[EXPERIMENTAL] Generic stereo codec with index header sending merged frames

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 262 matching lines @@
 namespace {
 
 bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
   rtc::Optional<VideoCodecType> codecType =
       PayloadNameToCodecType(payload_name);
   if (codecType &&
       (*codecType == kVideoCodecVP8 || *codecType == kVideoCodecVP9)) {
     return true;
   }
   RTC_DCHECK((codecType && *codecType == kVideoCodecH264) ||
+             (codecType && *codecType == kVideoCodecStereo) ||
              payload_name == "FAKE")
       << "unknown payload_name " << payload_name;
   return false;
 }
 
 int CalculateMaxPadBitrateBps(std::vector<VideoStream> streams,
                               int min_transmit_bitrate_bps,
                               bool pad_to_min_bitrate) {
   int pad_up_to_bitrate_bps = 0;
   // Calculate max padding bitrate for a multi layer codec.
@@ skipping 1077 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc