Stop feeding render data to AEC after audio processor has stopped.

BUG=389140
TEST=

Review URL: https://codereview.chromium.org/360703002

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@280611 0039d316-1c4b-4281-b951-d872f2087c98

3 files changed, 37 insertions, 22 deletions

diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc
index 12e8d34..d2e0695 100644
--- a/content/renderer/media/media_stream_audio_processor.cc
+++ b/content/renderer/media/media_stream_audio_processor.cc
@@ -173,7 +173,8 @@ MediaStreamAudioProcessor::MediaStreamAudioProcessor(
     : render_delay_ms_(0),
       playout_data_source_(playout_data_source),
       audio_mirroring_(false),
-      typing_detected_(false) {
+      typing_detected_(false),
+      stopped_(false) {
   capture_thread_checker_.DetachFromThread();
   render_thread_checker_.DetachFromThread();
   InitializeAudioProcessingModule(constraints, effects);
@@ -189,11 +190,7 @@ MediaStreamAudioProcessor::MediaStreamAudioProcessor(
 
 MediaStreamAudioProcessor::~MediaStreamAudioProcessor() {
   DCHECK(main_thread_checker_.CalledOnValidThread());
-  if (aec_dump_message_filter_) {
-    aec_dump_message_filter_->RemoveDelegate(this);
-    aec_dump_message_filter_ = NULL;
-  }
-  StopAudioProcessing();
+  Stop();
 }
 
 void MediaStreamAudioProcessor::OnCaptureFormatChanged(
@@ -242,6 +239,29 @@ bool MediaStreamAudioProcessor::ProcessAndConsumeData(
   return true;
 }
 
+void MediaStreamAudioProcessor::Stop() {
+  DCHECK(main_thread_checker_.CalledOnValidThread());
+  if (stopped_)
+    return;
+
+  stopped_ = true;
+
+  if (aec_dump_message_filter_) {
+    aec_dump_message_filter_->RemoveDelegate(this);
+    aec_dump_message_filter_ = NULL;
+  }
+
+  if (!audio_processing_.get())
+    return;
+
+  StopEchoCancellationDump(audio_processing_.get());
+
+  if (playout_data_source_) {
+    playout_data_source_->RemovePlayoutSink(this);
+    playout_data_source_ = NULL;
+  }
+}
+
 const media::AudioParameters& MediaStreamAudioProcessor::InputFormat() const {
   return capture_converter_->source_parameters();
 }
@@ -516,16 +536,4 @@ int MediaStreamAudioProcessor::ProcessData(webrtc::AudioFrame* audio_frame,
       0 : agc->stream_analog_level();
 }
 
-void MediaStreamAudioProcessor::StopAudioProcessing() {
-  if (!audio_processing_.get())
-    return;
-
-  StopEchoCancellationDump(audio_processing_.get());
-
-  if (playout_data_source_)
-    playout_data_source_->RemovePlayoutSink(this);
-
-  audio_processing_.reset();
-}
-
 }  // namespace content
diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h
index 7232ae6..08f8ed0 100644
--- a/content/renderer/media/media_stream_audio_processor.h
+++ b/content/renderer/media/media_stream_audio_processor.h
@@ -89,6 +89,10 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
                              int* new_volume,
                              int16** out);
 
+  // Stops the audio processor, no more AEC dump or render data after calling
+  // this method.
+  void Stop();
+
   // The audio format of the input to the processor.
   const media::AudioParameters& InputFormat() const;
 
@@ -146,9 +150,6 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
                   int volume,
                   bool key_pressed);
 
-  // Called when the processor is going away.
-  void StopAudioProcessing();
-
   // Cached value for the render delay latency. This member is accessed by
   // both the capture audio thread and the render audio thread.
   base::subtle::Atomic32 render_delay_ms_;
@@ -175,7 +176,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
 
   // Raw pointer to the WebRtcPlayoutDataSource, which is valid for the
   // lifetime of RenderThread.
-  WebRtcPlayoutDataSource* const playout_data_source_;
+  WebRtcPlayoutDataSource* playout_data_source_;
 
   // Used to DCHECK that the destructor is called on the main render thread.
   base::ThreadChecker main_thread_checker_;
@@ -199,6 +200,9 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
 
   // Communication with browser for AEC dump.
   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
+
+  // Flag to avoid executing Stop() more than once.
+  bool stopped_;
 };
 
 }  // namespace content
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 6212a8e..056cb2b 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -425,6 +425,9 @@ void WebRtcAudioCapturer::Stop() {
 
   if (source.get())
     source->Stop();
+
+  // Stop the audio processor to avoid feeding render data into the processor.
+  audio_processor_->Stop();
 }
 
 void WebRtcAudioCapturer::SetVolume(int volume) {