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

content/renderer/media/webrtc_local_audio_renderer.cc

 // Copyright (c) 2012 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_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/metrics/histogram.h"
@@ skipping 38 matching lines @@
     DVLOG(2) << "loopback FIFO is empty";
   }
 
   return audio_bus->frames();
 }
 
 void WebRtcLocalAudioRenderer::OnRenderError() {
   NOTIMPLEMENTED();
 }
 
-// content::WebRtcAudioCapturerSink implementation
-int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
-                                          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) {
+// content::MediaStreamAudioSink implementation
+void WebRtcLocalAudioRenderer::OnData(const int16* audio_data,
+                                      int sample_rate,
+                                      int number_of_channels,
+                                      int number_of_frames) {
   DCHECK(capture_thread_checker_.CalledOnValidThread());
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
   base::AutoLock auto_lock(thread_lock_);
   if (!playing_ || !volume_ || !loopback_fifo_)
-    return 0;
+    return;
 
   // Push captured audio to FIFO so it can be read by a local sink.
   if (loopback_fifo_->frames() + number_of_frames <=
       loopback_fifo_->max_frames()) {
     scoped_ptr<media::AudioBus> audio_source = media::AudioBus::Create(
         number_of_channels, number_of_frames);
     audio_source->FromInterleaved(audio_data,
                                   audio_source->frames(),
                                   sizeof(audio_data[0]));
     loopback_fifo_->Push(audio_source.get());
 
     const base::TimeTicks now = base::TimeTicks::Now();
     total_render_time_ += now - last_render_time_;
     last_render_time_ = now;
   } else {
     DVLOG(1) << "FIFO is full";
   }
-
-  return 0;
 }
 
-void WebRtcLocalAudioRenderer::SetCaptureFormat(
+void WebRtcLocalAudioRenderer::OnSetFormat(
     const media::AudioParameters& params) {
-  DVLOG(1) << "WebRtcLocalAudioRenderer::SetCaptureFormat()";
+  DVLOG(1) << "WebRtcLocalAudioRenderer::OnSetFormat()";
   // If the source is restarted, we might have changed to another capture
   // thread.
   capture_thread_checker_.DetachFromThread();
   DCHECK(capture_thread_checker_.CalledOnValidThread());
 
   // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
   // the new format.
   {
     base::AutoLock auto_lock(thread_lock_);
     if (source_params_ == params)
@@ skipping 33 matching lines @@
   // Post a task on the main render thread to reconfigure the |sink_| with the
   // new format.
   message_loop_->PostTask(
       FROM_HERE,
       base::Bind(&WebRtcLocalAudioRenderer::ReconfigureSink, this,
                  params));
 }
 
 // WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer implementation.
 WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer(
-    WebRtcLocalAudioTrack* audio_track,
+    const blink::WebMediaStreamTrack& audio_track,
     int source_render_view_id,
     int session_id,
     int frames_per_buffer)
     : audio_track_(audio_track),
       source_render_view_id_(source_render_view_id),
       session_id_(session_id),
       message_loop_(base::MessageLoopProxy::current()),
       playing_(false),
       frames_per_buffer_(frames_per_buffer),
       volume_(0.0),
       sink_started_(false) {
-  DCHECK(audio_track);
   DVLOG(1) << "WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer()";
 }
 
 WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer() {
   DCHECK(message_loop_->BelongsToCurrentThread());
   DCHECK(!sink_.get());
   DVLOG(1) << "WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer()";
 }
 
 void WebRtcLocalAudioRenderer::Start() {
   DVLOG(1) << "WebRtcLocalAudioRenderer::Start()";
   DCHECK(message_loop_->BelongsToCurrentThread());
 
-  if (!audio_track_)
-    return;  // Stop() has been called, so never start again.
-
   // We get audio data from |audio_track_|...
-  audio_track_->AddSink(this);
+  MediaStreamAudioSink::AddToAudioTrack(this, audio_track_);
   // ...and |sink_| will get audio data from us.
   DCHECK(!sink_.get());
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_);
 
   base::AutoLock auto_lock(thread_lock_);
   last_render_time_ = base::TimeTicks::Now();
   playing_ = false;
 }
 
 void WebRtcLocalAudioRenderer::Stop() {
@@ skipping 14 matching lines @@
     sink_ = NULL;
   }
 
   if (!sink_started_) {
     UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
                               kSinkNeverStarted, kSinkStatesMax);
   }
   sink_started_ = false;
 
   // Ensure that the capturer stops feeding us with captured audio.
-  // Note that, we do not stop the capturer here since it may still be used by
-  // the WebRTC ADM.
-  if (audio_track_) {
-    audio_track_->RemoveSink(this);
-    audio_track_ = NULL;
-  }
+  MediaStreamAudioSink::RemoveFromAudioTrack(this, audio_track_);
 }
 
 void WebRtcLocalAudioRenderer::Play() {
   DVLOG(1) << "WebRtcLocalAudioRenderer::Play()";
   DCHECK(message_loop_->BelongsToCurrentThread());
 
   if (!sink_.get())
     return;
 
   {
@@ skipping 88 matching lines @@
   // parameters.  Then, invoke MaybeStartSink() to restart everything again.
   if (sink_started_) {
     sink_->Stop();
     sink_started_ = false;
   }
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_view_id_);
   MaybeStartSink();
 }
 
 }  // namespace content