Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/webrtc_local_audio_source_provider.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 
 #include "base/logging.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_fifo.h"
@@ skipping 25 matching lines @@
         media::CHANNEL_LAYOUT_STEREO, 2, 0, sample_rate, 16,
         kWebAudioRenderBufferSize);
   }
 }
 
 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
   if (audio_converter_.get())
     audio_converter_->RemoveInput(this);
 }
 
-void WebRtcLocalAudioSourceProvider::SetCaptureFormat(
+void WebRtcLocalAudioSourceProvider::OnSetFormat(
     const media::AudioParameters& params) {
   // We need detach the thread here because it will be a new capture thread
-  // calling SetCaptureFormat() and CaptureData() if the source is restarted.
+  // calling OnSetFormat() and OnData() if the source is restarted.
   capture_thread_checker_.DetachFromThread();
   DCHECK(capture_thread_checker_.CalledOnValidThread());
   DCHECK(params.IsValid());
   DCHECK(sink_params_.IsValid());
 
   base::AutoLock auto_lock(lock_);
   source_params_ = params;
   // Create the audio converter with |disable_fifo| as false so that the
   // converter will request source_params.frames_per_buffer() each time.
   // This will not increase the complexity as there is only one client to
   // the converter.
   audio_converter_.reset(
       new media::AudioConverter(params, sink_params_, false));
   audio_converter_->AddInput(this);
   fifo_.reset(new media::AudioFifo(
       params.channels(),
       kMaxNumberOfBuffers * params.frames_per_buffer()));
   input_bus_ = media::AudioBus::Create(params.channels(),
                                        params.frames_per_buffer());
 }
 
-int WebRtcLocalAudioSourceProvider::CaptureData(
-    const std::vector<int>& channels,
+void WebRtcLocalAudioSourceProvider::OnData(
     const int16* audio_data,
     int sample_rate,
     int number_of_channels,
-    int number_of_frames,
-    int audio_delay_milliseconds,
-    int current_volume,
-    bool need_audio_processing,
-    bool key_pressed) {
+    int number_of_frames) {
   DCHECK(capture_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   if (!is_enabled_)
-    return 0;
+    return;
 
   DCHECK(fifo_.get());
 
   // TODO(xians): A better way to handle the interleaved and deinterleaved
   // format switching, see issue/317710.
   DCHECK(input_bus_->frames() == number_of_frames);
   DCHECK(input_bus_->channels() == number_of_channels);
   input_bus_->FromInterleaved(audio_data, number_of_frames, 2);
 
   if (fifo_->frames() + number_of_frames <= fifo_->max_frames()) {
     fifo_->Push(input_bus_.get());
   } else {
     // This can happen if the data in FIFO is too slowed to be consumed or
     // WebAudio stops consuming data.
     DLOG(WARNING) << "Local source provicer FIFO is full" << fifo_->frames();
   }
-
-  return 0;
 }
 
 void WebRtcLocalAudioSourceProvider::setClient(
     blink::WebAudioSourceProviderClient* client) {
   NOTREACHED();
 }
 
 void WebRtcLocalAudioSourceProvider::provideInput(
     const WebVector<float*>& audio_data, size_t number_of_frames) {
   DCHECK_EQ(number_of_frames, kWebAudioRenderBufferSize);
@@ skipping 27 matching lines @@
 
   return 1.0;
 }
 
 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
     const media::AudioParameters& sink_params) {
   sink_params_ = sink_params;
 }
 
 }  // namespace content