Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/codecs/audio_format.cc

Index: webrtc/modules/audio_coding/codecs/audio_format.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_format.cc b/webrtc/modules/audio_coding/codecs/audio_format.cc
index ebd7cb030b4fdbcec19eaef2cbc09cff856ec376..a43203440bfdc3f24e152d80639441dcff79900f 100644
--- a/webrtc/modules/audio_coding/codecs/audio_format.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_format.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_coding/codecs/audio_format.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 
 namespace webrtc {
@@ -63,4 +64,16 @@ std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
   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