Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/utility/source/coder.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 27 matching lines @@
 
 AudioCoder::~AudioCoder() {}
 
 int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
   const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
                        codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
   return success ? 0 : -1;
 }
 
 int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
-  if (acm_->RegisterReceiveCodec(codec_inst, [&] {
-        return rent_a_codec_.RentIsacDecoder(codec_inst.plfreq);
-      }) == -1) {
+  if (!acm_->RegisterReceiveCodec(codec_inst.pltype,
+                                  CodecInstToSdp(codec_inst))) {
     return -1;
   }
   memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
   return 0;
 }
 
 int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
                            uint32_t samp_freq_hz,
                            const int8_t* incoming_payload,
                            size_t payload_length) {
@@ skipping 46 matching lines @@
                              uint32_t /* time_stamp */,
                              const uint8_t* payload_data,
                              size_t payload_size,
                              const RTPFragmentationHeader* /* fragmentation*/) {
   memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
   encoded_length_in_bytes_ = payload_size;
   return 0;
 }
 
 }  // namespace webrtc