Adding getParameters/setParameters APIs to RtpReceiver.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 888 matching lines @@
     ERROR_PLAY_SRTP_ERROR,                // Generic SRTP failure.
     ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
     ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
   };
 
   VoiceMediaChannel() {}
   VoiceMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
   virtual ~VoiceMediaChannel() {}
   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
-  virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0;
-  virtual bool SetRtpParameters(uint32_t ssrc,
-                                const webrtc::RtpParameters& parameters) = 0;
+  virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
+  virtual webrtc::RtpParameters GetRtpReceiveParameters(
+      uint32_t ssrc) const = 0;
+  virtual bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
   // Starts or stops playout of received audio.
   virtual bool SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
   virtual void SetSend(bool send) = 0;
   // Configure stream for sending.
   virtual bool SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const AudioOptions* options,
                             AudioSource* source) = 0;
   // Gets current energy levels for all incoming streams.
@@ skipping 56 matching lines @@
     ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
     ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
   };
 
   VideoMediaChannel() {}
   VideoMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
   virtual ~VideoMediaChannel() {}
 
   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
   virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
-  virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0;
-  virtual bool SetRtpParameters(uint32_t ssrc,
-                                const webrtc::RtpParameters& parameters) = 0;
+  virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
+  virtual webrtc::RtpParameters GetRtpReceiveParameters(
+      uint32_t ssrc) const = 0;
+  virtual bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
   // Gets the currently set codecs/payload types to be used for outgoing media.
   virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
   // Starts or stops transmission (and potentially capture) of local video.
   virtual bool SetSend(bool send) = 0;
   // Configure stream for sending.
   virtual bool SetVideoSend(uint32_t ssrc,
                             bool enable,
                             const VideoOptions* options) = 0;
   // Sets the sink object to be used for the specified stream.
   // If SSRC is 0, the renderer is used for the 'default' stream.
@@ skipping 125 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_