content/renderer/media/webaudio_capturer_source.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/webaudio_capturer_source.h"
   6
   7 #include "base/logging.h"
   8 #include "base/time/time.h"
   9 #include "content/renderer/media/webrtc_local_audio_track.h"
  10
  11 using media::AudioBus;
  12 using media::AudioFifo;
  13 using media::AudioParameters;
  14 using media::ChannelLayout;
  15 using media::CHANNEL_LAYOUT_MONO;
  16 using media::CHANNEL_LAYOUT_STEREO;
  17
  18 static const int kMaxNumberOfBuffersInFifo = 5;
  19
  20 namespace content {
  21
  22 WebAudioCapturerSource::WebAudioCapturerSource(
  23     const blink::WebMediaStreamSource& blink_source)
  24     : track_(NULL),
  25       audio_format_changed_(false),
  26       blink_source_(blink_source) {
  27 }
  28
  29 WebAudioCapturerSource::~WebAudioCapturerSource() {
  30   DCHECK(thread_checker_.CalledOnValidThread());
  31   removeFromBlinkSource();
  32 }
  33
  34 void WebAudioCapturerSource::setFormat(
  35     size_t number_of_channels, float sample_rate) {
  36   DCHECK(thread_checker_.CalledOnValidThread());
  37   DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
  38            << sample_rate << ")";
  39   if (number_of_channels > 2) {
  40     // TODO(xians): Handle more than just the mono and stereo cases.
  41     LOG(WARNING) << "WebAudioCapturerSource::setFormat() : unhandled format.";
  42     return;
  43   }
  44
  45   ChannelLayout channel_layout =
  46       number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
  47
  48   base::AutoLock auto_lock(lock_);
  49   // Set the format used by this WebAudioCapturerSource. We are using 10ms data
  50   // as buffer size since that is the native buffer size of WebRtc packet
  51   // running on.
  52   params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  53                 channel_layout, number_of_channels, sample_rate, 16,
  54                 sample_rate / 100);
  55   audio_format_changed_ = true;
  56
  57   wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
  58   capture_bus_ = AudioBus::Create(params_);
  59   fifo_.reset(new AudioFifo(
  60       params_.channels(),
  61       kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
  62 }
  63
  64 void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
  65   DCHECK(thread_checker_.CalledOnValidThread());
  66   DCHECK(track);
  67   base::AutoLock auto_lock(lock_);
  68   track_ = track;
  69 }
  70
  71 void WebAudioCapturerSource::Stop() {
  72   DCHECK(thread_checker_.CalledOnValidThread());
  73   base::AutoLock auto_lock(lock_);
  74   track_ = NULL;
  75   removeFromBlinkSource();
  76 }
  77
  78 void WebAudioCapturerSource::consumeAudio(
  79     const blink::WebVector<const float*>& audio_data,
  80     size_t number_of_frames) {
  81   base::AutoLock auto_lock(lock_);
  82   if (!track_)
  83     return;
  84
  85   // Update the downstream client if the audio format has been changed.
  86   if (audio_format_changed_) {
  87     track_->OnSetFormat(params_);
  88     audio_format_changed_ = false;
  89   }
  90
  91   wrapper_bus_->set_frames(number_of_frames);
  92
  93   // Make sure WebKit is honoring what it told us up front
  94   // about the channels.
  95   DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
  96
  97   for (size_t i = 0; i < audio_data.size(); ++i)
  98     wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
  99
 100   // Handle mismatch between WebAudio buffer-size and WebRTC.
 101   int available = fifo_->max_frames() - fifo_->frames();
 102   if (available < static_cast<int>(number_of_frames)) {
 103     NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
 104     return;
 105   }
 106
 107   // Compute the estimated capture time of the first sample frame of audio that
 108   // will be consumed from the FIFO in the loop below.
 109   base::TimeTicks estimated_capture_time = base::TimeTicks::Now() -
 110       fifo_->frames() * base::TimeDelta::FromSeconds(1) / params_.sample_rate();
 111
 112   fifo_->Push(wrapper_bus_.get());
 113   while (fifo_->frames() >= capture_bus_->frames()) {
 114     fifo_->Consume(capture_bus_.get(), 0, capture_bus_->frames());
 115     track_->Capture(*capture_bus_, estimated_capture_time, false);
 116
 117     // Advance the estimated capture time for the next FIFO consume operation.
 118     estimated_capture_time +=
 119         capture_bus_->frames() * base::TimeDelta::FromSeconds(1) /
 120             params_.sample_rate();
 121   }
 122 }
 123
 124 // If registered as audio consumer in |blink_source_|, deregister from
 125 // |blink_source_| and stop keeping a reference to |blink_source_|.
 126 // Failure to call this method when stopping the track might leave an invalid
 127 // WebAudioCapturerSource reference still registered as an audio consumer on
 128 // the blink side.
 129 void WebAudioCapturerSource::removeFromBlinkSource() {
 130   if (!blink_source_.isNull()) {
 131     blink_source_.removeAudioConsumer(this);
 132     blink_source_.reset();
 133   }
 134 }
 135
 136 }  // namespace content