Enable platform echo cancellation through the AudioRecord path.

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 822c13aea90d8552fdcddb93c49a78c9e2822976..12206bd165f80d53ad363c696721f2096df0a576 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -104,7 +104,8 @@ scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer() {
 }
 
 void WebRtcAudioCapturer::Reconfigure(int sample_rate,
-                                      media::ChannelLayout channel_layout) {
+                                      media::ChannelLayout channel_layout,
+                                      bool use_platform_aec) {
   DCHECK(thread_checker_.CalledOnValidThread());
   int buffer_size = GetBufferSize(sample_rate);
   DVLOG(1) << "Using WebRTC input buffer size: " << buffer_size;
@@ -116,6 +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);
+  params.set_use_platform_aec(use_platform_aec);
 
   {
     base::AutoLock auto_lock(lock_);
@@ -134,7 +136,8 @@ bool WebRtcAudioCapturer::Initialize(int render_view_id,
                                      int session_id,
                                      const std::string& device_id,
                                      int paired_output_sample_rate,
-                                     int paired_output_frames_per_buffer) {
+                                     int paired_output_frames_per_buffer,
+                                     bool use_platform_aec) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
 
@@ -200,7 +203,8 @@ bool WebRtcAudioCapturer::Initialize(int render_view_id,
   // providing an alternative media::AudioCapturerSource.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
                     channel_layout,
-                    static_cast<float>(sample_rate));
+                    static_cast<float>(sample_rate),
+                    use_platform_aec);
 
   return true;
 }
@@ -286,10 +290,12 @@ void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
 void WebRtcAudioCapturer::SetCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
-    float sample_rate) {
+    float sample_rate,
+    bool use_platform_aec) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
-           << "sample_rate=" << sample_rate << ")";
+           << "sample_rate=" << sample_rate << "," << "use_platform_aec="
+           << use_platform_aec << ")";
   scoped_refptr<media::AudioCapturerSource> old_source;
   bool restart_source = false;
   {
@@ -312,7 +318,7 @@ void WebRtcAudioCapturer::SetCapturerSource(
   // Dispatch the new parameters both to the sink(s) and to the new source.
   // The idea is to get rid of any dependency of the microphone parameters
   // which would normally be used by default.
-  Reconfigure(sample_rate, channel_layout);
+  Reconfigure(sample_rate, channel_layout, use_platform_aec);
 
   // Make sure to grab the new parameters in case they were reconfigured.
   media::AudioParameters params = audio_parameters();
@@ -351,7 +357,8 @@ void WebRtcAudioCapturer::EnablePeerConnectionMode() {
   // WebRtc native buffer size.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
                     params.channel_layout(),
-                    static_cast<float>(params.sample_rate()));
+                    static_cast<float>(params.sample_rate()),
+                    params.use_platform_aec());
 }
 
 void WebRtcAudioCapturer::Start() {