mirror of
https://github.com/RetroDECK/Duckstation.git
synced 2024-11-25 07:05:40 +00:00
dep/cubeb: Sync to 19fcbef
And apply PR #740 (Re-enable and polish IAudioClient3 to achieve lower latencies). `*latency_frames = min_period;` in wasapi_get_min_latency was changed to `*latency_frames = hns_to_frames(params.rate, min_period_rt);`, as otherwise it reports in mixer frames, not stream frames.
This commit is contained in:
parent
c803c4fbef
commit
872cee908c
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@ -37,15 +37,15 @@ target_include_directories(cubeb
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PUBLIC $<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}/include> $<INSTALL_INTERFACE:include>
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)
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add_library(speex OBJECT subprojects/speex/resample.c)
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set_target_properties(speex PROPERTIES POSITION_INDEPENDENT_CODE TRUE)
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target_include_directories(speex INTERFACE subprojects)
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target_compile_definitions(speex PUBLIC
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OUTSIDE_SPEEX
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FLOATING_POINT
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EXPORT=
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RANDOM_PREFIX=speex
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)
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add_library(speex OBJECT subprojects/speex/resample.c)
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set_target_properties(speex PROPERTIES POSITION_INDEPENDENT_CODE TRUE)
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target_include_directories(speex INTERFACE subprojects)
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target_compile_definitions(speex PUBLIC
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OUTSIDE_SPEEX
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FLOATING_POINT
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EXPORT=
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RANDOM_PREFIX=speex
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)
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# $<BUILD_INTERFACE:> required because of https://gitlab.kitware.com/cmake/cmake/-/issues/15415
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target_link_libraries(cubeb PRIVATE $<BUILD_INTERFACE:speex>)
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@ -58,17 +58,20 @@ find_package(Threads)
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target_link_libraries(cubeb PRIVATE Threads::Threads)
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if(LAZY_LOAD_LIBS)
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if(NOT APPLE AND NOT WIN32)
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# Skip checks on MacOS because it takes ages in XCode.
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check_include_files(pulse/pulseaudio.h USE_PULSE)
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check_include_files(alsa/asoundlib.h USE_ALSA)
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check_include_files(jack/jack.h USE_JACK)
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check_include_files(sndio.h USE_SNDIO)
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if(NOT APPLE AND NOT WIN32) # Skip checks on MacOS because it takes ages in XCode.
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check_include_files(pulse/pulseaudio.h USE_PULSE)
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check_include_files(alsa/asoundlib.h USE_ALSA)
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check_include_files(jack/jack.h USE_JACK)
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check_include_files(sndio.h USE_SNDIO)
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if(USE_PULSE OR USE_ALSA OR USE_JACK OR USE_SNDIO)
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target_link_libraries(cubeb PRIVATE ${CMAKE_DL_LIBS})
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if(USE_PULSE OR USE_ALSA OR USE_JACK OR USE_SNDIO OR USE_AAUDIO)
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target_link_libraries(cubeb PRIVATE ${CMAKE_DL_LIBS})
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if(ANDROID)
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target_compile_definitions(cubeb PRIVATE __ANDROID_UNAVAILABLE_SYMBOLS_ARE_WEAK__)
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endif()
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endif()
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endif()
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elseif(NOT APPLE AND NOT WIN32)
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@ -123,56 +126,56 @@ if(USE_SNDIO)
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endif()
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if(APPLE)
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check_include_files(AudioUnit/AudioUnit.h USE_AUDIOUNIT)
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if(USE_AUDIOUNIT)
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target_sources(cubeb PRIVATE
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src/cubeb_audiounit.cpp
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src/cubeb_osx_run_loop.cpp)
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target_compile_definitions(cubeb PRIVATE USE_AUDIOUNIT)
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target_link_libraries(cubeb PRIVATE "-framework AudioUnit" "-framework CoreAudio" "-framework CoreServices")
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endif()
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check_include_files(AudioUnit/AudioUnit.h USE_AUDIOUNIT)
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if(USE_AUDIOUNIT)
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target_sources(cubeb PRIVATE
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src/cubeb_audiounit.cpp
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src/cubeb_osx_run_loop.cpp)
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target_compile_definitions(cubeb PRIVATE USE_AUDIOUNIT)
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target_link_libraries(cubeb PRIVATE "-framework AudioUnit" "-framework CoreAudio" "-framework CoreServices")
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endif()
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endif()
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if(WIN32)
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check_include_files(audioclient.h USE_WASAPI)
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if(USE_WASAPI)
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target_sources(cubeb PRIVATE
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src/cubeb_wasapi.cpp)
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target_compile_definitions(cubeb PRIVATE USE_WASAPI)
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target_link_libraries(cubeb PRIVATE avrt ole32 ksuser)
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endif()
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check_include_files(audioclient.h USE_WASAPI)
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if(USE_WASAPI)
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target_sources(cubeb PRIVATE
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src/cubeb_wasapi.cpp)
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target_compile_definitions(cubeb PRIVATE USE_WASAPI)
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target_link_libraries(cubeb PRIVATE ole32 ksuser)
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endif()
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check_include_files("windows.h;mmsystem.h" USE_WINMM)
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if(USE_WINMM)
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target_sources(cubeb PRIVATE
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src/cubeb_winmm.c)
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target_compile_definitions(cubeb PRIVATE USE_WINMM)
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target_link_libraries(cubeb PRIVATE winmm)
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endif()
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check_include_files("windows.h;mmsystem.h" USE_WINMM)
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if(USE_WINMM)
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target_sources(cubeb PRIVATE
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src/cubeb_winmm.c)
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target_compile_definitions(cubeb PRIVATE USE_WINMM)
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target_link_libraries(cubeb PRIVATE winmm)
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endif()
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endif()
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if(NOT WIN32 AND NOT APPLE)
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check_include_files(sys/soundcard.h HAVE_SYS_SOUNDCARD_H)
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if(HAVE_SYS_SOUNDCARD_H)
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try_compile(USE_OSS "${PROJECT_BINARY_DIR}/compile_tests"
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${PROJECT_SOURCE_DIR}/cmake/compile_tests/oss_is_v4.c)
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if(USE_OSS)
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# strlcpy is not available on BSD systems that use glibc,
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# like Debian kfreebsd, so try using libbsd if available
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include(CheckSymbolExists)
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check_symbol_exists(strlcpy string.h HAVE_STRLCPY)
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if(NOT HAVE_STRLCPY)
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pkg_check_modules(libbsd-overlay IMPORTED_TARGET libbsd-overlay)
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if(libbsd-overlay_FOUND)
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target_link_libraries(cubeb PRIVATE PkgConfig::libbsd-overlay)
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set(HAVE_STRLCPY true)
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endif()
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endif()
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if (HAVE_STRLCPY)
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target_sources(cubeb PRIVATE
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src/cubeb_oss.c)
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target_compile_definitions(cubeb PRIVATE USE_OSS)
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check_include_files(sys/soundcard.h HAVE_SYS_SOUNDCARD_H)
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if(HAVE_SYS_SOUNDCARD_H)
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try_compile(USE_OSS "${PROJECT_BINARY_DIR}/compile_tests"
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${PROJECT_SOURCE_DIR}/cmake/compile_tests/oss_is_v4.c)
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if(USE_OSS)
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# strlcpy is not available on BSD systems that use glibc,
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# like Debian kfreebsd, so try using libbsd if available
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include(CheckSymbolExists)
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check_symbol_exists(strlcpy string.h HAVE_STRLCPY)
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if(NOT HAVE_STRLCPY)
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pkg_check_modules(libbsd-overlay IMPORTED_TARGET libbsd-overlay)
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if(libbsd-overlay_FOUND)
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target_link_libraries(cubeb PRIVATE PkgConfig::libbsd-overlay)
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set(HAVE_STRLCPY true)
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endif()
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endif()
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if (HAVE_STRLCPY)
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target_sources(cubeb PRIVATE
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src/cubeb_oss.c)
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target_compile_definitions(cubeb PRIVATE USE_OSS)
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endif()
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endif()
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endif()
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endif()
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@ -258,11 +258,23 @@ typedef enum {
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the jack backend. */
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} cubeb_stream_prefs;
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/**
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* Input stream audio processing parameters. Only applicable with
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* CUBEB_STREAM_PREF_VOICE.
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*/
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typedef enum {
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CUBEB_INPUT_PROCESSING_PARAM_NONE = 0x00,
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CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION = 0x01,
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CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION = 0x02,
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CUBEB_INPUT_PROCESSING_PARAM_AUTOMATIC_GAIN_CONTROL = 0x04,
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CUBEB_INPUT_PROCESSING_PARAM_VOICE_ISOLATION = 0x08,
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} cubeb_input_processing_params;
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/** Stream format initialization parameters. */
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typedef struct {
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cubeb_sample_format format; /**< Requested sample format. One of
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#cubeb_sample_format. */
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uint32_t rate; /**< Requested sample rate. Valid range is [1000, 192000]. */
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uint32_t rate; /**< Requested sample rate. Valid range is [1000, 384000]. */
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uint32_t channels; /**< Requested channel count. Valid range is [1, 8]. */
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cubeb_channel_layout
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layout; /**< Requested channel layout. This must be consistent with the
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@ -519,6 +531,18 @@ cubeb_get_min_latency(cubeb * context, cubeb_stream_params * params,
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CUBEB_EXPORT int
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cubeb_get_preferred_sample_rate(cubeb * context, uint32_t * rate);
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/** Get the supported input processing features for this backend. See
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cubeb_stream_set_input_processing for how to set them for a particular input
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stream.
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@param context A pointer to the cubeb context.
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@param params Out parameter for the input processing params supported by
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this backend.
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@retval CUBEB_OK
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@retval CUBEB_ERROR_NOT_SUPPORTED */
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CUBEB_EXPORT int
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cubeb_get_supported_input_processing_params(
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cubeb * context, cubeb_input_processing_params * params);
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/** Destroy an application context. This must be called after all stream have
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* been destroyed.
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@param context A pointer to the cubeb context.*/
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@ -646,6 +670,30 @@ CUBEB_EXPORT int
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cubeb_stream_get_current_device(cubeb_stream * stm,
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cubeb_device ** const device);
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/** Set input mute state for this stream. Some platforms notify the user when an
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application is accessing audio input. When all inputs are muted they can
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prove to the user that the application is not actively capturing any input.
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@param stream the stream for which to set input mute state
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@param muted whether the input should mute or not
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@retval CUBEB_OK
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@retval CUBEB_ERROR_INVALID_PARAMETER if this stream does not have an input
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device
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@retval CUBEB_ERROR_NOT_SUPPORTED */
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CUBEB_EXPORT int
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cubeb_stream_set_input_mute(cubeb_stream * stream, int mute);
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/** Set what input processing features to enable for this stream.
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@param stream the stream for which to set input processing features.
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@param params what input processing features to use
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@retval CUBEB_OK
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@retval CUBEB_ERROR if params could not be applied
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@retval CUBEB_ERROR_INVALID_PARAMETER if a given param is not supported by
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this backend, or if this stream does not have an input device
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@retval CUBEB_ERROR_NOT_SUPPORTED */
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CUBEB_EXPORT int
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cubeb_stream_set_input_processing_params(cubeb_stream * stream,
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cubeb_input_processing_params params);
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/** Destroy a cubeb_device structure.
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@param stream the stream passed in cubeb_stream_get_current_device
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@param devices the devices to destroy
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@ -40,6 +40,8 @@ struct cubeb_ops {
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int (*get_min_latency)(cubeb * context, cubeb_stream_params params,
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uint32_t * latency_ms);
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int (*get_preferred_sample_rate)(cubeb * context, uint32_t * rate);
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int (*get_supported_input_processing_params)(
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cubeb * context, cubeb_input_processing_params * params);
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int (*enumerate_devices)(cubeb * context, cubeb_device_type type,
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cubeb_device_collection * collection);
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int (*device_collection_destroy)(cubeb * context,
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@ -62,6 +64,9 @@ struct cubeb_ops {
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int (*stream_set_name)(cubeb_stream * stream, char const * stream_name);
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int (*stream_get_current_device)(cubeb_stream * stream,
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cubeb_device ** const device);
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int (*stream_set_input_mute)(cubeb_stream * stream, int mute);
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int (*stream_set_input_processing_params)(
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cubeb_stream * stream, cubeb_input_processing_params params);
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int (*stream_device_destroy)(cubeb_stream * stream, cubeb_device * device);
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int (*stream_register_device_changed_callback)(
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cubeb_stream * stream,
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@ -67,7 +67,7 @@ validate_stream_params(cubeb_stream_params * input_stream_params,
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XASSERT(input_stream_params || output_stream_params);
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if (output_stream_params) {
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if (output_stream_params->rate < 1000 ||
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output_stream_params->rate > 192000 ||
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output_stream_params->rate > 768000 ||
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output_stream_params->channels < 1 ||
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output_stream_params->channels > UINT8_MAX) {
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return CUBEB_ERROR_INVALID_FORMAT;
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@ -75,7 +75,7 @@ validate_stream_params(cubeb_stream_params * input_stream_params,
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}
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if (input_stream_params) {
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if (input_stream_params->rate < 1000 ||
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input_stream_params->rate > 192000 ||
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input_stream_params->rate > 768000 ||
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input_stream_params->channels < 1 ||
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input_stream_params->channels > UINT8_MAX) {
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return CUBEB_ERROR_INVALID_FORMAT;
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@ -239,9 +239,6 @@ cubeb_get_backend_names()
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#if defined(USE_SNDIO)
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"sndio",
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#endif
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#if defined(USE_SUN)
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"sun",
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#endif
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#if defined(USE_OSS)
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"oss",
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#endif
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|
@ -304,6 +301,21 @@ cubeb_get_preferred_sample_rate(cubeb * context, uint32_t * rate)
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return context->ops->get_preferred_sample_rate(context, rate);
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}
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int
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cubeb_get_supported_input_processing_params(
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cubeb * context, cubeb_input_processing_params * params)
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{
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if (!context || !params) {
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return CUBEB_ERROR_INVALID_PARAMETER;
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}
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if (!context->ops->get_supported_input_processing_params) {
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return CUBEB_ERROR_NOT_SUPPORTED;
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}
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return context->ops->get_supported_input_processing_params(context, params);
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}
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void
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cubeb_destroy(cubeb * context)
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{
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@ -463,6 +475,36 @@ cubeb_stream_get_current_device(cubeb_stream * stream,
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return stream->context->ops->stream_get_current_device(stream, device);
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}
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int
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cubeb_stream_set_input_mute(cubeb_stream * stream, int mute)
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{
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if (!stream) {
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return CUBEB_ERROR_INVALID_PARAMETER;
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}
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if (!stream->context->ops->stream_set_input_mute) {
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return CUBEB_ERROR_NOT_SUPPORTED;
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}
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return stream->context->ops->stream_set_input_mute(stream, mute);
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}
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int
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cubeb_stream_set_input_processing_params(cubeb_stream * stream,
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cubeb_input_processing_params params)
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{
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if (!stream) {
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return CUBEB_ERROR_INVALID_PARAMETER;
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}
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if (!stream->context->ops->stream_set_input_processing_params) {
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return CUBEB_ERROR_NOT_SUPPORTED;
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}
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return stream->context->ops->stream_set_input_processing_params(stream,
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params);
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}
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int
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cubeb_stream_device_destroy(cubeb_stream * stream, cubeb_device * device)
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{
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|
|
|
@ -1472,6 +1472,7 @@ static struct cubeb_ops const alsa_ops = {
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.get_max_channel_count = alsa_get_max_channel_count,
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.get_min_latency = alsa_get_min_latency,
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.get_preferred_sample_rate = alsa_get_preferred_sample_rate,
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.get_supported_input_processing_params = NULL,
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.enumerate_devices = alsa_enumerate_devices,
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.device_collection_destroy = alsa_device_collection_destroy,
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.destroy = alsa_destroy,
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@ -1485,6 +1486,8 @@ static struct cubeb_ops const alsa_ops = {
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.stream_set_volume = alsa_stream_set_volume,
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.stream_set_name = NULL,
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.stream_get_current_device = NULL,
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.stream_set_input_mute = NULL,
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.stream_set_input_processing_params = NULL,
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.stream_device_destroy = NULL,
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.stream_register_device_changed_callback = NULL,
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.register_device_collection_changed = NULL};
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|
|
|
@ -377,6 +377,15 @@ cubeb_channel_to_channel_label(cubeb_channel channel)
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}
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}
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||||
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bool
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is_common_sample_rate(Float64 sample_rate)
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||||
{
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||||
/* Some commonly used sample rates and their multiples and divisors. */
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return sample_rate == 8000 || sample_rate == 16000 || sample_rate == 22050 ||
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sample_rate == 32000 || sample_rate == 44100 || sample_rate == 48000 ||
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sample_rate == 88200 || sample_rate == 96000;
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}
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||||
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#if TARGET_OS_IPHONE
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typedef UInt32 AudioDeviceID;
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typedef UInt32 AudioObjectID;
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@ -2502,6 +2511,12 @@ audiounit_configure_output(cubeb_stream * stm)
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return CUBEB_ERROR;
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}
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stm->output_hw_rate = output_hw_desc.mSampleRate;
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if (!is_common_sample_rate(stm->output_desc.mSampleRate)) {
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/* For uncommon sample rates, we may run into issues with the OS
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resampler if we don't do the resampling ourselves, so set the
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AudioUnit sample rate to the hardware rate and resample. */
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stm->output_desc.mSampleRate = stm->output_hw_rate;
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}
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LOG("(%p) Output device sampling rate: %.2f", stm,
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output_hw_desc.mSampleRate);
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stm->context->channels = output_hw_desc.mChannelsPerFrame;
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@ -2709,11 +2724,16 @@ audiounit_setup_stream(cubeb_stream * stm)
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input_unconverted_params.rate = stm->input_hw_rate;
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}
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||||
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/* Create resampler. Output params are unchanged
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||||
* because we do not need conversion on the output. */
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cubeb_stream_params output_unconverted_params;
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if (has_output(stm)) {
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output_unconverted_params = stm->output_stream_params;
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||||
output_unconverted_params.rate = stm->output_desc.mSampleRate;
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||||
}
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||||
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/* Create resampler. */
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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,
|
||||
has_output(stm) ? &output_unconverted_params : NULL, target_sample_rate,
|
||||
stm->data_callback, stm->user_ptr, CUBEB_RESAMPLER_QUALITY_DESKTOP,
|
||||
CUBEB_RESAMPLER_RECLOCK_NONE));
|
||||
if (!stm->resampler) {
|
||||
|
@ -3665,6 +3685,7 @@ cubeb_ops const audiounit_ops = {
|
|||
/*.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,
|
||||
/*.get_supported_input_processing_params =*/NULL,
|
||||
/*.enumerate_devices =*/audiounit_enumerate_devices,
|
||||
/*.device_collection_destroy =*/audiounit_device_collection_destroy,
|
||||
/*.destroy =*/audiounit_destroy,
|
||||
|
@ -3678,6 +3699,8 @@ cubeb_ops const audiounit_ops = {
|
|||
/*.stream_set_volume =*/audiounit_stream_set_volume,
|
||||
/*.stream_set_name =*/NULL,
|
||||
/*.stream_get_current_device =*/audiounit_stream_get_current_device,
|
||||
/*.stream_set_input_mute =*/NULL,
|
||||
/*.stream_set_input_processing_params =*/NULL,
|
||||
/*.stream_device_destroy =*/audiounit_stream_device_destroy,
|
||||
/*.stream_register_device_changed_callback =*/
|
||||
audiounit_stream_register_device_changed_callback,
|
||||
|
|
|
@ -160,6 +160,7 @@ static struct cubeb_ops const cbjack_ops = {
|
|||
.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,
|
||||
.get_supported_input_processing_params = NULL,
|
||||
.enumerate_devices = cbjack_enumerate_devices,
|
||||
.device_collection_destroy = cbjack_device_collection_destroy,
|
||||
.destroy = cbjack_destroy,
|
||||
|
@ -173,6 +174,8 @@ static struct cubeb_ops const cbjack_ops = {
|
|||
.stream_set_volume = cbjack_stream_set_volume,
|
||||
.stream_set_name = NULL,
|
||||
.stream_get_current_device = cbjack_stream_get_current_device,
|
||||
.stream_set_input_mute = NULL,
|
||||
.stream_set_input_processing_params = NULL,
|
||||
.stream_device_destroy = cbjack_stream_device_destroy,
|
||||
.stream_register_device_changed_callback = NULL,
|
||||
.register_device_collection_changed = NULL};
|
||||
|
@ -431,8 +434,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
|
|||
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 (buffer_out) {
|
||||
for (long f = 0; f < nframes; f++) {
|
||||
buffer_out[f] = 0.f;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
@ -440,8 +445,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
|
|||
// 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;
|
||||
if (buffer_in) {
|
||||
for (long f = 0; f < nframes; f++) {
|
||||
buffer_in[f] = 0.f;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
@ -493,8 +500,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
|
|||
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 (buffer_out) {
|
||||
for (long f = 0; f < nframes; f++) {
|
||||
buffer_out[f] = 0.f;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
@ -502,8 +511,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
|
|||
// 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;
|
||||
if (buffer_in) {
|
||||
for (long f = 0; f < nframes; f++) {
|
||||
buffer_in[f] = 0.f;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
@ -542,20 +553,26 @@ cbjack_deinterleave_playback_refill_float(cubeb_stream * stream, float ** in,
|
|||
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 (buffer) {
|
||||
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 (buffer) {
|
||||
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 (buffer) {
|
||||
buffer[f] = 0.f;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
|
@ -1335,6 +1335,7 @@ static struct cubeb_ops const oss_ops = {
|
|||
.get_max_channel_count = oss_get_max_channel_count,
|
||||
.get_min_latency = oss_get_min_latency,
|
||||
.get_preferred_sample_rate = oss_get_preferred_sample_rate,
|
||||
.get_supported_input_processing_params = NULL,
|
||||
.enumerate_devices = oss_enumerate_devices,
|
||||
.device_collection_destroy = oss_device_collection_destroy,
|
||||
.destroy = oss_destroy,
|
||||
|
@ -1348,6 +1349,8 @@ static struct cubeb_ops const oss_ops = {
|
|||
.stream_set_volume = oss_stream_set_volume,
|
||||
.stream_set_name = NULL,
|
||||
.stream_get_current_device = oss_get_current_device,
|
||||
.stream_set_input_mute = NULL,
|
||||
.stream_set_input_processing_params = NULL,
|
||||
.stream_device_destroy = oss_stream_device_destroy,
|
||||
.stream_register_device_changed_callback = NULL,
|
||||
.register_device_collection_changed = NULL};
|
||||
|
|
|
@ -1690,6 +1690,7 @@ static struct cubeb_ops const pulse_ops = {
|
|||
.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,
|
||||
.get_supported_input_processing_params = NULL,
|
||||
.enumerate_devices = pulse_enumerate_devices,
|
||||
.device_collection_destroy = pulse_device_collection_destroy,
|
||||
.destroy = pulse_destroy,
|
||||
|
@ -1703,6 +1704,8 @@ static struct cubeb_ops const pulse_ops = {
|
|||
.stream_set_volume = pulse_stream_set_volume,
|
||||
.stream_set_name = pulse_stream_set_name,
|
||||
.stream_get_current_device = pulse_stream_get_current_device,
|
||||
.stream_set_input_mute = NULL,
|
||||
.stream_set_input_processing_params = NULL,
|
||||
.stream_device_destroy = pulse_stream_device_destroy,
|
||||
.stream_register_device_changed_callback = NULL,
|
||||
.register_device_collection_changed =
|
||||
|
|
|
@ -667,6 +667,7 @@ static struct cubeb_ops const sndio_ops = {
|
|||
.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,
|
||||
.get_supported_input_processing_params = NULL,
|
||||
.enumerate_devices = sndio_enumerate_devices,
|
||||
.device_collection_destroy = sndio_device_collection_destroy,
|
||||
.destroy = sndio_destroy,
|
||||
|
@ -679,6 +680,8 @@ static struct cubeb_ops const sndio_ops = {
|
|||
.stream_set_volume = sndio_stream_set_volume,
|
||||
.stream_set_name = NULL,
|
||||
.stream_get_current_device = NULL,
|
||||
.stream_set_input_mute = NULL,
|
||||
.stream_set_input_processing_params = NULL,
|
||||
.stream_device_destroy = NULL,
|
||||
.stream_register_device_changed_callback = NULL,
|
||||
.register_device_collection_changed = NULL};
|
||||
|
|
|
@ -42,6 +42,13 @@ public:
|
|||
{
|
||||
return (shared_state.load(std::memory_order_relaxed) & BACK_DIRTY_BIT) != 0;
|
||||
}
|
||||
// Reset state and indices to initial values.
|
||||
void invalidate()
|
||||
{
|
||||
shared_state.store(0, std::memory_order_release);
|
||||
input_idx = 1;
|
||||
output_idx = 2;
|
||||
}
|
||||
|
||||
private:
|
||||
// Publish a value to the consumer. Returns true if the data was overwritten
|
||||
|
|
|
@ -182,7 +182,9 @@ public:
|
|||
if (length_ + length > capacity_) {
|
||||
reserve(length_ + length);
|
||||
}
|
||||
PodCopy(data_ + length_, elements, length);
|
||||
if (data_) {
|
||||
PodCopy(data_ + length_, elements, length);
|
||||
}
|
||||
length_ += length;
|
||||
}
|
||||
|
||||
|
@ -195,12 +197,14 @@ public:
|
|||
if (length_ + length > capacity_) {
|
||||
reserve(length + length_);
|
||||
}
|
||||
PodZero(data_ + length_, length);
|
||||
if (data_) {
|
||||
PodZero(data_ + length_, length);
|
||||
}
|
||||
length_ += length;
|
||||
}
|
||||
|
||||
/** Prepend `length` zero-ed elements to the end of the array, resizing the
|
||||
* array if needed.
|
||||
/** Prepend `length` zero-ed elements to the front of the array, resizing and
|
||||
* shifting the array if needed.
|
||||
* @parameter length the number of elements to prepend to the array.
|
||||
*/
|
||||
void push_front_silence(size_t length)
|
||||
|
@ -208,8 +212,10 @@ public:
|
|||
if (length_ + length > capacity_) {
|
||||
reserve(length + length_);
|
||||
}
|
||||
PodMove(data_ + length, data_, length_);
|
||||
PodZero(data_, length);
|
||||
if (data_) {
|
||||
PodMove(data_ + length, data_, length_);
|
||||
PodZero(data_, length);
|
||||
}
|
||||
length_ += length;
|
||||
}
|
||||
|
||||
|
@ -227,6 +233,9 @@ public:
|
|||
if (length > length_) {
|
||||
return false;
|
||||
}
|
||||
if (!data_) {
|
||||
return true;
|
||||
}
|
||||
if (elements) {
|
||||
PodCopy(elements, data_, length);
|
||||
}
|
||||
|
|
|
@ -37,6 +37,31 @@
|
|||
#include "cubeb_tracing.h"
|
||||
#include "cubeb_utils.h"
|
||||
|
||||
// Some people have reported glitches with IAudioClient3 capture streams:
|
||||
// http://blog.nirbheek.in/2018/03/low-latency-audio-on-windows-with.html
|
||||
// https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
|
||||
#define ALLOW_AUDIO_CLIENT_3_FOR_INPUT 0
|
||||
// IAudioClient3::GetSharedModeEnginePeriod() seem to return min latencies
|
||||
// bigger than IAudioClient::GetDevicePeriod(), which is confusing (10ms vs
|
||||
// 3ms), though the default latency is usually the same and we should use the
|
||||
// IAudioClient3 function anyway, as it's more correct
|
||||
#define USE_AUDIO_CLIENT_3_MIN_PERIOD 1
|
||||
// If this is true, we allow IAudioClient3 the creation of sessions with a
|
||||
// latency above the default one (usually 10ms).
|
||||
// Whether we should default this to true or false depend on many things:
|
||||
// -Does creating a shared IAudioClient3 session (not locked to a format)
|
||||
// actually forces all the IAudioClient(1) sessions to have the same latency?
|
||||
// I could find no proof of that.
|
||||
// -Does creating a shared IAudioClient3 session with a latency >= the default
|
||||
// one actually improve the latency (as in how late the audio is) at all?
|
||||
// -Maybe we could expose this as cubeb stream pref
|
||||
// (e.g. take priority over other apps)?
|
||||
#define ALLOW_AUDIO_CLIENT_3_LATENCY_OVER_DEFAULT 1
|
||||
// If this is true and the user specified a target latency >= the IAudioClient3
|
||||
// max one, then we reject it and fall back to IAudioClient(1). There wouldn't
|
||||
// be much point in having a low latency if that's not what the user wants.
|
||||
#define REJECT_AUDIO_CLIENT_3_LATENCY_OVER_MAX 0
|
||||
|
||||
// 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
|
||||
|
@ -204,6 +229,11 @@ struct auto_stream_ref {
|
|||
cubeb_stream * stm;
|
||||
};
|
||||
|
||||
using set_mm_thread_characteristics_function =
|
||||
decltype(&AvSetMmThreadCharacteristicsW);
|
||||
using revert_mm_thread_characteristics_function =
|
||||
decltype(&AvRevertMmThreadCharacteristics);
|
||||
|
||||
extern cubeb_ops const wasapi_ops;
|
||||
|
||||
static com_heap_ptr<wchar_t>
|
||||
|
@ -301,6 +331,13 @@ struct cubeb {
|
|||
nullptr;
|
||||
void * output_collection_changed_user_ptr = nullptr;
|
||||
UINT64 performance_counter_frequency;
|
||||
/* Library dynamically opened to increase the render thread priority, and
|
||||
the two function pointers we need. */
|
||||
HMODULE mmcss_module = nullptr;
|
||||
set_mm_thread_characteristics_function set_mm_thread_characteristics =
|
||||
nullptr;
|
||||
revert_mm_thread_characteristics_function revert_mm_thread_characteristics =
|
||||
nullptr;
|
||||
};
|
||||
|
||||
class wasapi_endpoint_notification_client;
|
||||
|
@ -1350,10 +1387,8 @@ refill_callback_output(cubeb_stream * stm)
|
|||
|
||||
long got = refill(stm, nullptr, 0, output_buffer, output_frames);
|
||||
|
||||
if (got != output_frames) {
|
||||
ALOGV("Output callback: output frames requested: %Iu, got %ld", output_frames,
|
||||
got);
|
||||
}
|
||||
ALOGV("Output callback: output frames requested: %Iu, got %ld", output_frames,
|
||||
got);
|
||||
if (got < 0) {
|
||||
return false;
|
||||
}
|
||||
|
@ -1403,7 +1438,8 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
|
|||
|
||||
/* We could consider using "Pro Audio" here for WebAudio and
|
||||
maybe WebRTC. */
|
||||
mmcss_handle = AvSetMmThreadCharacteristicsA("Audio", &mmcss_task_index);
|
||||
mmcss_handle =
|
||||
stm->context->set_mm_thread_characteristics(L"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",
|
||||
|
@ -1511,7 +1547,7 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
|
|||
}
|
||||
|
||||
if (mmcss_handle) {
|
||||
AvRevertMmThreadCharacteristics(mmcss_handle);
|
||||
stm->context->revert_mm_thread_characteristics(mmcss_handle);
|
||||
}
|
||||
|
||||
if (FAILED(hr)) {
|
||||
|
@ -1524,6 +1560,18 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
|
|||
void
|
||||
wasapi_destroy(cubeb * context);
|
||||
|
||||
HANDLE WINAPI
|
||||
set_mm_thread_characteristics_noop(LPCWSTR, LPDWORD mmcss_task_index)
|
||||
{
|
||||
return (HANDLE)1;
|
||||
}
|
||||
|
||||
BOOL WINAPI
|
||||
revert_mm_thread_characteristics_noop(HANDLE mmcss_handle)
|
||||
{
|
||||
return true;
|
||||
}
|
||||
|
||||
HRESULT
|
||||
register_notification_client(cubeb_stream * stm)
|
||||
{
|
||||
|
@ -1759,6 +1807,31 @@ wasapi_init(cubeb ** context, char const * context_name)
|
|||
ctx->performance_counter_frequency = 0;
|
||||
}
|
||||
|
||||
ctx->mmcss_module = LoadLibraryW(L"Avrt.dll");
|
||||
|
||||
bool success = false;
|
||||
if (ctx->mmcss_module) {
|
||||
ctx->set_mm_thread_characteristics =
|
||||
reinterpret_cast<set_mm_thread_characteristics_function>(
|
||||
GetProcAddress(ctx->mmcss_module, "AvSetMmThreadCharacteristicsW"));
|
||||
ctx->revert_mm_thread_characteristics =
|
||||
reinterpret_cast<revert_mm_thread_characteristics_function>(
|
||||
GetProcAddress(ctx->mmcss_module,
|
||||
"AvRevertMmThreadCharacteristics"));
|
||||
success = ctx->set_mm_thread_characteristics &&
|
||||
ctx->revert_mm_thread_characteristics;
|
||||
}
|
||||
if (!success) {
|
||||
// This is not a fatal error, but we might end up glitching when
|
||||
// the system is under high load.
|
||||
LOG("Could not load avrt.dll or fetch AvSetMmThreadCharacteristicsW "
|
||||
"AvRevertMmThreadCharacteristics: %lx",
|
||||
GetLastError());
|
||||
ctx->set_mm_thread_characteristics = &set_mm_thread_characteristics_noop;
|
||||
ctx->revert_mm_thread_characteristics =
|
||||
&revert_mm_thread_characteristics_noop;
|
||||
}
|
||||
|
||||
*context = ctx;
|
||||
|
||||
return CUBEB_OK;
|
||||
|
@ -1815,6 +1888,10 @@ wasapi_destroy(cubeb * context)
|
|||
}
|
||||
}
|
||||
|
||||
if (context->mmcss_module) {
|
||||
FreeLibrary(context->mmcss_module);
|
||||
}
|
||||
|
||||
delete context;
|
||||
}
|
||||
|
||||
|
@ -1872,6 +1949,44 @@ wasapi_get_min_latency(cubeb * ctx, cubeb_stream_params params,
|
|||
return CUBEB_ERROR;
|
||||
}
|
||||
|
||||
#if USE_AUDIO_CLIENT_3_MIN_PERIOD
|
||||
// This is unreliable as we can't know the actual mixer format cubeb will
|
||||
// ask for later on (nor we can branch on ALLOW_AUDIO_CLIENT_3_FOR_INPUT),
|
||||
// and the min latency can change based on that.
|
||||
com_ptr<IAudioClient3> client3;
|
||||
hr = device->Activate(__uuidof(IAudioClient3), CLSCTX_INPROC_SERVER, NULL,
|
||||
client3.receive_vpp());
|
||||
if (SUCCEEDED(hr)) {
|
||||
WAVEFORMATEX * mix_format = nullptr;
|
||||
hr = client3->GetMixFormat(&mix_format);
|
||||
|
||||
if (SUCCEEDED(hr)) {
|
||||
uint32_t default_period = 0, fundamental_period = 0, min_period = 0,
|
||||
max_period = 0;
|
||||
hr = client3->GetSharedModeEnginePeriod(mix_format, &default_period,
|
||||
&fundamental_period, &min_period,
|
||||
&max_period);
|
||||
|
||||
auto sample_rate = mix_format->nSamplesPerSec;
|
||||
CoTaskMemFree(mix_format);
|
||||
if (SUCCEEDED(hr)) {
|
||||
// Print values in the same format as IAudioDevice::GetDevicePeriod()
|
||||
REFERENCE_TIME min_period_rt(frames_to_hns(sample_rate, min_period));
|
||||
REFERENCE_TIME default_period_rt(
|
||||
frames_to_hns(sample_rate, default_period));
|
||||
LOG("default device period: %I64d, minimum device period: %I64d",
|
||||
default_period_rt, min_period_rt);
|
||||
|
||||
*latency_frames = hns_to_frames(params.rate, min_period_rt);
|
||||
|
||||
LOG("Minimum latency in frames: %u", *latency_frames);
|
||||
|
||||
return CUBEB_OK;
|
||||
}
|
||||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
com_ptr<IAudioClient> client;
|
||||
hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
|
||||
client.receive_vpp());
|
||||
|
@ -1891,18 +2006,8 @@ wasapi_get_min_latency(cubeb * ctx, cubeb_stream_params params,
|
|||
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
|
||||
#if 0
|
||||
*latency_frames = hns_to_frames(params.rate, minimum_period);
|
||||
#else
|
||||
// The minimum_period is only relevant in exclusive streams.
|
||||
*latency_frames = hns_to_frames(params.rate, default_period);
|
||||
#endif
|
||||
|
||||
LOG("Minimum latency in frames: %u", *latency_frames);
|
||||
|
||||
|
@ -1992,7 +2097,10 @@ handle_channel_layout(cubeb_stream * stm, EDataFlow direction,
|
|||
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. */
|
||||
the suggestion, since it seems to always be IEEE_FLOAT.
|
||||
This fallback doesn't update the bit depth, so if a device
|
||||
only supported bit depths cubeb doesn't support, so IAudioClient3
|
||||
streams might fail */
|
||||
LOG("Using WASAPI suggested format: channels: %d", closest->nChannels);
|
||||
XASSERT(closest->wFormatTag == WAVE_FORMAT_EXTENSIBLE);
|
||||
WAVEFORMATEXTENSIBLE * closest_pcm =
|
||||
|
@ -2036,12 +2144,12 @@ initialize_iaudioclient2(com_ptr<IAudioClient> & audio_client)
|
|||
return CUBEB_OK;
|
||||
}
|
||||
|
||||
#if 0
|
||||
bool
|
||||
initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
|
||||
cubeb_stream * stm,
|
||||
const com_heap_ptr<WAVEFORMATEX> & mix_format,
|
||||
DWORD flags, EDataFlow direction)
|
||||
DWORD flags, EDataFlow direction,
|
||||
REFERENCE_TIME latency_hns)
|
||||
{
|
||||
com_ptr<IAudioClient3> audio_client3;
|
||||
audio_client->QueryInterface<IAudioClient3>(audio_client3.receive());
|
||||
|
@ -2057,24 +2165,22 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
|
|||
return false;
|
||||
}
|
||||
|
||||
// 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.
|
||||
// might 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, if it created with the
|
||||
// AUDCLNT_STREAMOPTIONS_MATCH_FORMAT option, the audio engine would be
|
||||
// locked to that format, so we have to match it (a custom one would fail).
|
||||
// - We don't lock the WASAPI engine to a format, as it's antisocial towards
|
||||
// other apps, especially if we locked to a latency >= than its default.
|
||||
// - If the user requested latency is >= the default one, we might still
|
||||
// accept it (without locking the format) depending on
|
||||
// ALLOW_AUDIO_CLIENT_3_LATENCY_OVER_DEFAULT, as we might want to prioritize
|
||||
// to lower our latency over other apps
|
||||
// (there might still be latency advantages compared to IAudioDevice(1)).
|
||||
|
||||
HRESULT hr;
|
||||
uint32_t default_period = 0, fundamental_period = 0, min_period = 0,
|
||||
|
@ -2086,28 +2192,59 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
|
|||
LOG("Could not get shared mode engine period: error: %lx", hr);
|
||||
return false;
|
||||
}
|
||||
uint32_t requested_latency = stm->latency;
|
||||
uint32_t requested_latency =
|
||||
hns_to_frames(mix_format->nSamplesPerSec, latency_hns);
|
||||
#if !ALLOW_AUDIO_CLIENT_3_LATENCY_OVER_DEFAULT
|
||||
if (requested_latency >= default_period) {
|
||||
LOG("Requested latency %i greater than default latency %i, not using "
|
||||
"IAudioClient3",
|
||||
LOG("Requested latency %i equal or greater than default latency %i,"
|
||||
" not using IAudioClient3",
|
||||
requested_latency, default_period);
|
||||
return false;
|
||||
}
|
||||
#elif REJECT_AUDIO_CLIENT_3_LATENCY_OVER_MAX
|
||||
if (requested_latency > max_period) {
|
||||
// Fallback to IAudioClient(1) as it's more accepting of large latencies
|
||||
LOG("Requested latency %i greater than max latency %i,"
|
||||
" not using IAudioClient3",
|
||||
requested_latency, max_period);
|
||||
return false;
|
||||
}
|
||||
#endif
|
||||
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;
|
||||
// The period is required to be a multiple of the fundamental period
|
||||
// (and >= min and <= max, which should still be true)
|
||||
requested_latency -= requested_latency % fundamental_period;
|
||||
if (requested_latency < min_period) {
|
||||
requested_latency = min_period;
|
||||
}
|
||||
requested_latency -= (requested_latency - min_period) % fundamental_period;
|
||||
// Likely unnecessary, but won't hurt
|
||||
if (requested_latency > max_period) {
|
||||
requested_latency = max_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,
|
||||
DWORD new_flags = flags;
|
||||
// Always add these flags to IAudioClient3, they might help
|
||||
// if the stream doesn't have the same format as the audio engine.
|
||||
new_flags |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
|
||||
new_flags |= AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
|
||||
|
||||
hr = audio_client3->InitializeSharedAudioStream(new_flags, requested_latency,
|
||||
mix_format.get(), NULL);
|
||||
// This error should be returned first even if
|
||||
// the period was locked (AUDCLNT_E_ENGINE_PERIODICITY_LOCKED)
|
||||
if (hr == AUDCLNT_E_INVALID_STREAM_FLAG) {
|
||||
LOG("Got AUDCLNT_E_INVALID_STREAM_FLAG, removing some flags");
|
||||
hr = audio_client3->InitializeSharedAudioStream(flags, requested_latency,
|
||||
mix_format.get(), NULL);
|
||||
}
|
||||
|
||||
if (SUCCEEDED(hr)) {
|
||||
return true;
|
||||
} else if (hr == AUDCLNT_E_ENGINE_PERIODICITY_LOCKED) {
|
||||
|
@ -2119,23 +2256,38 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
|
|||
}
|
||||
|
||||
uint32_t current_period = 0;
|
||||
WAVEFORMATEX * current_format = nullptr;
|
||||
WAVEFORMATEX * current_format_ptr = nullptr;
|
||||
// We have to pass a valid WAVEFORMATEX** and not nullptr, otherwise
|
||||
// GetCurrentSharedModeEnginePeriod will return E_POINTER
|
||||
hr = audio_client3->GetCurrentSharedModeEnginePeriod(¤t_format,
|
||||
hr = audio_client3->GetCurrentSharedModeEnginePeriod(¤t_format_ptr,
|
||||
¤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);
|
||||
com_heap_ptr<WAVEFORMATEX> current_format(current_format_ptr);
|
||||
if (current_format->nSamplesPerSec != mix_format->nSamplesPerSec) {
|
||||
// Unless some other external app locked the shared mode engine period
|
||||
// within our audio initialization, this is unlikely to happen, though we
|
||||
// can't respect the user selected latency, so we fallback on IAudioClient
|
||||
LOG("IAudioClient3::GetCurrentSharedModeEnginePeriod() returned a "
|
||||
"different mixer format (nSamplesPerSec) from "
|
||||
"IAudioClient::GetMixFormat(); not using IAudioClient3");
|
||||
return false;
|
||||
}
|
||||
|
||||
#if REJECT_AUDIO_CLIENT_3_LATENCY_OVER_MAX
|
||||
// Reject IAudioClient3 if we can't respect the user target latency.
|
||||
// We don't need to check against default_latency anymore,
|
||||
// as the current_period is already the best one we could get.
|
||||
if (old_requested_latency > current_period) {
|
||||
LOG("Requested latency %i greater than currently locked shared mode "
|
||||
"latency %i, not using IAudioClient3",
|
||||
old_requested_latency, current_period);
|
||||
return false;
|
||||
}
|
||||
#endif
|
||||
|
||||
hr = audio_client3->InitializeSharedAudioStream(flags, current_period,
|
||||
mix_format.get(), NULL);
|
||||
if (SUCCEEDED(hr)) {
|
||||
|
@ -2147,7 +2299,6 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
|
|||
LOG("Could not initialize shared stream with IAudioClient3: error: %lx", hr);
|
||||
return false;
|
||||
}
|
||||
#endif
|
||||
|
||||
#define DIRECTION_NAME (direction == eCapture ? "capture" : "render")
|
||||
|
||||
|
@ -2171,6 +2322,12 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
|
|||
return CUBEB_ERROR;
|
||||
}
|
||||
|
||||
#if ALLOW_AUDIO_CLIENT_3_FOR_INPUT
|
||||
constexpr bool allow_audio_client_3 = true;
|
||||
#else
|
||||
const bool allow_audio_client_3 = direction == eRender;
|
||||
#endif
|
||||
|
||||
stm->stream_reset_lock.assert_current_thread_owns();
|
||||
// If user doesn't specify a particular device, we can choose another one when
|
||||
// the given devid is unavailable.
|
||||
|
@ -2207,17 +2364,14 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
|
|||
|
||||
/* Get a client. We will get all other interfaces we need from
|
||||
* this pointer. */
|
||||
#if 0 // See https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
|
||||
hr = device->Activate(__uuidof(IAudioClient3),
|
||||
CLSCTX_INPROC_SERVER,
|
||||
NULL, audio_client.receive_vpp());
|
||||
if (hr == E_NOINTERFACE) {
|
||||
#endif
|
||||
hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
|
||||
audio_client.receive_vpp());
|
||||
#if 0
|
||||
if (allow_audio_client_3) {
|
||||
hr = device->Activate(__uuidof(IAudioClient3), CLSCTX_INPROC_SERVER, NULL,
|
||||
audio_client.receive_vpp());
|
||||
}
|
||||
if (!allow_audio_client_3 || hr == E_NOINTERFACE) {
|
||||
hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
|
||||
audio_client.receive_vpp());
|
||||
}
|
||||
#endif
|
||||
|
||||
if (FAILED(hr)) {
|
||||
LOG("Could not activate the device to get an audio"
|
||||
|
@ -2346,16 +2500,15 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
|
|||
}
|
||||
}
|
||||
|
||||
#if 0 // See https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
|
||||
if (initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction)) {
|
||||
if (allow_audio_client_3 &&
|
||||
initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction,
|
||||
latency_hns)) {
|
||||
LOG("Initialized with IAudioClient3");
|
||||
} else {
|
||||
#endif
|
||||
hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, latency_hns, 0,
|
||||
mix_format.get(), NULL);
|
||||
#if 0
|
||||
hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, latency_hns,
|
||||
0, mix_format.get(), NULL);
|
||||
}
|
||||
#endif
|
||||
|
||||
if (FAILED(hr)) {
|
||||
LOG("Unable to initialize audio client for %s: %lx.", DIRECTION_NAME, hr);
|
||||
return CUBEB_ERROR;
|
||||
|
@ -3351,6 +3504,7 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
|
|||
CUBEB_DEVICE_FMT_S16NE);
|
||||
ret.default_format = CUBEB_DEVICE_FMT_F32NE;
|
||||
prop_variant fmtvar;
|
||||
WAVEFORMATEX * wfx = NULL;
|
||||
hr = propstore->GetValue(PKEY_AudioEngine_DeviceFormat, &fmtvar);
|
||||
if (SUCCEEDED(hr) && fmtvar.vt == VT_BLOB) {
|
||||
if (fmtvar.blob.cbSize == sizeof(PCMWAVEFORMAT)) {
|
||||
|
@ -3360,8 +3514,7 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
|
|||
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<WAVEFORMATEX *>(fmtvar.blob.pBlobData);
|
||||
wfx = reinterpret_cast<WAVEFORMATEX *>(fmtvar.blob.pBlobData);
|
||||
|
||||
if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX) + wfx->cbSize ||
|
||||
wfx->wFormatTag == WAVE_FORMAT_PCM) {
|
||||
|
@ -3371,9 +3524,30 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
|
|||
}
|
||||
}
|
||||
|
||||
if (SUCCEEDED(dev->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER,
|
||||
NULL, client.receive_vpp())) &&
|
||||
SUCCEEDED(client->GetDevicePeriod(&def_period, &min_period))) {
|
||||
#if USE_AUDIO_CLIENT_3_MIN_PERIOD
|
||||
// Here we assume an IAudioClient3 stream will successfully
|
||||
// be initialized later (it might fail)
|
||||
#if ALLOW_AUDIO_CLIENT_3_FOR_INPUT
|
||||
constexpr bool allow_audio_client_3 = true;
|
||||
#else
|
||||
const bool allow_audio_client_3 = flow == eRender;
|
||||
#endif
|
||||
com_ptr<IAudioClient3> client3;
|
||||
uint32_t def, fun, min, max;
|
||||
if (allow_audio_client_3 && wfx &&
|
||||
SUCCEEDED(dev->Activate(__uuidof(IAudioClient3), CLSCTX_INPROC_SERVER,
|
||||
NULL, client3.receive_vpp())) &&
|
||||
SUCCEEDED(
|
||||
client3->GetSharedModeEnginePeriod(wfx, &def, &fun, &min, &max))) {
|
||||
ret.latency_lo = min;
|
||||
// This latency might actually be used as "default" and not "max" later on,
|
||||
// so we return the default (we never really want to use the max anyway)
|
||||
ret.latency_hi = def;
|
||||
} else
|
||||
#endif
|
||||
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 {
|
||||
|
@ -3562,6 +3736,7 @@ cubeb_ops const wasapi_ops = {
|
|||
/*.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,
|
||||
/*.get_supported_input_processing_params =*/NULL,
|
||||
/*.enumerate_devices =*/wasapi_enumerate_devices,
|
||||
/*.device_collection_destroy =*/wasapi_device_collection_destroy,
|
||||
/*.destroy =*/wasapi_destroy,
|
||||
|
@ -3575,6 +3750,8 @@ cubeb_ops const wasapi_ops = {
|
|||
/*.stream_set_volume =*/wasapi_stream_set_volume,
|
||||
/*.stream_set_name =*/NULL,
|
||||
/*.stream_get_current_device =*/NULL,
|
||||
/*.stream_set_input_mute =*/NULL,
|
||||
/*.stream_set_input_processing_params =*/NULL,
|
||||
/*.stream_device_destroy =*/NULL,
|
||||
/*.stream_register_device_changed_callback =*/NULL,
|
||||
/*.register_device_collection_changed =*/
|
||||
|
|
|
@ -1192,6 +1192,7 @@ static struct cubeb_ops const winmm_ops = {
|
|||
/*.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,
|
||||
/*.get_supported_input_processing_params =*/NULL,
|
||||
/*.enumerate_devices =*/winmm_enumerate_devices,
|
||||
/*.device_collection_destroy =*/winmm_device_collection_destroy,
|
||||
/*.destroy =*/winmm_destroy,
|
||||
|
@ -1205,6 +1206,8 @@ static struct cubeb_ops const winmm_ops = {
|
|||
/*.stream_set_volume =*/winmm_stream_set_volume,
|
||||
/*.stream_set_name =*/NULL,
|
||||
/*.stream_get_current_device =*/NULL,
|
||||
/*.stream_set_input_mute =*/NULL,
|
||||
/*.stream_set_input_processing_params =*/NULL,
|
||||
/*.stream_device_destroy =*/NULL,
|
||||
/*.stream_register_device_changed_callback=*/NULL,
|
||||
/*.register_device_collection_changed =*/NULL};
|
||||
|
|
Loading…
Reference in a new issue