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| 1 // Copyright 2016 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/webrtc_audio_sink.h" | |
| 6 | |
| 7 #include <algorithm> | |
| 8 #include <limits> | |
| 9 | |
| 10 #include "base/bind.h" | |
| 11 #include "base/bind_helpers.h" | |
| 12 #include "base/location.h" | |
| 13 #include "base/logging.h" | |
| 14 #include "base/message_loop/message_loop.h" | |
| 15 | |
| 16 namespace content { | |
| 17 | |
| 18 WebRtcAudioSink::WebRtcAudioSink( | |
| 19 const std::string& label, | |
| 20 scoped_refptr<webrtc::AudioSourceInterface> track_source, | |
| 21 scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner) | |
| 22 : adapter_(new rtc::RefCountedObject<Adapter>( | |
| 23 label, std::move(track_source), std::move(signaling_task_runner))), | |
| 24 fifo_(base::Bind(&WebRtcAudioSink::DeliverRebufferedAudio, | |
| 25 base::Unretained(this))) { | |
| 26 DVLOG(1) << "WebRtcAudioSink::WebRtcAudioSink()"; | |
| 27 } | |
| 28 | |
| 29 WebRtcAudioSink::~WebRtcAudioSink() { | |
| 30 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 31 DVLOG(1) << "WebRtcAudioSink::~WebRtcAudioSink()"; | |
| 32 } | |
| 33 | |
| 34 void WebRtcAudioSink::SetAudioProcessor( | |
| 35 scoped_refptr<MediaStreamAudioProcessor> processor) { | |
| 36 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 37 DCHECK(processor.get()); | |
| 38 adapter_->set_processor(std::move(processor)); | |
| 39 } | |
| 40 | |
| 41 void WebRtcAudioSink::SetLevel( | |
| 42 scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) { | |
| 43 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 44 DCHECK(level.get()); | |
| 45 adapter_->set_level(std::move(level)); | |
| 46 } | |
| 47 | |
| 48 void WebRtcAudioSink::OnEnabledChanged(bool enabled) { | |
| 49 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 50 adapter_->signaling_task_runner()->PostTask( | |
| 51 FROM_HERE, | |
| 52 base::Bind( | |
| 53 base::IgnoreResult(&WebRtcAudioSink::Adapter::set_enabled), | |
| 54 adapter_, enabled)); | |
| 55 } | |
| 56 | |
| 57 void WebRtcAudioSink::OnData(const media::AudioBus& audio_bus, | |
| 58 base::TimeTicks estimated_capture_time) { | |
| 59 DCHECK(audio_thread_checker_.CalledOnValidThread()); | |
| 60 // The following will result in zero, one, or multiple synchronous calls to | |
| 61 // DeliverRebufferedAudio(). | |
| 62 fifo_.Push(audio_bus); | |
| 63 } | |
| 64 | |
| 65 void WebRtcAudioSink::OnSetFormat(const media::AudioParameters& params) { | |
| 66 // On a format change, the thread delivering audio might have also changed. | |
| 67 audio_thread_checker_.DetachFromThread(); | |
| 68 DCHECK(audio_thread_checker_.CalledOnValidThread()); | |
| 69 | |
| 70 DCHECK(params.IsValid()); | |
| 71 params_ = params; | |
| 72 fifo_.Reset(params_.frames_per_buffer()); | |
| 73 const int num_pcm16_data_elements = | |
| 74 params_.frames_per_buffer() * params_.channels(); | |
| 75 interleaved_data_.reset(new int16_t[num_pcm16_data_elements]); | |
| 76 } | |
| 77 | |
| 78 void WebRtcAudioSink::DeliverRebufferedAudio(const media::AudioBus& audio_bus, | |
| 79 int frame_delay) { | |
| 80 DCHECK(audio_thread_checker_.CalledOnValidThread()); | |
| 81 DCHECK(params_.IsValid()); | |
| 82 | |
| 83 // TODO(miu): Why doesn't a WebRTC sink care about reference time passed to | |
| 84 // OnData(), and the |frame_delay| here? How is AV sync achieved otherwise? | |
| 85 | |
| 86 // TODO(henrika): Remove this conversion once the interface in libjingle | |
| 87 // supports float vectors. | |
| 88 audio_bus.ToInterleaved(audio_bus.frames(), | |
| 89 sizeof(interleaved_data_[0]), | |
| 90 interleaved_data_.get()); | |
| 91 adapter_->DeliverPCMToWebRtcSinks(interleaved_data_.get(), | |
| 92 params_.sample_rate(), | |
| 93 audio_bus.channels(), | |
| 94 audio_bus.frames()); | |
| 95 } | |
| 96 | |
| 97 namespace { | |
| 98 // TODO(miu): MediaStreamAudioProcessor destructor requires this nonsense. | |
| 99 void DereferenceOnMainThread( | |
| 100 const scoped_refptr<MediaStreamAudioProcessor>& processor) {} | |
| 101 } // namespace | |
| 102 | |
| 103 WebRtcAudioSink::Adapter::Adapter( | |
| 104 const std::string& label, | |
| 105 scoped_refptr<webrtc::AudioSourceInterface> source, | |
| 106 scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner) | |
| 107 : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label), | |
| 108 source_(std::move(source)), | |
| 109 signaling_task_runner_(std::move(signaling_task_runner)), | |
| 110 main_task_runner_(base::MessageLoop::current()->task_runner()) { | |
| 111 DCHECK(signaling_task_runner_); | |
| 112 } | |
| 113 | |
| 114 WebRtcAudioSink::Adapter::~Adapter() { | |
| 115 if (audio_processor_) { | |
| 116 main_task_runner_->PostTask( | |
| 117 FROM_HERE, | |
| 118 base::Bind(&DereferenceOnMainThread, std::move(audio_processor_))); | |
| 119 } | |
| 120 } | |
| 121 | |
| 122 void WebRtcAudioSink::Adapter::DeliverPCMToWebRtcSinks( | |
| 123 const int16_t* audio_data, | |
| 124 int sample_rate, | |
| 125 size_t number_of_channels, | |
| 126 size_t number_of_frames) { | |
| 127 base::AutoLock auto_lock(lock_); | |
| 128 for (webrtc::AudioTrackSinkInterface* sink : sinks_) { | |
| 129 sink->OnData(audio_data, sizeof(int16_t) * 8, sample_rate, | |
| 130 number_of_channels, number_of_frames); | |
| 131 } | |
| 132 } | |
| 133 | |
| 134 std::string WebRtcAudioSink::Adapter::kind() const { | |
| 135 return webrtc::MediaStreamTrackInterface::kAudioKind; | |
| 136 } | |
| 137 | |
| 138 bool WebRtcAudioSink::Adapter::set_enabled(bool enable) { | |
| 139 DCHECK(!signaling_task_runner_ || | |
| 140 signaling_task_runner_->RunsTasksOnCurrentThread()); | |
| 141 return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>:: | |
| 142 set_enabled(enable); | |
| 143 } | |
| 144 | |
| 145 void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) { | |
| 146 DCHECK(!signaling_task_runner_ || | |
| 147 signaling_task_runner_->RunsTasksOnCurrentThread()); | |
| 148 DCHECK(sink); | |
| 149 base::AutoLock auto_lock(lock_); | |
| 150 DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end()); | |
| 151 sinks_.push_back(sink); | |
| 152 } | |
| 153 | |
| 154 void WebRtcAudioSink::Adapter::RemoveSink( | |
| 155 webrtc::AudioTrackSinkInterface* sink) { | |
| 156 DCHECK(!signaling_task_runner_ || | |
| 157 signaling_task_runner_->RunsTasksOnCurrentThread()); | |
| 158 base::AutoLock auto_lock(lock_); | |
| 159 const auto it = std::find(sinks_.begin(), sinks_.end(), sink); | |
| 160 if (it != sinks_.end()) | |
| 161 sinks_.erase(it); | |
| 162 } | |
| 163 | |
| 164 bool WebRtcAudioSink::Adapter::GetSignalLevel(int* level) { | |
| 165 DCHECK(!signaling_task_runner_ || | |
| 166 signaling_task_runner_->RunsTasksOnCurrentThread()); | |
| 167 | |
| 168 // |level_| is only set once, so it's safe to read without first acquiring a | |
| 169 // mutex. | |
| 170 if (!level_) | |
| 171 return false; | |
| 172 const float signal_level = level_->GetCurrent(); | |
| 173 DCHECK_GE(signal_level, 0.0f); | |
| 174 DCHECK_LE(signal_level, 1.0f); | |
| 175 // Convert from float in range [0.0,1.0] to an int in range [0,32767]. | |
| 176 *level = static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() + | |
| 177 0.5f /* rounding to nearest int */); | |
| 178 return true; | |
| 179 } | |
| 180 | |
| 181 rtc::scoped_refptr<webrtc::AudioProcessorInterface> | |
| 182 WebRtcAudioSink::Adapter::GetAudioProcessor() { | |
| 183 DCHECK(!signaling_task_runner_ || | |
| 184 signaling_task_runner_->RunsTasksOnCurrentThread()); | |
| 185 return audio_processor_.get(); | |
| 186 } | |
| 187 | |
| 188 webrtc::AudioSourceInterface* WebRtcAudioSink::Adapter::GetSource() const { | |
| 189 DCHECK(!signaling_task_runner_ || | |
| 190 signaling_task_runner_->RunsTasksOnCurrentThread()); | |
| 191 return source_.get(); | |
| 192 } | |
| 193 | |
| 194 } // namespace content | |
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Chromium Code Reviews
Issue 1966043006:
Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master