| // 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/renderer/media/webrtc_audio_capturer.h" |
| #include "content/renderer/media/webrtc_audio_capturer_sink_owner.h" |
| #include "third_party/libjingle/source/talk/media/base/audiorenderer.h" |
| |
| namespace content { |
| |
| static const char kAudioTrackKind[] = "audio"; |
| |
| scoped_refptr<WebRtcLocalAudioTrack> WebRtcLocalAudioTrack::Create( |
| const std::string& id, |
| const scoped_refptr<WebRtcAudioCapturer>& capturer, |
| webrtc::AudioSourceInterface* track_source) { |
| talk_base::RefCountedObject<WebRtcLocalAudioTrack>* track = |
| new talk_base::RefCountedObject<WebRtcLocalAudioTrack>( |
| id, capturer, track_source); |
| return track; |
| } |
| |
| WebRtcLocalAudioTrack::WebRtcLocalAudioTrack( |
| const std::string& label, |
| const scoped_refptr<WebRtcAudioCapturer>& capturer, |
| webrtc::AudioSourceInterface* track_source) |
| : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label), |
| capturer_(capturer), |
| track_source_(track_source) { |
| DCHECK(capturer.get()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()"; |
| } |
| |
| WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DCHECK(sinks_.empty()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()"; |
| |
| // Users might not call Stop() on the track. |
| if (capturer_.get()) |
| Stop(); |
| } |
| |
| // Content::WebRtcAudioCapturerSink implementation. |
| void WebRtcLocalAudioTrack::CaptureData(const int16* audio_data, |
| int number_of_channels, |
| int number_of_frames, |
| int audio_delay_milliseconds, |
| double volume) { |
| SinkList sinks; |
| { |
| base::AutoLock auto_lock(lock_); |
| // When the track is diabled, we simply return here. |
| // TODO(xians): Figure out if we should feed zero to sinks instead, in |
| // order to inject VAD data in such case. |
| if (!enabled()) |
| return; |
| |
| sinks = sinks_; |
| } |
| |
| // Feed the data to the sinks. |
| for (SinkList::const_iterator it = sinks.begin(); |
| it != sinks.end(); |
| ++it) { |
| (*it)->CaptureData(audio_data, number_of_channels, number_of_frames, |
| audio_delay_milliseconds, volume); |
| } |
| } |
| |
| void WebRtcLocalAudioTrack::SetCaptureFormat( |
| const media::AudioParameters& params) { |
| base::AutoLock auto_lock(lock_); |
| params_ = params; |
| |
| // Update all the existing sinks with the new format. |
| for (SinkList::const_iterator it = sinks_.begin(); |
| it != sinks_.end(); ++it) |
| (*it)->SetCaptureFormat(params); |
| } |
| |
| // webrtc::AudioTrackInterface implementation. |
| webrtc::AudioSourceInterface* WebRtcLocalAudioTrack::GetSource() const { |
| return track_source_; |
| } |
| |
| std::string WebRtcLocalAudioTrack::kind() const { |
| return kAudioTrackKind; |
| } |
| |
| void WebRtcLocalAudioTrack::AddSink(WebRtcAudioCapturerSink* sink) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()"; |
| base::AutoLock auto_lock(lock_); |
| sink->SetCaptureFormat(params_); |
| |
| // Verify that |sink| is not already added to the list. |
| DCHECK(std::find_if( |
| sinks_.begin(), sinks_.end(), |
| WebRtcAudioCapturerSinkOwner::WrapsSink(sink)) == sinks_.end()); |
| |
| // Create (and add to the list) a new WebRtcAudioCapturerSinkOwner which owns |
| // the |sink| and delagates all calls to the WebRtcAudioCapturerSink |
| // interface. |
| sinks_.push_back(new WebRtcAudioCapturerSinkOwner(sink)); |
| } |
| |
| void WebRtcLocalAudioTrack::RemoveSink( |
| WebRtcAudioCapturerSink* sink) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()"; |
| |
| base::AutoLock auto_lock(lock_); |
| |
| // Get iterator to the first element for which WrapsSink(sink) returns true. |
| SinkList::iterator it = std::find_if( |
| sinks_.begin(), sinks_.end(), |
| WebRtcAudioCapturerSinkOwner::WrapsSink(sink)); |
| if (it != sinks_.end()) { |
| // 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. |
| (*it)->Reset(); |
| sinks_.erase(it); |
| } |
| } |
| |
| void WebRtcLocalAudioTrack::Start() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::Start()"; |
| if (capturer_.get()) |
| capturer_->AddSink(this); |
| } |
| |
| void WebRtcLocalAudioTrack::Stop() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebRtcLocalAudioTrack::Stop()"; |
| if (capturer_.get()) { |
| capturer_->RemoveSink(this); |
| capturer_ = NULL; |
| } |
| } |
| |
| } // namespace content |