Support the Aec dump for the APM in 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 8313fcac5f4228ebf10ff50856e2a12ab2f41edb..e18032b1b28a75f5bab40f342856810b32089507 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -187,9 +187,7 @@ bool WebRtcAudioCapturer::Initialize() {
   // Create and configure the default audio capturing source.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id_),
                     channel_layout,
-                    static_cast<float>(device_info_.device.input.sample_rate),
-                    device_info_.device.input.effects,
-                    constraints_);
+                    static_cast<float>(device_info_.device.input.sample_rate));
 
   // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware
   // information from the capturer.
@@ -205,6 +203,8 @@ WebRtcAudioCapturer::WebRtcAudioCapturer(
     const blink::WebMediaConstraints& constraints,
     WebRtcAudioDeviceImpl* audio_device)
     : constraints_(constraints),
+      audio_processor_(new MediaStreamAudioProcessor(
+          constraints, device_info.device.input.effects, audio_device)),
       running_(false),
       render_view_id_(render_view_id),
       device_info_(device_info),
@@ -261,9 +261,7 @@ void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
 void WebRtcAudioCapturer::SetCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
-    float sample_rate,
-    int effects,
-    const blink::WebMediaConstraints& constraints) {
+    float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
            << "sample_rate=" << sample_rate << ")";
@@ -294,15 +292,16 @@ void WebRtcAudioCapturer::SetCapturerSource(
   int buffer_size = GetBufferSize(sample_rate);
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                                 channel_layout, 0, sample_rate,
-                                16, buffer_size, effects);
-  scoped_refptr<MediaStreamAudioProcessor> new_audio_processor(
-      new MediaStreamAudioProcessor(params, constraints, effects,
-                                    audio_device_));
+                                16, buffer_size,
+                                device_info_.device.input.effects);
+
   {
     base::AutoLock auto_lock(lock_);
-    audio_processor_ = new_audio_processor;
-    need_audio_processing_ = NeedsAudioProcessing(constraints, effects);
+    // Notify the |audio_processor_| of the new format.
+    audio_processor_->OnCaptureFormatChanged(params);
 
+    need_audio_processing_ = NeedsAudioProcessing(
+        constraints_, device_info_.device.input.effects);
     // Notify all tracks about the new format.
     tracks_.TagAll();
   }
@@ -345,12 +344,11 @@ void WebRtcAudioCapturer::EnablePeerConnectionMode() {
   // WebRtc native buffer size.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
                     input_params.channel_layout(),
-                    static_cast<float>(input_params.sample_rate()),
-                    input_params.effects(),
-                    constraints_);
+                    static_cast<float>(input_params.sample_rate()));
 }
 
 void WebRtcAudioCapturer::Start() {
+  DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Start()";
   base::AutoLock auto_lock(lock_);
   if (running_ || !source_)
@@ -364,6 +362,7 @@ void WebRtcAudioCapturer::Start() {
 }
 
 void WebRtcAudioCapturer::Stop() {
+  DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Stop()";
   scoped_refptr<media::AudioCapturerSource> source;
   TrackList::ItemList tracks;
@@ -382,6 +381,9 @@ void WebRtcAudioCapturer::Stop() {
   if (audio_device_)
     audio_device_->RemoveAudioCapturer(this);
 
+  // Stop the Aec dump.
+  StopAecDump();
+
   for (TrackList::ItemList::const_iterator it = tracks.begin();
        it != tracks.end();
        ++it) {
@@ -432,7 +434,6 @@ void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
   TrackList::ItemList tracks_to_notify_format;
   int current_volume = 0;
   base::TimeDelta audio_delay;
-  scoped_refptr<MediaStreamAudioProcessor> audio_processor;
   bool need_audio_processing = true;
   {
     base::AutoLock auto_lock(lock_);
@@ -449,20 +450,19 @@ void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
     key_pressed_ = key_pressed;
     tracks = tracks_.Items();
     tracks_.RetrieveAndClearTags(&tracks_to_notify_format);
-    audio_processor = audio_processor_;
 
     // Set the flag to turn on the audio processing in PeerConnection level.
     // Note that, we turn off the audio processing in PeerConnection if the
     // processor has already processed the data.
     need_audio_processing = need_audio_processing_ ?
-        !audio_processor->has_audio_processing() : false;
+        !audio_processor_->has_audio_processing() : false;
   }
 
-  DCHECK(audio_processor->InputFormat().IsValid());
+  DCHECK(audio_processor_->InputFormat().IsValid());
   DCHECK_EQ(audio_source->channels(),
-            audio_processor->InputFormat().channels());
+            audio_processor_->InputFormat().channels());
   DCHECK_EQ(audio_source->frames(),
-           audio_processor->InputFormat().frames_per_buffer());
+            audio_processor_->InputFormat().frames_per_buffer());
 
   // Notify the tracks on when the format changes. This will do nothing if
   // |tracks_to_notify_format| is empty.
@@ -473,13 +473,13 @@ void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
   }
 
   // Push the data to the processor for processing.
-  audio_processor->PushCaptureData(audio_source);
+  audio_processor_->PushCaptureData(audio_source);
 
   // Process and consume the data in the processor until there is not enough
   // data in the processor.
   int16* output = NULL;
   int new_volume = 0;
-  while (audio_processor->ProcessAndConsumeData(
+  while (audio_processor_->ProcessAndConsumeData(
       audio_delay, current_volume, key_pressed, &new_volume, &output)) {
     // Feed the post-processed data to the tracks.
     for (TrackList::ItemList::const_iterator it = tracks.begin();
@@ -561,9 +561,19 @@ void WebRtcAudioCapturer::SetCapturerSourceForTesting(
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSource(source, params.channel_layout(),
-                    static_cast<float>(params.sample_rate()),
-                    params.effects(),
-                    constraints_);
+                    static_cast<float>(params.sample_rate()));
+}
+
+void WebRtcAudioCapturer::StartAecDump(
+    const base::PlatformFile& aec_dump_file) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK_NE(aec_dump_file, base::kInvalidPlatformFileValue);
+  audio_processor_->StartAecDump(aec_dump_file);
+}
+
+void WebRtcAudioCapturer::StopAecDump() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  audio_processor_->StopAecDump();
 }
 
 }  // namespace content