content: Cleanup class/struct forward declarations

content/renderer/media/webrtc/processed_local_audio_source.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 
 #include "base/atomicops.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "content/common/media/media_stream_options.h"
 #include "content/renderer/media/media_stream_audio_level_calculator.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "media/base/audio_capturer_source.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 
 namespace media {
 class AudioBus;
 }
 
-namespace webrtc {
-class AudioSourceInterface;
-}
-
 namespace content {
 
 class PeerConnectionDependencyFactory;
 
 // Represents a local source of audio data that is routed through the WebRTC
 // audio pipeline for post-processing (e.g., for echo cancellation during a
 // video conferencing call). Owns a media::AudioCapturerSource and the
 // MediaStreamProcessor that modifies its audio. Modified audio is delivered to
 // one or more MediaStreamAudioTracks.
 class CONTENT_EXPORT ProcessedLocalAudioSource final
@@ skipping 97 matching lines @@
   MediaStreamAudioLevelCalculator level_calculator_;
 
   bool allow_invalid_render_frame_id_for_testing_;
 
   DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_