Fix RtpReceiver.GetParameters when SSRCs aren't signaled.

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 955 matching lines @@
   VoiceMediaChannel() {}
   explicit 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 GetRtpSendParameters(uint32_t ssrc) const = 0;
   virtual bool SetRtpSendParameters(
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) = 0;
+  // Get the receive parameters for the incoming stream identified by |ssrc|.
+  // If |ssrc| is 0, retrieve the receive parameters for the default receive
+  // stream, which is used when SSRCs are not signaled. Note that calling with
+  // an |ssrc| of 0 will return encoding parameters with an unset |ssrc|
+  // member.
   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 void SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
   virtual void SetSend(bool send) = 0;
   // Configure stream for sending.
@@ skipping 60 matching lines @@
   explicit 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 GetRtpSendParameters(uint32_t ssrc) const = 0;
   virtual bool SetRtpSendParameters(
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) = 0;
+  // Get the receive parameters for the incoming stream identified by |ssrc|.
+  // If |ssrc| is 0, retrieve the receive parameters for the default receive
+  // stream, which is used when SSRCs are not signaled. Note that calling with
+  // an |ssrc| of 0 will return encoding parameters with an unset |ssrc|
+  // member.
   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 and register a source.
   // The |ssrc| must correspond to a registered send stream.
   virtual bool SetVideoSend(
       uint32_t ssrc,
       bool enable,
       const VideoOptions* options,
       rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0;
   // Sets the sink object to be used for the specified stream.
-  // If SSRC is 0, the renderer is used for the 'default' stream.
+  // If SSRC is 0, the sink is used for the 'default' stream.
   virtual bool SetSink(uint32_t ssrc,
                        rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0;
   // Gets quality stats for the channel.
   virtual bool GetStats(VideoMediaInfo* info) = 0;
 };
 
 enum DataMessageType {
   // Chrome-Internal use only.  See SctpDataMediaChannel for the actual PPID
   // values.
   DMT_NONE = 0,
@@ skipping 111 matching lines @@
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_