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(
  38         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
  39         media::CHANNEL_LAYOUT_STEREO, 2, 0, sample_rate, 16,
  40         kWebAudioRenderBufferSize);
  41   }
  42
  43   // Connect the source provider to the track as a sink.
  44   MediaStreamAudioSink::AddToAudioTrack(this, track_);
  45 }
  46
  47 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
  48   if (audio_converter_.get())
  49     audio_converter_->RemoveInput(this);
  50
  51   // If the track is still active, it is necessary to notify the track before
  52   // the source provider goes away.
  53   if (!track_stopped_)
  54     MediaStreamAudioSink::RemoveFromAudioTrack(this, track_);
  55 }
  56
  57 void WebRtcLocalAudioSourceProvider::OnSetFormat(
  58     const media::AudioParameters& params) {
  59   // We need detach the thread here because it will be a new capture thread
  60   // calling OnSetFormat() and OnData() if the source is restarted.
  61   capture_thread_checker_.DetachFromThread();
  62   DCHECK(capture_thread_checker_.CalledOnValidThread());
  63   DCHECK(params.IsValid());
  64   DCHECK(sink_params_.IsValid());
  65
  66   base::AutoLock auto_lock(lock_);
  67   source_params_ = params;
  68   // Create the audio converter with |disable_fifo| as false so that the
  69   // converter will request source_params.frames_per_buffer() each time.
  70   // This will not increase the complexity as there is only one client to
  71   // the converter.
  72   audio_converter_.reset(
  73       new media::AudioConverter(params, sink_params_, false));
  74   audio_converter_->AddInput(this);
  75   fifo_.reset(new media::AudioFifo(
  76       params.channels(),
  77       kMaxNumberOfBuffers * params.frames_per_buffer()));
  78   input_bus_ = media::AudioBus::Create(params.channels(),
  79                                        params.frames_per_buffer());
  80 }
  81
  82 void WebRtcLocalAudioSourceProvider::OnReadyStateChanged(
  83       blink::WebMediaStreamSource::ReadyState state) {
  84   if (state ==  blink::WebMediaStreamSource::ReadyStateEnded)
  85     track_stopped_ = true;
  86 }
  87
  88 void WebRtcLocalAudioSourceProvider::OnData(
  89     const int16* audio_data,
  90     int sample_rate,
  91     int number_of_channels,
  92     int number_of_frames) {
  93   DCHECK(capture_thread_checker_.CalledOnValidThread());
  94   base::AutoLock auto_lock(lock_);
  95   if (!is_enabled_)
  96     return;
  97
  98   DCHECK(fifo_.get());
  99
 100   // TODO(xians): A better way to handle the interleaved and deinterleaved
 101   // format switching, see issue/317710.
 102   DCHECK(input_bus_->frames() == number_of_frames);
 103   DCHECK(input_bus_->channels() == number_of_channels);
 104   input_bus_->FromInterleaved(audio_data, number_of_frames, 2);
 105
 106   if (fifo_->frames() + number_of_frames <= fifo_->max_frames()) {
 107     fifo_->Push(input_bus_.get());
 108   } else {
 109     // This can happen if the data in FIFO is too slowly consumed or
 110     // WebAudio stops consuming data.
 111     DVLOG(3) << "Local source provicer FIFO is full" << fifo_->frames();
 112   }
 113 }
 114
 115 void WebRtcLocalAudioSourceProvider::setClient(
 116     blink::WebAudioSourceProviderClient* client) {
 117   NOTREACHED();
 118 }
 119
 120 void WebRtcLocalAudioSourceProvider::provideInput(
 121     const WebVector<float*>& audio_data, size_t number_of_frames) {
 122   DCHECK_EQ(number_of_frames, kWebAudioRenderBufferSize);
 123   if (!output_wrapper_ ||
 124       static_cast<size_t>(output_wrapper_->channels()) != audio_data.size()) {
 125     output_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size());
 126   }
 127
 128   output_wrapper_->set_frames(number_of_frames);
 129   for (size_t i = 0; i < audio_data.size(); ++i)
 130     output_wrapper_->SetChannelData(i, audio_data[i]);
 131
 132   base::AutoLock auto_lock(lock_);
 133   if (!audio_converter_)
 134     return;
 135
 136   is_enabled_ = true;
 137   audio_converter_->Convert(output_wrapper_.get());
 138 }
 139
 140 double WebRtcLocalAudioSourceProvider::ProvideInput(
 141     media::AudioBus* audio_bus, base::TimeDelta buffer_delay) {
 142   if (fifo_->frames() >= audio_bus->frames()) {
 143     fifo_->Consume(audio_bus, 0, audio_bus->frames());
 144   } else {
 145     audio_bus->Zero();
 146     DVLOG(1) << "WARNING: Underrun, FIFO has data " << fifo_->frames()
 147              << " samples but " << audio_bus->frames()
 148              << " samples are needed";
 149   }
 150
 151   return 1.0;
 152 }
 153
 154 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
 155     const media::AudioParameters& sink_params) {
 156   sink_params_ = sink_params;
 157 }
 158
 159 }  // namespace content