Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/include/rtp_receiver.h

 /*
  *  Copyright (c) 2012 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_RECEIVER_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
+struct CodecInst;
+class VideoCodec;

[danilchap, 2016/11/23 17:12:50]

maybe put VideoCodec after RTPPayloadRegistry for alphabetical order.

[magjed_webrtc, 2016/11/24 09:39:40]

Done.

 class RTPPayloadRegistry;
 
 class TelephoneEventHandler {
  public:
   virtual ~TelephoneEventHandler() {}
 
   // The following three methods implement the TelephoneEventHandler interface.
   // Forward DTMFs to decoder for playout.
   virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
 
@@ skipping 20 matching lines @@
       RtpFeedback* incoming_messages_callback,
       RTPPayloadRegistry* rtp_payload_registry);
 
   virtual ~RtpReceiver() {}
 
   // Returns a TelephoneEventHandler if available.
   virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
 
   // Registers a receive payload in the payload registry and notifies the media
   // receiver strategy.
-  virtual int32_t RegisterReceivePayload(
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      const int8_t payload_type,
-      const uint32_t frequency,
-      const size_t channels,
-      const uint32_t rate) = 0;
+  virtual int32_t RegisterReceivePayload(const CodecInst& audio_codec) = 0;
 
   // De-registers |payload_type| from the payload registry.
   virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
 
   // Parses the media specific parts of an RTP packet and updates the receiver
   // state. This for instance means that any changes in SSRC and payload type is
   // detected and acted upon.
   virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
                                  const uint8_t* payload,
                                  size_t payload_length,
@@ skipping 12 matching lines @@
 
   // Returns the current remote CSRCs.
   virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
 
   // Returns the current energy of the RTP stream received.
   virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_