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, channel_layout,
  53                 sample_rate, 16, sample_rate / 100);
  54   audio_format_changed_ = true;
  55
  56   wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
  57   capture_bus_ = AudioBus::Create(params_);
  58   fifo_.reset(new AudioFifo(
  59       params_.channels(),
  60       kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
  61 }
  62
  63 void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
  64   DCHECK(thread_checker_.CalledOnValidThread());
  65   DCHECK(track);
  66   base::AutoLock auto_lock(lock_);
  67   track_ = track;
  68 }
  69
  70 void WebAudioCapturerSource::Stop() {
  71   DCHECK(thread_checker_.CalledOnValidThread());
  72   {
  73     base::AutoLock auto_lock(lock_);
  74     track_ = NULL;
  75   }
  76   // removeFromBlinkSource() should not be called while |lock_| is acquired,
  77   // as it could result in a deadlock.
  78   removeFromBlinkSource();
  79 }
  80
  81 void WebAudioCapturerSource::consumeAudio(
  82     const blink::WebVector<const float*>& audio_data,
  83     size_t number_of_frames) {
  84   base::AutoLock auto_lock(lock_);
  85   if (!track_)
  86     return;
  87
  88   // Update the downstream client if the audio format has been changed.
  89   if (audio_format_changed_) {
  90     track_->OnSetFormat(params_);
  91     audio_format_changed_ = false;
  92   }
  93
  94   wrapper_bus_->set_frames(number_of_frames);
  95
  96   // Make sure WebKit is honoring what it told us up front
  97   // about the channels.
  98   DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
  99
 100   for (size_t i = 0; i < audio_data.size(); ++i)
 101     wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
 102
 103   // Handle mismatch between WebAudio buffer-size and WebRTC.
 104   int available = fifo_->max_frames() - fifo_->frames();
 105   if (available < static_cast<int>(number_of_frames)) {
 106     NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
 107     return;
 108   }
 109
 110   // Compute the estimated capture time of the first sample frame of audio that
 111   // will be consumed from the FIFO in the loop below.
 112   base::TimeTicks estimated_capture_time = base::TimeTicks::Now() -
 113       fifo_->frames() * base::TimeDelta::FromSeconds(1) / params_.sample_rate();
 114
 115   fifo_->Push(wrapper_bus_.get());
 116   while (fifo_->frames() >= capture_bus_->frames()) {
 117     fifo_->Consume(capture_bus_.get(), 0, capture_bus_->frames());
 118     track_->Capture(*capture_bus_, estimated_capture_time, false);
 119
 120     // Advance the estimated capture time for the next FIFO consume operation.
 121     estimated_capture_time +=
 122         capture_bus_->frames() * base::TimeDelta::FromSeconds(1) /
 123             params_.sample_rate();
 124   }
 125 }
 126
 127 // If registered as audio consumer in |blink_source_|, deregister from
 128 // |blink_source_| and stop keeping a reference to |blink_source_|.
 129 // Failure to call this method when stopping the track might leave an invalid
 130 // WebAudioCapturerSource reference still registered as an audio consumer on
 131 // the blink side.
 132 void WebAudioCapturerSource::removeFromBlinkSource() {
 133   if (!blink_source_.isNull()) {
 134     blink_source_.removeAudioConsumer(this);
 135     blink_source_.reset();
 136   }
 137 }
 138
 139 }  // namespace content