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AudioTransport::NeedMorePlayData is no longer called from different threads using OpenSL ES on Android

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BUG=webrtc:6476
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2410033002
Cr-Commit-Position: refs/heads/master@{#14599}
This commit is contained in:
henrika 2016-10-11 06:15:41 -07:00 committed by Commit bot
parent 99df6c03c3
commit 14acf658ad
2 changed files with 20 additions and 8 deletions
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23
webrtc/modules/audio_device/android/opensles_player.cc
View File

@ -120,7 +120,7 @@ int OpenSLESPlayer::StartPlayout() {
// EnqueuePlayoutData. Most likely not worth the risk of adding a glitch.
last_play_time_ = rtc::Time();
for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
EnqueuePlayoutData();
EnqueuePlayoutData(true);
}
// Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
// For a player object, when the object is in the SL_PLAYSTATE_PLAYING
@ -376,10 +376,10 @@ void OpenSLESPlayer::FillBufferQueue() {
ALOGW("Buffer callback in non-playing state!");
return;
}
EnqueuePlayoutData();
EnqueuePlayoutData(false);
}
void OpenSLESPlayer::EnqueuePlayoutData() {
void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
// Check delta time between two successive callbacks and provide a warning
// if it becomes very large.
// TODO(henrika): using 150ms as upper limit but this value is rather random.
@ -389,11 +389,20 @@ void OpenSLESPlayer::EnqueuePlayoutData() {
ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
}
last_play_time_ = current_time;
// Read audio data from the WebRTC source using the FineAudioBuffer object
// to adjust for differences in buffer size between WebRTC (10ms) and native
// OpenSL ES.
SLint8* audio_ptr = audio_buffers_[buffer_index_].get();
fine_audio_buffer_->GetPlayoutData(audio_ptr);
if (silence) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// Avoid aquiring real audio data from WebRTC and fill the buffer with
// zeros instead. Used to prime the buffer with silence and to avoid asking
// for audio data from two different threads.
memset(audio_ptr, 0, audio_parameters_.GetBytesPerBuffer());
} else {
RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
// Read audio data from the WebRTC source using the FineAudioBuffer object
// to adjust for differences in buffer size between WebRTC (10ms) and native
// OpenSL ES.
fine_audio_buffer_->GetPlayoutData(audio_ptr);
}
// Enqueue the decoded audio buffer for playback.
SLresult err = (*simple_buffer_queue_)
->Enqueue(simple_buffer_queue_, audio_ptr,

5
webrtc/modules/audio_device/android/opensles_player.h
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@ -86,7 +86,10 @@ class OpenSLESPlayer {
// Reads audio data in PCM format using the AudioDeviceBuffer.
// Can be called both on the main thread (during Start()) and from the
// internal audio thread while output streaming is active.
void EnqueuePlayoutData();
// If the |silence| flag is set, the audio is filled with zeros instead of
// asking the WebRTC layer for real audio data. This procedure is also known
// as audio priming.
void EnqueuePlayoutData(bool silence);
// Allocate memory for audio buffers which will be used to render audio
// via the SLAndroidSimpleBufferQueueItf interface.