content/renderer/media/webrtc_local_audio_track.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_track.h"
   6
   7 #include <limits>
   8
   9 #include "content/public/renderer/media_stream_audio_sink.h"
  10 #include "content/renderer/media/media_stream_audio_level_calculator.h"
  11 #include "content/renderer/media/media_stream_audio_processor.h"
  12 #include "content/renderer/media/media_stream_audio_sink_owner.h"
  13 #include "content/renderer/media/media_stream_audio_track_sink.h"
  14 #include "content/renderer/media/webaudio_capturer_source.h"
  15 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
  16 #include "content/renderer/media/webrtc_audio_capturer.h"
  17
  18 namespace content {
  19
  20 WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
  21     WebRtcLocalAudioTrackAdapter* adapter,
  22     const scoped_refptr<WebRtcAudioCapturer>& capturer,
  23     WebAudioCapturerSource* webaudio_source)
  24     : MediaStreamTrack(true),
  25       adapter_(adapter),
  26       capturer_(capturer),
  27       webaudio_source_(webaudio_source) {
  28   DCHECK(capturer.get() || webaudio_source);
  29   signal_thread_checker_.DetachFromThread();
  30
  31   adapter_->Initialize(this);
  32
  33   DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
  34 }
  35
  36 WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
  37   DCHECK(main_render_thread_checker_.CalledOnValidThread());
  38   DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
  39   // Users might not call Stop() on the track.
  40   Stop();
  41 }
  42
  43 media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
  44   DCHECK(main_render_thread_checker_.CalledOnValidThread());
  45   if (webaudio_source_.get()) {
  46     return media::AudioParameters();
  47   } else {
  48     return capturer_->GetOutputFormat();
  49   }
  50 }
  51
  52 void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
  53                                     base::TimeTicks estimated_capture_time,
  54                                     bool force_report_nonzero_energy) {
  55   DCHECK(capture_thread_checker_.CalledOnValidThread());
  56   DCHECK(!estimated_capture_time.is_null());
  57
  58   // Calculate the signal level regardless of whether the track is disabled or
  59   // enabled.  If |force_report_nonzero_energy| is true, |audio_bus| contains
  60   // post-processed data that may be all zeros even though the signal contained
  61   // energy before the processing.  In this case, report nonzero energy even if
  62   // the energy of the data in |audio_bus| is zero.
  63   const float minimum_signal_level = force_report_nonzero_energy ?
  64       1.0f / std::numeric_limits<int16>::max() : 0.0f;
  65   const float signal_level = std::max(
  66       minimum_signal_level,
  67       std::min(1.0f, level_calculator_->Calculate(audio_bus)));
  68   const int signal_level_as_pcm16 =
  69       static_cast<int>(signal_level * std::numeric_limits<int16>::max() +
  70                        0.5f /* rounding to nearest int */);
  71   adapter_->SetSignalLevel(signal_level_as_pcm16);
  72
  73   scoped_refptr<WebRtcAudioCapturer> capturer;
  74   SinkList::ItemList sinks;
  75   SinkList::ItemList sinks_to_notify_format;
  76   {
  77     base::AutoLock auto_lock(lock_);
  78     capturer = capturer_;
  79     sinks = sinks_.Items();
  80     sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
  81   }
  82
  83   // Notify the tracks on when the format changes. This will do nothing if
  84   // |sinks_to_notify_format| is empty.
  85   for (const auto& sink : sinks_to_notify_format)
  86     sink->OnSetFormat(audio_parameters_);
  87
  88   // Feed the data to the sinks.
  89   // TODO(jiayl): we should not pass the real audio data down if the track is
  90   // disabled. This is currently done so to feed input to WebRTC typing
  91   // detection and should be changed when audio processing is moved from
  92   // WebRTC to the track.
  93   std::vector<int> voe_channels = adapter_->VoeChannels();
  94   for (const auto& sink : sinks)
  95     sink->OnData(audio_bus, estimated_capture_time);
  96 }
  97
  98 void WebRtcLocalAudioTrack::OnSetFormat(
  99     const media::AudioParameters& params) {
 100   DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
 101   // If the source is restarted, we might have changed to another capture
 102   // thread.
 103   capture_thread_checker_.DetachFromThread();
 104   DCHECK(capture_thread_checker_.CalledOnValidThread());
 105
 106   audio_parameters_ = params;
 107   level_calculator_.reset(new MediaStreamAudioLevelCalculator());
 108
 109   base::AutoLock auto_lock(lock_);
 110   // Remember to notify all sinks of the new format.
 111   sinks_.TagAll();
 112 }
 113
 114 void WebRtcLocalAudioTrack::SetAudioProcessor(
 115     const scoped_refptr<MediaStreamAudioProcessor>& processor) {
 116   // if the |processor| does not have audio processing, which can happen if
 117   // kDisableAudioTrackProcessing is set set or all the constraints in
 118   // the |processor| are turned off. In such case, we pass NULL to the
 119   // adapter to indicate that no stats can be gotten from the processor.
 120   adapter_->SetAudioProcessor(processor->has_audio_processing() ?
 121       processor : NULL);
 122 }
 123
 124 void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
 125   // This method is called from webrtc, on the signaling thread, when the local
 126   // description is set and from the main thread from WebMediaPlayerMS::load
 127   // (via WebRtcLocalAudioRenderer::Start).
 128   DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
 129          signal_thread_checker_.CalledOnValidThread());
 130   DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
 131   base::AutoLock auto_lock(lock_);
 132
 133   // Verify that |sink| is not already added to the list.
 134   DCHECK(!sinks_.Contains(
 135       MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)));
 136
 137   // Create (and add to the list) a new MediaStreamAudioTrackSink
 138   // which owns the |sink| and delagates all calls to the
 139   // MediaStreamAudioSink interface. It will be tagged in the list, so
 140   // we remember to call OnSetFormat() on the new sink.
 141   scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
 142       new MediaStreamAudioSinkOwner(sink));
 143   sinks_.AddAndTag(sink_owner.get());
 144 }
 145
 146 void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
 147   // See AddSink for additional context.  When local audio is stopped from
 148   // webrtc, we'll be called here on the signaling thread.
 149   DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
 150          signal_thread_checker_.CalledOnValidThread());
 151   DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
 152
 153   scoped_refptr<MediaStreamAudioTrackSink> removed_item;
 154   {
 155     base::AutoLock auto_lock(lock_);
 156     removed_item = sinks_.Remove(
 157         MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
 158   }
 159
 160   // Clear the delegate to ensure that no more capture callbacks will
 161   // be sent to this sink. Also avoids a possible crash which can happen
 162   // if this method is called while capturing is active.
 163   if (removed_item.get())
 164     removed_item->Reset();
 165 }
 166
 167 void WebRtcLocalAudioTrack::Start() {
 168   DCHECK(main_render_thread_checker_.CalledOnValidThread());
 169   DVLOG(1) << "WebRtcLocalAudioTrack::Start()";
 170   if (webaudio_source_.get()) {
 171     // If the track is hooking up with WebAudio, do NOT add the track to the
 172     // capturer as its sink otherwise two streams in different clock will be
 173     // pushed through the same track.
 174     webaudio_source_->Start(this);
 175   } else if (capturer_.get()) {
 176     capturer_->AddTrack(this);
 177   }
 178
 179   SinkList::ItemList sinks;
 180   {
 181     base::AutoLock auto_lock(lock_);
 182     sinks = sinks_.Items();
 183   }
 184   for (SinkList::ItemList::const_iterator it = sinks.begin();
 185        it != sinks.end();
 186        ++it) {
 187     (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateLive);
 188   }
 189 }
 190
 191 void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
 192   DCHECK(main_render_thread_checker_.CalledOnValidThread());
 193   if (adapter_.get())
 194     adapter_->set_enabled(enabled);
 195 }
 196
 197 void WebRtcLocalAudioTrack::Stop() {
 198   DCHECK(main_render_thread_checker_.CalledOnValidThread());
 199   DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
 200   if (!capturer_.get() && !webaudio_source_.get())
 201     return;
 202
 203   if (webaudio_source_.get()) {
 204     // Called Stop() on the |webaudio_source_| explicitly so that
 205     // |webaudio_source_| won't push more data to the track anymore.
 206     // Also note that the track is not registered as a sink to the |capturer_|
 207     // in such case and no need to call RemoveTrack().
 208     webaudio_source_->Stop();
 209   } else {
 210     // It is necessary to call RemoveTrack on the |capturer_| to avoid getting
 211     // audio callback after Stop().
 212     capturer_->RemoveTrack(this);
 213   }
 214
 215   // Protect the pointers using the lock when accessing |sinks_| and
 216   // setting the |capturer_| to NULL.
 217   SinkList::ItemList sinks;
 218   {
 219     base::AutoLock auto_lock(lock_);
 220     sinks = sinks_.Items();
 221     sinks_.Clear();
 222     webaudio_source_ = NULL;
 223     capturer_ = NULL;
 224   }
 225
 226   for (SinkList::ItemList::const_iterator it = sinks.begin();
 227        it != sinks.end();
 228        ++it){
 229     (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
 230     (*it)->Reset();
 231   }
 232 }
 233
 234 webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
 235   DCHECK(main_render_thread_checker_.CalledOnValidThread());
 236   return adapter_.get();
 237 }
 238
 239 }  // namespace content