (latency_ms < context->minimum_latency_ms) { + latency_ms = context->minimum_latency_ms; + } + + bufsz = (size_t) (stm->params.rate / 1000.0 * latency_ms * bytes_per_frame(stm->params) / NBUFS); + if (bufsz % bytes_per_frame(stm->params) != 0) { + bufsz += bytes_per_frame(stm->params) - (bufsz % bytes_per_frame(stm->params)); + } + XASSERT(bufsz % bytes_per_frame(stm->params) == 0); + + stm->buffer_size = bufsz; + + InitializeCriticalSection(&stm->lock); + + stm->event = CreateEvent(NULL, FALSE, FALSE, NULL); + if (!stm->event) { + winmm_stream_destroy(stm); + return CUBEB_ERROR; + } + + stm->soft_volume = -1.0; + + /* winmm_buffer_callback will be called during waveOutOpen, so all + other initialization must be complete before calling it. */ + r = waveOutOpen(&stm->waveout, WAVE_MAPPER, &wfx.Format, + (DWORD_PTR) winmm_buffer_callback, (DWORD_PTR) stm, + CALLBACK_FUNCTION); + if (r != MMSYSERR_NOERROR) { + winmm_stream_destroy(stm); + return CUBEB_ERROR; + } + + r = waveOutPause(stm->waveout); + if (r != MMSYSERR_NOERROR) { + winmm_stream_destroy(stm); + return CUBEB_ERROR; + } + + for (i = 0; i < NBUFS; ++i) { + WAVEHDR * hdr = &stm->buffers[i]; + + hdr->lpData = calloc(1, bufsz); + XASSERT(hdr->lpData); + hdr->dwBufferLength = bufsz; + hdr->dwFlags = 0; + + r = waveOutPrepareHeader(stm->waveout, hdr, sizeof(*hdr)); + if (r != MMSYSERR_NOERROR) { + winmm_stream_destroy(stm); + return CUBEB_ERROR; + } + + winmm_refill_stream(stm); + } + + *stream = stm; + + return CUBEB_OK; +} + +static void +winmm_stream_destroy(cubeb_stream * stm) +{ + int i; + + if (stm->waveout) { + MMTIME time; + MMRESULT r; + int device_valid; + int enqueued; + + EnterCriticalSection(&stm->lock); + stm->shutdown = 1; + + waveOutReset(stm->waveout); + + /* Don't need this value, we just want the result to detect invalid + handle/no device errors than waveOutReset doesn't seem to report. */ + time.wType = TIME_SAMPLES; + r = waveOutGetPosition(stm->waveout, &time, sizeof(time)); + device_valid = !(r == MMSYSERR_INVALHANDLE || r == MMSYSERR_NODRIVER); + + enqueued = NBUFS - stm->free_buffers; + LeaveCriticalSection(&stm->lock); + + /* Wait for all blocks to complete. */ + while (device_valid && enqueued > 0) { + DWORD rv = WaitForSingleObject(stm->event, INFINITE); + XASSERT(rv == WAIT_OBJECT_0); + + EnterCriticalSection(&stm->lock); + enqueued = NBUFS - stm->free_buffers; + LeaveCriticalSection(&stm->lock); + } + + EnterCriticalSection(&stm->lock); + + for (i = 0; i < NBUFS; ++i) { + if (stm->buffers[i].dwFlags & WHDR_PREPARED) { + waveOutUnprepareHeader(stm->waveout, &stm->buffers[i], sizeof(stm->buffers[i])); + } + } + + waveOutClose(stm->waveout); + + LeaveCriticalSection(&stm->lock); + } + + if (stm->event) { + CloseHandle(stm->event); + } + + DeleteCriticalSection(&stm->lock); + + for (i = 0; i < NBUFS; ++i) { + free(stm->buffers[i].lpData); + } + + EnterCriticalSection(&stm->context->lock); + XASSERT(stm->context->active_streams >= 1); + stm->context->active_streams -= 1; + LeaveCriticalSection(&stm->context->lock); + + free(stm); +} + +static int +winmm_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) +{ + XASSERT(ctx && max_channels); + + /* We don't support more than two channels in this backend. */ + *max_channels = 2; + + return CUBEB_OK; +} + +static int +winmm_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency) +{ + // 100ms minimum, if we are not in a bizarre configuration. + *latency = ctx->minimum_latency_ms * params.rate / 1000; + + return CUBEB_OK; +} + +static int +winmm_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) +{ + WAVEOUTCAPS woc; + MMRESULT r; + + r = waveOutGetDevCaps(WAVE_MAPPER, &woc, sizeof(WAVEOUTCAPS)); + if (r != MMSYSERR_NOERROR) { + return CUBEB_ERROR; + } + + /* Check if we support 48kHz, but not 44.1kHz. */ + if (!(woc.dwFormats & WAVE_FORMAT_4S16) && + woc.dwFormats & WAVE_FORMAT_48S16) { + *rate = 48000; + return CUBEB_OK; + } + /* Prefer 44.1kHz between 44.1kHz and 48kHz. */ + *rate = 44100; + + return CUBEB_OK; +} + +static int +winmm_stream_start(cubeb_stream * stm) +{ + MMRESULT r; + + EnterCriticalSection(&stm->lock); + r = waveOutRestart(stm->waveout); + LeaveCriticalSection(&stm->lock); + + if (r != MMSYSERR_NOERROR) { + return CUBEB_ERROR; + } + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED); + + return CUBEB_OK; +} + +static int +winmm_stream_stop(cubeb_stream * stm) +{ + MMRESULT r; + + EnterCriticalSection(&stm->lock); + r = waveOutPause(stm->waveout); + LeaveCriticalSection(&stm->lock); + + if (r != MMSYSERR_NOERROR) { + return CUBEB_ERROR; + } + + stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED); + + return CUBEB_OK; +} + +static int +winmm_stream_get_position(cubeb_stream * stm, uint64_t * position) +{ + MMRESULT r; + MMTIME time; + + EnterCriticalSection(&stm->lock); + time.wType = TIME_SAMPLES; + r = waveOutGetPosition(stm->waveout, &time, sizeof(time)); + LeaveCriticalSection(&stm->lock); + + if (r != MMSYSERR_NOERROR || time.wType != TIME_SAMPLES) { + return CUBEB_ERROR; + } + + *position = time.u.sample; + + return CUBEB_OK; +} + +static int +winmm_stream_get_latency(cubeb_stream * stm, uint32_t * latency) +{ + MMRESULT r; + MMTIME time; + uint64_t written; + + EnterCriticalSection(&stm->lock); + time.wType = TIME_SAMPLES; + r = waveOutGetPosition(stm->waveout, &time, sizeof(time)); + written = stm->written; + LeaveCriticalSection(&stm->lock); + + if (r != MMSYSERR_NOERROR || time.wType != TIME_SAMPLES) { + return CUBEB_ERROR; + } + + XASSERT(written - time.u.sample <= UINT32_MAX); + *latency = (uint32_t) (written - time.u.sample); + + return CUBEB_OK; +} + +static int +winmm_stream_set_volume(cubeb_stream * stm, float volume) +{ + EnterCriticalSection(&stm->lock); + stm->soft_volume = volume; + LeaveCriticalSection(&stm->lock); + return CUBEB_OK; +} + +#define MM_11025HZ_MASK (WAVE_FORMAT_1M08 | WAVE_FORMAT_1M16 | WAVE_FORMAT_1S08 | WAVE_FORMAT_1S16) +#define MM_22050HZ_MASK (WAVE_FORMAT_2M08 | WAVE_FORMAT_2M16 | WAVE_FORMAT_2S08 | WAVE_FORMAT_2S16) +#define MM_44100HZ_MASK (WAVE_FORMAT_4M08 | WAVE_FORMAT_4M16 | WAVE_FORMAT_4S08 | WAVE_FORMAT_4S16) +#define MM_48000HZ_MASK (WAVE_FORMAT_48M08 | WAVE_FORMAT_48M16 | WAVE_FORMAT_48S08 | WAVE_FORMAT_48S16) +#define MM_96000HZ_MASK (WAVE_FORMAT_96M08 | WAVE_FORMAT_96M16 | WAVE_FORMAT_96S08 | WAVE_FORMAT_96S16) +static void +winmm_calculate_device_rate(cubeb_device_info * info, DWORD formats) +{ + if (formats & MM_11025HZ_MASK) { + info->min_rate = 11025; + info->default_rate = 11025; + info->max_rate = 11025; + } + if (formats & MM_22050HZ_MASK) { + if (info->min_rate == 0) info->min_rate = 22050; + info->max_rate = 22050; + info->default_rate = 22050; + } + if (formats & MM_44100HZ_MASK) { + if (info->min_rate == 0) info->min_rate = 44100; + info->max_rate = 44100; + info->default_rate = 44100; + } + if (formats & MM_48000HZ_MASK) { + if (info->min_rate == 0) info->min_rate = 48000; + info->max_rate = 48000; + info->default_rate = 48000; + } + if (formats & MM_96000HZ_MASK) { + if (info->min_rate == 0) { + info->min_rate = 96000; + info->default_rate = 96000; + } + info->max_rate = 96000; + } +} + +#define MM_S16_MASK (WAVE_FORMAT_1M16 | WAVE_FORMAT_1S16 | WAVE_FORMAT_2M16 | WAVE_FORMAT_2S16 | WAVE_FORMAT_4M16 | \ + WAVE_FORMAT_4S16 | WAVE_FORMAT_48M16 | WAVE_FORMAT_48S16 | WAVE_FORMAT_96M16 | WAVE_FORMAT_96S16) +static int +winmm_query_supported_formats(UINT devid, DWORD formats, + cubeb_device_fmt * supfmt, cubeb_device_fmt * deffmt) +{ + WAVEFORMATEXTENSIBLE wfx; + + if (formats & MM_S16_MASK) + *deffmt = *supfmt = CUBEB_DEVICE_FMT_S16LE; + else + *deffmt = *supfmt = 0; + + ZeroMemory(&wfx, sizeof(WAVEFORMATEXTENSIBLE)); + wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; + wfx.Format.nChannels = 2; + wfx.Format.nSamplesPerSec = 44100; + wfx.Format.wBitsPerSample = 32; + wfx.Format.nBlockAlign = (wfx.Format.wBitsPerSample * wfx.Format.nChannels) / 8; + wfx.Format.nAvgBytesPerSec = wfx.Format.nSamplesPerSec * wfx.Format.nBlockAlign; + wfx.Format.cbSize = 22; + wfx.Samples.wValidBitsPerSample = wfx.Format.wBitsPerSample; + wfx.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; + wfx.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; + if (waveOutOpen(NULL, devid, &wfx.Format, 0, 0, WAVE_FORMAT_QUERY) == MMSYSERR_NOERROR) + *supfmt = (cubeb_device_fmt)(*supfmt | CUBEB_DEVICE_FMT_F32LE); + + return (*deffmt != 0) ? CUBEB_OK : CUBEB_ERROR; +} + +static char * +guid_to_cstr(LPGUID guid) +{ + char * ret = malloc(40); + if (!ret) { + return NULL; + } + _snprintf_s(ret, 40, _TRUNCATE, + "{%08X-%04X-%04X-%02X%02X-%02X%02X%02X%02X%02X%02X}", + guid->Data1, guid->Data2, guid->Data3, + guid->Data4[0], guid->Data4[1], guid->Data4[2], guid->Data4[3], + guid->Data4[4], guid->Data4[5], guid->Data4[6], guid->Data4[7]); + return ret; +} + +static cubeb_device_pref +winmm_query_preferred_out_device(UINT devid) +{ + DWORD mmpref = WAVE_MAPPER, compref = WAVE_MAPPER, status; + cubeb_device_pref ret = CUBEB_DEVICE_PREF_NONE; + + if (waveOutMessage((HWAVEOUT) WAVE_MAPPER, DRVM_MAPPER_PREFERRED_GET, + (DWORD_PTR)&mmpref, (DWORD_PTR)&status) == MMSYSERR_NOERROR && + devid == mmpref) + ret |= CUBEB_DEVICE_PREF_MULTIMEDIA | CUBEB_DEVICE_PREF_NOTIFICATION; + + if (waveOutMessage((HWAVEOUT) WAVE_MAPPER, DRVM_MAPPER_CONSOLEVOICECOM_GET, + (DWORD_PTR)&compref, (DWORD_PTR)&status) == MMSYSERR_NOERROR && + devid == compref) + ret |= CUBEB_DEVICE_PREF_VOICE; + + return ret; +} + +static char * +device_id_idx(UINT devid) +{ + char * ret = malloc(16); + if (!ret) { + return NULL; + } + _snprintf_s(ret, 16, _TRUNCATE, "%u", devid); + return ret; +} + +static void +winmm_create_device_from_outcaps2(cubeb_device_info * ret, LPWAVEOUTCAPS2A caps, UINT devid) +{ + XASSERT(ret); + ret->devid = (cubeb_devid) devid; + ret->device_id = device_id_idx(devid); + ret->friendly_name = _strdup(caps->szPname); + ret->group_id = guid_to_cstr(&caps->ProductGuid); + ret->vendor_name = guid_to_cstr(&caps->ManufacturerGuid); + + ret->type = CUBEB_DEVICE_TYPE_OUTPUT; + ret->state = CUBEB_DEVICE_STATE_ENABLED; + ret->preferred = winmm_query_preferred_out_device(devid); + + ret->max_channels = caps->wChannels; + winmm_calculate_device_rate(ret, caps->dwFormats); + winmm_query_supported_formats(devid, caps->dwFormats, + &ret->format, &ret->default_format); + + /* Hardcoded latency estimates... */ + ret->latency_lo = 100 * ret->default_rate / 1000; + ret->latency_hi = 200 * ret->default_rate / 1000; +} + +static void +winmm_create_device_from_outcaps(cubeb_device_info * ret, LPWAVEOUTCAPSA caps, UINT devid) +{ + XASSERT(ret); + ret->devid = (cubeb_devid) devid; + ret->device_id = device_id_idx(devid); + ret->friendly_name = _strdup(caps->szPname); + ret->group_id = NULL; + ret->vendor_name = NULL; + + ret->type = CUBEB_DEVICE_TYPE_OUTPUT; + ret->state = CUBEB_DEVICE_STATE_ENABLED; + ret->preferred = winmm_query_preferred_out_device(devid); + + ret->max_channels = caps->wChannels; + winmm_calculate_device_rate(ret, caps->dwFormats); + winmm_query_supported_formats(devid, caps->dwFormats, + &ret->format, &ret->default_format); + + /* Hardcoded latency estimates... */ + ret->latency_lo = 100 * ret->default_rate / 1000; + ret->latency_hi = 200 * ret->default_rate / 1000; +} + +static cubeb_device_pref +winmm_query_preferred_in_device(UINT devid) +{ + DWORD mmpref = WAVE_MAPPER, compref = WAVE_MAPPER, status; + cubeb_device_pref ret = CUBEB_DEVICE_PREF_NONE; + + if (waveInMessage((HWAVEIN) WAVE_MAPPER, DRVM_MAPPER_PREFERRED_GET, + (DWORD_PTR)&mmpref, (DWORD_PTR)&status) == MMSYSERR_NOERROR && + devid == mmpref) + ret |= CUBEB_DEVICE_PREF_MULTIMEDIA | CUBEB_DEVICE_PREF_NOTIFICATION; + + if (waveInMessage((HWAVEIN) WAVE_MAPPER, DRVM_MAPPER_CONSOLEVOICECOM_GET, + (DWORD_PTR)&compref, (DWORD_PTR)&status) == MMSYSERR_NOERROR && + devid == compref) + ret |= CUBEB_DEVICE_PREF_VOICE; + + return ret; +} + +static void +winmm_create_device_from_incaps2(cubeb_device_info * ret, LPWAVEINCAPS2A caps, UINT devid) +{ + XASSERT(ret); + ret->devid = (cubeb_devid) devid; + ret->device_id = device_id_idx(devid); + ret->friendly_name = _strdup(caps->szPname); + ret->group_id = guid_to_cstr(&caps->ProductGuid); + ret->vendor_name = guid_to_cstr(&caps->ManufacturerGuid); + + ret->type = CUBEB_DEVICE_TYPE_INPUT; + ret->state = CUBEB_DEVICE_STATE_ENABLED; + ret->preferred = winmm_query_preferred_in_device(devid); + + ret->max_channels = caps->wChannels; + winmm_calculate_device_rate(ret, caps->dwFormats); + winmm_query_supported_formats(devid, caps->dwFormats, + &ret->format, &ret->default_format); + + /* Hardcoded latency estimates... */ + ret->latency_lo = 100 * ret->default_rate / 1000; + ret->latency_hi = 200 * ret->default_rate / 1000; +} + +static void +winmm_create_device_from_incaps(cubeb_device_info * ret, LPWAVEINCAPSA caps, UINT devid) +{ + XASSERT(ret); + ret->devid = (cubeb_devid) devid; + ret->device_id = device_id_idx(devid); + ret->friendly_name = _strdup(caps->szPname); + ret->group_id = NULL; + ret->vendor_name = NULL; + + ret->type = CUBEB_DEVICE_TYPE_INPUT; + ret->state = CUBEB_DEVICE_STATE_ENABLED; + ret->preferred = winmm_query_preferred_in_device(devid); + + ret->max_channels = caps->wChannels; + winmm_calculate_device_rate(ret, caps->dwFormats); + winmm_query_supported_formats(devid, caps->dwFormats, + &ret->format, &ret->default_format); + + /* Hardcoded latency estimates... */ + ret->latency_lo = 100 * ret->default_rate / 1000; + ret->latency_hi = 200 * ret->default_rate / 1000; +} + +static int +winmm_enumerate_devices(cubeb * context, cubeb_device_type type, + cubeb_device_collection * collection) +{ + UINT i, incount, outcount, total; + cubeb_device_info * devices; + cubeb_device_info * dev; + + outcount = waveOutGetNumDevs(); + incount = waveInGetNumDevs(); + total = outcount + incount; + + devices = calloc(total, sizeof(cubeb_device_info)); + collection->count = 0; + + if (type & CUBEB_DEVICE_TYPE_OUTPUT) { + WAVEOUTCAPSA woc; + WAVEOUTCAPS2A woc2; + + ZeroMemory(&woc, sizeof(woc)); + ZeroMemory(&woc2, sizeof(woc2)); + + for (i = 0; i < outcount; i++) { + dev = &devices[collection->count]; + if (waveOutGetDevCapsA(i, (LPWAVEOUTCAPSA)&woc2, sizeof(woc2)) == MMSYSERR_NOERROR) { + winmm_create_device_from_outcaps2(dev, &woc2, i); + collection->count += 1; + } else if (waveOutGetDevCapsA(i, &woc, sizeof(woc)) == MMSYSERR_NOERROR) { + winmm_create_device_from_outcaps(dev, &woc, i); + collection->count += 1; + } + } + } + + if (type & CUBEB_DEVICE_TYPE_INPUT) { + WAVEINCAPSA wic; + WAVEINCAPS2A wic2; + + ZeroMemory(&wic, sizeof(wic)); + ZeroMemory(&wic2, sizeof(wic2)); + + for (i = 0; i < incount; i++) { + dev = &devices[collection->count]; + if (waveInGetDevCapsA(i, (LPWAVEINCAPSA)&wic2, sizeof(wic2)) == MMSYSERR_NOERROR) { + winmm_create_device_from_incaps2(dev, &wic2, i); + collection->count += 1; + } else if (waveInGetDevCapsA(i, &wic, sizeof(wic)) == MMSYSERR_NOERROR) { + winmm_create_device_from_incaps(dev, &wic, i); + collection->count += 1; + } + } + } + + collection->device = devices; + + return CUBEB_OK; +} + +static int +winmm_device_collection_destroy(cubeb * ctx, + cubeb_device_collection * collection) +{ + uint32_t i; + XASSERT(collection); + + (void) ctx; + + for (i = 0; i < collection->count; i++) { + free((void *) collection->device[i].device_id); + free((void *) collection->device[i].friendly_name); + free((void *) collection->device[i].group_id); + free((void *) collection->device[i].vendor_name); + } + + free(collection->device); + return CUBEB_OK; +} + +static struct cubeb_ops const winmm_ops = { + /*.init =*/ winmm_init, + /*.get_backend_id =*/ winmm_get_backend_id, + /*.get_max_channel_count=*/ winmm_get_max_channel_count, + /*.get_min_latency=*/ winmm_get_min_latency, + /*.get_preferred_sample_rate =*/ winmm_get_preferred_sample_rate, + /*.enumerate_devices =*/ winmm_enumerate_devices, + /*.device_collection_destroy =*/ winmm_device_collection_destroy, + /*.destroy =*/ winmm_destroy, + /*.stream_init =*/ winmm_stream_init, + /*.stream_destroy =*/ winmm_stream_destroy, + /*.stream_start =*/ winmm_stream_start, + /*.stream_stop =*/ winmm_stream_stop, + /*.stream_reset_default_device =*/ NULL, + /*.stream_get_position =*/ winmm_stream_get_position, + /*.stream_get_latency = */ winmm_stream_get_latency, + /*.stream_set_volume =*/ winmm_stream_set_volume, + /*.stream_get_current_device =*/ NULL, + /*.stream_device_destroy =*/ NULL, + /*.stream_register_device_changed_callback=*/ NULL, + /*.register_device_collection_changed =*/ NULL +}; diff --git a/dep/cubeb/src/speex/arch.h b/dep/cubeb/src/speex/arch.h new file mode 100644 index 000000000..73a45a069 --- /dev/null +++ b/dep/cubeb/src/speex/arch.h @@ -0,0 +1,235 @@ +/* Copyright (C) 2003 Jean-Marc Valin */ +/** + @file arch.h + @brief Various architecture definitions Speex +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef ARCH_H +#define ARCH_H + +/* A couple test to catch stupid option combinations */ +#ifdef FIXED_POINT + +#ifdef FLOATING_POINT +#error You cannot compile as floating point and fixed point at the same time +#endif +#ifdef _USE_SSE +#error SSE is only for floating-point +#endif +#if ((defined (ARM4_ASM)||defined (ARM4_ASM)) && defined(BFIN_ASM)) || (defined (ARM4_ASM)&&defined(ARM5E_ASM)) +#error Make up your mind. What CPU do you have? +#endif +#ifdef VORBIS_PSYCHO +#error Vorbis-psy model currently not implemented in fixed-point +#endif + +#else + +#ifndef FLOATING_POINT +#error You now need to define either FIXED_POINT or FLOATING_POINT +#endif +#if defined (ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM) +#error I suppose you can have a [ARM4/ARM5E/Blackfin] that has float instructions? +#endif +#ifdef FIXED_POINT_DEBUG +#error "Don't you think enabling fixed-point is a good thing to do if you want to debug that?" +#endif + + +#endif + +#ifndef OUTSIDE_SPEEX +#include "speex/speexdsp_types.h" +#endif + +#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */ +#define ABS16(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 16-bit value. */ +#define MIN16(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 16-bit value. */ +#define MAX16(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 16-bit value. */ +#define ABS32(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 32-bit value. */ +#define MIN32(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 32-bit value. */ +#define MAX32(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 32-bit value. */ + +#ifdef FIXED_POINT + +typedef spx_int16_t spx_word16_t; +typedef spx_int32_t spx_word32_t; +typedef spx_word32_t spx_mem_t; +typedef spx_word16_t spx_coef_t; +typedef spx_word16_t spx_lsp_t; +typedef spx_word32_t spx_sig_t; + +#define Q15ONE 32767 + +#define LPC_SCALING 8192 +#define SIG_SCALING 16384 +#define LSP_SCALING 8192. +#define GAMMA_SCALING 32768. +#define GAIN_SCALING 64 +#define GAIN_SCALING_1 0.015625 + +#define LPC_SHIFT 13 +#define LSP_SHIFT 13 +#define SIG_SHIFT 14 +#define GAIN_SHIFT 6 + +#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x))) + +#define VERY_SMALL 0 +#define VERY_LARGE32 ((spx_word32_t)2147483647) +#define VERY_LARGE16 ((spx_word16_t)32767) +#define Q15_ONE ((spx_word16_t)32767) + + +#ifdef FIXED_DEBUG +#include "fixed_debug.h" +#else + +#include "fixed_generic.h" + +#ifdef ARM5E_ASM +#include "fixed_arm5e.h" +#elif defined (ARM4_ASM) +#include "fixed_arm4.h" +#elif defined (BFIN_ASM) +#include "fixed_bfin.h" +#endif + +#endif + + +#else + +typedef float spx_mem_t; +typedef float spx_coef_t; +typedef float spx_lsp_t; +typedef float spx_sig_t; +typedef float spx_word16_t; +typedef float spx_word32_t; + +#define Q15ONE 1.0f +#define LPC_SCALING 1.f +#define SIG_SCALING 1.f +#define LSP_SCALING 1.f +#define GAMMA_SCALING 1.f +#define GAIN_SCALING 1.f +#define GAIN_SCALING_1 1.f + + +#define VERY_SMALL 1e-15f +#define VERY_LARGE32 1e15f +#define VERY_LARGE16 1e15f +#define Q15_ONE ((spx_word16_t)1.f) + +#define QCONST16(x,bits) (x) +#define QCONST32(x,bits) (x) + +#define NEG16(x) (-(x)) +#define NEG32(x) (-(x)) +#define EXTRACT16(x) (x) +#define EXTEND32(x) (x) +#define SHR16(a,shift) (a) +#define SHL16(a,shift) (a) +#define SHR32(a,shift) (a) +#define SHL32(a,shift) (a) +#define PSHR16(a,shift) (a) +#define PSHR32(a,shift) (a) +#define VSHR32(a,shift) (a) +#define SATURATE16(x,a) (x) +#define SATURATE32(x,a) (x) +#define SATURATE32PSHR(x,shift,a) (x) + +#define PSHR(a,shift) (a) +#define SHR(a,shift) (a) +#define SHL(a,shift) (a) +#define SATURATE(x,a) (x) + +#define ADD16(a,b) ((a)+(b)) +#define SUB16(a,b) ((a)-(b)) +#define ADD32(a,b) ((a)+(b)) +#define SUB32(a,b) ((a)-(b)) +#define MULT16_16_16(a,b) ((a)*(b)) +#define MULT16_16(a,b) ((spx_word32_t)(a)*(spx_word32_t)(b)) +#define MAC16_16(c,a,b) ((c)+(spx_word32_t)(a)*(spx_word32_t)(b)) + +#define MULT16_32_Q11(a,b) ((a)*(b)) +#define MULT16_32_Q13(a,b) ((a)*(b)) +#define MULT16_32_Q14(a,b) ((a)*(b)) +#define MULT16_32_Q15(a,b) ((a)*(b)) +#define MULT16_32_P15(a,b) ((a)*(b)) + +#define MAC16_32_Q11(c,a,b) ((c)+(a)*(b)) +#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b)) + +#define MAC16_16_Q11(c,a,b) ((c)+(a)*(b)) +#define MAC16_16_Q13(c,a,b) ((c)+(a)*(b)) +#define MAC16_16_P13(c,a,b) ((c)+(a)*(b)) +#define MULT16_16_Q11_32(a,b) ((a)*(b)) +#define MULT16_16_Q13(a,b) ((a)*(b)) +#define MULT16_16_Q14(a,b) ((a)*(b)) +#define MULT16_16_Q15(a,b) ((a)*(b)) +#define MULT16_16_P15(a,b) ((a)*(b)) +#define MULT16_16_P13(a,b) ((a)*(b)) +#define MULT16_16_P14(a,b) ((a)*(b)) + +#define DIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b)) +#define PDIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b)) +#define DIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b)) +#define PDIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b)) + +#define WORD2INT(x) ((x) < -32767.5f ? -32768 : \ + ((x) > 32766.5f ? 32767 : (spx_int16_t)floor(.5 + (x)))) +#endif + + +#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + +/* 2 on TI C5x DSP */ +#define BYTES_PER_CHAR 2 +#define BITS_PER_CHAR 16 +#define LOG2_BITS_PER_CHAR 4 + +#else + +#define BYTES_PER_CHAR 1 +#define BITS_PER_CHAR 8 +#define LOG2_BITS_PER_CHAR 3 + +#endif + + + +#ifdef FIXED_DEBUG +extern long long spx_mips; +#endif + + +#endif diff --git a/dep/cubeb/src/speex/fixed_generic.h b/dep/cubeb/src/speex/fixed_generic.h new file mode 100644 index 000000000..12d27aac5 --- /dev/null +++ b/dep/cubeb/src/speex/fixed_generic.h @@ -0,0 +1,110 @@ +/* Copyright (C) 2003 Jean-Marc Valin */ +/** + @file fixed_generic.h + @brief Generic fixed-point operations +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_GENERIC_H +#define FIXED_GENERIC_H + +#define QCONST16(x,bits) ((spx_word16_t)(.5+(x)*(((spx_word32_t)1)<<(bits)))) +#define QCONST32(x,bits) ((spx_word32_t)(.5+(x)*(((spx_word32_t)1)<<(bits)))) + +#define NEG16(x) (-(x)) +#define NEG32(x) (-(x)) +#define EXTRACT16(x) ((spx_word16_t)(x)) +#define EXTEND32(x) ((spx_word32_t)(x)) +#define SHR16(a,shift) ((a) >> (shift)) +#define SHL16(a,shift) ((a) << (shift)) +#define SHR32(a,shift) ((a) >> (shift)) +#define SHL32(a,shift) ((a) << (shift)) +#define PSHR16(a,shift) (SHR16((a)+((1<<((shift))>>1)),shift)) +#define PSHR32(a,shift) (SHR32((a)+((EXTEND32(1)<<((shift))>>1)),shift)) +#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift))) +#define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) +#define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) + +#define SATURATE32PSHR(x,shift,a) (((x)>=(SHL32(a,shift))) ? (a) : \ + (x)<=-(SHL32(a,shift)) ? -(a) : \ + (PSHR32(x, shift))) + +#define SHR(a,shift) ((a) >> (shift)) +#define SHL(a,shift) ((spx_word32_t)(a) << (shift)) +#define PSHR(a,shift) (SHR((a)+((EXTEND32(1)<<((shift))>>1)),shift)) +#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) + + +#define ADD16(a,b) ((spx_word16_t)((spx_word16_t)(a)+(spx_word16_t)(b))) +#define SUB16(a,b) ((spx_word16_t)(a)-(spx_word16_t)(b)) +#define ADD32(a,b) ((spx_word32_t)(a)+(spx_word32_t)(b)) +#define SUB32(a,b) ((spx_word32_t)(a)-(spx_word32_t)(b)) + + +/* result fits in 16 bits */ +#define MULT16_16_16(a,b) ((((spx_word16_t)(a))*((spx_word16_t)(b)))) + +/* (spx_word32_t)(spx_word16_t) gives TI compiler a hint that it's 16x16->32 multiply */ +#define MULT16_16(a,b) (((spx_word32_t)(spx_word16_t)(a))*((spx_word32_t)(spx_word16_t)(b))) + +#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b)))) +#define MULT16_32_Q12(a,b) ADD32(MULT16_16((a),SHR((b),12)), SHR(MULT16_16((a),((b)&0x00000fff)),12)) +#define MULT16_32_Q13(a,b) ADD32(MULT16_16((a),SHR((b),13)), SHR(MULT16_16((a),((b)&0x00001fff)),13)) +#define MULT16_32_Q14(a,b) ADD32(MULT16_16((a),SHR((b),14)), SHR(MULT16_16((a),((b)&0x00003fff)),14)) + +#define MULT16_32_Q11(a,b) ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11)) +#define MAC16_32_Q11(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11))) + +#define MULT16_32_P15(a,b) ADD32(MULT16_16((a),SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15)) +#define MULT16_32_Q15(a,b) ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)) +#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))) + + +#define MAC16_16_Q11(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),11))) +#define MAC16_16_Q13(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),13))) +#define MAC16_16_P13(c,a,b) (ADD32((c),SHR(ADD32(4096,MULT16_16((a),(b))),13))) + +#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11)) +#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13)) +#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14)) +#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15)) + +#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13)) +#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14)) +#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15)) + +#define MUL_16_32_R15(a,bh,bl) ADD32(MULT16_16((a),(bh)), SHR(MULT16_16((a),(bl)),15)) + +#define DIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a))/((spx_word16_t)(b)))) +#define PDIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word16_t)(b)))) +#define DIV32(a,b) (((spx_word32_t)(a))/((spx_word32_t)(b))) +#define PDIV32(a,b) (((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word32_t)(b))) + +#endif diff --git a/dep/cubeb/src/speex/resample.c b/dep/cubeb/src/speex/resample.c new file mode 100644 index 000000000..10cb06593 --- /dev/null +++ b/dep/cubeb/src/speex/resample.c @@ -0,0 +1,1240 @@ +/* Copyright (C) 2007-2008 Jean-Marc Valin + Copyright (C) 2008 Thorvald Natvig + + File: resample.c + Arbitrary resampling code + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +/* + The design goals of this code are: + - Very fast algorithm + - SIMD-friendly algorithm + - Low memory requirement + - Good *perceptual* quality (and not best SNR) + + Warning: This resampler is relatively new. Although I think I got rid of + all the major bugs and I don't expect the API to change anymore, there + may be something I've missed. So use with caution. + + This algorithm is based on this original resampling algorithm: + Smith, Julius O. Digital Audio Resampling Home Page + Center for Computer Research in Music and Acoustics (CCRMA), + Stanford University, 2007. + Web published at http://ccrma.stanford.edu/~jos/resample/. + + There is one main difference, though. This resampler uses cubic + interpolation instead of linear interpolation in the above paper. This + makes the table much smaller and makes it possible to compute that table + on a per-stream basis. In turn, being able to tweak the table for each + stream makes it possible to both reduce complexity on simple ratios + (e.g. 2/3), and get rid of the rounding operations in the inner loop. + The latter both reduces CPU time and makes the algorithm more SIMD-friendly. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#ifdef OUTSIDE_SPEEX +#include +static void *speex_alloc (int size) {return calloc(size,1);} +static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);} +static void speex_free (void *ptr) {free(ptr);} +#include "speex_resampler.h" +#include "arch.h" +#else /* OUTSIDE_SPEEX */ + +#include "speex/speex_resampler.h" +#include "arch.h" +#include "os_support.h" +#endif /* OUTSIDE_SPEEX */ + +#include "stack_alloc.h" +#include +#include + +#ifndef M_PI +#define M_PI 3.14159265358979323846 +#endif + +#define IMAX(a,b) ((a) > (b) ? (a) : (b)) +#define IMIN(a,b) ((a) < (b) ? (a) : (b)) + +#ifndef NULL +#define NULL 0 +#endif + +#ifndef UINT32_MAX +#define UINT32_MAX 4294967296U +#endif + +#ifdef _USE_SSE +#include "resample_sse.h" +#endif + +#ifdef _USE_NEON +#include "resample_neon.h" +#endif + +/* Numer of elements to allocate on the stack */ +#ifdef VAR_ARRAYS +#define FIXED_STACK_ALLOC 8192 +#else +#define FIXED_STACK_ALLOC 1024 +#endif + +typedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *); + +struct SpeexResamplerState_ { + spx_uint32_t in_rate; + spx_uint32_t out_rate; + spx_uint32_t num_rate; + spx_uint32_t den_rate; + + int quality; + spx_uint32_t nb_channels; + spx_uint32_t filt_len; + spx_uint32_t mem_alloc_size; + spx_uint32_t buffer_size; + int int_advance; + int frac_advance; + float cutoff; + spx_uint32_t oversample; + int initialised; + int started; + + /* These are per-channel */ + spx_int32_t *last_sample; + spx_uint32_t *samp_frac_num; + spx_uint32_t *magic_samples; + + spx_word16_t *mem; + spx_word16_t *sinc_table; + spx_uint32_t sinc_table_length; + resampler_basic_func resampler_ptr; + + int in_stride; + int out_stride; +} ; + +static const double kaiser12_table[68] = { + 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076, + 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014, + 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601, + 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014, + 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490, + 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546, + 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178, + 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947, + 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058, + 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438, + 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734, + 0.00001000, 0.00000000}; +/* +static const double kaiser12_table[36] = { + 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741, + 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762, + 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274, + 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466, + 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291, + 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000}; +*/ +static const double kaiser10_table[36] = { + 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446, + 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347, + 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962, + 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451, + 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739, + 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000}; + +static const double kaiser8_table[36] = { + 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200, + 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126, + 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272, + 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758, + 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490, + 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000}; + +static const double kaiser6_table[36] = { + 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003, + 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565, + 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561, + 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058, + 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600, + 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000}; + +struct FuncDef { + const double *table; + int oversample; +}; + +static const struct FuncDef _KAISER12 = {kaiser12_table, 64}; +#define KAISER12 (&_KAISER12) +/*static struct FuncDef _KAISER12 = {kaiser12_table, 32}; +#define KAISER12 (&_KAISER12)*/ +static const struct FuncDef _KAISER10 = {kaiser10_table, 32}; +#define KAISER10 (&_KAISER10) +static const struct FuncDef _KAISER8 = {kaiser8_table, 32}; +#define KAISER8 (&_KAISER8) +static const struct FuncDef _KAISER6 = {kaiser6_table, 32}; +#define KAISER6 (&_KAISER6) + +struct QualityMapping { + int base_length; + int oversample; + float downsample_bandwidth; + float upsample_bandwidth; + const struct FuncDef *window_func; +}; + + +/* This table maps conversion quality to internal parameters. There are two + reasons that explain why the up-sampling bandwidth is larger than the + down-sampling bandwidth: + 1) When up-sampling, we can assume that the spectrum is already attenuated + close to the Nyquist rate (from an A/D or a previous resampling filter) + 2) Any aliasing that occurs very close to the Nyquist rate will be masked + by the sinusoids/noise just below the Nyquist rate (guaranteed only for + up-sampling). +*/ +static const struct QualityMapping quality_map[11] = { + { 8, 4, 0.830f, 0.860f, KAISER6 }, /* Q0 */ + { 16, 4, 0.850f, 0.880f, KAISER6 }, /* Q1 */ + { 32, 4, 0.882f, 0.910f, KAISER6 }, /* Q2 */ /* 82.3% cutoff ( ~60 dB stop) 6 */ + { 48, 8, 0.895f, 0.917f, KAISER8 }, /* Q3 */ /* 84.9% cutoff ( ~80 dB stop) 8 */ + { 64, 8, 0.921f, 0.940f, KAISER8 }, /* Q4 */ /* 88.7% cutoff ( ~80 dB stop) 8 */ + { 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */ + { 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */ + {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */ + {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */ + {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */ + {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */ +}; +/*8,24,40,56,80,104,128,160,200,256,320*/ +static double compute_func(float x, const struct FuncDef *func) +{ + float y, frac; + double interp[4]; + int ind; + y = x*func->oversample; + ind = (int)floor(y); + frac = (y-ind); + /* CSE with handle the repeated powers */ + interp[3] = -0.1666666667*frac + 0.1666666667*(frac*frac*frac); + interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac); + /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/ + interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac); + /* Just to make sure we don't have rounding problems */ + interp[1] = 1.f-interp[3]-interp[2]-interp[0]; + + /*sum = frac*accum[1] + (1-frac)*accum[2];*/ + return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3]; +} + +#if 0 +#include +int main(int argc, char **argv) +{ + int i; + for (i=0;i<256;i++) + { + printf ("%f\n", compute_func(i/256., KAISER12)); + } + return 0; +} +#endif + +#ifdef FIXED_POINT +/* The slow way of computing a sinc for the table. Should improve that some day */ +static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func) +{ + /*fprintf (stderr, "%f ", x);*/ + float xx = x * cutoff; + if (fabs(x)<1e-6f) + return WORD2INT(32768.*cutoff); + else if (fabs(x) > .5f*N) + return 0; + /*FIXME: Can it really be any slower than this? */ + return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func)); +} +#else +/* The slow way of computing a sinc for the table. Should improve that some day */ +static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func) +{ + /*fprintf (stderr, "%f ", x);*/ + float xx = x * cutoff; + if (fabs(x)<1e-6) + return cutoff; + else if (fabs(x) > .5*N) + return 0; + /*FIXME: Can it really be any slower than this? */ + return cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func); +} +#endif + +#ifdef FIXED_POINT +static void cubic_coef(spx_word16_t x, spx_word16_t interp[4]) +{ + /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation + but I know it's MMSE-optimal on a sinc */ + spx_word16_t x2, x3; + x2 = MULT16_16_P15(x, x); + x3 = MULT16_16_P15(x, x2); + interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15); + interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1)); + interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15); + /* Just to make sure we don't have rounding problems */ + interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3]; + if (interp[2]<32767) + interp[2]+=1; +} +#else +static void cubic_coef(spx_word16_t frac, spx_word16_t interp[4]) +{ + /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation + but I know it's MMSE-optimal on a sinc */ + interp[0] = -0.16667f*frac + 0.16667f*frac*frac*frac; + interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac; + /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/ + interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac; + /* Just to make sure we don't have rounding problems */ + interp[2] = 1.-interp[0]-interp[1]-interp[3]; +} +#endif + +static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len) +{ + const int N = st->filt_len; + int out_sample = 0; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + const spx_word16_t *sinc_table = st->sinc_table; + const int out_stride = st->out_stride; + const int int_advance = st->int_advance; + const int frac_advance = st->frac_advance; + const spx_uint32_t den_rate = st->den_rate; + spx_word32_t sum; + + while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) + { + const spx_word16_t *sinct = & sinc_table[samp_frac_num*N]; + const spx_word16_t *iptr = & in[last_sample]; + +#ifndef OVERRIDE_INNER_PRODUCT_SINGLE + int j; + sum = 0; + for(j=0;j= den_rate) + { + samp_frac_num -= den_rate; + last_sample++; + } + } + + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} + +#ifdef FIXED_POINT +#else +/* This is the same as the previous function, except with a double-precision accumulator */ +static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len) +{ + const int N = st->filt_len; + int out_sample = 0; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + const spx_word16_t *sinc_table = st->sinc_table; + const int out_stride = st->out_stride; + const int int_advance = st->int_advance; + const int frac_advance = st->frac_advance; + const spx_uint32_t den_rate = st->den_rate; + double sum; + + while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) + { + const spx_word16_t *sinct = & sinc_table[samp_frac_num*N]; + const spx_word16_t *iptr = & in[last_sample]; + +#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE + int j; + double accum[4] = {0,0,0,0}; + + for(j=0;j= den_rate) + { + samp_frac_num -= den_rate; + last_sample++; + } + } + + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} +#endif + +static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len) +{ + const int N = st->filt_len; + int out_sample = 0; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + const int out_stride = st->out_stride; + const int int_advance = st->int_advance; + const int frac_advance = st->frac_advance; + const spx_uint32_t den_rate = st->den_rate; + spx_word32_t sum; + + while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) + { + const spx_word16_t *iptr = & in[last_sample]; + + const int offset = samp_frac_num*st->oversample/st->den_rate; +#ifdef FIXED_POINT + const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate); +#else + const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate; +#endif + spx_word16_t interp[4]; + + +#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE + int j; + spx_word32_t accum[4] = {0,0,0,0}; + + for(j=0;jsinc_table[4+(j+1)*st->oversample-offset-2]); + accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]); + accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]); + accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]); + } + + cubic_coef(frac, interp); + sum = MULT16_32_Q15(interp[0],SHR32(accum[0], 1)) + MULT16_32_Q15(interp[1],SHR32(accum[1], 1)) + MULT16_32_Q15(interp[2],SHR32(accum[2], 1)) + MULT16_32_Q15(interp[3],SHR32(accum[3], 1)); + sum = SATURATE32PSHR(sum, 15, 32767); +#else + cubic_coef(frac, interp); + sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp); +#endif + + out[out_stride * out_sample++] = sum; + last_sample += int_advance; + samp_frac_num += frac_advance; + if (samp_frac_num >= den_rate) + { + samp_frac_num -= den_rate; + last_sample++; + } + } + + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} + +#ifdef FIXED_POINT +#else +/* This is the same as the previous function, except with a double-precision accumulator */ +static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len) +{ + const int N = st->filt_len; + int out_sample = 0; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + const int out_stride = st->out_stride; + const int int_advance = st->int_advance; + const int frac_advance = st->frac_advance; + const spx_uint32_t den_rate = st->den_rate; + spx_word32_t sum; + + while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) + { + const spx_word16_t *iptr = & in[last_sample]; + + const int offset = samp_frac_num*st->oversample/st->den_rate; +#ifdef FIXED_POINT + const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate); +#else + const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate; +#endif + spx_word16_t interp[4]; + + +#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE + int j; + double accum[4] = {0,0,0,0}; + + for(j=0;jsinc_table[4+(j+1)*st->oversample-offset-2]); + accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]); + accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]); + accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]); + } + + cubic_coef(frac, interp); + sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]); +#else + cubic_coef(frac, interp); + sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp); +#endif + + out[out_stride * out_sample++] = PSHR32(sum,15); + last_sample += int_advance; + samp_frac_num += frac_advance; + if (samp_frac_num >= den_rate) + { + samp_frac_num -= den_rate; + last_sample++; + } + } + + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} +#endif + +/* This resampler is used to produce zero output in situations where memory + for the filter could not be allocated. The expected numbers of input and + output samples are still processed so that callers failing to check error + codes are not surprised, possibly getting into infinite loops. */ +static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len) +{ + int out_sample = 0; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + const int out_stride = st->out_stride; + const int int_advance = st->int_advance; + const int frac_advance = st->frac_advance; + const spx_uint32_t den_rate = st->den_rate; + + while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) + { + out[out_stride * out_sample++] = 0; + last_sample += int_advance; + samp_frac_num += frac_advance; + if (samp_frac_num >= den_rate) + { + samp_frac_num -= den_rate; + last_sample++; + } + } + + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} + +static int _muldiv(spx_uint32_t *result, spx_uint32_t value, spx_uint32_t mul, spx_uint32_t div) +{ + speex_assert(result); + spx_uint32_t major = value / div; + spx_uint32_t remainder = value % div; + /* TODO: Could use 64 bits operation to check for overflow. But only guaranteed in C99+ */ + if (remainder > UINT32_MAX / mul || major > UINT32_MAX / mul + || major * mul > UINT32_MAX - remainder * mul / div) + return RESAMPLER_ERR_OVERFLOW; + *result = remainder * mul / div + major * mul; + return RESAMPLER_ERR_SUCCESS; +} + +static int update_filter(SpeexResamplerState *st) +{ + spx_uint32_t old_length = st->filt_len; + spx_uint32_t old_alloc_size = st->mem_alloc_size; + int use_direct; + spx_uint32_t min_sinc_table_length; + spx_uint32_t min_alloc_size; + + st->int_advance = st->num_rate/st->den_rate; + st->frac_advance = st->num_rate%st->den_rate; + st->oversample = quality_map[st->quality].oversample; + st->filt_len = quality_map[st->quality].base_length; + + if (st->num_rate > st->den_rate) + { + /* down-sampling */ + st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate; + if (_muldiv(&st->filt_len,st->filt_len,st->num_rate,st->den_rate) != RESAMPLER_ERR_SUCCESS) + goto fail; + /* Round up to make sure we have a multiple of 8 for SSE */ + st->filt_len = ((st->filt_len-1)&(~0x7))+8; + if (2*st->den_rate < st->num_rate) + st->oversample >>= 1; + if (4*st->den_rate < st->num_rate) + st->oversample >>= 1; + if (8*st->den_rate < st->num_rate) + st->oversample >>= 1; + if (16*st->den_rate < st->num_rate) + st->oversample >>= 1; + if (st->oversample < 1) + st->oversample = 1; + } else { + /* up-sampling */ + st->cutoff = quality_map[st->quality].upsample_bandwidth; + } + + /* Choose the resampling type that requires the least amount of memory */ +#ifdef RESAMPLE_FULL_SINC_TABLE + use_direct = 1; + if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len) + goto fail; +#else + use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8 + && INT_MAX/sizeof(spx_word16_t)/st->den_rate >= st->filt_len; +#endif + if (use_direct) + { + min_sinc_table_length = st->filt_len*st->den_rate; + } else { + if ((INT_MAX/sizeof(spx_word16_t)-8)/st->oversample < st->filt_len) + goto fail; + + min_sinc_table_length = st->filt_len*st->oversample+8; + } + if (st->sinc_table_length < min_sinc_table_length) + { + spx_word16_t *sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,min_sinc_table_length*sizeof(spx_word16_t)); + if (!sinc_table) + goto fail; + + st->sinc_table = sinc_table; + st->sinc_table_length = min_sinc_table_length; + } + if (use_direct) + { + spx_uint32_t i; + for (i=0;iden_rate;i++) + { + spx_int32_t j; + for (j=0;jfilt_len;j++) + { + st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func); + } + } +#ifdef FIXED_POINT + st->resampler_ptr = resampler_basic_direct_single; +#else + if (st->quality>8) + st->resampler_ptr = resampler_basic_direct_double; + else + st->resampler_ptr = resampler_basic_direct_single; +#endif + /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/ + } else { + spx_int32_t i; + for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++) + st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func); +#ifdef FIXED_POINT + st->resampler_ptr = resampler_basic_interpolate_single; +#else + if (st->quality>8) + st->resampler_ptr = resampler_basic_interpolate_double; + else + st->resampler_ptr = resampler_basic_interpolate_single; +#endif + /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/ + } + + /* Here's the place where we update the filter memory to take into account + the change in filter length. It's probably the messiest part of the code + due to handling of lots of corner cases. */ + + /* Adding buffer_size to filt_len won't overflow here because filt_len + could be multiplied by sizeof(spx_word16_t) above. */ + min_alloc_size = st->filt_len-1 + st->buffer_size; + if (min_alloc_size > st->mem_alloc_size) + { + spx_word16_t *mem; + if (INT_MAX/sizeof(spx_word16_t)/st->nb_channels < min_alloc_size) + goto fail; + else if (!(mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*min_alloc_size * sizeof(*mem)))) + goto fail; + + st->mem = mem; + st->mem_alloc_size = min_alloc_size; + } + if (!st->started) + { + spx_uint32_t i; + for (i=0;inb_channels*st->mem_alloc_size;i++) + st->mem[i] = 0; + /*speex_warning("reinit filter");*/ + } else if (st->filt_len > old_length) + { + spx_uint32_t i; + /* Increase the filter length */ + /*speex_warning("increase filter size");*/ + for (i=st->nb_channels;i--;) + { + spx_uint32_t j; + spx_uint32_t olen = old_length; + /*if (st->magic_samples[i])*/ + { + /* Try and remove the magic samples as if nothing had happened */ + + /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */ + olen = old_length + 2*st->magic_samples[i]; + for (j=old_length-1+st->magic_samples[i];j--;) + st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j]; + for (j=0;jmagic_samples[i];j++) + st->mem[i*st->mem_alloc_size+j] = 0; + st->magic_samples[i] = 0; + } + if (st->filt_len > olen) + { + /* If the new filter length is still bigger than the "augmented" length */ + /* Copy data going backward */ + for (j=0;jmem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)]; + /* Then put zeros for lack of anything better */ + for (;jfilt_len-1;j++) + st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0; + /* Adjust last_sample */ + st->last_sample[i] += (st->filt_len - olen)/2; + } else { + /* Put back some of the magic! */ + st->magic_samples[i] = (olen - st->filt_len)/2; + for (j=0;jfilt_len-1+st->magic_samples[i];j++) + st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]]; + } + } + } else if (st->filt_len < old_length) + { + spx_uint32_t i; + /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic" + samples so they can be used directly as input the next time(s) */ + for (i=0;inb_channels;i++) + { + spx_uint32_t j; + spx_uint32_t old_magic = st->magic_samples[i]; + st->magic_samples[i] = (old_length - st->filt_len)/2; + /* We must copy some of the memory that's no longer used */ + /* Copy data going backward */ + for (j=0;jfilt_len-1+st->magic_samples[i]+old_magic;j++) + st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]]; + st->magic_samples[i] += old_magic; + } + } + return RESAMPLER_ERR_SUCCESS; + +fail: + st->resampler_ptr = resampler_basic_zero; + /* st->mem may still contain consumed input samples for the filter. + Restore filt_len so that filt_len - 1 still points to the position after + the last of these samples. */ + st->filt_len = old_length; + return RESAMPLER_ERR_ALLOC_FAILED; +} + +EXPORT SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err) +{ + return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err); +} + +EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err) +{ + SpeexResamplerState *st; + int filter_err; + + if (nb_channels == 0 || ratio_num == 0 || ratio_den == 0 || quality > 10 || quality < 0) + { + if (err) + *err = RESAMPLER_ERR_INVALID_ARG; + return NULL; + } + st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState)); + if (!st) + { + if (err) + *err = RESAMPLER_ERR_ALLOC_FAILED; + return NULL; + } + st->initialised = 0; + st->started = 0; + st->in_rate = 0; + st->out_rate = 0; + st->num_rate = 0; + st->den_rate = 0; + st->quality = -1; + st->sinc_table_length = 0; + st->mem_alloc_size = 0; + st->filt_len = 0; + st->mem = 0; + st->resampler_ptr = 0; + + st->cutoff = 1.f; + st->nb_channels = nb_channels; + st->in_stride = 1; + st->out_stride = 1; + + st->buffer_size = 160; + + /* Per channel data */ + if (!(st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t)))) + goto fail; + if (!(st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t)))) + goto fail; + if (!(st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t)))) + goto fail; + + speex_resampler_set_quality(st, quality); + speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate); + + filter_err = update_filter(st); + if (filter_err == RESAMPLER_ERR_SUCCESS) + { + st->initialised = 1; + } else { + speex_resampler_destroy(st); + st = NULL; + } + if (err) + *err = filter_err; + + return st; + +fail: + if (err) + *err = RESAMPLER_ERR_ALLOC_FAILED; + speex_resampler_destroy(st); + return NULL; +} + +EXPORT void speex_resampler_destroy(SpeexResamplerState *st) +{ + speex_free(st->mem); + speex_free(st->sinc_table); + speex_free(st->last_sample); + speex_free(st->magic_samples); + speex_free(st->samp_frac_num); + speex_free(st); +} + +static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t channel_index, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len) +{ + int j=0; + const int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size; + spx_uint32_t ilen; + + st->started = 1; + + /* Call the right resampler through the function ptr */ + out_sample = st->resampler_ptr(st, channel_index, mem, in_len, out, out_len); + + if (st->last_sample[channel_index] < (spx_int32_t)*in_len) + *in_len = st->last_sample[channel_index]; + *out_len = out_sample; + st->last_sample[channel_index] -= *in_len; + + ilen = *in_len; + + for(j=0;jmagic_samples[channel_index]; + spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size; + const int N = st->filt_len; + + speex_resampler_process_native(st, channel_index, &tmp_in_len, *out, &out_len); + + st->magic_samples[channel_index] -= tmp_in_len; + + /* If we couldn't process all "magic" input samples, save the rest for next time */ + if (st->magic_samples[channel_index]) + { + spx_uint32_t i; + for (i=0;imagic_samples[channel_index];i++) + mem[N-1+i]=mem[N-1+i+tmp_in_len]; + } + *out += out_len*st->out_stride; + return out_len; +} + +#ifdef FIXED_POINT +EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len) +#else +EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len) +#endif +{ + int j; + spx_uint32_t ilen = *in_len; + spx_uint32_t olen = *out_len; + spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size; + const int filt_offs = st->filt_len - 1; + const spx_uint32_t xlen = st->mem_alloc_size - filt_offs; + const int istride = st->in_stride; + + if (st->magic_samples[channel_index]) + olen -= speex_resampler_magic(st, channel_index, &out, olen); + if (! st->magic_samples[channel_index]) { + while (ilen && olen) { + spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen; + spx_uint32_t ochunk = olen; + + if (in) { + for(j=0;jout_stride; + if (in) + in += ichunk * istride; + } + } + *in_len -= ilen; + *out_len -= olen; + return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS; +} + +#ifdef FIXED_POINT +EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len) +#else +EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len) +#endif +{ + int j; + const int istride_save = st->in_stride; + const int ostride_save = st->out_stride; + spx_uint32_t ilen = *in_len; + spx_uint32_t olen = *out_len; + spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size; + const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1); +#ifdef VAR_ARRAYS + const unsigned int ylen = (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC; + VARDECL(spx_word16_t *ystack); + ALLOC(ystack, ylen, spx_word16_t); +#else + const unsigned int ylen = FIXED_STACK_ALLOC; + spx_word16_t ystack[FIXED_STACK_ALLOC]; +#endif + + st->out_stride = 1; + + while (ilen && olen) { + spx_word16_t *y = ystack; + spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen; + spx_uint32_t ochunk = (olen > ylen) ? ylen : olen; + spx_uint32_t omagic = 0; + + if (st->magic_samples[channel_index]) { + omagic = speex_resampler_magic(st, channel_index, &y, ochunk); + ochunk -= omagic; + olen -= omagic; + } + if (! st->magic_samples[channel_index]) { + if (in) { + for(j=0;jfilt_len-1]=WORD2INT(in[j*istride_save]); +#else + x[j+st->filt_len-1]=in[j*istride_save]; +#endif + } else { + for(j=0;jfilt_len-1]=0; + } + + speex_resampler_process_native(st, channel_index, &ichunk, y, &ochunk); + } else { + ichunk = 0; + ochunk = 0; + } + + for (j=0;jout_stride = ostride_save; + *in_len -= ilen; + *out_len -= olen; + + return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS; +} + +EXPORT int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len) +{ + spx_uint32_t i; + int istride_save, ostride_save; + spx_uint32_t bak_out_len = *out_len; + spx_uint32_t bak_in_len = *in_len; + istride_save = st->in_stride; + ostride_save = st->out_stride; + st->in_stride = st->out_stride = st->nb_channels; + for (i=0;inb_channels;i++) + { + *out_len = bak_out_len; + *in_len = bak_in_len; + if (in != NULL) + speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len); + else + speex_resampler_process_float(st, i, NULL, in_len, out+i, out_len); + } + st->in_stride = istride_save; + st->out_stride = ostride_save; + return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS; +} + +EXPORT int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len) +{ + spx_uint32_t i; + int istride_save, ostride_save; + spx_uint32_t bak_out_len = *out_len; + spx_uint32_t bak_in_len = *in_len; + istride_save = st->in_stride; + ostride_save = st->out_stride; + st->in_stride = st->out_stride = st->nb_channels; + for (i=0;inb_channels;i++) + { + *out_len = bak_out_len; + *in_len = bak_in_len; + if (in != NULL) + speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len); + else + speex_resampler_process_int(st, i, NULL, in_len, out+i, out_len); + } + st->in_stride = istride_save; + st->out_stride = ostride_save; + return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS; +} + +EXPORT int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate) +{ + return speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate); +} + +EXPORT void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate) +{ + *in_rate = st->in_rate; + *out_rate = st->out_rate; +} + +static inline spx_uint32_t _gcd(spx_uint32_t a, spx_uint32_t b) +{ + while (b != 0) + { + spx_uint32_t temp = a; + + a = b; + b = temp % b; + } + return a; +} + +EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate) +{ + spx_uint32_t fact; + spx_uint32_t old_den; + spx_uint32_t i; + + if (ratio_num == 0 || ratio_den == 0) + return RESAMPLER_ERR_INVALID_ARG; + + if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den) + return RESAMPLER_ERR_SUCCESS; + + old_den = st->den_rate; + st->in_rate = in_rate; + st->out_rate = out_rate; + st->num_rate = ratio_num; + st->den_rate = ratio_den; + + fact = _gcd (st->num_rate, st->den_rate); + + st->num_rate /= fact; + st->den_rate /= fact; + + if (old_den > 0) + { + for (i=0;inb_channels;i++) + { + if (_muldiv(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS) + return RESAMPLER_ERR_OVERFLOW; + /* Safety net */ + if (st->samp_frac_num[i] >= st->den_rate) + st->samp_frac_num[i] = st->den_rate-1; + } + } + + if (st->initialised) + return update_filter(st); + return RESAMPLER_ERR_SUCCESS; +} + +EXPORT void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den) +{ + *ratio_num = st->num_rate; + *ratio_den = st->den_rate; +} + +EXPORT int speex_resampler_set_quality(SpeexResamplerState *st, int quality) +{ + if (quality > 10 || quality < 0) + return RESAMPLER_ERR_INVALID_ARG; + if (st->quality == quality) + return RESAMPLER_ERR_SUCCESS; + st->quality = quality; + if (st->initialised) + return update_filter(st); + return RESAMPLER_ERR_SUCCESS; +} + +EXPORT void speex_resampler_get_quality(SpeexResamplerState *st, int *quality) +{ + *quality = st->quality; +} + +EXPORT void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride) +{ + st->in_stride = stride; +} + +EXPORT void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride) +{ + *stride = st->in_stride; +} + +EXPORT void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride) +{ + st->out_stride = stride; +} + +EXPORT void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride) +{ + *stride = st->out_stride; +} + +EXPORT int speex_resampler_get_input_latency(SpeexResamplerState *st) +{ + return st->filt_len / 2; +} + +EXPORT int speex_resampler_get_output_latency(SpeexResamplerState *st) +{ + return ((st->filt_len / 2) * st->den_rate + (st->num_rate >> 1)) / st->num_rate; +} + +EXPORT int speex_resampler_skip_zeros(SpeexResamplerState *st) +{ + spx_uint32_t i; + for (i=0;inb_channels;i++) + st->last_sample[i] = st->filt_len/2; + return RESAMPLER_ERR_SUCCESS; +} + +EXPORT int speex_resampler_reset_mem(SpeexResamplerState *st) +{ + spx_uint32_t i; + for (i=0;inb_channels;i++) + { + st->last_sample[i] = 0; + st->magic_samples[i] = 0; + st->samp_frac_num[i] = 0; + } + for (i=0;inb_channels*(st->filt_len-1);i++) + st->mem[i] = 0; + return RESAMPLER_ERR_SUCCESS; +} + +EXPORT const char *speex_resampler_strerror(int err) +{ + switch (err) + { + case RESAMPLER_ERR_SUCCESS: + return "Success."; + case RESAMPLER_ERR_ALLOC_FAILED: + return "Memory allocation failed."; + case RESAMPLER_ERR_BAD_STATE: + return "Bad resampler state."; + case RESAMPLER_ERR_INVALID_ARG: + return "Invalid argument."; + case RESAMPLER_ERR_PTR_OVERLAP: + return "Input and output buffers overlap."; + default: + return "Unknown error. Bad error code or strange version mismatch."; + } +} diff --git a/dep/cubeb/src/speex/resample_neon.h b/dep/cubeb/src/speex/resample_neon.h new file mode 100644 index 000000000..0acbd27b9 --- /dev/null +++ b/dep/cubeb/src/speex/resample_neon.h @@ -0,0 +1,201 @@ +/* Copyright (C) 2007-2008 Jean-Marc Valin + * Copyright (C) 2008 Thorvald Natvig + * Copyright (C) 2011 Texas Instruments + * author Jyri Sarha + */ +/** + @file resample_neon.h + @brief Resampler functions (NEON version) +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#include + +#ifdef FIXED_POINT +#ifdef __thumb2__ +static inline int32_t saturate_32bit_to_16bit(int32_t a) { + int32_t ret; + asm ("ssat %[ret], #16, %[a]" + : [ret] "=&r" (ret) + : [a] "r" (a) + : ); + return ret; +} +#else +static inline int32_t saturate_32bit_to_16bit(int32_t a) { + int32_t ret; + asm ("vmov.s32 d0[0], %[a]\n" + "vqmovn.s32 d0, q0\n" + "vmov.s16 %[ret], d0[0]\n" + : [ret] "=&r" (ret) + : [a] "r" (a) + : "q0"); + return ret; +} +#endif +#undef WORD2INT +#define WORD2INT(x) (saturate_32bit_to_16bit(x)) + +#define OVERRIDE_INNER_PRODUCT_SINGLE +/* Only works when len % 4 == 0 */ +static inline int32_t inner_product_single(const int16_t *a, const int16_t *b, unsigned int len) +{ + int32_t ret; + uint32_t remainder = len % 16; + len = len - remainder; + + asm volatile (" cmp %[len], #0\n" + " bne 1f\n" + " vld1.16 {d16}, [%[b]]!\n" + " vld1.16 {d20}, [%[a]]!\n" + " subs %[remainder], %[remainder], #4\n" + " vmull.s16 q0, d16, d20\n" + " beq 5f\n" + " b 4f\n" + "1:" + " vld1.16 {d16, d17, d18, d19}, [%[b]]!\n" + " vld1.16 {d20, d21, d22, d23}, [%[a]]!\n" + " subs %[len], %[len], #16\n" + " vmull.s16 q0, d16, d20\n" + " vmlal.s16 q0, d17, d21\n" + " vmlal.s16 q0, d18, d22\n" + " vmlal.s16 q0, d19, d23\n" + " beq 3f\n" + "2:" + " vld1.16 {d16, d17, d18, d19}, [%[b]]!\n" + " vld1.16 {d20, d21, d22, d23}, [%[a]]!\n" + " subs %[len], %[len], #16\n" + " vmlal.s16 q0, d16, d20\n" + " vmlal.s16 q0, d17, d21\n" + " vmlal.s16 q0, d18, d22\n" + " vmlal.s16 q0, d19, d23\n" + " bne 2b\n" + "3:" + " cmp %[remainder], #0\n" + " beq 5f\n" + "4:" + " vld1.16 {d16}, [%[b]]!\n" + " vld1.16 {d20}, [%[a]]!\n" + " subs %[remainder], %[remainder], #4\n" + " vmlal.s16 q0, d16, d20\n" + " bne 4b\n" + "5:" + " vaddl.s32 q0, d0, d1\n" + " vadd.s64 d0, d0, d1\n" + " vqmovn.s64 d0, q0\n" + " vqrshrn.s32 d0, q0, #15\n" + " vmov.s16 %[ret], d0[0]\n" + : [ret] "=&r" (ret), [a] "+r" (a), [b] "+r" (b), + [len] "+r" (len), [remainder] "+r" (remainder) + : + : "cc", "q0", + "d16", "d17", "d18", "d19", + "d20", "d21", "d22", "d23"); + + return ret; +} +#elif defined(FLOATING_POINT) + +static inline int32_t saturate_float_to_16bit(float a) { + int32_t ret; + asm ("vmov.f32 d0[0], %[a]\n" + "vcvt.s32.f32 d0, d0, #15\n" + "vqrshrn.s32 d0, q0, #15\n" + "vmov.s16 %[ret], d0[0]\n" + : [ret] "=&r" (ret) + : [a] "r" (a) + : "q0"); + return ret; +} +#undef WORD2INT +#define WORD2INT(x) (saturate_float_to_16bit(x)) + +#define OVERRIDE_INNER_PRODUCT_SINGLE +/* Only works when len % 4 == 0 */ +static inline float inner_product_single(const float *a, const float *b, unsigned int len) +{ + float ret; + uint32_t remainder = len % 16; + len = len - remainder; + + asm volatile (" cmp %[len], #0\n" + " bne 1f\n" + " vld1.32 {q4}, [%[b]]!\n" + " vld1.32 {q8}, [%[a]]!\n" + " subs %[remainder], %[remainder], #4\n" + " vmul.f32 q0, q4, q8\n" + " bne 4f\n" + " b 5f\n" + "1:" + " vld1.32 {q4, q5}, [%[b]]!\n" + " vld1.32 {q8, q9}, [%[a]]!\n" + " vld1.32 {q6, q7}, [%[b]]!\n" + " vld1.32 {q10, q11}, [%[a]]!\n" + " subs %[len], %[len], #16\n" + " vmul.f32 q0, q4, q8\n" + " vmul.f32 q1, q5, q9\n" + " vmul.f32 q2, q6, q10\n" + " vmul.f32 q3, q7, q11\n" + " beq 3f\n" + "2:" + " vld1.32 {q4, q5}, [%[b]]!\n" + " vld1.32 {q8, q9}, [%[a]]!\n" + " vld1.32 {q6, q7}, [%[b]]!\n" + " vld1.32 {q10, q11}, [%[a]]!\n" + " subs %[len], %[len], #16\n" + " vmla.f32 q0, q4, q8\n" + " vmla.f32 q1, q5, q9\n" + " vmla.f32 q2, q6, q10\n" + " vmla.f32 q3, q7, q11\n" + " bne 2b\n" + "3:" + " vadd.f32 q4, q0, q1\n" + " vadd.f32 q5, q2, q3\n" + " cmp %[remainder], #0\n" + " vadd.f32 q0, q4, q5\n" + " beq 5f\n" + "4:" + " vld1.32 {q6}, [%[b]]!\n" + " vld1.32 {q10}, [%[a]]!\n" + " subs %[remainder], %[remainder], #4\n" + " vmla.f32 q0, q6, q10\n" + " bne 4b\n" + "5:" + " vadd.f32 d0, d0, d1\n" + " vpadd.f32 d0, d0, d0\n" + " vmov.f32 %[ret], d0[0]\n" + : [ret] "=&r" (ret), [a] "+r" (a), [b] "+r" (b), + [len] "+l" (len), [remainder] "+l" (remainder) + : + : "cc", "q0", "q1", "q2", "q3", "q4", "q5", "q6", "q7", "q8", + "q9", "q10", "q11"); + return ret; +} +#endif diff --git a/dep/cubeb/src/speex/resample_sse.h b/dep/cubeb/src/speex/resample_sse.h new file mode 100644 index 000000000..fed5b8276 --- /dev/null +++ b/dep/cubeb/src/speex/resample_sse.h @@ -0,0 +1,128 @@ +/* Copyright (C) 2007-2008 Jean-Marc Valin + * Copyright (C) 2008 Thorvald Natvig + */ +/** + @file resample_sse.h + @brief Resampler functions (SSE version) +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#include + +#define OVERRIDE_INNER_PRODUCT_SINGLE +static inline float inner_product_single(const float *a, const float *b, unsigned int len) +{ + int i; + float ret; + __m128 sum = _mm_setzero_ps(); + for (i=0;i +#define OVERRIDE_INNER_PRODUCT_DOUBLE + +static inline double inner_product_double(const float *a, const float *b, unsigned int len) +{ + int i; + double ret; + __m128d sum = _mm_setzero_pd(); + __m128 t; + for (i=0;i +# else +# ifdef HAVE_ALLOCA_H +# include +# else +# include +# endif +# endif +#endif + +/** + * @def ALIGN(stack, size) + * + * Aligns the stack to a 'size' boundary + * + * @param stack Stack + * @param size New size boundary + */ + +/** + * @def PUSH(stack, size, type) + * + * Allocates 'size' elements of type 'type' on the stack + * + * @param stack Stack + * @param size Number of elements + * @param type Type of element + */ + +/** + * @def VARDECL(var) + * + * Declare variable on stack + * + * @param var Variable to declare + */ + +/** + * @def ALLOC(var, size, type) + * + * Allocate 'size' elements of 'type' on stack + * + * @param var Name of variable to allocate + * @param size Number of elements + * @param type Type of element + */ + +#ifdef ENABLE_VALGRIND + +#include + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) + +#define PUSH(stack, size, type) (VALGRIND_MAKE_NOACCESS(stack, 1000),ALIGN((stack),sizeof(type)),VALGRIND_MAKE_WRITABLE(stack, ((size)*sizeof(type))),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type)))) + +#else + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) + +#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type)))) + +#endif + +#if defined(VAR_ARRAYS) +#define VARDECL(var) +#define ALLOC(var, size, type) type var[size] +#elif defined(USE_ALLOCA) +#define VARDECL(var) var +#define ALLOC(var, size, type) var = alloca(sizeof(type)*(size)) +#else +#define VARDECL(var) var +#define ALLOC(var, size, type) var = PUSH(stack, size, type) +#endif + + +#endif diff --git a/duckstation.sln b/duckstation.sln index 9681c0464..ea78763ad 100644 --- a/duckstation.sln +++ b/duckstation.sln @@ -27,6 +27,8 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "tinyxml2", "dep\tinyxml2\ti EndProject Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "duckstation-qt", "src\duckstation-qt\duckstation-qt.vcxproj", "{28F14272-0EC4-41BB-849F-182ADB81AF70}" EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "cubeb", "dep\cubeb\cubeb.vcxproj", "{72F9423C-91EE-4487-AAC6-555ED6F61AA1}" +EndProject Global GlobalSection(SolutionConfigurationPlatforms) = preSolution Debug|x64 = Debug|x64 @@ -211,6 +213,22 @@ Global {28F14272-0EC4-41BB-849F-182ADB81AF70}.ReleaseLTCG|x64.ActiveCfg = ReleaseLTCG|x64 {28F14272-0EC4-41BB-849F-182ADB81AF70}.ReleaseLTCG|x86.ActiveCfg = ReleaseLTCG|Win32 {28F14272-0EC4-41BB-849F-182ADB81AF70}.ReleaseLTCG|x86.Build.0 = ReleaseLTCG|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Debug|x64.ActiveCfg = Debug|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Debug|x64.Build.0 = Debug|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Debug|x86.ActiveCfg = Debug|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Debug|x86.Build.0 = Debug|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.DebugFast|x64.ActiveCfg = DebugFast|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.DebugFast|x64.Build.0 = DebugFast|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.DebugFast|x86.ActiveCfg = DebugFast|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.DebugFast|x86.Build.0 = DebugFast|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Release|x64.ActiveCfg = Release|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Release|x64.Build.0 = Release|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Release|x86.ActiveCfg = Release|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.Release|x86.Build.0 = Release|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.ReleaseLTCG|x64.ActiveCfg = ReleaseLTCG|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.ReleaseLTCG|x64.Build.0 = ReleaseLTCG|x64 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.ReleaseLTCG|x86.ActiveCfg = ReleaseLTCG|Win32 + {72F9423C-91EE-4487-AAC6-555ED6F61AA1}.ReleaseLTCG|x86.Build.0 = ReleaseLTCG|Win32 EndGlobalSection GlobalSection(SolutionProperties) = preSolution HideSolutionNode = FALSE @@ -223,6 +241,7 @@ Global {ACE32F47-2960-4FB3-9F77-2C375625BF61} = {BA490C0E-497D-4634-A21E-E65012006385} {3773F4CC-614E-4028-8595-22E08CA649E3} = {BA490C0E-497D-4634-A21E-E65012006385} {933118A9-68C5-47B4-B151-B03C93961623} = {BA490C0E-497D-4634-A21E-E65012006385} + {72F9423C-91EE-4487-AAC6-555ED6F61AA1} = {BA490C0E-497D-4634-A21E-E65012006385} EndGlobalSection GlobalSection(ExtensibilityGlobals) = postSolution SolutionGuid = {26E40B32-7C1D-48D0-95F4-1A500E054028}