Fold AudioEncoderMutable into AudioEncoder

webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc

 /*
  *  Copyright (c) 2014 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 47 matching lines @@
   codec.pltype = payload_type;
   codec.pacsize = frame_size_samples;
   codec_registered_ = (acm_->RegisterSendCodec(codec) == 0);
   input_frame_.num_channels_ = channels;
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   return codec_registered_;
 }
 
 bool AcmSendTestOldApi::RegisterExternalCodec(
-    AudioEncoderMutable* external_speech_encoder) {
+    AudioEncoder* external_speech_encoder) {
   acm_->RegisterExternalSendCodec(external_speech_encoder);
   input_frame_.num_channels_ = external_speech_encoder->NumChannels();
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   return codec_registered_ = true;
 }
 
 Packet* AcmSendTestOldApi::NextPacket() {
   assert(codec_registered_);
   if (filter_.test(static_cast<size_t>(payload_type_))) {
@@ skipping 69 matching lines @@
          last_payload_vec_.size());
   Packet* packet =
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
   assert(packet);
   assert(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc