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_audio_capturer.h"
  10 #include "content/renderer/media/webrtc_local_audio_track.h"
  11
  12 using media::AudioBus;
  13 using media::AudioFifo;
  14 using media::AudioParameters;
  15 using media::ChannelLayout;
  16 using media::CHANNEL_LAYOUT_MONO;
  17 using media::CHANNEL_LAYOUT_STEREO;
  18
  19 static const int kMaxNumberOfBuffersInFifo = 5;
  20
  21 namespace content {
  22
  23 WebAudioCapturerSource::WebAudioCapturerSource()
  24     : track_(NULL),
  25       capturer_(NULL),
  26       audio_format_changed_(false) {
  27 }
  28
  29 WebAudioCapturerSource::~WebAudioCapturerSource() {
  30 }
  31
  32 void WebAudioCapturerSource::setFormat(
  33     size_t number_of_channels, float sample_rate) {
  34   DCHECK(thread_checker_.CalledOnValidThread());
  35   DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
  36            << sample_rate << ")";
  37   if (number_of_channels > 2) {
  38     // TODO(xians): Handle more than just the mono and stereo cases.
  39     LOG(WARNING) << "WebAudioCapturerSource::setFormat() : unhandled format.";
  40     return;
  41   }
  42
  43   ChannelLayout channel_layout =
  44       number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
  45
  46   base::AutoLock auto_lock(lock_);
  47   // Set the format used by this WebAudioCapturerSource. We are using 10ms data
  48   // as buffer size since that is the native buffer size of WebRtc packet
  49   // running on.
  50   params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  51                 channel_layout, number_of_channels, 0, sample_rate, 16,
  52                 sample_rate / 100);
  53   audio_format_changed_ = true;
  54
  55   wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
  56   capture_bus_ = AudioBus::Create(params_);
  57   audio_data_.reset(
  58       new int16[params_.frames_per_buffer() * params_.channels()]);
  59   fifo_.reset(new AudioFifo(
  60       params_.channels(),
  61       kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
  62 }
  63
  64 void WebAudioCapturerSource::Start(
  65     WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer) {
  66   DCHECK(thread_checker_.CalledOnValidThread());
  67   DCHECK(track);
  68   base::AutoLock auto_lock(lock_);
  69   track_ = track;
  70   capturer_ = capturer;
  71 }
  72
  73 void WebAudioCapturerSource::Stop() {
  74   DCHECK(thread_checker_.CalledOnValidThread());
  75   base::AutoLock auto_lock(lock_);
  76   track_ = NULL;
  77   capturer_ = NULL;
  78 }
  79
  80 void WebAudioCapturerSource::consumeAudio(
  81     const blink::WebVector<const float*>& audio_data,
  82     size_t number_of_frames) {
  83   base::AutoLock auto_lock(lock_);
  84   if (!track_)
  85     return;
  86
  87   // Update the downstream client if the audio format has been changed.
  88   if (audio_format_changed_) {
  89     track_->OnSetFormat(params_);
  90     audio_format_changed_ = false;
  91   }
  92
  93   wrapper_bus_->set_frames(number_of_frames);
  94
  95   // Make sure WebKit is honoring what it told us up front
  96   // about the channels.
  97   DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
  98
  99   for (size_t i = 0; i < audio_data.size(); ++i)
 100     wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
 101
 102   // Handle mismatch between WebAudio buffer-size and WebRTC.
 103   int available = fifo_->max_frames() - fifo_->frames();
 104   if (available < static_cast<int>(number_of_frames)) {
 105     NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
 106     return;
 107   }
 108
 109   fifo_->Push(wrapper_bus_.get());
 110   int capture_frames = params_.frames_per_buffer();
 111   base::TimeDelta delay;
 112   int volume = 0;
 113   bool key_pressed = false;
 114   if (capturer_) {
 115     capturer_->GetAudioProcessingParams(&delay, &volume, &key_pressed);
 116   }
 117
 118   // Turn off audio processing if the delay value is 0, since in such case,
 119   // it indicates the data is not from microphone.
 120   // TODO(xians): remove the flag when supporting one APM per audio track.
 121   // See crbug/264611 for details.
 122   bool need_audio_processing = (delay.InMilliseconds() != 0);
 123   while (fifo_->frames() >= capture_frames) {
 124     fifo_->Consume(capture_bus_.get(), 0, capture_frames);
 125     // TODO(xians): Avoid this interleave/deinterleave operation.
 126     capture_bus_->ToInterleaved(capture_bus_->frames(),
 127                                 params_.bits_per_sample() / 8,
 128                                 audio_data_.get());
 129     track_->Capture(audio_data_.get(), delay, volume, key_pressed,
 130                     need_audio_processing);
 131   }
 132 }
 133
 134 }  // namespace content