diff options
Diffstat (limited to 'webrtc/modules/audio_device/android/opensles_player.cc')
| -rw-r--r-- | webrtc/modules/audio_device/android/opensles_player.cc | 37 |
1 files changed, 27 insertions, 10 deletions
diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc index 40967c5fb9..d2bff4905e 100644 --- a/webrtc/modules/audio_device/android/opensles_player.cc +++ b/webrtc/modules/audio_device/android/opensles_player.cc @@ -15,6 +15,7 @@ #include "webrtc/base/arraysize.h" #include "webrtc/base/checks.h" #include "webrtc/base/format_macros.h" +#include "webrtc/base/timeutils.h" #include "webrtc/modules/audio_device/android/audio_manager.h" #include "webrtc/modules/audio_device/fine_audio_buffer.h" @@ -38,7 +39,6 @@ namespace webrtc { OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager) : audio_parameters_(audio_manager->GetPlayoutAudioParameters()), - stream_type_(audio_manager->OutputStreamType()), audio_device_buffer_(NULL), initialized_(false), playing_(false), @@ -47,11 +47,9 @@ OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager) engine_(nullptr), player_(nullptr), simple_buffer_queue_(nullptr), - volume_(nullptr) { + volume_(nullptr), + last_play_time_(0) { ALOGD("ctor%s", GetThreadInfo().c_str()); - RTC_DCHECK(stream_type_ == SL_ANDROID_STREAM_VOICE || - stream_type_ == SL_ANDROID_STREAM_RING || - stream_type_ == SL_ANDROID_STREAM_MEDIA) << stream_type_; // Use native audio output parameters provided by the audio manager and // define the PCM format structure. pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(), @@ -99,6 +97,7 @@ int OpenSLESPlayer::InitPlayout() { CreateMix(); initialized_ = true; buffer_index_ = 0; + last_play_time_ = rtc::Time(); return 0; } @@ -180,15 +179,15 @@ void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { const int sample_rate_hz = audio_parameters_.sample_rate(); ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz); audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz); - const int channels = audio_parameters_.channels(); - ALOGD("SetPlayoutChannels(%d)", channels); + const size_t channels = audio_parameters_.channels(); + ALOGD("SetPlayoutChannels(%" PRIuS ")", channels); audio_device_buffer_->SetPlayoutChannels(channels); RTC_CHECK(audio_device_buffer_); AllocateDataBuffers(); } SLDataFormat_PCM OpenSLESPlayer::CreatePCMConfiguration( - int channels, + size_t channels, int sample_rate, size_t bits_per_sample) { ALOGD("CreatePCMConfiguration"); @@ -237,7 +236,16 @@ void OpenSLESPlayer::AllocateDataBuffers() { RTC_DCHECK(thread_checker_.CalledOnValidThread()); RTC_DCHECK(!simple_buffer_queue_); RTC_CHECK(audio_device_buffer_); - bytes_per_buffer_ = audio_parameters_.GetBytesPerBuffer(); + // Don't use the lowest possible size as native buffer size. Instead, + // use 10ms to better match the frame size that WebRTC uses. It will result + // in a reduced risk for audio glitches and also in a more "clean" sequence + // of callbacks from the OpenSL ES thread in to WebRTC when asking for audio + // to render. + ALOGD("lowest possible buffer size: %" PRIuS, + audio_parameters_.GetBytesPerBuffer()); + bytes_per_buffer_ = audio_parameters_.GetBytesPerFrame() * + audio_parameters_.frames_per_10ms_buffer(); + RTC_DCHECK_GE(bytes_per_buffer_, audio_parameters_.GetBytesPerBuffer()); ALOGD("native buffer size: %" PRIuS, bytes_per_buffer_); // Create a modified audio buffer class which allows us to ask for any number // of samples (and not only multiple of 10ms) to match the native OpenSL ES @@ -351,7 +359,7 @@ bool OpenSLESPlayer::CreateAudioPlayer() { false); // Set audio player configuration to SL_ANDROID_STREAM_VOICE which // corresponds to android.media.AudioManager.STREAM_VOICE_CALL. - SLint32 stream_type = stream_type_; + SLint32 stream_type = SL_ANDROID_STREAM_VOICE; RETURN_ON_ERROR( (*player_config) ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE, @@ -422,6 +430,15 @@ void OpenSLESPlayer::FillBufferQueue() { } void OpenSLESPlayer::EnqueuePlayoutData() { + // Check delta time between two successive callbacks and provide a warning + // if it becomes very large. + // TODO(henrika): using 100ms as upper limit but this value is rather random. + const uint32_t current_time = rtc::Time(); + const uint32_t diff = current_time - last_play_time_; + if (diff > 100) { + 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. |