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| 1 // Copyright 2015 The Chromium Authors. All rights reserved. | |
| 2 // Use of this source code is governed by a BSD-style license that can be | |
| 3 // found in the LICENSE file. | |
| 4 | |
| 5 #include "content/renderer/media/webrtc/media_stream_remote_audio_track.h" | |
| 6 | |
| 7 #include <stddef.h> | |
| 8 | |
| 9 #include <list> | |
| 10 | |
| 11 #include "base/logging.h" | |
| 12 #include "content/public/renderer/media_stream_audio_sink.h" | |
| 13 #include "third_party/webrtc/api/mediastreaminterface.h" | |
| 14 | |
| 15 namespace content { | |
| 16 | |
| 17 class MediaStreamRemoteAudioSource::AudioSink | |
| 18 : public webrtc::AudioTrackSinkInterface { | |
| 19 public: | |
| 20 AudioSink() { | |
| 21 } | |
| 22 ~AudioSink() override { | |
| 23 DCHECK(sinks_.empty()); | |
| 24 } | |
| 25 | |
| 26 void Add(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track, | |
| 27 bool enabled) { | |
| 28 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 29 SinkInfo info(sink, track, enabled); | |
| 30 base::AutoLock lock(lock_); | |
| 31 sinks_.push_back(info); | |
| 32 } | |
| 33 | |
| 34 void Remove(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track) { | |
| 35 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 36 base::AutoLock lock(lock_); | |
| 37 sinks_.remove_if([&sink, &track](const SinkInfo& info) { | |
| 38 return info.sink == sink && info.track == track; | |
| 39 }); | |
| 40 } | |
| 41 | |
| 42 void SetEnabled(MediaStreamAudioTrack* track, bool enabled) { | |
| 43 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 44 base::AutoLock lock(lock_); | |
| 45 for (SinkInfo& info : sinks_) { | |
| 46 if (info.track == track) | |
| 47 info.enabled = enabled; | |
| 48 } | |
| 49 } | |
| 50 | |
| 51 void RemoveAll(MediaStreamAudioTrack* track) { | |
| 52 base::AutoLock lock(lock_); | |
| 53 sinks_.remove_if([&track](const SinkInfo& info) { | |
| 54 return info.track == track; | |
| 55 }); | |
| 56 } | |
| 57 | |
| 58 bool IsNeeded() const { | |
| 59 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 60 return !sinks_.empty(); | |
| 61 } | |
| 62 | |
| 63 private: | |
| 64 void OnData(const void* audio_data, int bits_per_sample, int sample_rate, | |
| 65 size_t number_of_channels, size_t number_of_frames) override { | |
| 66 if (!audio_bus_ || | |
| 67 static_cast<size_t>(audio_bus_->channels()) != number_of_channels || | |
| 68 static_cast<size_t>(audio_bus_->frames()) != number_of_frames) { | |
| 69 audio_bus_ = media::AudioBus::Create(number_of_channels, | |
| 70 number_of_frames); | |
| 71 } | |
| 72 | |
| 73 audio_bus_->FromInterleaved(audio_data, number_of_frames, | |
| 74 bits_per_sample / 8); | |
| 75 | |
| 76 bool format_changed = false; | |
| 77 if (params_.format() != media::AudioParameters::AUDIO_PCM_LOW_LATENCY || | |
| 78 static_cast<size_t>(params_.channels()) != number_of_channels || | |
| 79 params_.sample_rate() != sample_rate || | |
| 80 static_cast<size_t>(params_.frames_per_buffer()) != number_of_frames) { | |
| 81 params_ = media::AudioParameters( | |
| 82 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
| 83 media::GuessChannelLayout(number_of_channels), | |
| 84 sample_rate, 16, number_of_frames); | |
| 85 format_changed = true; | |
| 86 } | |
| 87 | |
| 88 // TODO(tommi): We should get the timestamp from WebRTC. | |
| 89 base::TimeTicks estimated_capture_time(base::TimeTicks::Now()); | |
| 90 | |
| 91 base::AutoLock lock(lock_); | |
| 92 for (const SinkInfo& info : sinks_) { | |
| 93 if (info.enabled) { | |
| 94 if (format_changed) | |
| 95 info.sink->OnSetFormat(params_); | |
| 96 info.sink->OnData(*audio_bus_.get(), estimated_capture_time); | |
| 97 } | |
| 98 } | |
| 99 } | |
| 100 | |
| 101 mutable base::Lock lock_; | |
| 102 struct SinkInfo { | |
| 103 SinkInfo(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track, | |
| 104 bool enabled) : sink(sink), track(track), enabled(enabled) {} | |
| 105 MediaStreamAudioSink* sink; | |
| 106 MediaStreamAudioTrack* track; | |
| 107 bool enabled; | |
| 108 }; | |
| 109 std::list<SinkInfo> sinks_; | |
| 110 base::ThreadChecker thread_checker_; | |
| 111 media::AudioParameters params_; // Only used on the callback thread. | |
| 112 scoped_ptr<media::AudioBus> audio_bus_; // Only used on the callback thread. | |
| 113 }; | |
| 114 | |
| 115 MediaStreamRemoteAudioTrack::MediaStreamRemoteAudioTrack( | |
| 116 const blink::WebMediaStreamSource& source, bool enabled) | |
| 117 : MediaStreamAudioTrack(false), source_(source), enabled_(enabled) { | |
| 118 DCHECK(source.extraData()); // Make sure the source has a native source. | |
| 119 } | |
| 120 | |
| 121 MediaStreamRemoteAudioTrack::~MediaStreamRemoteAudioTrack() { | |
| 122 DCHECK(main_render_thread_checker_.CalledOnValidThread()); | |
| 123 source()->RemoveAll(this); | |
| 124 } | |
| 125 | |
| 126 void MediaStreamRemoteAudioTrack::SetEnabled(bool enabled) { | |
| 127 DCHECK(main_render_thread_checker_.CalledOnValidThread()); | |
| 128 | |
| 129 // This affects the shared state of the source for whether or not it's a part | |
| 130 // of the mixed audio that's rendered for remote tracks from WebRTC. | |
| 131 // All tracks from the same source will share this state and thus can step | |
| 132 // on each other's toes. | |
| 133 // This is also why we can't check the |enabled_| state for equality with | |
| 134 // |enabled| before setting the mixing enabled state. |enabled_| and the | |
| 135 // shared state might not be the same. | |
| 136 source()->SetEnabledForMixing(enabled); | |
| 137 | |
| 138 enabled_ = enabled; | |
| 139 source()->SetSinksEnabled(this, enabled); | |
| 140 } | |
| 141 | |
| 142 void MediaStreamRemoteAudioTrack::Stop() { | |
| 143 DCHECK(main_render_thread_checker_.CalledOnValidThread()); | |
| 144 // Stop means that a track should be stopped permanently. But | |
| 145 // since there is no proper way of doing that on a remote track, we can | |
| 146 // at least disable the track. Blink will not call down to the content layer | |
| 147 // after a track has been stopped. | |
| 148 SetEnabled(false); | |
| 149 } | |
| 150 | |
| 151 void MediaStreamRemoteAudioTrack::AddSink(MediaStreamAudioSink* sink) { | |
| 152 DCHECK(main_render_thread_checker_.CalledOnValidThread()); | |
| 153 return source()->AddSink(sink, this, enabled_); | |
| 154 } | |
| 155 | |
| 156 void MediaStreamRemoteAudioTrack::RemoveSink(MediaStreamAudioSink* sink) { | |
| 157 DCHECK(main_render_thread_checker_.CalledOnValidThread()); | |
| 158 return source()->RemoveSink(sink, this); | |
| 159 } | |
| 160 | |
| 161 media::AudioParameters MediaStreamRemoteAudioTrack::GetOutputFormat() const { | |
| 162 DCHECK(main_render_thread_checker_.CalledOnValidThread()); | |
| 163 // This method is not implemented on purpose and should be removed. | |
| 164 // TODO(tommi): See comment for GetOutputFormat in MediaStreamAudioTrack. | |
| 165 NOTIMPLEMENTED(); | |
| 166 return media::AudioParameters(); | |
| 167 } | |
| 168 | |
| 169 webrtc::AudioTrackInterface* MediaStreamRemoteAudioTrack::GetAudioAdapter() { | |
| 170 DCHECK(main_render_thread_checker_.CalledOnValidThread()); | |
| 171 return source()->GetAudioAdapter(); | |
| 172 } | |
| 173 | |
| 174 MediaStreamRemoteAudioSource* MediaStreamRemoteAudioTrack::source() const { | |
| 175 return static_cast<MediaStreamRemoteAudioSource*>(source_.extraData()); | |
| 176 } | |
| 177 | |
| 178 MediaStreamRemoteAudioSource::MediaStreamRemoteAudioSource( | |
| 179 const scoped_refptr<webrtc::AudioTrackInterface>& track) : track_(track) {} | |
| 180 | |
| 181 MediaStreamRemoteAudioSource::~MediaStreamRemoteAudioSource() { | |
| 182 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 183 } | |
| 184 | |
| 185 void MediaStreamRemoteAudioSource::SetEnabledForMixing(bool enabled) { | |
| 186 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 187 track_->set_enabled(enabled); | |
| 188 } | |
| 189 | |
| 190 void MediaStreamRemoteAudioSource::AddSink(MediaStreamAudioSink* sink, | |
| 191 MediaStreamAudioTrack* track, | |
| 192 bool enabled) { | |
| 193 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 194 if (!sink_) { | |
| 195 sink_.reset(new AudioSink()); | |
| 196 track_->AddSink(sink_.get()); | |
| 197 } | |
| 198 | |
| 199 sink_->Add(sink, track, enabled); | |
| 200 } | |
| 201 | |
| 202 void MediaStreamRemoteAudioSource::RemoveSink(MediaStreamAudioSink* sink, | |
| 203 MediaStreamAudioTrack* track) { | |
| 204 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 205 DCHECK(sink_); | |
| 206 | |
| 207 sink_->Remove(sink, track); | |
| 208 | |
| 209 if (!sink_->IsNeeded()) { | |
| 210 track_->RemoveSink(sink_.get()); | |
| 211 sink_.reset(); | |
| 212 } | |
| 213 } | |
| 214 | |
| 215 void MediaStreamRemoteAudioSource::SetSinksEnabled(MediaStreamAudioTrack* track, | |
| 216 bool enabled) { | |
| 217 if (sink_) | |
| 218 sink_->SetEnabled(track, enabled); | |
| 219 } | |
| 220 | |
| 221 void MediaStreamRemoteAudioSource::RemoveAll(MediaStreamAudioTrack* track) { | |
| 222 if (sink_) | |
| 223 sink_->RemoveAll(track); | |
| 224 } | |
| 225 | |
| 226 webrtc::AudioTrackInterface* MediaStreamRemoteAudioSource::GetAudioAdapter() { | |
| 227 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 228 return track_.get(); | |
| 229 } | |
| 230 | |
| 231 } // namespace content | |
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Chromium Code Reviews
Issue 1647773002:
MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master