Update MediaStreamTrack::Stop to latest draft.

content/renderer/media/webrtc_audio_capturer.cc

Index: content/renderer/media/webrtc_audio_capturer.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 7363b3e7d13fe436bebf1882e357d1f36026e9e4..e942f8f5e4f2ce97bb73cbaa02b11f9d5f30b68a 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -13,6 +13,7 @@
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
+#include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "content/renderer/media/webrtc_logging.h"
@@ -87,7 +88,7 @@ class WebRtcAudioCapturer::TrackOwner
     // This can be reentrant so reset |delegate_| before calling out.
     WebRtcLocalAudioTrack* temp = delegate_;
     delegate_ = NULL;
-    temp->Stop();
+    temp->StopTrack();
   }
 
   // Wrapper which allows to use std::find_if() when adding and removing
@@ -123,9 +124,10 @@ class WebRtcAudioCapturer::TrackOwner
 scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer(
     int render_view_id, const StreamDeviceInfo& device_info,
     const blink::WebMediaConstraints& constraints,
-    WebRtcAudioDeviceImpl* audio_device) {
+    WebRtcAudioDeviceImpl* audio_device,
+    MediaStreamAudioSource* audio_source) {
   scoped_refptr<WebRtcAudioCapturer> capturer = new WebRtcAudioCapturer(
-      render_view_id, device_info, constraints, audio_device);
+      render_view_id, device_info, constraints, audio_device, audio_source);
   if (capturer->Initialize())
     return capturer;
 
@@ -208,7 +210,8 @@ WebRtcAudioCapturer::WebRtcAudioCapturer(
     int render_view_id,
     const StreamDeviceInfo& device_info,
     const blink::WebMediaConstraints& constraints,
-    WebRtcAudioDeviceImpl* audio_device)
+    WebRtcAudioDeviceImpl* audio_device,
+    MediaStreamAudioSource* audio_source)
     : constraints_(constraints),
       audio_processor_(
           new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
@@ -221,7 +224,8 @@ WebRtcAudioCapturer::WebRtcAudioCapturer(
       peer_connection_mode_(false),
       key_pressed_(false),
       need_audio_processing_(false),
-      audio_device_(audio_device) {
+      audio_device_(audio_device),
+      audio_source_(audio_source) {
   DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
 }
 
@@ -255,16 +259,30 @@ void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) {
 
 void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
   DCHECK(thread_checker_.CalledOnValidThread());
-  base::AutoLock auto_lock(lock_);
+  DVLOG(1) << "WebRtcAudioCapturer::RemoveTrack()";
+  bool stop_source = false;
+  {
+    base::AutoLock auto_lock(lock_);
 
-  scoped_refptr<TrackOwner> removed_item =
-      tracks_.Remove(TrackOwner::TrackWrapper(track));
+    scoped_refptr<TrackOwner> removed_item =
+        tracks_.Remove(TrackOwner::TrackWrapper(track));
 
-  // Clear the delegate to ensure that no more capture callbacks will
-  // be sent to this sink. Also avoids a possible crash which can happen
-  // if this method is called while capturing is active.
-  if (removed_item.get())
-    removed_item->Reset();
+    // Clear the delegate to ensure that no more capture callbacks will
+    // be sent to this sink. Also avoids a possible crash which can happen
+    // if this method is called while capturing is active.
+    if (removed_item.get()) {
+      removed_item->Reset();
+      stop_source = tracks_.IsEmpty();
+    }
+  }
+  if (stop_source) {
+    // Since WebRtcAudioCapturer does not inherit MediaStreamAudioSource,
+    // and instead MediaStreamAudioSource is composed of a WebRtcAudioCapturer,
+    // we have to call StopSource on the MediaStreamSource. This will call
+    // MediaStreamAudioSource::DoStopSource which in turn call
+    // WebRtcAudioCapturerer::Stop();
+    audio_source_->StopSource();

[no longer working on chromium, 2014/04/02 13:50:56]

you will trigger WebRtcAudioCapturerer::Stop() here, but if a new track is added
to the capturer after the Stop(), the new track will call Start() again. You can
look at AddTrack for details.
We talked about that before, and you said that the source should not be started
again after being stopped, is this still the case?

[perkj_chrome, 2014/04/03 11:58:34]

We discussed offline and decided to start the capturere during initialization.
It stopped either when the last track is removed or on destruction. It can never
be restarted.

+  }
 }
 
 void WebRtcAudioCapturer::SetCapturerSource(