content/renderer/media/webrtc_local_audio_source_provider.cc

   1 // Copyright 2013 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/webrtc_local_audio_source_provider.h"
   6
   7 #include "base/logging.h"
   8 #include "content/renderer/render_thread_impl.h"
   9 #include "media/audio/audio_parameters.h"
  10 #include "media/base/audio_fifo.h"
  11 #include "media/base/audio_hardware_config.h"
  12 #include "third_party/WebKit/public/platform/WebAudioSourceProviderClient.h"
  13
  14 using blink::WebVector;
  15
  16 namespace content {
  17
  18 static const size_t kMaxNumberOfBuffers = 10;
  19
  20 // Size of the buffer that WebAudio processes each time, it is the same value
  21 // as AudioNode::ProcessingSizeInFrames in WebKit.
  22 // static
  23 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128;
  24
  25 WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
  26     const blink::WebMediaStreamTrack& track)
  27     : is_enabled_(false),
  28       track_(track),
  29       track_stopped_(false) {
  30   // Get the native audio output hardware sample-rate for the sink.
  31   // We need to check if RenderThreadImpl is valid here since the unittests
  32   // do not have one and they will inject their own |sink_params_| for testing.
  33   if (RenderThreadImpl::current()) {
  34     media::AudioHardwareConfig* hardware_config =
  35         RenderThreadImpl::current()->GetAudioHardwareConfig();
  36     int sample_rate = hardware_config->GetOutputSampleRate();
  37     sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  38                        media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
  39                        kWebAudioRenderBufferSize);
  40   }
  41
  42   // Connect the source provider to the track as a sink.
  43   MediaStreamAudioSink::AddToAudioTrack(this, track_);
  44 }
  45
  46 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
  47   if (audio_converter_.get())
  48     audio_converter_->RemoveInput(this);
  49
  50   // If the track is still active, it is necessary to notify the track before
  51   // the source provider goes away.
  52   if (!track_stopped_)
  53     MediaStreamAudioSink::RemoveFromAudioTrack(this, track_);
  54 }
  55
  56 void WebRtcLocalAudioSourceProvider::OnSetFormat(
  57     const media::AudioParameters& params) {
  58   // We need detach the thread here because it will be a new capture thread
  59   // calling OnSetFormat() and OnData() if the source is restarted.
  60   capture_thread_checker_.DetachFromThread();
  61   DCHECK(capture_thread_checker_.CalledOnValidThread());
  62   DCHECK(params.IsValid());
  63   DCHECK(sink_params_.IsValid());
  64
  65   base::AutoLock auto_lock(lock_);
  66   source_params_ = params;
  67   // Create the audio converter with |disable_fifo| as false so that the
  68   // converter will request source_params.frames_per_buffer() each time.
  69   // This will not increase the complexity as there is only one client to
  70   // the converter.
  71   audio_converter_.reset(
  72       new media::AudioConverter(params, sink_params_, false));
  73   audio_converter_->AddInput(this);
  74   fifo_.reset(new media::AudioFifo(
  75       params.channels(),
  76       kMaxNumberOfBuffers * params.frames_per_buffer()));
  77 }
  78
  79 void WebRtcLocalAudioSourceProvider::OnReadyStateChanged(
  80       blink::WebMediaStreamSource::ReadyState state) {
  81   if (state ==  blink::WebMediaStreamSource::ReadyStateEnded)
  82     track_stopped_ = true;
  83 }
  84
  85 void WebRtcLocalAudioSourceProvider::OnData(
  86     const media::AudioBus& audio_bus,
  87     base::TimeTicks estimated_capture_time) {
  88   DCHECK(capture_thread_checker_.CalledOnValidThread());
  89   DCHECK_EQ(audio_bus.channels(), source_params_.channels());
  90   DCHECK_EQ(audio_bus.frames(), source_params_.frames_per_buffer());
  91   DCHECK(!estimated_capture_time.is_null());
  92
  93   base::AutoLock auto_lock(lock_);
  94   if (!is_enabled_)
  95     return;
  96
  97   DCHECK(fifo_.get());
  98
  99   if (fifo_->frames() + audio_bus.frames() <= fifo_->max_frames()) {
 100     fifo_->Push(&audio_bus);
 101   } else {
 102     // This can happen if the data in FIFO is too slowly consumed or
 103     // WebAudio stops consuming data.
 104     DVLOG(3) << "Local source provicer FIFO is full" << fifo_->frames();
 105   }
 106 }
 107
 108 void WebRtcLocalAudioSourceProvider::setClient(
 109     blink::WebAudioSourceProviderClient* client) {
 110   NOTREACHED();
 111 }
 112
 113 void WebRtcLocalAudioSourceProvider::provideInput(
 114     const WebVector<float*>& audio_data, size_t number_of_frames) {
 115   DCHECK_EQ(number_of_frames, kWebAudioRenderBufferSize);
 116   if (!output_wrapper_ ||
 117       static_cast<size_t>(output_wrapper_->channels()) != audio_data.size()) {
 118     output_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size());
 119   }
 120
 121   output_wrapper_->set_frames(number_of_frames);
 122   for (size_t i = 0; i < audio_data.size(); ++i)
 123     output_wrapper_->SetChannelData(i, audio_data[i]);
 124
 125   base::AutoLock auto_lock(lock_);
 126   if (!audio_converter_)
 127     return;
 128
 129   is_enabled_ = true;
 130   audio_converter_->Convert(output_wrapper_.get());
 131 }
 132
 133 double WebRtcLocalAudioSourceProvider::ProvideInput(
 134     media::AudioBus* audio_bus, base::TimeDelta buffer_delay) {
 135   if (fifo_->frames() >= audio_bus->frames()) {
 136     fifo_->Consume(audio_bus, 0, audio_bus->frames());
 137   } else {
 138     audio_bus->Zero();
 139     DVLOG(1) << "WARNING: Underrun, FIFO has data " << fifo_->frames()
 140              << " samples but " << audio_bus->frames()
 141              << " samples are needed";
 142   }
 143
 144   return 1.0;
 145 }
 146
 147 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
 148     const media::AudioParameters& sink_params) {
 149   sink_params_ = sink_params;
 150 }
 151
 152 }  // namespace content