move the APM to chrome.

content/renderer/media/media_stream_dependency_factory.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/media_stream_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 47 matching lines @@
   { webrtc::MediaConstraintsInterface::kEchoCancellation,
     webrtc::MediaConstraintsInterface::kValueTrue },
 #if defined(OS_CHROMEOS) || defined(OS_MACOSX)
   // Enable the extended filter mode AEC on platforms with known echo issues.
   { webrtc::MediaConstraintsInterface::kExperimentalEchoCancellation,
     webrtc::MediaConstraintsInterface::kValueTrue },
 #endif
   { webrtc::MediaConstraintsInterface::kAutoGainControl,
     webrtc::MediaConstraintsInterface::kValueTrue },
   { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl,
-    webrtc::MediaConstraintsInterface::kValueTrue },
+    webrtc::MediaConstraintsInterface::kValueFalse },
   { webrtc::MediaConstraintsInterface::kNoiseSuppression,
     webrtc::MediaConstraintsInterface::kValueTrue },
   { webrtc::MediaConstraintsInterface::kHighpassFilter,
     webrtc::MediaConstraintsInterface::kValueTrue },
 };
 
 // Merge |constraints| with |kDefaultAudioConstraints|. For any key which exists
 // in both, the value from |constraints| is maintained, including its
 // mandatory/optional status. New values from |kDefaultAudioConstraints| will
 // be added with mandatory status.
@@ skipping 355 matching lines @@
     scoped_refptr<WebRtcAudioCapturer> capturer;
     if (GetWebRtcAudioDevice())
       capturer = GetWebRtcAudioDevice()->GetDefaultCapturer();
 
     scoped_refptr<webrtc::AudioTrackInterface> audio_track(
         CreateLocalAudioTrack(track_id,
                               capturer,
                               webaudio_source.get(),
                               source_data->local_audio_source(),
                               &track_constraints));
+    // Set the source provider to the webkit.
+    WebKit::WebMediaStreamTrack writable_track = track;
+    writable_track.setSourceProvider(static_cast<WebRtcLocalAudioTrack*>(
+        audio_track.get())->audio_source_provider());
     audio_track->set_enabled(track.isEnabled());
-    if (capturer.get()) {
-      WebKit::WebMediaStreamTrack writable_track = track;
-      writable_track.setSourceProvider(capturer->audio_source_provider());
-    }
     return native_stream->AddTrack(audio_track.get());
   } else {
     DCHECK(source.type() == WebKit::WebMediaStreamSource::TypeVideo);
     scoped_refptr<webrtc::VideoTrackInterface> video_track(
         CreateLocalVideoTrack(track_id, source_data->video_source()));
     video_track->set_enabled(track.isEnabled());
     return native_stream->AddTrack(video_track.get());
   }
 }
 
@@ skipping 206 matching lines @@
     const std::string& id,
     const scoped_refptr<WebRtcAudioCapturer>& capturer,
     WebAudioCapturerSource* webaudio_source,
     webrtc::AudioSourceInterface* source,
     const webrtc::MediaConstraintsInterface* constraints) {
   // TODO(xians): Merge |source| to the capturer(). We can't do this today
   // because only one capturer() is supported while one |source| is created
   // for each audio track.
   scoped_refptr<WebRtcLocalAudioTrack> audio_track(
       WebRtcLocalAudioTrack::Create(id, capturer, webaudio_source,
-                                    source, constraints));
+                                    source, constraints,
+                                    GetWebRtcAudioDevice()));
 
   // Add the WebRtcAudioDevice as the sink to the local audio track.
   audio_track->AddSink(GetWebRtcAudioDevice());
   // Start the audio track. This will hook the |audio_track| to the capturer
   // as the sink of the audio, and only start the source of the capturer if
   // it is the first audio track connecting to the capturer.
   audio_track->Start();
   return audio_track;
 }
 
@@ skipping 153 matching lines @@
   }
 
   // Add the capturer to the WebRtcAudioDeviceImpl if it is a new capturer.
   if (is_new_capturer)
     GetWebRtcAudioDevice()->AddAudioCapturer(capturer);
 
   return capturer;
 }
 
 }  // namespace content