/* ex: set tabstop=2 shiftwidth=2 expandtab: * Copyright © 2019 Jan Kelling * * This program is made available under an ISC-style license. See the * accompanying file LICENSE for details. */ #include "cubeb-internal.h" #include "cubeb/cubeb.h" #include "cubeb_android.h" #include "cubeb_log.h" #include "cubeb_resampler.h" #include #include #include #include #include #include #include #include #include #include #include #ifdef DISABLE_LIBAAUDIO_DLOPEN #define WRAP(x) x #else #define WRAP(x) (*cubeb_##x) #define LIBAAUDIO_API_VISIT(X) \ X(AAudio_convertResultToText) \ X(AAudio_convertStreamStateToText) \ X(AAudio_createStreamBuilder) \ X(AAudioStreamBuilder_openStream) \ X(AAudioStreamBuilder_setChannelCount) \ X(AAudioStreamBuilder_setBufferCapacityInFrames) \ X(AAudioStreamBuilder_setDirection) \ X(AAudioStreamBuilder_setFormat) \ X(AAudioStreamBuilder_setSharingMode) \ X(AAudioStreamBuilder_setPerformanceMode) \ X(AAudioStreamBuilder_setSampleRate) \ X(AAudioStreamBuilder_delete) \ X(AAudioStreamBuilder_setDataCallback) \ X(AAudioStreamBuilder_setErrorCallback) \ X(AAudioStream_close) \ X(AAudioStream_read) \ X(AAudioStream_requestStart) \ X(AAudioStream_requestPause) \ X(AAudioStream_setBufferSizeInFrames) \ X(AAudioStream_getTimestamp) \ X(AAudioStream_requestFlush) \ X(AAudioStream_requestStop) \ X(AAudioStream_getPerformanceMode) \ X(AAudioStream_getSharingMode) \ X(AAudioStream_getBufferSizeInFrames) \ X(AAudioStream_getBufferCapacityInFrames) \ X(AAudioStream_getSampleRate) \ X(AAudioStream_waitForStateChange) \ X(AAudioStream_getFramesRead) \ X(AAudioStream_getState) \ X(AAudioStream_getFramesWritten) \ X(AAudioStream_getFramesPerBurst) \ X(AAudioStreamBuilder_setInputPreset) \ X(AAudioStreamBuilder_setUsage) // not needed or added later on // X(AAudioStreamBuilder_setFramesPerDataCallback) \ // X(AAudioStreamBuilder_setDeviceId) \ // X(AAudioStreamBuilder_setSamplesPerFrame) \ // X(AAudioStream_getSamplesPerFrame) \ // X(AAudioStream_getDeviceId) \ // X(AAudioStream_write) \ // X(AAudioStream_getChannelCount) \ // X(AAudioStream_getFormat) \ // X(AAudioStream_getXRunCount) \ // X(AAudioStream_isMMapUsed) \ // X(AAudioStreamBuilder_setContentType) \ // X(AAudioStreamBuilder_setSessionId) \ // X(AAudioStream_getUsage) \ // X(AAudioStream_getContentType) \ // X(AAudioStream_getInputPreset) \ // X(AAudioStream_getSessionId) \ // END: not needed or added later on #define MAKE_TYPEDEF(x) static decltype(x) * cubeb_##x; LIBAAUDIO_API_VISIT(MAKE_TYPEDEF) #undef MAKE_TYPEDEF #endif const uint8_t MAX_STREAMS = 16; using unique_lock = std::unique_lock; using lock_guard = std::lock_guard; enum class stream_state { INIT = 0, STOPPED, STOPPING, STARTED, STARTING, DRAINING, ERROR, SHUTDOWN, }; struct cubeb_stream { /* Note: Must match cubeb_stream layout in cubeb.c. */ cubeb * context{}; void * user_ptr{}; std::atomic in_use{false}; std::atomic state{stream_state::INIT}; AAudioStream * ostream{}; AAudioStream * istream{}; cubeb_data_callback data_callback{}; cubeb_state_callback state_callback{}; cubeb_resampler * resampler{}; // mutex synchronizes access to the stream from the state thread // and user-called functions. Everything that is accessed in the // aaudio data (or error) callback is synchronized only via atomics. std::mutex mutex; std::unique_ptr in_buf; unsigned in_frame_size{}; // size of one input frame cubeb_sample_format out_format{}; std::atomic volume{1.f}; unsigned out_channels{}; unsigned out_frame_size{}; int64_t latest_output_latency = 0; int64_t latest_input_latency = 0; bool voice_input; bool voice_output; uint64_t previous_clock; }; struct cubeb { struct cubeb_ops const * ops{}; void * libaaudio{}; struct { // The state thread: it waits for state changes and stops // drained streams. std::thread thread; std::thread notifier; std::mutex mutex; std::condition_variable cond; std::atomic join{false}; std::atomic waiting{false}; } state; // streams[i].in_use signals whether a stream is used struct cubeb_stream streams[MAX_STREAMS]; }; // Only allowed from state thread, while mutex on stm is locked static void shutdown(cubeb_stream * stm) { if (stm->istream) { WRAP(AAudioStream_requestStop)(stm->istream); } if (stm->ostream) { WRAP(AAudioStream_requestStop)(stm->ostream); } stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR); stm->state.store(stream_state::SHUTDOWN); } // Returns whether the given state is one in which we wait for // an asynchronous change static bool waiting_state(stream_state state) { switch (state) { case stream_state::DRAINING: case stream_state::STARTING: case stream_state::STOPPING: return true; default: return false; } } static void update_state(cubeb_stream * stm) { // Fast path for streams that don't wait for state change or are invalid enum stream_state old_state = stm->state.load(); if (old_state == stream_state::INIT || old_state == stream_state::STARTED || old_state == stream_state::STOPPED || old_state == stream_state::SHUTDOWN) { return; } // If the main thread currently operates on this thread, we don't // have to wait for it unique_lock lock(stm->mutex, std::try_to_lock); if (!lock.owns_lock()) { return; } // check again: if this is true now, the stream was destroyed or // changed between our fast path check and locking the mutex old_state = stm->state.load(); if (old_state == stream_state::INIT || old_state == stream_state::STARTED || old_state == stream_state::STOPPED || old_state == stream_state::SHUTDOWN) { return; } // We compute the new state the stream has and then compare_exchange it // if it has changed. This way we will never just overwrite state // changes that were set from the audio thread in the meantime, // such as a DRAINING or error state. enum stream_state new_state; do { if (old_state == stream_state::SHUTDOWN) { return; } if (old_state == stream_state::ERROR) { shutdown(stm); return; } new_state = old_state; aaudio_stream_state_t istate = 0; aaudio_stream_state_t ostate = 0; // We use waitForStateChange (with zero timeout) instead of just // getState since only the former internally updates the state. // See the docs of aaudio getState/waitForStateChange for details, // why we are passing STATE_UNKNOWN. aaudio_result_t res; if (stm->istream) { res = WRAP(AAudioStream_waitForStateChange)( stm->istream, AAUDIO_STREAM_STATE_UNKNOWN, &istate, 0); if (res != AAUDIO_OK) { LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res)); return; } assert(istate); } if (stm->ostream) { res = WRAP(AAudioStream_waitForStateChange)( stm->ostream, AAUDIO_STREAM_STATE_UNKNOWN, &ostate, 0); if (res != AAUDIO_OK) { LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res)); return; } assert(ostate); } // handle invalid stream states if (istate == AAUDIO_STREAM_STATE_PAUSING || istate == AAUDIO_STREAM_STATE_PAUSED || istate == AAUDIO_STREAM_STATE_FLUSHING || istate == AAUDIO_STREAM_STATE_FLUSHED || istate == AAUDIO_STREAM_STATE_UNKNOWN || istate == AAUDIO_STREAM_STATE_DISCONNECTED) { const char * name = WRAP(AAudio_convertStreamStateToText)(istate); LOG("Unexpected android input stream state %s", name); shutdown(stm); return; } if (ostate == AAUDIO_STREAM_STATE_PAUSING || ostate == AAUDIO_STREAM_STATE_PAUSED || ostate == AAUDIO_STREAM_STATE_FLUSHING || ostate == AAUDIO_STREAM_STATE_FLUSHED || ostate == AAUDIO_STREAM_STATE_UNKNOWN || ostate == AAUDIO_STREAM_STATE_DISCONNECTED) { const char * name = WRAP(AAudio_convertStreamStateToText)(istate); LOG("Unexpected android output stream state %s", name); shutdown(stm); return; } switch (old_state) { case stream_state::STARTING: if ((!istate || istate == AAUDIO_STREAM_STATE_STARTED) && (!ostate || ostate == AAUDIO_STREAM_STATE_STARTED)) { stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED); new_state = stream_state::STARTED; } break; case stream_state::DRAINING: // The DRAINING state means that we want to stop the streams but // may not have done so yet. // The aaudio docs state that returning STOP from the callback isn't // enough, the stream has to be stopped from another thread // afterwards. // No callbacks are triggered anymore when requestStop returns. // That is important as we otherwise might read from a closed istream // for a duplex stream. // Therefor it is important to close ostream first. if (ostate && ostate != AAUDIO_STREAM_STATE_STOPPING && ostate != AAUDIO_STREAM_STATE_STOPPED) { res = WRAP(AAudioStream_requestStop)(stm->ostream); if (res != AAUDIO_OK) { LOG("AAudioStream_requestStop: %s", WRAP(AAudio_convertResultToText)(res)); return; } } if (istate && istate != AAUDIO_STREAM_STATE_STOPPING && istate != AAUDIO_STREAM_STATE_STOPPED) { res = WRAP(AAudioStream_requestStop)(stm->istream); if (res != AAUDIO_OK) { LOG("AAudioStream_requestStop: %s", WRAP(AAudio_convertResultToText)(res)); return; } } // we always wait until both streams are stopped until we // send CUBEB_STATE_DRAINED. Then we can directly transition // our logical state to STOPPED, not triggering // an additional CUBEB_STATE_STOPPED callback (which might // be unexpected for the user). if ((!ostate || ostate == AAUDIO_STREAM_STATE_STOPPED) && (!istate || istate == AAUDIO_STREAM_STATE_STOPPED)) { new_state = stream_state::STOPPED; stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED); } break; case stream_state::STOPPING: assert(!istate || istate == AAUDIO_STREAM_STATE_STOPPING || istate == AAUDIO_STREAM_STATE_STOPPED); assert(!ostate || ostate == AAUDIO_STREAM_STATE_STOPPING || ostate == AAUDIO_STREAM_STATE_STOPPED); if ((!istate || istate == AAUDIO_STREAM_STATE_STOPPED) && (!ostate || ostate == AAUDIO_STREAM_STATE_STOPPED)) { stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED); new_state = stream_state::STOPPED; } break; default: assert(false && "Unreachable: invalid state"); } } while (old_state != new_state && !stm->state.compare_exchange_strong(old_state, new_state)); } // See https://nyorain.github.io/lock-free-wakeup.html for a note // why this is needed. The audio thread notifies the state thread about // state changes and must not block. The state thread on the other hand should // sleep until there is work to be done. So we need a lockfree producer // and blocking producer. This can only be achieved safely with a new thread // that only serves as notifier backup (in case the notification happens // right between the state thread checking and going to sleep in which case // this thread will kick in and signal it right again). static void notifier_thread(cubeb * ctx) { unique_lock lock(ctx->state.mutex); while (!ctx->state.join.load()) { ctx->state.cond.wait(lock); if (ctx->state.waiting.load()) { // This must signal our state thread since there is no other // thread currently waiting on the condition variable. // The state change thread is guaranteed to be waiting since // we hold the mutex it locks when awake. ctx->state.cond.notify_one(); } } // make sure other thread joins as well ctx->state.cond.notify_one(); LOG("Exiting notifier thread"); } static void state_thread(cubeb * ctx) { unique_lock lock(ctx->state.mutex); bool waiting = false; while (!ctx->state.join.load()) { waiting |= ctx->state.waiting.load(); if (waiting) { ctx->state.waiting.store(false); waiting = false; for (unsigned i = 0u; i < MAX_STREAMS; ++i) { cubeb_stream * stm = &ctx->streams[i]; update_state(stm); waiting |= waiting_state(atomic_load(&stm->state)); } // state changed from another thread, update again immediately if (ctx->state.waiting.load()) { waiting = true; continue; } // Not waiting for any change anymore: we can wait on the // condition variable without timeout if (!waiting) { continue; } // while any stream is waiting for state change we sleep with regular // timeouts. But we wake up immediately if signaled. // This might seem like a poor man's implementation of state change // waiting but (as of october 2020), the implementation of // AAudioStream_waitForStateChange is just sleeping with regular // timeouts as well: // https://android.googlesource.com/platform/frameworks/av/+/refs/heads/master/media/libaaudio/src/core/AudioStream.cpp auto dur = std::chrono::milliseconds(5); ctx->state.cond.wait_for(lock, dur); } else { ctx->state.cond.wait(lock); } } // make sure other thread joins as well ctx->state.cond.notify_one(); LOG("Exiting state thread"); } static char const * aaudio_get_backend_id(cubeb * /* ctx */) { return "aaudio"; } static int aaudio_get_max_channel_count(cubeb * ctx, uint32_t * max_channels) { assert(ctx && max_channels); // NOTE: we might get more, AAudio docs don't specify anything. *max_channels = 2; return CUBEB_OK; } static void aaudio_destroy(cubeb * ctx) { assert(ctx); #ifndef NDEBUG // make sure all streams were destroyed for (unsigned i = 0u; i < MAX_STREAMS; ++i) { assert(!ctx->streams[i].in_use.load()); } #endif // broadcast joining to both threads // they will additionally signal each other before joining ctx->state.join.store(true); ctx->state.cond.notify_all(); if (ctx->state.thread.joinable()) { ctx->state.thread.join(); } if (ctx->state.notifier.joinable()) { ctx->state.notifier.join(); } if (ctx->libaaudio) { dlclose(ctx->libaaudio); } delete ctx; } static void apply_volume(cubeb_stream * stm, void * audio_data, uint32_t num_frames) { float volume = stm->volume.load(); // optimization: we don't have to change anything in this case if (volume == 1.f) { return; } switch (stm->out_format) { case CUBEB_SAMPLE_S16NE: for (uint32_t i = 0u; i < num_frames * stm->out_channels; ++i) { (static_cast(audio_data))[i] *= volume; } break; case CUBEB_SAMPLE_FLOAT32NE: for (uint32_t i = 0u; i < num_frames * stm->out_channels; ++i) { (static_cast(audio_data))[i] *= volume; } break; default: assert(false && "Unreachable: invalid stream out_format"); } } // Returning AAUDIO_CALLBACK_RESULT_STOP seems to put the stream in // an invalid state. Seems like an AAudio bug/bad documentation. // We therefore only return it on error. static aaudio_data_callback_result_t aaudio_duplex_data_cb(AAudioStream * astream, void * user_data, void * audio_data, int32_t num_frames) { cubeb_stream * stm = (cubeb_stream *)user_data; assert(stm->ostream == astream); assert(stm->istream); assert(num_frames >= 0); stream_state state = atomic_load(&stm->state); // int istate = WRAP(AAudioStream_getState)(stm->istream); // int ostate = WRAP(AAudioStream_getState)(stm->ostream); // ALOGV("aaudio duplex data cb on stream %p: state %ld (in: %d, out: %d), // num_frames: %ld", // (void*) stm, state, istate, ostate, num_frames); // all other states may happen since the callback might be called // from within requestStart assert(state != stream_state::SHUTDOWN); // This might happen when we started draining but not yet actually // stopped the stream from the state thread. if (state == stream_state::DRAINING) { std::memset(audio_data, 0x0, num_frames * stm->out_frame_size); return AAUDIO_CALLBACK_RESULT_CONTINUE; } // The aaudio docs state that AAudioStream_read must not be called on // the stream associated with a callback. But we call it on the input stream // while this callback is for the output stream so this is ok. // We also pass timeout 0, giving us strong non-blocking guarantees. // This is exactly how it's done in the aaudio duplex example code snippet. long in_num_frames = WRAP(AAudioStream_read)(stm->istream, stm->in_buf.get(), num_frames, 0); if (in_num_frames < 0) { // error stm->state.store(stream_state::ERROR); LOG("AAudioStream_read: %s", WRAP(AAudio_convertResultToText)(in_num_frames)); return AAUDIO_CALLBACK_RESULT_STOP; } // This can happen shortly after starting the stream. AAudio might immediately // begin to buffer output but not have any input ready yet. We could // block AAudioStream_read (passing a timeout > 0) but that leads to issues // since blocking in this callback is a bad idea in general and it might break // the stream when it is stopped by another thread shortly after being // started. We therefore simply send silent input to the application, as shown // in the AAudio duplex stream code example. if (in_num_frames < num_frames) { // LOG("AAudioStream_read returned not enough frames: %ld instead of %d", // in_num_frames, num_frames); unsigned left = num_frames - in_num_frames; char * buf = stm->in_buf.get() + in_num_frames * stm->in_frame_size; std::memset(buf, 0x0, left * stm->in_frame_size); in_num_frames = num_frames; } long done_frames = cubeb_resampler_fill(stm->resampler, stm->in_buf.get(), &in_num_frames, audio_data, num_frames); if (done_frames < 0 || done_frames > num_frames) { LOG("Error in data callback or resampler: %ld", done_frames); stm->state.store(stream_state::ERROR); return AAUDIO_CALLBACK_RESULT_STOP; } else if (done_frames < num_frames) { stm->state.store(stream_state::DRAINING); stm->context->state.waiting.store(true); stm->context->state.cond.notify_one(); char * begin = static_cast(audio_data) + done_frames * stm->out_frame_size; std::memset(begin, 0x0, (num_frames - done_frames) * stm->out_frame_size); } apply_volume(stm, audio_data, done_frames); return AAUDIO_CALLBACK_RESULT_CONTINUE; } static aaudio_data_callback_result_t aaudio_output_data_cb(AAudioStream * astream, void * user_data, void * audio_data, int32_t num_frames) { cubeb_stream * stm = (cubeb_stream *)user_data; assert(stm->ostream == astream); assert(!stm->istream); assert(num_frames >= 0); stream_state state = stm->state.load(); // int ostate = WRAP(AAudioStream_getState)(stm->ostream); // ALOGV("aaudio output data cb on stream %p: state %ld (%d), num_frames: // %ld", // (void*) stm, state, ostate, num_frames); // all other states may happen since the callback might be called // from within requestStart assert(state != stream_state::SHUTDOWN); // This might happen when we started draining but not yet actually // stopped the stream from the state thread. if (state == stream_state::DRAINING) { std::memset(audio_data, 0x0, num_frames * stm->out_frame_size); return AAUDIO_CALLBACK_RESULT_CONTINUE; } long done_frames = cubeb_resampler_fill(stm->resampler, NULL, NULL, audio_data, num_frames); if (done_frames < 0 || done_frames > num_frames) { LOG("Error in data callback or resampler: %ld", done_frames); stm->state.store(stream_state::ERROR); return AAUDIO_CALLBACK_RESULT_STOP; } else if (done_frames < num_frames) { stm->state.store(stream_state::DRAINING); stm->context->state.waiting.store(true); stm->context->state.cond.notify_one(); char * begin = static_cast(audio_data) + done_frames * stm->out_frame_size; std::memset(begin, 0x0, (num_frames - done_frames) * stm->out_frame_size); } apply_volume(stm, audio_data, done_frames); return AAUDIO_CALLBACK_RESULT_CONTINUE; } static aaudio_data_callback_result_t aaudio_input_data_cb(AAudioStream * astream, void * user_data, void * audio_data, int32_t num_frames) { cubeb_stream * stm = (cubeb_stream *)user_data; assert(stm->istream == astream); assert(!stm->ostream); assert(num_frames >= 0); stream_state state = stm->state.load(); // int istate = WRAP(AAudioStream_getState)(stm->istream); // ALOGV("aaudio input data cb on stream %p: state %ld (%d), num_frames: %ld", // (void*) stm, state, istate, num_frames); // all other states may happen since the callback might be called // from within requestStart assert(state != stream_state::SHUTDOWN); // This might happen when we started draining but not yet actually // STOPPED the stream from the state thread. if (state == stream_state::DRAINING) { return AAUDIO_CALLBACK_RESULT_CONTINUE; } long input_frame_count = num_frames; long done_frames = cubeb_resampler_fill(stm->resampler, audio_data, &input_frame_count, NULL, 0); if (done_frames < 0 || done_frames > num_frames) { LOG("Error in data callback or resampler: %ld", done_frames); stm->state.store(stream_state::ERROR); return AAUDIO_CALLBACK_RESULT_STOP; } else if (done_frames < input_frame_count) { // we don't really drain an input stream, just have to // stop it from the state thread. That is signaled via the // DRAINING state. stm->state.store(stream_state::DRAINING); stm->context->state.waiting.store(true); stm->context->state.cond.notify_one(); } return AAUDIO_CALLBACK_RESULT_CONTINUE; } static void aaudio_error_cb(AAudioStream * astream, void * user_data, aaudio_result_t error) { cubeb_stream * stm = static_cast(user_data); assert(stm->ostream == astream || stm->istream == astream); LOG("AAudio error callback: %s", WRAP(AAudio_convertResultToText)(error)); stm->state.store(stream_state::ERROR); } static int realize_stream(AAudioStreamBuilder * sb, const cubeb_stream_params * params, AAudioStream ** stream, unsigned * frame_size) { aaudio_result_t res; assert(params->rate); assert(params->channels); WRAP(AAudioStreamBuilder_setSampleRate)(sb, params->rate); WRAP(AAudioStreamBuilder_setChannelCount)(sb, params->channels); aaudio_format_t fmt; switch (params->format) { case CUBEB_SAMPLE_S16NE: fmt = AAUDIO_FORMAT_PCM_I16; *frame_size = sizeof(int16_t) * params->channels; break; case CUBEB_SAMPLE_FLOAT32NE: fmt = AAUDIO_FORMAT_PCM_FLOAT; *frame_size = sizeof(float) * params->channels; break; default: return CUBEB_ERROR_INVALID_FORMAT; } WRAP(AAudioStreamBuilder_setFormat)(sb, fmt); res = WRAP(AAudioStreamBuilder_openStream)(sb, stream); if (res == AAUDIO_ERROR_INVALID_FORMAT) { LOG("AAudio device doesn't support output format %d", fmt); return CUBEB_ERROR_INVALID_FORMAT; } else if (params->rate && res == AAUDIO_ERROR_INVALID_RATE) { // The requested rate is not supported. // Just try again with default rate, we create a resampler anyways WRAP(AAudioStreamBuilder_setSampleRate)(sb, AAUDIO_UNSPECIFIED); res = WRAP(AAudioStreamBuilder_openStream)(sb, stream); LOG("Requested rate of %u is not supported, inserting resampler", params->rate); } // When the app has no permission to record audio // (android.permission.RECORD_AUDIO) but requested and input stream, this will // return INVALID_ARGUMENT. if (res != AAUDIO_OK) { LOG("AAudioStreamBuilder_openStream: %s", WRAP(AAudio_convertResultToText)(res)); return CUBEB_ERROR; } return CUBEB_OK; } static void aaudio_stream_destroy(cubeb_stream * stm) { lock_guard lock(stm->mutex); assert(stm->state == stream_state::STOPPED || stm->state == stream_state::STOPPING || stm->state == stream_state::INIT || stm->state == stream_state::DRAINING || stm->state == stream_state::ERROR || stm->state == stream_state::SHUTDOWN); aaudio_result_t res; // No callbacks are triggered anymore when requestStop returns. // That is important as we otherwise might read from a closed istream // for a duplex stream. if (stm->ostream) { if (stm->state != stream_state::STOPPED && stm->state != stream_state::STOPPING && stm->state != stream_state::SHUTDOWN) { res = WRAP(AAudioStream_requestStop)(stm->ostream); if (res != AAUDIO_OK) { LOG("AAudioStreamBuilder_requestStop: %s", WRAP(AAudio_convertResultToText)(res)); } } WRAP(AAudioStream_close)(stm->ostream); stm->ostream = NULL; } if (stm->istream) { if (stm->state != stream_state::STOPPED && stm->state != stream_state::STOPPING && stm->state != stream_state::SHUTDOWN) { res = WRAP(AAudioStream_requestStop)(stm->istream); if (res != AAUDIO_OK) { LOG("AAudioStreamBuilder_requestStop: %s", WRAP(AAudio_convertResultToText)(res)); } } WRAP(AAudioStream_close)(stm->istream); stm->istream = NULL; } if (stm->resampler) { cubeb_resampler_destroy(stm->resampler); stm->resampler = NULL; } stm->in_buf = {}; stm->in_frame_size = {}; stm->out_format = {}; stm->out_channels = {}; stm->out_frame_size = {}; stm->state.store(stream_state::INIT); stm->in_use.store(false); } static int aaudio_stream_init_impl(cubeb_stream * stm, cubeb_devid input_device, cubeb_stream_params * input_stream_params, cubeb_devid output_device, cubeb_stream_params * output_stream_params, unsigned int latency_frames) { assert(stm->state.load() == stream_state::INIT); stm->in_use.store(true); aaudio_result_t res; AAudioStreamBuilder * sb; res = WRAP(AAudio_createStreamBuilder)(&sb); if (res != AAUDIO_OK) { LOG("AAudio_createStreamBuilder: %s", WRAP(AAudio_convertResultToText)(res)); return CUBEB_ERROR; } // make sure the builder is always destroyed struct StreamBuilderDestructor { void operator()(AAudioStreamBuilder * sb) { WRAP(AAudioStreamBuilder_delete)(sb); } }; std::unique_ptr sbPtr(sb); WRAP(AAudioStreamBuilder_setErrorCallback)(sb, aaudio_error_cb, stm); WRAP(AAudioStreamBuilder_setBufferCapacityInFrames)(sb, latency_frames); AAudioStream_dataCallback in_data_callback{}; AAudioStream_dataCallback out_data_callback{}; if (output_stream_params && input_stream_params) { out_data_callback = aaudio_duplex_data_cb; in_data_callback = NULL; } else if (input_stream_params) { in_data_callback = aaudio_input_data_cb; } else if (output_stream_params) { out_data_callback = aaudio_output_data_cb; } else { LOG("Tried to open stream without input or output parameters"); return CUBEB_ERROR; } #ifdef CUBEB_AAUDIO_EXCLUSIVE_STREAM LOG("AAudio setting exclusive share mode for stream"); WRAP(AAudioStreamBuilder_setSharingMode)(sb, AAUDIO_SHARING_MODE_EXCLUSIVE); #endif if (latency_frames <= POWERSAVE_LATENCY_FRAMES_THRESHOLD) { LOG("AAudio setting low latency mode for stream"); WRAP(AAudioStreamBuilder_setPerformanceMode) (sb, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY); } else { LOG("AAudio setting power saving mode for stream"); WRAP(AAudioStreamBuilder_setPerformanceMode) (sb, AAUDIO_PERFORMANCE_MODE_POWER_SAVING); } unsigned frame_size; // initialize streams // output uint32_t target_sample_rate = 0; cubeb_stream_params out_params; if (output_stream_params) { int output_preset = stm->voice_output ? AAUDIO_USAGE_VOICE_COMMUNICATION : AAUDIO_USAGE_MEDIA; WRAP(AAudioStreamBuilder_setUsage)(sb, output_preset); WRAP(AAudioStreamBuilder_setDirection)(sb, AAUDIO_DIRECTION_OUTPUT); WRAP(AAudioStreamBuilder_setDataCallback)(sb, out_data_callback, stm); int res_err = realize_stream(sb, output_stream_params, &stm->ostream, &frame_size); if (res_err) { return res_err; } // output debug information aaudio_sharing_mode_t sm = WRAP(AAudioStream_getSharingMode)(stm->ostream); aaudio_performance_mode_t pm = WRAP(AAudioStream_getPerformanceMode)(stm->ostream); int bcap = WRAP(AAudioStream_getBufferCapacityInFrames)(stm->ostream); int bsize = WRAP(AAudioStream_getBufferSizeInFrames)(stm->ostream); int rate = WRAP(AAudioStream_getSampleRate)(stm->ostream); LOG("AAudio output stream sharing mode: %d", sm); LOG("AAudio output stream performance mode: %d", pm); LOG("AAudio output stream buffer capacity: %d", bcap); LOG("AAudio output stream buffer size: %d", bsize); LOG("AAudio output stream buffer rate: %d", rate); target_sample_rate = output_stream_params->rate; out_params = *output_stream_params; out_params.rate = rate; stm->out_channels = output_stream_params->channels; stm->out_format = output_stream_params->format; stm->out_frame_size = frame_size; stm->volume.store(1.f); } // input cubeb_stream_params in_params; if (input_stream_params) { // Match what the OpenSL backend does for now, we could use UNPROCESSED and // VOICE_COMMUNICATION here, but we'd need to make it clear that // application-level AEC and other voice processing should be disabled // there. int input_preset = stm->voice_input ? AAUDIO_INPUT_PRESET_VOICE_RECOGNITION : AAUDIO_INPUT_PRESET_CAMCORDER; WRAP(AAudioStreamBuilder_setInputPreset)(sb, input_preset); WRAP(AAudioStreamBuilder_setDirection)(sb, AAUDIO_DIRECTION_INPUT); WRAP(AAudioStreamBuilder_setDataCallback)(sb, in_data_callback, stm); int res_err = realize_stream(sb, input_stream_params, &stm->istream, &frame_size); if (res_err) { return res_err; } // output debug information aaudio_sharing_mode_t sm = WRAP(AAudioStream_getSharingMode)(stm->istream); aaudio_performance_mode_t pm = WRAP(AAudioStream_getPerformanceMode)(stm->istream); int bcap = WRAP(AAudioStream_getBufferCapacityInFrames)(stm->istream); int bsize = WRAP(AAudioStream_getBufferSizeInFrames)(stm->istream); int rate = WRAP(AAudioStream_getSampleRate)(stm->istream); LOG("AAudio input stream sharing mode: %d", sm); LOG("AAudio input stream performance mode: %d", pm); LOG("AAudio input stream buffer capacity: %d", bcap); LOG("AAudio input stream buffer size: %d", bsize); LOG("AAudio input stream buffer rate: %d", rate); stm->in_buf.reset(new char[bcap * frame_size]()); assert(!target_sample_rate || target_sample_rate == input_stream_params->rate); target_sample_rate = input_stream_params->rate; in_params = *input_stream_params; in_params.rate = rate; stm->in_frame_size = frame_size; } // initialize resampler stm->resampler = cubeb_resampler_create( stm, input_stream_params ? &in_params : NULL, output_stream_params ? &out_params : NULL, target_sample_rate, stm->data_callback, stm->user_ptr, CUBEB_RESAMPLER_QUALITY_DEFAULT, CUBEB_RESAMPLER_RECLOCK_NONE); if (!stm->resampler) { LOG("Failed to create resampler"); return CUBEB_ERROR; } // the stream isn't started initially. We don't need to differentiate // between a stream that was just initialized and one that played // already but was stopped. stm->state.store(stream_state::STOPPED); LOG("Cubeb stream (%p) INIT success", (void *)stm); return CUBEB_OK; } static int aaudio_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) { assert(!input_device); assert(!output_device); // atomically find a free stream. cubeb_stream * stm = NULL; unique_lock lock; for (unsigned i = 0u; i < MAX_STREAMS; ++i) { // This check is only an optimization, we don't strictly need it // since we check again after locking the mutex. if (ctx->streams[i].in_use.load()) { continue; } // if this fails, another thread initialized this stream // between our check of in_use and this. lock = unique_lock(ctx->streams[i].mutex, std::try_to_lock); if (!lock.owns_lock()) { continue; } if (ctx->streams[i].in_use.load()) { lock = {}; continue; } stm = &ctx->streams[i]; break; } if (!stm) { LOG("Error: maximum number of streams reached"); return CUBEB_ERROR; } stm->context = ctx; stm->user_ptr = user_ptr; stm->data_callback = data_callback; stm->state_callback = state_callback; stm->voice_input = input_stream_params && !!(input_stream_params->prefs & CUBEB_STREAM_PREF_VOICE); stm->voice_output = output_stream_params && !!(output_stream_params->prefs & CUBEB_STREAM_PREF_VOICE); stm->previous_clock = 0; LOG("cubeb stream prefs: voice_input: %s voice_output: %s", stm->voice_input ? "true" : "false", stm->voice_output ? "true" : "false"); int err = aaudio_stream_init_impl(stm, input_device, input_stream_params, output_device, output_stream_params, latency_frames); if (err != CUBEB_OK) { // This is needed since aaudio_stream_destroy will lock the mutex again. // It's no problem that there is a gap in between as the stream isn't // actually in u se. lock.unlock(); aaudio_stream_destroy(stm); return err; } *stream = stm; return CUBEB_OK; } static int aaudio_stream_start(cubeb_stream * stm) { assert(stm && stm->in_use.load()); lock_guard lock(stm->mutex); stream_state state = stm->state.load(); int istate = stm->istream ? WRAP(AAudioStream_getState)(stm->istream) : 0; int ostate = stm->ostream ? WRAP(AAudioStream_getState)(stm->ostream) : 0; LOGV("STARTING stream %p: %d (%d %d)", (void *)stm, state, istate, ostate); switch (state) { case stream_state::STARTED: case stream_state::STARTING: LOG("cubeb stream %p already STARTING/STARTED", (void *)stm); return CUBEB_OK; case stream_state::ERROR: case stream_state::SHUTDOWN: return CUBEB_ERROR; case stream_state::INIT: assert(false && "Invalid stream"); return CUBEB_ERROR; case stream_state::STOPPED: case stream_state::STOPPING: case stream_state::DRAINING: break; } aaudio_result_t res; // Important to start istream before ostream. // As soon as we start ostream, the callbacks might be triggered an we // might read from istream (on duplex). If istream wasn't started yet // this is a problem. if (stm->istream) { res = WRAP(AAudioStream_requestStart)(stm->istream); if (res != AAUDIO_OK) { LOG("AAudioStream_requestStart (istream): %s", WRAP(AAudio_convertResultToText)(res)); stm->state.store(stream_state::ERROR); return CUBEB_ERROR; } } if (stm->ostream) { res = WRAP(AAudioStream_requestStart)(stm->ostream); if (res != AAUDIO_OK) { LOG("AAudioStream_requestStart (ostream): %s", WRAP(AAudio_convertResultToText)(res)); stm->state.store(stream_state::ERROR); return CUBEB_ERROR; } } int ret = CUBEB_OK; bool success; while (!(success = stm->state.compare_exchange_strong( state, stream_state::STARTING))) { // we land here only if the state has changed in the meantime switch (state) { // If an error ocurred in the meantime, we can't change that. // The stream will be stopped when shut down. case stream_state::ERROR: ret = CUBEB_ERROR; break; // The only situation in which the state could have switched to draining // is if the callback was already fired and requested draining. Don't // overwrite that. It's not an error either though. case stream_state::DRAINING: break; // If the state switched [DRAINING -> STOPPING] or [DRAINING/STOPPING -> // STOPPED] in the meantime, we can simply overwrite that since we restarted // the stream. case stream_state::STOPPING: case stream_state::STOPPED: continue; // There is no situation in which the state could have been valid before // but now in shutdown mode, since we hold the streams mutex. // There is also no way that it switched *into* STARTING or // STARTED mode. default: assert(false && "Invalid state change"); ret = CUBEB_ERROR; break; } break; } if (success) { stm->context->state.waiting.store(true); stm->context->state.cond.notify_one(); } return ret; } static int aaudio_stream_stop(cubeb_stream * stm) { assert(stm && stm->in_use.load()); lock_guard lock(stm->mutex); stream_state state = stm->state.load(); int istate = stm->istream ? WRAP(AAudioStream_getState)(stm->istream) : 0; int ostate = stm->ostream ? WRAP(AAudioStream_getState)(stm->ostream) : 0; LOGV("STOPPING stream %p: %d (%d %d)", (void *)stm, state, istate, ostate); switch (state) { case stream_state::STOPPED: case stream_state::STOPPING: case stream_state::DRAINING: LOG("cubeb stream %p already STOPPING/STOPPED", (void *)stm); return CUBEB_OK; case stream_state::ERROR: case stream_state::SHUTDOWN: return CUBEB_ERROR; case stream_state::INIT: assert(false && "Invalid stream"); return CUBEB_ERROR; case stream_state::STARTED: case stream_state::STARTING: break; } aaudio_result_t res; // No callbacks are triggered anymore when requestStop returns. // That is important as we otherwise might read from a closed istream // for a duplex stream. // Therefor it is important to close ostream first. if (stm->ostream) { // Could use pause + flush here as well, the public cubeb interface // doesn't state behavior. res = WRAP(AAudioStream_requestStop)(stm->ostream); if (res != AAUDIO_OK) { LOG("AAudioStream_requestStop (ostream): %s", WRAP(AAudio_convertResultToText)(res)); stm->state.store(stream_state::ERROR); return CUBEB_ERROR; } } if (stm->istream) { res = WRAP(AAudioStream_requestStop)(stm->istream); if (res != AAUDIO_OK) { LOG("AAudioStream_requestStop (istream): %s", WRAP(AAudio_convertResultToText)(res)); stm->state.store(stream_state::ERROR); return CUBEB_ERROR; } } int ret = CUBEB_OK; bool success; while (!(success = atomic_compare_exchange_strong(&stm->state, &state, stream_state::STOPPING))) { // we land here only if the state has changed in the meantime switch (state) { // If an error ocurred in the meantime, we can't change that. // The stream will be STOPPED when shut down. case stream_state::ERROR: ret = CUBEB_ERROR; break; // If it was switched to DRAINING in the meantime, it was or // will be STOPPED soon anyways. We don't interfere with // the DRAINING process, no matter in which state. // Not an error case stream_state::DRAINING: case stream_state::STOPPING: case stream_state::STOPPED: break; // If the state switched from STARTING to STARTED in the meantime // we can simply overwrite that since we just STOPPED it. case stream_state::STARTED: continue; // There is no situation in which the state could have been valid before // but now in shutdown mode, since we hold the streams mutex. // There is also no way that it switched *into* STARTING mode. default: assert(false && "Invalid state change"); ret = CUBEB_ERROR; break; } break; } if (success) { stm->context->state.waiting.store(true); stm->context->state.cond.notify_one(); } return ret; } static int aaudio_stream_get_position(cubeb_stream * stm, uint64_t * position) { assert(stm && stm->in_use.load()); lock_guard lock(stm->mutex); stream_state state = stm->state.load(); AAudioStream * stream = stm->ostream ? stm->ostream : stm->istream; switch (state) { case stream_state::ERROR: case stream_state::SHUTDOWN: return CUBEB_ERROR; case stream_state::DRAINING: case stream_state::STOPPED: case stream_state::STOPPING: // getTimestamp is only valid when the stream is playing. // Simply return the number of frames passed to aaudio *position = WRAP(AAudioStream_getFramesRead)(stream); if (*position < stm->previous_clock) { *position = stm->previous_clock; } else { stm->previous_clock = *position; } return CUBEB_OK; case stream_state::INIT: assert(false && "Invalid stream"); return CUBEB_ERROR; case stream_state::STARTED: case stream_state::STARTING: break; } int64_t pos; int64_t ns; aaudio_result_t res; res = WRAP(AAudioStream_getTimestamp)(stream, CLOCK_MONOTONIC, &pos, &ns); if (res != AAUDIO_OK) { // When the audio stream is not running, invalid_state is returned and we // simply fall back to the method we use for non-playing streams. if (res == AAUDIO_ERROR_INVALID_STATE) { *position = WRAP(AAudioStream_getFramesRead)(stream); if (*position < stm->previous_clock) { *position = stm->previous_clock; } else { stm->previous_clock = *position; } return CUBEB_OK; } LOG("AAudioStream_getTimestamp: %s", WRAP(AAudio_convertResultToText)(res)); return CUBEB_ERROR; } *position = pos; if (*position < stm->previous_clock) { *position = stm->previous_clock; } else { stm->previous_clock = *position; } return CUBEB_OK; } static int aaudio_stream_get_latency(cubeb_stream * stm, uint32_t * latency) { int64_t pos; int64_t ns; aaudio_result_t res; if (!stm->ostream) { LOG("error: aaudio_stream_get_latency on input-only stream"); return CUBEB_ERROR; } res = WRAP(AAudioStream_getTimestamp)(stm->ostream, CLOCK_MONOTONIC, &pos, &ns); if (res != AAUDIO_OK) { LOG("aaudio_stream_get_latency, AAudioStream_getTimestamp: %s, returning " "memoized value", WRAP(AAudio_convertResultToText)(res)); // Expected when the stream is paused. *latency = stm->latest_output_latency; return CUBEB_OK; } int64_t read = WRAP(AAudioStream_getFramesRead)(stm->ostream); *latency = stm->latest_output_latency = read - pos; LOG("aaudio_stream_get_latency, %u", *latency); return CUBEB_OK; } static int aaudio_stream_get_input_latency(cubeb_stream * stm, uint32_t * latency) { int64_t pos; int64_t ns; aaudio_result_t res; if (!stm->istream) { LOG("error: aaudio_stream_get_input_latency on an ouput-only stream"); return CUBEB_ERROR; } res = WRAP(AAudioStream_getTimestamp)(stm->istream, CLOCK_MONOTONIC, &pos, &ns); if (res != AAUDIO_OK) { // Expected when the stream is paused. LOG("aaudio_stream_get_input_latency, AAudioStream_getTimestamp: %s, " "returning memoized value", WRAP(AAudio_convertResultToText)(res)); *latency = stm->latest_input_latency; return CUBEB_OK; } int64_t written = WRAP(AAudioStream_getFramesWritten)(stm->istream); *latency = stm->latest_input_latency = written - pos; LOG("aaudio_stream_get_input_latency, %u", *latency); return CUBEB_OK; } static int aaudio_stream_set_volume(cubeb_stream * stm, float volume) { assert(stm && stm->in_use.load() && stm->ostream); stm->volume.store(volume); return CUBEB_OK; } aaudio_data_callback_result_t dummy_callback(AAudioStream * stream, void * userData, void * audioData, int32_t numFrames) { return AAUDIO_CALLBACK_RESULT_STOP; } // Returns a dummy stream with all default settings static AAudioStream * init_dummy_stream() { AAudioStreamBuilder * streamBuilder; aaudio_result_t res; res = WRAP(AAudio_createStreamBuilder)(&streamBuilder); if (res != AAUDIO_OK) { LOG("init_dummy_stream: AAudio_createStreamBuilder: %s", WRAP(AAudio_convertResultToText)(res)); return nullptr; } WRAP(AAudioStreamBuilder_setDataCallback) (streamBuilder, dummy_callback, nullptr); WRAP(AAudioStreamBuilder_setPerformanceMode) (streamBuilder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY); AAudioStream * stream; res = WRAP(AAudioStreamBuilder_openStream)(streamBuilder, &stream); if (res != AAUDIO_OK) { LOG("init_dummy_stream: AAudioStreamBuilder_openStream %s", WRAP(AAudio_convertResultToText)(res)); return nullptr; } WRAP(AAudioStreamBuilder_delete)(streamBuilder); return stream; } static void destroy_dummy_stream(AAudioStream * stream) { WRAP(AAudioStream_close)(stream); } static int aaudio_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames) { AAudioStream * stream = init_dummy_stream(); if (!stream) { return CUBEB_ERROR; } // https://android.googlesource.com/platform/compatibility/cdd/+/refs/heads/master/5_multimedia/5_6_audio-latency.md *latency_frames = WRAP(AAudioStream_getFramesPerBurst)(stream); LOG("aaudio_get_min_latency: %u frames", *latency_frames); destroy_dummy_stream(stream); return CUBEB_OK; } int aaudio_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate) { AAudioStream * stream = init_dummy_stream(); if (!stream) { return CUBEB_ERROR; } *rate = WRAP(AAudioStream_getSampleRate)(stream); LOG("aaudio_get_preferred_sample_rate %uHz", *rate); destroy_dummy_stream(stream); return CUBEB_OK; } extern "C" int aaudio_init(cubeb ** context, char const * context_name); const static struct cubeb_ops aaudio_ops = { /*.init =*/aaudio_init, /*.get_backend_id =*/aaudio_get_backend_id, /*.get_max_channel_count =*/aaudio_get_max_channel_count, /* .get_min_latency =*/aaudio_get_min_latency, /*.get_preferred_sample_rate =*/aaudio_get_preferred_sample_rate, /*.enumerate_devices =*/NULL, /*.device_collection_destroy =*/NULL, /*.destroy =*/aaudio_destroy, /*.stream_init =*/aaudio_stream_init, /*.stream_destroy =*/aaudio_stream_destroy, /*.stream_start =*/aaudio_stream_start, /*.stream_stop =*/aaudio_stream_stop, /*.stream_get_position =*/aaudio_stream_get_position, /*.stream_get_latency =*/aaudio_stream_get_latency, /*.stream_get_input_latency =*/aaudio_stream_get_input_latency, /*.stream_set_volume =*/aaudio_stream_set_volume, /*.stream_set_name =*/NULL, /*.stream_get_current_device =*/NULL, /*.stream_device_destroy =*/NULL, /*.stream_register_device_changed_callback =*/NULL, /*.register_device_collection_changed =*/NULL}; extern "C" /*static*/ int aaudio_init(cubeb ** context, char const * /* context_name */) { // load api void * libaaudio = NULL; #ifndef DISABLE_LIBAAUDIO_DLOPEN libaaudio = dlopen("libaaudio.so", RTLD_NOW); if (!libaaudio) { return CUBEB_ERROR; } #define LOAD(x) \ { \ cubeb_##x = (decltype(x) *)(dlsym(libaaudio, #x)); \ if (!WRAP(x)) { \ LOG("AAudio: Failed to load %s", #x); \ dlclose(libaaudio); \ return CUBEB_ERROR; \ } \ } LIBAAUDIO_API_VISIT(LOAD); #undef LOAD #endif cubeb * ctx = new cubeb; ctx->ops = &aaudio_ops; ctx->libaaudio = libaaudio; ctx->state.thread = std::thread(state_thread, ctx); // NOTE: using platform-specific APIs we could set the priority of the // notifier thread lower than the priority of the state thread. // This way, it's more likely that the state thread will be woken up // by the condition variable signal when both are currently waiting ctx->state.notifier = std::thread(notifier_thread, ctx); *context = ctx; return CUBEB_OK; }