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

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 0f35b4f59f1118bd3fafdafbb9975b4c3160ca6b..6fb21eb2e30f11ae525cc9aecd652a70163d0fb1 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -164,7 +164,8 @@ bool WebRtcAudioCapturer::Initialize(int render_view_id,
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer()
-    : source_(NULL),
+    : default_sink_(NULL),
+      source_(NULL),
       running_(false),
       agc_is_enabled_(false),
       session_id_(0) {
@@ -175,14 +176,31 @@ WebRtcAudioCapturer::~WebRtcAudioCapturer() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(tracks_.empty());
   DCHECK(!running_);
+  DCHECK(!default_sink_);
   DVLOG(1) << "WebRtcAudioCapturer::~WebRtcAudioCapturer()";
 }
 
-void WebRtcAudioCapturer::AddSink(
-    WebRtcAudioCapturerSink* track) {
-  DCHECK(thread_checker_.CalledOnValidThread());
+void WebRtcAudioCapturer::SetDefaultSink(WebRtcAudioCapturerSink* sink) {
+  DVLOG(1) << "WebRtcAudioCapturer::SetDefaultSink()";
+  if (sink) {
+    DCHECK(!default_sink_);
+    default_sink_ = sink;
+    AddSink(sink);
+  } else {
+    DCHECK(default_sink_);
+    RemoveSink(default_sink_);
+    default_sink_ = NULL;
+  }
+}
+
+void WebRtcAudioCapturer::AddSink(WebRtcAudioCapturerSink* track) {
   DCHECK(track);
   DVLOG(1) << "WebRtcAudioCapturer::AddSink()";
+
+  // Start the source if a real audio track is connected to the capturer.

[henrika (OOO until Aug 14), 2013/06/05 09:09:20]

Can you please extend the comment and give an example on what is not a "real"
audio track.

[no longer working on chromium, 2013/06/05 16:29:45]

Done.

+  if (track != default_sink_)
+    Start();
+
   base::AutoLock auto_lock(lock_);
   // Verify that |track| is not already added to the list.

[henrika (OOO until Aug 14), 2013/06/05 09:09:20]

What happens if it is (in release mode)?

[no longer working on chromium, 2013/06/05 16:29:45]

It is a push_back, then we will have two reference to the same audio track in
the list.

   DCHECK(std::find_if(
@@ -208,19 +226,33 @@ void WebRtcAudioCapturer::RemoveSink(
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::RemoveSink()";
 
-  base::AutoLock auto_lock(lock_);
+  bool stop_source = false;
+  {
+    base::AutoLock auto_lock(lock_);
 
-  // Get iterator to the first element for which WrapsSink(track) returns true.
-  TrackList::iterator it = std::find_if(
-      tracks_.begin(), tracks_.end(),
-      WebRtcAudioCapturerSinkOwner::WrapsSink(track));
-  if (it != tracks_.end()) {
-    // 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.
-    (*it)->Reset();
-    tracks_.erase(it);
+    // Get iterator to the first element for which WrapsSink(track) returns
+    // true.
+    TrackList::iterator it = std::find_if(
+        tracks_.begin(), tracks_.end(),
+        WebRtcAudioCapturerSinkOwner::WrapsSink(track));
+    if (it != tracks_.end()) {
+      // 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.
+      (*it)->Reset();
+      tracks_.erase(it);
+    }
+
+    if (tracks_.size() == 1 && default_sink_ &&

[henrika (OOO until Aug 14), 2013/06/05 09:09:20]

You must really add clear comments here explaining exactly what has to be
fulfilled for Stop to be called. Can't tracks_.size() be 0 if only one track had
been added and you have called erase above? OK, covered in else.

[no longer working on chromium, 2013/06/05 16:29:45]

Done.

+        (*tracks_.begin())->IsEqual(default_sink_)) {
+      stop_source = true;
+    } else {
+      stop_source = tracks_.empty();
+    }
   }
+
+  if (stop_source)
+    Stop();
 }
 
 void WebRtcAudioCapturer::SetCapturerSource(