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/location.h"
   8 #include "base/logging.h"
   9 #include "base/metrics/histogram.h"
  10 #include "base/synchronization/lock.h"
  11 #include "base/thread_task_runner_handle.h"
  12 #include "base/trace_event/trace_event.h"
  13 #include "content/renderer/media/audio_device_factory.h"
  14 #include "content/renderer/media/media_stream_dispatcher.h"
  15 #include "content/renderer/media/webrtc_audio_capturer.h"
  16 #include "content/renderer/media/webrtc_audio_renderer.h"
  17 #include "content/renderer/render_frame_impl.h"
  18 #include "media/audio/audio_output_device.h"
  19 #include "media/base/audio_bus.h"
  20 #include "media/base/audio_shifter.h"
  21
  22 namespace content {
  23
  24 namespace {
  25
  26 enum LocalRendererSinkStates {
  27   kSinkStarted = 0,
  28   kSinkNeverStarted,
  29   kSinkStatesMax  // Must always be last!
  30 };
  31
  32 }  // namespace
  33
  34 // media::AudioRendererSink::RenderCallback implementation
  35 int WebRtcLocalAudioRenderer::Render(
  36     media::AudioBus* audio_bus, int audio_delay_milliseconds) {
  37   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
  38   base::AutoLock auto_lock(thread_lock_);
  39
  40   if (!playing_ || !volume_ || !audio_shifter_) {
  41     audio_bus->Zero();
  42     return 0;
  43   }
  44
  45   audio_shifter_->Pull(
  46       audio_bus,
  47       base::TimeTicks::Now() -
  48       base::TimeDelta::FromMilliseconds(audio_delay_milliseconds));
  49
  50   return audio_bus->frames();
  51 }
  52
  53 void WebRtcLocalAudioRenderer::OnRenderError() {
  54   NOTIMPLEMENTED();
  55 }
  56
  57 // content::MediaStreamAudioSink implementation
  58 void WebRtcLocalAudioRenderer::OnData(const media::AudioBus& audio_bus,
  59                                       base::TimeTicks estimated_capture_time) {
  60   DCHECK(capture_thread_checker_.CalledOnValidThread());
  61   DCHECK(!estimated_capture_time.is_null());
  62
  63   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
  64
  65   base::AutoLock auto_lock(thread_lock_);
  66   if (!playing_ || !volume_ || !audio_shifter_)
  67     return;
  68
  69   scoped_ptr<media::AudioBus> audio_data(
  70       media::AudioBus::Create(audio_bus.channels(), audio_bus.frames()));
  71   audio_bus.CopyTo(audio_data.get());
  72   audio_shifter_->Push(audio_data.Pass(), estimated_capture_time);
  73   const base::TimeTicks now = base::TimeTicks::Now();
  74   total_render_time_ += now - last_render_time_;
  75   last_render_time_ = now;
  76 }
  77
  78 void WebRtcLocalAudioRenderer::OnSetFormat(
  79     const media::AudioParameters& params) {
  80   DVLOG(1) << "WebRtcLocalAudioRenderer::OnSetFormat()";
  81   // If the source is restarted, we might have changed to another capture
  82   // thread.
  83   capture_thread_checker_.DetachFromThread();
  84   DCHECK(capture_thread_checker_.CalledOnValidThread());
  85
  86   // Post a task on the main render thread to reconfigure the |sink_| with the
  87   // new format.
  88   task_runner_->PostTask(
  89       FROM_HERE,
  90       base::Bind(&WebRtcLocalAudioRenderer::ReconfigureSink, this, 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       task_runner_(base::ThreadTaskRunnerHandle::Get()),
 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(task_runner_->BelongsToCurrentThread());
 112   DCHECK(!sink_.get());
 113   DVLOG(1) << "WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer()";
 114 }
 115
 116 void WebRtcLocalAudioRenderer::Start() {
 117   DVLOG(1) << "WebRtcLocalAudioRenderer::Start()";
 118   DCHECK(task_runner_->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(task_runner_->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(task_runner_->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(task_runner_->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(task_runner_->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 media::OutputDevice* WebRtcLocalAudioRenderer::GetOutputDevice() {
 211   DVLOG(1) << __FUNCTION__;
 212   DCHECK(task_runner_->BelongsToCurrentThread());
 213   return sink_.get();
 214 }
 215
 216 base::TimeDelta WebRtcLocalAudioRenderer::GetCurrentRenderTime() const {
 217   DCHECK(task_runner_->BelongsToCurrentThread());
 218   base::AutoLock auto_lock(thread_lock_);
 219   if (!sink_.get())
 220     return base::TimeDelta();
 221   return total_render_time();
 222 }
 223
 224 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const {
 225   return true;
 226 }
 227
 228 void WebRtcLocalAudioRenderer::MaybeStartSink() {
 229   DCHECK(task_runner_->BelongsToCurrentThread());
 230   DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink()";
 231
 232   if (!sink_.get() || !source_params_.IsValid())
 233     return;
 234
 235   {
 236     // Clear up the old data in the FIFO.
 237     base::AutoLock auto_lock(thread_lock_);
 238     audio_shifter_->Flush();
 239   }
 240
 241   if (!sink_params_.IsValid() || !playing_ || !volume_ || sink_started_)
 242     return;
 243
 244   DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_.";
 245   sink_->InitializeWithSessionId(sink_params_, this, session_id_);
 246   sink_->Start();
 247   sink_started_ = true;
 248   UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
 249                             kSinkStarted, kSinkStatesMax);
 250 }
 251
 252 void WebRtcLocalAudioRenderer::ReconfigureSink(
 253     const media::AudioParameters& params) {
 254   DCHECK(task_runner_->BelongsToCurrentThread());
 255
 256   DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()";
 257
 258   int implicit_ducking_effect = 0;
 259   RenderFrameImpl* const frame =
 260       RenderFrameImpl::FromRoutingID(source_render_frame_id_);
 261   MediaStreamDispatcher* const dispatcher = frame ?
 262       frame->GetMediaStreamDispatcher() : NULL;
 263   if (dispatcher && dispatcher->IsAudioDuckingActive()) {
 264     DVLOG(1) << "Forcing DUCKING to be ON for output";
 265     implicit_ducking_effect = media::AudioParameters::DUCKING;
 266   } else {
 267     DVLOG(1) << "DUCKING not forced ON for output";
 268   }
 269
 270   if (source_params_.Equals(params))
 271     return;
 272
 273   // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
 274   // the new format.
 275
 276   source_params_ = params;
 277
 278   sink_params_ = media::AudioParameters(source_params_.format(),
 279       source_params_.channel_layout(), source_params_.sample_rate(),
 280       source_params_.bits_per_sample(),
 281       WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
 282                                                 frames_per_buffer_),
 283       // If DUCKING is enabled on the source, it needs to be enabled on the
 284       // sink as well.
 285       source_params_.effects() | implicit_ducking_effect);
 286
 287   {
 288     // Note: The max buffer is fairly large, but will rarely be used.
 289     // Cast needs the buffer to hold at least one second of audio.
 290     // The clock accuracy is set to 20ms because clock accuracy is
 291     // ~15ms on windows.
 292     media::AudioShifter* const new_shifter = new media::AudioShifter(
 293         base::TimeDelta::FromSeconds(2),
 294         base::TimeDelta::FromMilliseconds(20),
 295         base::TimeDelta::FromSeconds(20),
 296         source_params_.sample_rate(),
 297         params.channels());
 298
 299     base::AutoLock auto_lock(thread_lock_);
 300     audio_shifter_.reset(new_shifter);
 301   }
 302
 303   if (!sink_.get())
 304     return;  // WebRtcLocalAudioRenderer has not yet been started.
 305
 306   // Stop |sink_| and re-create a new one to be initialized with different audio
 307   // parameters.  Then, invoke MaybeStartSink() to restart everything again.
 308   if (sink_started_) {
 309     sink_->Stop();
 310     sink_started_ = false;
 311   }
 312
 313   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
 314   MaybeStartSink();
 315 }
 316
 317 }  // namespace content