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