Fold AudioEncoderMutable into AudioEncoder

webrtc/modules/audio_coding/main/interface/audio_coding_module.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 10 matching lines @@
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // forward declarations
 struct CodecInst;
 struct WebRtcRTPHeader;
 class AudioFrame;
 class RTPFragmentationHeader;
-class AudioEncoderMutable;
+class AudioEncoder;
 class AudioDecoder;
 
 #define WEBRTC_10MS_PCM_AUDIO 960  // 16 bits super wideband 48 kHz
 
 // Callback class used for sending data ready to be packetized
 class AudioPacketizationCallback {
  public:
   virtual ~AudioPacketizationCallback() {}
 
   virtual int32_t SendData(FrameType frame_type,
@@ skipping 176 matching lines @@
   //
   // Return value:
   //   -1 if failed to initialize,
   //    0 if succeeded.
   //
   virtual int32_t RegisterSendCodec(const CodecInst& send_codec) = 0;
 
   // Registers |external_speech_encoder| as encoder. The new encoder will
   // replace any previously registered speech encoder (internal or external).
   virtual void RegisterExternalSendCodec(
-      AudioEncoderMutable* external_speech_encoder) = 0;
+      AudioEncoder* external_speech_encoder) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SendCodec()
   // Get parameters for the codec currently registered as send codec.
   //
   // Output:
   //   -current_send_codec          : parameters of the send codec.
   //
   // Return value:
   //   -1 if failed to get send codec,
@@ skipping 803 matching lines @@
   virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
 
   // Returns the timing statistics for calls to Get10MsAudio.
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_