Merge 198717 "Fixed the crash when the WebRtcAudioCapturer::|buf..."

> Fixed the crash when the WebRtcAudioCapturer::|buffer_| is NULL, this can happen when the capturer is not correctly initialized.
> 
> BUG=238388
> TEST=http://simpl.info/webaudio
> R=henrika@chromium.org
> 
> Review URL: https://codereview.chromium.org/15031003

TBR=xians@chromium.org

Review URL: https://codereview.chromium.org/15014019

git-svn-id: svn://svn.chromium.org/chrome/branches/1500/src@199431 0039d316-1c4b-4281-b951-d872f2087c98

content/renderer/media/webrtc_audio_capturer.cc

@@ -201,12 +201,21 @@ WebRtcAudioCapturer::~WebRtcAudioCapturer() {
 void WebRtcAudioCapturer::AddSink(
     WebRtcAudioCapturerSink* track) {
   DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(track);
   DVLOG(1) << "WebRtcAudioCapturer::AddSink()";
   base::AutoLock auto_lock(lock_);
   // Verify that |track| is not already added to the list.
   DCHECK(std::find_if(
       tracks_.begin(), tracks_.end(),
       WebRtcAudioCapturerSinkOwner::WrapsSink(track)) == tracks_.end());
+
+  if (buffer_.get()) {
+    track->SetCaptureFormat(buffer_->params());
+  } else {
+    DLOG(WARNING) << "The format of the capturer has not been correctly "
+                  <<  "initialized";
+  }
+
   // Create (and add to the list) a new WebRtcAudioCapturerSinkOwner which owns
   // the |track| and delagates all calls to the WebRtcAudioCapturerSink
   // interface.
@@ -379,7 +388,9 @@ void WebRtcAudioCapturer::OnCaptureError() {
 
 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const {
   base::AutoLock auto_lock(lock_);
-  return buffer_->params();
+  // |buffer_| can be NULL when SetCapturerSource() or Initialize() has not
+  // been called.
+  return buffer_.get() ? buffer_->params() : media::AudioParameters();
 }
 
 }  // namespace content

content/renderer/media/webrtc_local_audio_track.cc

@@ -30,7 +30,6 @@ WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
       track_source_(track_source) {
   DCHECK(capturer);
   capturer_->AddSink(this);
-  params_ = capturer_->audio_parameters();
   DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
 }
 
@@ -72,6 +71,11 @@ void WebRtcLocalAudioTrack::SetCaptureFormat(
     const media::AudioParameters& params) {
   base::AutoLock auto_lock(lock_);
   params_ = params;
+
+  // Update all the existing sinks with the new format.
+  for (SinkList::const_iterator it = sinks_.begin();
+       it != sinks_.end(); ++it)
+    (*it)->SetCaptureFormat(params);
 }
 
 // webrtc::AudioTrackInterface implementation.