diff options
Diffstat (limited to 'chromium/content/renderer/media/webrtc_local_audio_track.cc')
| -rw-r--r-- | chromium/content/renderer/media/webrtc_local_audio_track.cc | 275 |
1 files changed, 57 insertions, 218 deletions
diff --git a/chromium/content/renderer/media/webrtc_local_audio_track.cc b/chromium/content/renderer/media/webrtc_local_audio_track.cc index 8afa06feab2..95f34f64ea3 100644 --- a/chromium/content/renderer/media/webrtc_local_audio_track.cc +++ b/chromium/content/renderer/media/webrtc_local_audio_track.cc @@ -5,150 +5,29 @@ #include "content/renderer/media/webrtc_local_audio_track.h" #include "content/public/renderer/media_stream_audio_sink.h" +#include "content/renderer/media/media_stream_audio_level_calculator.h" +#include "content/renderer/media/media_stream_audio_processor.h" #include "content/renderer/media/media_stream_audio_sink_owner.h" #include "content/renderer/media/media_stream_audio_track_sink.h" #include "content/renderer/media/peer_connection_audio_sink_owner.h" #include "content/renderer/media/webaudio_capturer_source.h" +#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" #include "content/renderer/media/webrtc_audio_capturer.h" -#include "content/renderer/media/webrtc_local_audio_source_provider.h" -#include "media/base/audio_fifo.h" -#include "third_party/libjingle/source/talk/media/base/audiorenderer.h" namespace content { -static const size_t kMaxNumberOfBuffersInFifo = 2; -static const char kAudioTrackKind[] = "audio"; - -namespace { - -using webrtc::MediaConstraintsInterface; - -// This helper function checks if any audio constraints are set that require -// audio processing to be applied. Right now this is a big, single switch for -// all of the properties, but in the future they'll be handled one by one. -bool NeedsAudioProcessing( - const webrtc::MediaConstraintsInterface* constraints) { - if (!constraints) - return false; - - static const char* kAudioProcessingProperties[] = { - MediaConstraintsInterface::kEchoCancellation, - MediaConstraintsInterface::kExperimentalEchoCancellation, - MediaConstraintsInterface::kAutoGainControl, - MediaConstraintsInterface::kExperimentalAutoGainControl, - MediaConstraintsInterface::kNoiseSuppression, - MediaConstraintsInterface::kHighpassFilter, - MediaConstraintsInterface::kTypingNoiseDetection, - }; - - for (size_t i = 0; i < arraysize(kAudioProcessingProperties); ++i) { - bool value = false; - if (webrtc::FindConstraint(constraints, kAudioProcessingProperties[i], - &value, NULL) && - value) { - return true; - } - } - - return false; -} - -} // namespace. - -// This is a temporary audio buffer with parameters used to send data to -// callbacks. -class WebRtcLocalAudioTrack::ConfiguredBuffer { - public: - ConfiguredBuffer() {} - virtual ~ConfiguredBuffer() {} - - void Configure(const media::AudioParameters& params) { - DCHECK(params.IsValid()); - - // PeerConnection uses 10ms as the sink buffer size as its native packet - // size. We use the native PeerConnection buffer size to achieve the best - // performance when a PeerConnection is connected with a track. - int sink_buffer_size = params.sample_rate() / 100; - if (params.frames_per_buffer() < sink_buffer_size) { - // When the source is running with a buffer size smaller than the peer - // connection buffer size, that means no PeerConnection is connected - // to the track, use the same buffer size as the incoming format to - // avoid extra FIFO for WebAudio. - sink_buffer_size = params.frames_per_buffer(); - } - params_.Reset(params.format(), params.channel_layout(), params.channels(), - params.input_channels(), params.sample_rate(), - params.bits_per_sample(), sink_buffer_size); - - audio_wrapper_ = media::AudioBus::Create(params_.channels(), - params_.frames_per_buffer()); - buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]); - - // The size of the FIFO should be at least twice of the source buffer size - // or twice of the sink buffer size. - int buffer_size = std::max( - kMaxNumberOfBuffersInFifo * params.frames_per_buffer(), - kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()); - fifo_.reset(new media::AudioFifo(params_.channels(), buffer_size)); - } - - void Push(media::AudioBus* audio_source) { - DCHECK(fifo_->frames() + audio_source->frames() <= fifo_->max_frames()); - fifo_->Push(audio_source); - } - - bool Consume() { - if (fifo_->frames() < audio_wrapper_->frames()) - return false; - - fifo_->Consume(audio_wrapper_.get(), 0, audio_wrapper_->frames()); - audio_wrapper_->ToInterleaved(audio_wrapper_->frames(), - params_.bits_per_sample() / 8, - buffer()); - return true; - } - - int16* buffer() const { return buffer_.get(); } - - // Format of the output audio buffer. - const media::AudioParameters& params() const { return params_; } - - private: - media::AudioParameters params_; - scoped_ptr<media::AudioBus> audio_wrapper_; - scoped_ptr<media::AudioFifo> fifo_; - scoped_ptr<int16[]> buffer_; -}; - -scoped_refptr<WebRtcLocalAudioTrack> WebRtcLocalAudioTrack::Create( - const std::string& id, - const scoped_refptr<WebRtcAudioCapturer>& capturer, - WebAudioCapturerSource* webaudio_source, - webrtc::AudioSourceInterface* track_source, - const webrtc::MediaConstraintsInterface* constraints) { - talk_base::RefCountedObject<WebRtcLocalAudioTrack>* track = - new talk_base::RefCountedObject<WebRtcLocalAudioTrack>( - id, capturer, webaudio_source, track_source, constraints); - return track; -} - WebRtcLocalAudioTrack::WebRtcLocalAudioTrack( - const std::string& label, + WebRtcLocalAudioTrackAdapter* adapter, const scoped_refptr<WebRtcAudioCapturer>& capturer, - WebAudioCapturerSource* webaudio_source, - webrtc::AudioSourceInterface* track_source, - const webrtc::MediaConstraintsInterface* constraints) - : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label), + WebAudioCapturerSource* webaudio_source) + : MediaStreamTrack(adapter, true), + adapter_(adapter), capturer_(capturer), - webaudio_source_(webaudio_source), - track_source_(track_source), - need_audio_processing_(NeedsAudioProcessing(constraints)), - buffer_(new ConfiguredBuffer()) { + webaudio_source_(webaudio_source) { DCHECK(capturer.get() || webaudio_source); - if (!webaudio_source_) { - source_provider_.reset(new WebRtcLocalAudioSourceProvider()); - AddSink(source_provider_.get()); - } + + adapter_->Initialize(this); + DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()"; } @@ -159,69 +38,58 @@ WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() { Stop(); } -void WebRtcLocalAudioTrack::Capture(media::AudioBus* audio_source, - int audio_delay_milliseconds, +void WebRtcLocalAudioTrack::Capture(const int16* audio_data, + base::TimeDelta delay, int volume, - bool key_pressed) { + bool key_pressed, + bool need_audio_processing) { DCHECK(capture_thread_checker_.CalledOnValidThread()); + + // Calculate the signal level regardless if the track is disabled or enabled. + int signal_level = level_calculator_->Calculate( + audio_data, audio_parameters_.channels(), + audio_parameters_.frames_per_buffer()); + adapter_->SetSignalLevel(signal_level); + scoped_refptr<WebRtcAudioCapturer> capturer; - std::vector<int> voe_channels; SinkList::ItemList sinks; SinkList::ItemList sinks_to_notify_format; - bool is_webaudio_source = false; { base::AutoLock auto_lock(lock_); capturer = capturer_; - voe_channels = voe_channels_; sinks = sinks_.Items(); sinks_.RetrieveAndClearTags(&sinks_to_notify_format); - is_webaudio_source = (webaudio_source_.get() != NULL); } // Notify the tracks on when the format changes. This will do nothing if // |sinks_to_notify_format| is empty. for (SinkList::ItemList::const_iterator it = sinks_to_notify_format.begin(); it != sinks_to_notify_format.end(); ++it) { - (*it)->OnSetFormat(buffer_->params()); + (*it)->OnSetFormat(audio_parameters_); } - // Push the data to the fifo. - buffer_->Push(audio_source); - - // When the source is WebAudio, turn off the audio processing if the delay - // value is 0 even though the constraint is set to true. In such case, it - // indicates the data is not from microphone. - // TODO(xians): remove the flag when supporting one APM per audio track. - // See crbug/264611 for details. - bool need_audio_processing = need_audio_processing_; - if (is_webaudio_source && need_audio_processing) - need_audio_processing = (audio_delay_milliseconds != 0); - - int current_volume = volume; - while (buffer_->Consume()) { - // Feed the data to the sinks. - // TODO (jiayl): we should not pass the real audio data down if the track is - // disabled. This is currently done so to feed input to WebRTC typing - // detection and should be changed when audio processing is moved from - // WebRTC to the track. - for (SinkList::ItemList::const_iterator it = sinks.begin(); - it != sinks.end(); - ++it) { - int new_volume = (*it)->OnData(buffer_->buffer(), - buffer_->params().sample_rate(), - buffer_->params().channels(), - buffer_->params().frames_per_buffer(), - voe_channels, - audio_delay_milliseconds, - current_volume, - need_audio_processing, - key_pressed); - if (new_volume != 0 && capturer.get()) { - // Feed the new volume to WebRtc while changing the volume on the - // browser. - capturer->SetVolume(new_volume); - current_volume = new_volume; - } + // Feed the data to the sinks. + // TODO(jiayl): we should not pass the real audio data down if the track is + // disabled. This is currently done so to feed input to WebRTC typing + // detection and should be changed when audio processing is moved from + // WebRTC to the track. + std::vector<int> voe_channels = adapter_->VoeChannels(); + for (SinkList::ItemList::const_iterator it = sinks.begin(); + it != sinks.end(); + ++it) { + int new_volume = (*it)->OnData(audio_data, + audio_parameters_.sample_rate(), + audio_parameters_.channels(), + audio_parameters_.frames_per_buffer(), + voe_channels, + delay.InMilliseconds(), + volume, + need_audio_processing, + key_pressed); + if (new_volume != 0 && capturer.get() && !webaudio_source_) { + // Feed the new volume to WebRtc while changing the volume on the + // browser. + capturer->SetVolume(new_volume); } } } @@ -234,49 +102,22 @@ void WebRtcLocalAudioTrack::OnSetFormat( capture_thread_checker_.DetachFromThread(); DCHECK(capture_thread_checker_.CalledOnValidThread()); - DCHECK(params.IsValid()); - buffer_->Configure(params); + audio_parameters_ = params; + level_calculator_.reset(new MediaStreamAudioLevelCalculator()); base::AutoLock auto_lock(lock_); // Remember to notify all sinks of the new format. sinks_.TagAll(); } -void WebRtcLocalAudioTrack::AddChannel(int channel_id) { - DVLOG(1) << "WebRtcLocalAudioTrack::AddChannel(channel_id=" - << channel_id << ")"; - base::AutoLock auto_lock(lock_); - if (std::find(voe_channels_.begin(), voe_channels_.end(), channel_id) != - voe_channels_.end()) { - // We need to handle the case when the same channel is connected to the - // track more than once. - return; - } - - voe_channels_.push_back(channel_id); -} - -void WebRtcLocalAudioTrack::RemoveChannel(int channel_id) { - DVLOG(1) << "WebRtcLocalAudioTrack::RemoveChannel(channel_id=" - << channel_id << ")"; - base::AutoLock auto_lock(lock_); - std::vector<int>::iterator iter = - std::find(voe_channels_.begin(), voe_channels_.end(), channel_id); - DCHECK(iter != voe_channels_.end()); - voe_channels_.erase(iter); -} - -// webrtc::AudioTrackInterface implementation. -webrtc::AudioSourceInterface* WebRtcLocalAudioTrack::GetSource() const { - return track_source_; -} - -cricket::AudioRenderer* WebRtcLocalAudioTrack::GetRenderer() { - return this; -} - -std::string WebRtcLocalAudioTrack::kind() const { - return kAudioTrackKind; +void WebRtcLocalAudioTrack::SetAudioProcessor( + const scoped_refptr<MediaStreamAudioProcessor>& processor) { + // if the |processor| does not have audio processing, which can happen if + // kDisableAudioTrackProcessing is set set or all the constraints in + // the |processor| are turned off. In such case, we pass NULL to the + // adapter to indicate that no stats can be gotten from the processor. + adapter_->SetAudioProcessor(processor->has_audio_processing() ? + processor : NULL); } void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) { @@ -354,11 +195,9 @@ void WebRtcLocalAudioTrack::Start() { // capturer as its sink otherwise two streams in different clock will be // pushed through the same track. webaudio_source_->Start(this, capturer_.get()); - return; - } - - if (capturer_.get()) + } else if (capturer_.get()) { capturer_->AddTrack(this); + } SinkList::ItemList sinks; { |