Add video send SSRC to RtpParameters, and don't allow changing SSRC.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 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 1542 matching lines @@
       parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
       rtp_parameters_(CreateRtpParametersWithOneEncoding()),
       allocated_encoder_(nullptr, cricket::VideoCodec(), false),
       sending_(false),
       last_frame_timestamp_us_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
   parameters_.conference_mode = send_params.conference_mode;
 
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
 
+  // ValidateStreamParams should prevent this from happening.
+  RTC_CHECK(!parameters_.config.rtp.ssrcs.empty());

[pthatcher1, 2016/12/14 19:01:38]

This might happen down the road when we support MID only and no signaled SSRCs. 
It might be easier to just change the code now to be:

if (!parameters_.config.rtp.ssrcs.empty()) {
  rtp_parameters_.encodings[0].ssrc =
rtc::Optional<uint32_t>(parameters_.config.rtp.ssrcs[0]);
}

[Taylor Brandstetter, 2016/12/15 00:04:26]

When that change occurs, we can update this RTC_CHECK and add a unit test for
the newly-supported behavior, but for now I think the check serves its purpose.

+  rtp_parameters_.encodings[0].ssrc =
+      rtc::Optional<uint32_t>(parameters_.config.rtp.ssrcs[0]);
+
   // RTX.
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
                  &parameters_.config.rtp.rtx.ssrcs);
 
   // FlexFEC.
   // TODO(brandtr): This code needs to be generalized when we add support for
   // multistream protection.
   if (IsFlexfecFieldTrialEnabled()) {
     uint32_t flexfec_ssrc;
     bool flexfec_enabled = false;
@@ skipping 282 matching lines @@
 }
 
 webrtc::RtpParameters
 WebRtcVideoChannel2::WebRtcVideoSendStream::GetRtpParameters() const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   return rtp_parameters_;
 }
 
 bool WebRtcVideoChannel2::WebRtcVideoSendStream::ValidateRtpParameters(
     const webrtc::RtpParameters& rtp_parameters) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   if (rtp_parameters.encodings.size() != 1) {
     LOG(LS_ERROR)
         << "Attempted to set RtpParameters without exactly one encoding";
     return false;
   }
+  if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
+    LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";

[pthatcher1, 2016/12/14 19:01:38]

You might as well use a for-loop here that will be robust for when we add more
than one encoding.

+    return false;
+  }
   return true;
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateSendState() {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   // TODO(deadbeef): Need to handle more than one encoding in the future.
   RTC_DCHECK(rtp_parameters_.encodings.size() == 1u);
   if (sending_ && rtp_parameters_.encodings[0].active) {
     RTC_DCHECK(stream_ != nullptr);
     stream_->Start();
@@ skipping 693 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             ulpfec_config.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket