Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/codecs/audio_format_conversion.cc

Index: webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
new file mode 100644
index 0000000000000000000000000000000000000000..ef9aa4479b9be95466f25ca24b1a5569c87778df
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
@@ -0,0 +1,30 @@
+/*
+ *  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_conversion.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+
+namespace webrtc {
+
+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, static_cast<int>(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, rtc::checked_cast<int>(ci.channels)};
+  }
+}
+
+}  // namespace webrtc