| // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #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" |
| |
| namespace content { |
| |
| WebRtcLocalAudioTrack::WebRtcLocalAudioTrack( |
| WebRtcLocalAudioTrackAdapter* adapter, |
| const scoped_refptr<WebRtcAudioCapturer>& capturer, |
| WebAudioCapturerSource* webaudio_source) |
| : MediaStreamTrack(adapter, true), |
| adapter_(adapter), |
| capturer_(capturer), |
| webaudio_source_(webaudio_source) { |
| DCHECK(capturer.get() || webaudio_source); |
| |
| adapter_->Initialize(this); |
| |
| DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()"; |
| } |
| |
| WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() { |
| DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()"; |
| // Users might not call Stop() on the track. |
| Stop(); |
| } |
| |
| void WebRtcLocalAudioTrack::Capture(const int16* audio_data, |
| base::TimeDelta delay, |
| int volume, |
| 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; |
| SinkList::ItemList sinks; |
| SinkList::ItemList sinks_to_notify_format; |
| { |
| base::AutoLock auto_lock(lock_); |
| capturer = capturer_; |
| sinks = sinks_.Items(); |
| sinks_.RetrieveAndClearTags(&sinks_to_notify_format); |
| } |
| |
| // 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(audio_parameters_); |
| } |
| |
| // 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); |
| } |
| } |
| } |
| |
| void WebRtcLocalAudioTrack::OnSetFormat( |
| const media::AudioParameters& params) { |
| DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()"; |
| // If the source is restarted, we might have changed to another capture |
| // thread. |
| capture_thread_checker_.DetachFromThread(); |
| DCHECK(capture_thread_checker_.CalledOnValidThread()); |
| |
| 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::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) { |
| DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()"; |
| base::AutoLock auto_lock(lock_); |
| |
| // Verify that |sink| is not already added to the list. |
| DCHECK(!sinks_.Contains( |
| MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink))); |
| |
| // Create (and add to the list) a new MediaStreamAudioTrackSink |
| // which owns the |sink| and delagates all calls to the |
| // MediaStreamAudioSink interface. It will be tagged in the list, so |
| // we remember to call OnSetFormat() on the new sink. |
| scoped_refptr<MediaStreamAudioTrackSink> sink_owner( |
| new MediaStreamAudioSinkOwner(sink)); |
| sinks_.AddAndTag(sink_owner); |
| } |
| |
| void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) { |
| DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()"; |
| |
| base::AutoLock auto_lock(lock_); |
| |
| scoped_refptr<MediaStreamAudioTrackSink> removed_item = sinks_.Remove( |
| MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)); |
| |
| // Clear the delegate to ensure that no more capture callbacks will |
| // be sent to this sink. Also avoids a possible crash which can happen |
| // if this method is called while capturing is active. |
| if (removed_item.get()) |
| removed_item->Reset(); |
| } |
| |
| void WebRtcLocalAudioTrack::AddSink(PeerConnectionAudioSink* sink) { |
| DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()"; |
| base::AutoLock auto_lock(lock_); |
| |
| // Verify that |sink| is not already added to the list. |
| DCHECK(!sinks_.Contains( |
| MediaStreamAudioTrackSink::WrapsPeerConnectionSink(sink))); |
| |
| // Create (and add to the list) a new MediaStreamAudioTrackSink |
| // which owns the |sink| and delagates all calls to the |
| // MediaStreamAudioSink interface. It will be tagged in the list, so |
| // we remember to call OnSetFormat() on the new sink. |
| scoped_refptr<MediaStreamAudioTrackSink> sink_owner( |
| new PeerConnectionAudioSinkOwner(sink)); |
| sinks_.AddAndTag(sink_owner); |
| } |
| |
| void WebRtcLocalAudioTrack::RemoveSink(PeerConnectionAudioSink* sink) { |
| DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()"; |
| |
| base::AutoLock auto_lock(lock_); |
| |
| scoped_refptr<MediaStreamAudioTrackSink> removed_item = sinks_.Remove( |
| MediaStreamAudioTrackSink::WrapsPeerConnectionSink(sink)); |
| // Clear the delegate to ensure that no more capture callbacks will |
| // be sent to this sink. Also avoids a possible crash which can happen |
| // if this method is called while capturing is active. |
| if (removed_item.get()) |
| removed_item->Reset(); |
| } |
| |
| void WebRtcLocalAudioTrack::Start() { |
| DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::Start()"; |
| if (webaudio_source_.get()) { |
| // If the track is hooking up with WebAudio, do NOT add the track to the |
| // capturer as its sink otherwise two streams in different clock will be |
| // pushed through the same track. |
| webaudio_source_->Start(this, capturer_.get()); |
| } else if (capturer_.get()) { |
| capturer_->AddTrack(this); |
| } |
| |
| SinkList::ItemList sinks; |
| { |
| base::AutoLock auto_lock(lock_); |
| sinks = sinks_.Items(); |
| } |
| for (SinkList::ItemList::const_iterator it = sinks.begin(); |
| it != sinks.end(); |
| ++it) { |
| (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateLive); |
| } |
| } |
| |
| void WebRtcLocalAudioTrack::Stop() { |
| DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::Stop()"; |
| if (!capturer_.get() && !webaudio_source_.get()) |
| return; |
| |
| if (webaudio_source_.get()) { |
| // Called Stop() on the |webaudio_source_| explicitly so that |
| // |webaudio_source_| won't push more data to the track anymore. |
| // Also note that the track is not registered as a sink to the |capturer_| |
| // in such case and no need to call RemoveTrack(). |
| webaudio_source_->Stop(); |
| } else { |
| // It is necessary to call RemoveTrack on the |capturer_| to avoid getting |
| // audio callback after Stop(). |
| capturer_->RemoveTrack(this); |
| } |
| |
| // Protect the pointers using the lock when accessing |sinks_| and |
| // setting the |capturer_| to NULL. |
| SinkList::ItemList sinks; |
| { |
| base::AutoLock auto_lock(lock_); |
| sinks = sinks_.Items(); |
| sinks_.Clear(); |
| webaudio_source_ = NULL; |
| capturer_ = NULL; |
| } |
| |
| for (SinkList::ItemList::const_iterator it = sinks.begin(); |
| it != sinks.end(); |
| ++it){ |
| (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded); |
| (*it)->Reset(); |
| } |
| } |
| |
| } // namespace content |