WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h

Index: webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index 668fe876f2e326b90b2ea118d7e99918ef68180f..4e95a9b6df00157820669204d4964b1f32a123ac 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -15,6 +15,8 @@
 #include <memory>
 #include <set>
 
+#include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/base/annotations.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/deprecation.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -43,6 +45,12 @@ class RTPPayloadRegistry {
 
   // TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
   // and simplify the code. http://crbug/webrtc/6743.
+
+  // Replace all audio receive payload types with the given map. Returns true
+  // on success.
+  RTC_WARN_UNUSED_RESULT(bool)
+  SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
+
   int32_t RegisterReceivePayload(const CodecInst& audio_codec,
                                  bool* created_new_payload_type);
   int32_t RegisterReceivePayload(const VideoCodec& video_codec);