Moving/renaming webrtc/common.h.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 24 matching lines @@
 #include "webrtc/voice_engine/shared_data.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace rtc {
 class TimestampWrapAroundHandler;
 }
 
 namespace webrtc {
 
 class AudioDeviceModule;
-class Config;
 class FileWrapper;
 class PacketRouter;
 class ProcessThread;
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class RTPReceiverAudio;
@@ skipping 112 matching lines @@
  public:
   friend class VoERtcpObserver;
 
   enum { KNumSocketThreads = 1 };
   enum { KNumberOfSocketBuffers = 8 };
   virtual ~Channel();
   static int32_t CreateChannel(
       Channel*& channel,
       int32_t channelId,
       uint32_t instanceId,
-      const Config& config,
-      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
+      const AudioCodingModule::Config& acm_config);
   Channel(int32_t channelId,
           uint32_t instanceId,
-          const Config& config,
-          const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
+          const AudioCodingModule::Config& acm_config);
   int32_t Init();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                OutputMixer& outputMixer,
                                TransmitMixer& transmitMixer,
                                ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
                                VoiceEngineObserver* voiceEngineObserver,
                                rtc::CriticalSection* callbackCritSect);
   int32_t UpdateLocalTimeStamp();
 
@@ skipping 390 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_