content/renderer/media/webrtc_local_audio_renderer.cc

   1 // Copyright (c) 2012 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_local_audio_renderer.h"
   6
   7 #include "base/logging.h"
   8 #include "base/message_loop/message_loop_proxy.h"
   9 #include "base/metrics/histogram.h"
  10 #include "base/synchronization/lock.h"
  11 #include "base/trace_event/trace_event.h"
  12 #include "content/renderer/media/audio_device_factory.h"
  13 #include "content/renderer/media/media_stream_dispatcher.h"
  14 #include "content/renderer/media/webrtc_audio_capturer.h"
  15 #include "content/renderer/media/webrtc_audio_renderer.h"
  16 #include "content/renderer/render_frame_impl.h"
  17 #include "media/audio/audio_output_device.h"
  18 #include "media/base/audio_bus.h"
  19 #include "media/base/audio_shifter.h"
  20
  21 namespace content {
  22
  23 namespace {
  24
  25 enum LocalRendererSinkStates {
  26   kSinkStarted = 0,
  27   kSinkNeverStarted,
  28   kSinkStatesMax  // Must always be last!
  29 };
  30
  31 }  // namespace
  32
  33 // media::AudioRendererSink::RenderCallback implementation
  34 int WebRtcLocalAudioRenderer::Render(
  35     media::AudioBus* audio_bus, int audio_delay_milliseconds) {
  36   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
  37   base::AutoLock auto_lock(thread_lock_);
  38
  39   if (!playing_ || !volume_ || !audio_shifter_) {
  40     audio_bus->Zero();
  41     return 0;
  42   }
  43
  44   audio_shifter_->Pull(
  45       audio_bus,
  46       base::TimeTicks::Now() -
  47       base::TimeDelta::FromMilliseconds(audio_delay_milliseconds));
  48
  49   return audio_bus->frames();
  50 }
  51
  52 void WebRtcLocalAudioRenderer::OnRenderError() {
  53   NOTIMPLEMENTED();
  54 }
  55
  56 // content::MediaStreamAudioSink implementation
  57 void WebRtcLocalAudioRenderer::OnData(const media::AudioBus& audio_bus,
  58                                       base::TimeTicks estimated_capture_time) {
  59   DCHECK(capture_thread_checker_.CalledOnValidThread());
  60   DCHECK(!estimated_capture_time.is_null());
  61
  62   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
  63
  64   base::AutoLock auto_lock(thread_lock_);
  65   if (!playing_ || !volume_ || !audio_shifter_)
  66     return;
  67
  68   scoped_ptr<media::AudioBus> audio_data(
  69       media::AudioBus::Create(audio_bus.channels(), audio_bus.frames()));
  70   audio_bus.CopyTo(audio_data.get());
  71   audio_shifter_->Push(audio_data.Pass(), estimated_capture_time);
  72   const base::TimeTicks now = base::TimeTicks::Now();
  73   total_render_time_ += now - last_render_time_;
  74   last_render_time_ = now;
  75 }
  76
  77 void WebRtcLocalAudioRenderer::OnSetFormat(
  78     const media::AudioParameters& params) {
  79   DVLOG(1) << "WebRtcLocalAudioRenderer::OnSetFormat()";
  80   // If the source is restarted, we might have changed to another capture
  81   // thread.
  82   capture_thread_checker_.DetachFromThread();
  83   DCHECK(capture_thread_checker_.CalledOnValidThread());
  84
  85   // Post a task on the main render thread to reconfigure the |sink_| with the
  86   // new format.
  87   message_loop_->PostTask(
  88       FROM_HERE,
  89       base::Bind(&WebRtcLocalAudioRenderer::ReconfigureSink, this,
  90                  params));
  91 }
  92
  93 // WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer implementation.
  94 WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer(
  95     const blink::WebMediaStreamTrack& audio_track,
  96     int source_render_frame_id,
  97     int session_id,
  98     int frames_per_buffer)
  99     : audio_track_(audio_track),
 100       source_render_frame_id_(source_render_frame_id),
 101       session_id_(session_id),
 102       message_loop_(base::MessageLoopProxy::current()),
 103       playing_(false),
 104       frames_per_buffer_(frames_per_buffer),
 105       volume_(0.0),
 106       sink_started_(false) {
 107   DVLOG(1) << "WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer()";
 108 }
 109
 110 WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer() {
 111   DCHECK(message_loop_->BelongsToCurrentThread());
 112   DCHECK(!sink_.get());
 113   DVLOG(1) << "WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer()";
 114 }
 115
 116 void WebRtcLocalAudioRenderer::Start() {
 117   DVLOG(1) << "WebRtcLocalAudioRenderer::Start()";
 118   DCHECK(message_loop_->BelongsToCurrentThread());
 119
 120   // We get audio data from |audio_track_|...
 121   MediaStreamAudioSink::AddToAudioTrack(this, audio_track_);
 122   // ...and |sink_| will get audio data from us.
 123   DCHECK(!sink_.get());
 124   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
 125
 126   base::AutoLock auto_lock(thread_lock_);
 127   last_render_time_ = base::TimeTicks::Now();
 128   playing_ = false;
 129 }
 130
 131 void WebRtcLocalAudioRenderer::Stop() {
 132   DVLOG(1) << "WebRtcLocalAudioRenderer::Stop()";
 133   DCHECK(message_loop_->BelongsToCurrentThread());
 134
 135   {
 136     base::AutoLock auto_lock(thread_lock_);
 137     playing_ = false;
 138     audio_shifter_.reset();
 139   }
 140
 141   // Stop the output audio stream, i.e, stop asking for data to render.
 142   // It is safer to call Stop() on the |sink_| to clean up the resources even
 143   // when the |sink_| is never started.
 144   if (sink_.get()) {
 145     sink_->Stop();
 146     sink_ = NULL;
 147   }
 148
 149   if (!sink_started_) {
 150     UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
 151                               kSinkNeverStarted, kSinkStatesMax);
 152   }
 153   sink_started_ = false;
 154
 155   // Ensure that the capturer stops feeding us with captured audio.
 156   MediaStreamAudioSink::RemoveFromAudioTrack(this, audio_track_);
 157 }
 158
 159 void WebRtcLocalAudioRenderer::Play() {
 160   DVLOG(1) << "WebRtcLocalAudioRenderer::Play()";
 161   DCHECK(message_loop_->BelongsToCurrentThread());
 162
 163   if (!sink_.get())
 164     return;
 165
 166   {
 167     base::AutoLock auto_lock(thread_lock_);
 168     // Resumes rendering by ensuring that WebRtcLocalAudioRenderer::Render()
 169     // now reads data from the local FIFO.
 170     playing_ = true;
 171     last_render_time_ = base::TimeTicks::Now();
 172   }
 173
 174   // Note: If volume_ is currently muted, the |sink_| will not be started yet.
 175   MaybeStartSink();
 176 }
 177
 178 void WebRtcLocalAudioRenderer::Pause() {
 179   DVLOG(1) << "WebRtcLocalAudioRenderer::Pause()";
 180   DCHECK(message_loop_->BelongsToCurrentThread());
 181
 182   if (!sink_.get())
 183     return;
 184
 185   base::AutoLock auto_lock(thread_lock_);
 186   // Temporarily suspends rendering audio.
 187   // WebRtcLocalAudioRenderer::Render() will return early during this state
 188   // and only zeros will be provided to the active sink.
 189   playing_ = false;
 190 }
 191
 192 void WebRtcLocalAudioRenderer::SetVolume(float volume) {
 193   DVLOG(1) << "WebRtcLocalAudioRenderer::SetVolume(" << volume << ")";
 194   DCHECK(message_loop_->BelongsToCurrentThread());
 195
 196   {
 197     base::AutoLock auto_lock(thread_lock_);
 198     // Cache the volume.
 199     volume_ = volume;
 200   }
 201
 202   // Lazily start the |sink_| when the local renderer is unmuted during
 203   // playing.
 204   MaybeStartSink();
 205
 206   if (sink_.get())
 207     sink_->SetVolume(volume);
 208 }
 209
 210 base::TimeDelta WebRtcLocalAudioRenderer::GetCurrentRenderTime() const {
 211   DCHECK(message_loop_->BelongsToCurrentThread());
 212   base::AutoLock auto_lock(thread_lock_);
 213   if (!sink_.get())
 214     return base::TimeDelta();
 215   return total_render_time();
 216 }
 217
 218 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const {
 219   return true;
 220 }
 221
 222 void WebRtcLocalAudioRenderer::MaybeStartSink() {
 223   DCHECK(message_loop_->BelongsToCurrentThread());
 224   DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink()";
 225
 226   if (!sink_.get() || !source_params_.IsValid())
 227     return;
 228
 229   {
 230     // Clear up the old data in the FIFO.
 231     base::AutoLock auto_lock(thread_lock_);
 232     audio_shifter_->Flush();
 233   }
 234
 235   if (!sink_params_.IsValid() || !playing_ || !volume_ || sink_started_)
 236     return;
 237
 238   DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_.";
 239   sink_->InitializeWithSessionId(sink_params_, this, session_id_);
 240   sink_->Start();
 241   sink_started_ = true;
 242   UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
 243                             kSinkStarted, kSinkStatesMax);
 244 }
 245
 246 void WebRtcLocalAudioRenderer::ReconfigureSink(
 247     const media::AudioParameters& params) {
 248   DCHECK(message_loop_->BelongsToCurrentThread());
 249
 250   DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()";
 251
 252   int implicit_ducking_effect = 0;
 253   RenderFrameImpl* const frame =
 254       RenderFrameImpl::FromRoutingID(source_render_frame_id_);
 255   MediaStreamDispatcher* const dispatcher = frame ?
 256       frame->GetMediaStreamDispatcher() : NULL;
 257   if (dispatcher && dispatcher->IsAudioDuckingActive()) {
 258     DVLOG(1) << "Forcing DUCKING to be ON for output";
 259     implicit_ducking_effect = media::AudioParameters::DUCKING;
 260   } else {
 261     DVLOG(1) << "DUCKING not forced ON for output";
 262   }
 263
 264   if (source_params_.Equals(params))
 265     return;
 266
 267   // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
 268   // the new format.
 269
 270   source_params_ = params;
 271
 272   sink_params_ = media::AudioParameters(source_params_.format(),
 273       source_params_.channel_layout(), source_params_.sample_rate(),
 274       source_params_.bits_per_sample(),
 275       WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
 276                                                 frames_per_buffer_),
 277       // If DUCKING is enabled on the source, it needs to be enabled on the
 278       // sink as well.
 279       source_params_.effects() | implicit_ducking_effect);
 280
 281   {
 282     // Note: The max buffer is fairly large, but will rarely be used.
 283     // Cast needs the buffer to hold at least one second of audio.
 284     // The clock accuracy is set to 20ms because clock accuracy is
 285     // ~15ms on windows.
 286     media::AudioShifter* const new_shifter = new media::AudioShifter(
 287         base::TimeDelta::FromSeconds(2),
 288         base::TimeDelta::FromMilliseconds(20),
 289         base::TimeDelta::FromSeconds(20),
 290         source_params_.sample_rate(),
 291         params.channels());
 292
 293     base::AutoLock auto_lock(thread_lock_);
 294     audio_shifter_.reset(new_shifter);
 295   }
 296
 297   if (!sink_.get())
 298     return;  // WebRtcLocalAudioRenderer has not yet been started.
 299
 300   // Stop |sink_| and re-create a new one to be initialized with different audio
 301   // parameters.  Then, invoke MaybeStartSink() to restart everything again.
 302   if (sink_started_) {
 303     sink_->Stop();
 304     sink_started_ = false;
 305   }
 306
 307   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
 308   MaybeStartSink();
 309 }
 310
 311 }  // namespace content