start/stop the source of the capturer when 1st audiotrack/last audiotrack is added/removed

content/renderer/media/webrtc_audio_device_impl.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_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 204 matching lines @@
 
 int32_t WebRtcAudioDeviceImpl::Init() {
   DVLOG(1) << "WebRtcAudioDeviceImpl::Init()";
   DCHECK(thread_checker_.CalledOnValidThread());
 
   if (initialized_)
     return 0;
 
   DCHECK(!capturer_.get());
   capturer_ = WebRtcAudioCapturer::CreateCapturer();
+
   // Add itself as an audio track to the |capturer_|. This is because WebRTC
   // supports only one ADM but multiple audio tracks, so the ADM can't be the
   // sink of certain audio track now.
   // TODO(xians): Register the ADM as the sink of the audio track if WebRTC
-  // supports one ADM for each audio track.
-  if (capturer_.get())
-    capturer_->AddSink(this);
+  // supports one ADM for each audio track. See http://crbug/247027.
+  capturer_->SetDefaultSink(this);
 
   // We need to return a success to continue the initialization of WebRtc VoE
   // because failure on the capturer_ initialization should not prevent WebRTC
   // from working. See issue http://crbug.com/144421 for details.
   initialized_ = true;
 
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::Terminate() {
@@ skipping 12 matching lines @@
     // Grab a local reference while we call Stop(), which will trigger a call to
     // RemoveAudioRenderer that clears our reference to the audio renderer.
     scoped_refptr<WebRtcAudioRenderer> local_renderer(renderer_);
     local_renderer->Stop();
     DCHECK(!renderer_.get());
   }
 
   if (capturer_.get()) {
     // |capturer_| is stopped by the media stream, so do not need to
     // call Stop() here.
-    capturer_->RemoveSink(this);
+    capturer_->SetDefaultSink(NULL);
     capturer_ = NULL;
   }
 
   initialized_ = false;
   return 0;
 }
 
 bool WebRtcAudioDeviceImpl::Initialized() const {
   return initialized_;
 }
@@ skipping 61 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::StartRecording() {
   DVLOG(1) << "WebRtcAudioDeviceImpl::StartRecording()";
   DCHECK(initialized_);
   LOG_IF(ERROR, !audio_transport_callback_) << "Audio transport is missing";
   if (!audio_transport_callback_) {
     return -1;
   }
 
+  {
+    base::AutoLock auto_lock(lock_);
+    if (recording_)
+      return 0;
+
+    recording_ = true;
+  }
+
   start_capture_time_ = base::Time::Now();
 
-  base::AutoLock auto_lock(lock_);
-  recording_ = true;
-
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::StopRecording() {
   DVLOG(1) << "WebRtcAudioDeviceImpl::StopRecording()";
-  if (!recording_) {
-    return 0;
+  {
+    base::AutoLock auto_lock(lock_);
+    if (!recording_)
+      return 0;
+
+    recording_ = false;
   }
 
   // Add histogram data to be uploaded as part of an UMA logging event.
   // This histogram keeps track of total recording times.
   if (!start_capture_time_.is_null()) {
     base::TimeDelta capture_time = base::Time::Now() - start_capture_time_;
     UMA_HISTOGRAM_LONG_TIMES("WebRTC.AudioCaptureTime", capture_time);
   }
 
-  base::AutoLock auto_lock(lock_);
-  recording_ = false;
-
   return 0;
 }
 
 bool WebRtcAudioDeviceImpl::Recording() const {
   base::AutoLock auto_lock(lock_);
   return recording_;
 }
 
 int32_t WebRtcAudioDeviceImpl::SetAGC(bool enable) {
   DVLOG(1) <<  "WebRtcAudioDeviceImpl::SetAGC(enable=" << enable << ")";
@@ skipping 111 matching lines @@
     return false;
 
   if (!renderer->Initialize(this))
     return false;
 
   renderer_ = renderer;
   return true;
 }
 
 }  // namespace content