WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
 
 #include <map>
 #include <memory>
 #include <set>
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/deprecation.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 
 struct CodecInst;
 class VideoCodec;
 
 // TODO(magjed): Remove once external code is updated.
 class RTPPayloadStrategy {
  public:
   static RTPPayloadStrategy* CreateStrategy(bool handling_audio) {
     return nullptr;
   }
 };
 
 class RTPPayloadRegistry {
  public:
   RTPPayloadRegistry();
   ~RTPPayloadRegistry();
   // TODO(magjed): Remove once external code is updated.
   explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
       : 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.
+  void 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);
 
   int32_t DeRegisterReceivePayload(int8_t payload_type);
 
   int32_t ReceivePayloadType(const CodecInst& audio_codec,
                              int8_t* payload_type) const;
   int32_t ReceivePayloadType(const VideoCodec& video_codec,
                              int8_t* payload_type) const;
@@ skipping 87 matching lines @@
   // video, DCHECK that no instance is used for both audio and video.
 #if RTC_DCHECK_IS_ON
   bool used_for_audio_ GUARDED_BY(crit_sect_) = false;
   bool used_for_video_ GUARDED_BY(crit_sect_) = false;
 #endif
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_