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

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  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 1366 matching lines @@
   // Accessor to the VoE channel ID.
   int channel() const {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     return config_.voe_channel_id;
   }
 
   const webrtc::RtpParameters& rtp_parameters() const {
     return rtp_parameters_;
   }
 
+  bool ValidateRtpParameters(const webrtc::RtpParameters& rtp_parameters) {
+    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";
+      return false;
+    }
+    return true;
+  }
+
   bool SetRtpParameters(const webrtc::RtpParameters& parameters) {
-    RTC_CHECK_EQ(1UL, parameters.encodings.size());
+    if (!ValidateRtpParameters(parameters)) {
+      return false;
+    }
     auto send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
                                         parameters.encodings[0].max_bitrate_bps,
                                         send_codec_spec_.codec_inst);
     if (!send_rate) {
       return false;
     }
 
     rtp_parameters_ = parameters;
 
     // parameters.encodings[0].encodings[0].max_bitrate_bps could have changed.
@@ skipping 298 matching lines @@
   for (const AudioCodec& codec : send_codecs_) {
     rtp_params.codecs.push_back(codec.ToCodecParameters());
   }
   return rtp_params;
 }
 
 bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  if (!ValidateRtpParameters(parameters)) {
-    return false;
-  }
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
     LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
                     << "with ssrc " << ssrc << " which doesn't exist.";
     return false;
   }
 
   // TODO(deadbeef): Handle setting parameters with a list of codecs in a
   // different order (which should change the send codec).
   webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
@@ skipping 33 matching lines @@
     rtp_params.codecs.push_back(codec.ToCodecParameters());
   }
   rtp_params.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
   return rtp_params;
 }
 
 bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  if (!ValidateRtpParameters(parameters)) {
-    return false;
-  }
   auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
     LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
                     << "with ssrc " << ssrc << " which doesn't exist.";
     return false;
   }
 
   webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
   if (current_parameters != parameters) {
     LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
                   << "unsupported.";
     return false;
   }
   return true;
 }
 
-bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
-    const webrtc::RtpParameters& rtp_parameters) {
-  if (rtp_parameters.encodings.size() != 1) {
-    LOG(LS_ERROR)
-        << "Attempted to set RtpParameters without exactly one encoding";
-    return false;
-  }
-  return true;
-}
-
 bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "Setting voice channel options: "
                << options.ToString();
 
   // We retain all of the existing options, and apply the given ones
   // on top.  This means there is no way to "clear" options such that
   // they go back to the engine default.
   options_.SetAll(options);
   if (!engine()->ApplyOptions(options_)) {
@@ skipping 860 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE