Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize

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

 /*
  *  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 715 matching lines @@
     aux_rtp_header_->type.Audio.channel = 1;
   }
 
   aux_rtp_header_->header.timestamp = timestamp;
   IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_);
   // Get ready for the next payload.
   aux_rtp_header_->header.sequenceNumber++;
   return 0;
 }
 
-// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine.
-int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
-
-  if (!HaveValidEncoder("SetISACMaxRate")) {
-    return -1;
-  }
-
-  codec_manager_.CurrentEncoder()->SetMaxBitrate(max_bit_per_sec);
-  return 0;
-}
-
-// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine.
-int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
-
-  if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
-    return -1;
-  }
-
-  codec_manager_.CurrentEncoder()->SetMaxPayloadSize(max_size_bytes);
-  return 0;
-}
-
 int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   if (!HaveValidEncoder("SetOpusApplication")) {
     return -1;
   }
   if (!codec_manager_.CurrentEncoderIsOpus())
     return -1;
   AudioEncoder::Application app;
   switch (application) {
     case kVoip:
@@ skipping 387 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc