Home | History | Annotate | Download | only in media 
      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
    135