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      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   fifo_.reset(new AudioFifo(
     58       params_.channels(),
     59       kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
     60 }
     61 
     62 void WebAudioCapturerSource::Start(
     63     WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer) {
     64   DCHECK(thread_checker_.CalledOnValidThread());
     65   DCHECK(track);
     66   base::AutoLock auto_lock(lock_);
     67   track_ = track;
     68   capturer_ = capturer;
     69 }
     70 
     71 void WebAudioCapturerSource::Stop() {
     72   DCHECK(thread_checker_.CalledOnValidThread());
     73   base::AutoLock auto_lock(lock_);
     74   track_ = NULL;
     75   capturer_ = NULL;
     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   fifo_->Push(wrapper_bus_.get());
    108   int capture_frames = params_.frames_per_buffer();
    109   base::TimeDelta delay;
    110   int volume = 0;
    111   bool key_pressed = false;
    112   if (capturer_) {
    113     capturer_->GetAudioProcessingParams(&delay, &volume, &key_pressed);
    114   }
    115   while (fifo_->frames() >= capture_frames) {
    116     fifo_->Consume(capture_bus_.get(), 0, capture_frames);
    117     track_->Capture(capture_bus_.get(), delay.InMilliseconds(),
    118                     volume, key_pressed);
    119   }
    120 }
    121 
    122 }  // namespace content
    123