Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/codecs/audio_format.cc

 /*
  *  Copyright (c) 2016 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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_format.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 
 namespace webrtc {
 
 SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
 SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
 
 SdpAudioFormat::SdpAudioFormat(const char* name,
                                int clockrate_hz,
                                int num_channels)
@@ skipping 33 matching lines @@
   os << ", parameters: {";
   const char* sep = "";
   for (const auto& kv : saf.parameters) {
     os << sep << kv.first << ": " << kv.second;
     sep = ", ";
   }
   os << "}}";
   return os;
 }
 
+SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
+  if (STR_CASE_CMP(ci.plname, "g722") == 0 && ci.plfreq == 16000) {
+    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
+    return {"g722", 8000, ci.channels};
+  } else if (STR_CASE_CMP(ci.plname, "opus") == 0 && ci.plfreq == 48000) {
+    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
+    return {"opus", 48000, 2, {{"stereo", ci.channels == 1 ? "0" : "1"}}};
+  } else {
+    return {ci.plname, ci.plfreq, ci.channels};
+  }
+}
+
 }  // namespace webrtc