move the APM to chrome.

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 aa3ecf8627f84ca4165d481afd14b50388faf291..00b3b8baf465542b106739d70988f98677e9a7e8 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -63,6 +63,8 @@ class WebRtcAudioCapturer::TrackOwner
 
   void Reset() {
     base::AutoLock lock(lock_);
+    if (delegate_)
+      delegate_->Stop();
     delegate_ = NULL;
   }
 
@@ -101,8 +103,9 @@ scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer() {
   return capturer;
 }
 
-void WebRtcAudioCapturer::Reconfigure(int sample_rate,
-                                      media::ChannelLayout channel_layout) {
+void WebRtcAudioCapturer::Reconfigure(
+    int sample_rate,
+    media::ChannelLayout channel_layout) {
   DCHECK(thread_checker_.CalledOnValidThread());
   int buffer_size = GetBufferSize(sample_rate);
   DVLOG(1) << "Using WebRTC input buffer size: " << buffer_size;
@@ -114,6 +117,7 @@ void WebRtcAudioCapturer::Reconfigure(int sample_rate,
   int bits_per_sample = 16;
   media::AudioParameters params(format, channel_layout, sample_rate,
                                 bits_per_sample, buffer_size);
+  DCHECK(params.IsValid());
 
   TrackList tracks;
   {
@@ -128,14 +132,15 @@ void WebRtcAudioCapturer::Reconfigure(int sample_rate,
     (*it)->SetCaptureFormat(params);
 }
 
-bool WebRtcAudioCapturer::Initialize(int render_view_id,
-                                     media::ChannelLayout channel_layout,
-                                     int sample_rate,
-                                     int buffer_size,
-                                     int session_id,
-                                     const std::string& device_id,
-                                     int paired_output_sample_rate,
-                                     int paired_output_frames_per_buffer) {
+bool WebRtcAudioCapturer::Initialize(
+    int render_view_id,
+    media::ChannelLayout channel_layout,
+    int sample_rate,
+    int buffer_size,
+    int session_id,
+    const std::string& device_id,
+    int paired_output_sample_rate,
+    int paired_output_frames_per_buffer) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK_GE(render_view_id, 0);
   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
@@ -187,9 +192,9 @@ bool WebRtcAudioCapturer::Initialize(int render_view_id,
   // Create and configure the default audio capturing source. The |source_|
   // will be overwritten if an external client later calls SetCapturerSource()
   // providing an alternative media::AudioCapturerSource.
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
-                    channel_layout,
-                    static_cast<float>(sample_rate));
+  InitializeCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
+                           channel_layout,
+                           static_cast<float>(sample_rate));
 
   return true;
 }
@@ -202,11 +207,11 @@ WebRtcAudioCapturer::WebRtcAudioCapturer()
       hardware_buffer_size_(0),
       session_id_(0),
       volume_(0),
-      source_provider_(new WebRtcLocalAudioSourceProvider()),
       peer_connection_mode_(false),
       output_sample_rate_(0),
-      output_frames_per_buffer_(0) {
-  DCHECK(source_provider_.get());
+      output_frames_per_buffer_(0),
+      audio_delay_ms_(0),
+      key_pressed_(false) {
   DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
 }
 
@@ -230,7 +235,9 @@ void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) {
   DCHECK(std::find_if(tracks_.begin(), tracks_.end(),
                       TrackOwner::TrackWrapper(track)) == tracks_.end());
 
-  track->SetCaptureFormat(params_);
+  if (params_.IsValid())
+    track->SetCaptureFormat(params_);
+
   tracks_.push_back(new WebRtcAudioCapturer::TrackOwner(track));
 }
 
@@ -260,13 +267,13 @@ void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
     Stop();
 }
 
-void WebRtcAudioCapturer::SetCapturerSource(
+void WebRtcAudioCapturer::InitializeCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
     float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
-           << "sample_rate=" << sample_rate << ")";
+  DVLOG(1) << "InitializeCapturerSource(channel_layout=" << channel_layout
+           << "," << "sample_rate=" << sample_rate << ")";
   scoped_refptr<media::AudioCapturerSource> old_source;
   bool restart_source = false;
   {
@@ -293,7 +300,6 @@ void WebRtcAudioCapturer::SetCapturerSource(
 
   // Make sure to grab the new parameters in case they were reconfigured.
   media::AudioParameters params = audio_parameters();
-  source_provider_->Initialize(params);
   if (source.get())
     source->Initialize(params, this, session_id_);
 
@@ -327,9 +333,9 @@ void WebRtcAudioCapturer::EnablePeerConnectionMode() {
 
   // Create a new audio stream as source which will open the hardware using
   // WebRtc native buffer size.
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
-                    params.channel_layout(),
-                    static_cast<float>(params.sample_rate()));
+  InitializeCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
+                           params.channel_layout(),
+                           static_cast<float>(params.sample_rate()));
 }
 
 void WebRtcAudioCapturer::Start() {
@@ -352,17 +358,22 @@ void WebRtcAudioCapturer::Start() {
 void WebRtcAudioCapturer::Stop() {
   DVLOG(1) << "WebRtcAudioCapturer::Stop()";
   scoped_refptr<media::AudioCapturerSource> source;
+  TrackList tracks;
   {
     base::AutoLock auto_lock(lock_);
     if (!running_)
       return;
 
     source = source_;
+    tracks = tracks_;
     running_ = false;
   }
 
   if (source.get())
     source->Stop();
+
+  for (TrackList::const_iterator it = tracks.begin(); it != tracks.end(); ++it)
+    (*it)->Reset();
 }
 
 void WebRtcAudioCapturer::SetVolume(int volume) {
@@ -426,11 +437,6 @@ void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
     tracks = tracks_;
   }
 
-  // Deliver captured data to source provider, which stores the data into FIFO
-  // for WebAudio to fetch.
-  source_provider_->DeliverData(audio_source, audio_delay_milliseconds,
-                                current_volume, key_pressed);
-
   // Feed the data to the tracks.
   for (TrackList::const_iterator it = tracks.begin();
        it != tracks.end();
@@ -478,4 +484,12 @@ int WebRtcAudioCapturer::GetBufferSize(int sample_rate) const {
   return (sample_rate / 100);
 }
 
+void WebRtcAudioCapturer::GetAudioProcessingParams(
+    int* delay_ms, int* volume, bool* key_pressed) {
+  base::AutoLock auto_lock(lock_);
+  *delay_ms = audio_delay_ms_;
+  *volume = volume_;
+  *key_pressed = key_pressed_;
+}
+
 }  // namespace content