Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.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.
  */
 
@@ skipping 19 matching lines @@
     return false;
   const AudioPayload& audio_payload = payload.typeSpecific.Audio;
   const uint32_t rate = std::max(0, audio_codec.rate);
   return audio_payload.frequency == static_cast<uint32_t>(audio_codec.plfreq) &&
          audio_payload.channels == audio_codec.channels &&
          (audio_payload.rate == rate || audio_payload.rate == 0 || rate == 0);
 }
 
 bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                          const VideoCodec& video_codec) {
-  return !payload.audio && _stricmp(payload.name, video_codec.plName) == 0;
+  if (payload.audio || _stricmp(payload.name, video_codec.plName) != 0)
+    return false;
+  // For H264, profiles must match as well.
+  if (video_codec.codecType == kVideoCodecH264) {
+    return video_codec.H264().profile ==
+           payload.typeSpecific.Video.h264_profile;
+  }
+  return true;
 }
 
 RtpUtility::Payload CreatePayloadType(const CodecInst& audio_codec) {
   RtpUtility::Payload payload;
   payload.name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
   strncpy(payload.name, audio_codec.plname, RTP_PAYLOAD_NAME_SIZE - 1);
   RTC_DCHECK_GE(audio_codec.plfreq, 1000);
   payload.typeSpecific.Audio.frequency = audio_codec.plfreq;
   payload.typeSpecific.Audio.channels = audio_codec.channels;
   payload.typeSpecific.Audio.rate = std::max(0, audio_codec.rate);
@@ skipping 16 matching lines @@
       return kRtpVideoGeneric;
   }
 }
 
 RtpUtility::Payload CreatePayloadType(const VideoCodec& video_codec) {
   RtpUtility::Payload payload;
   payload.name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
   strncpy(payload.name, video_codec.plName, RTP_PAYLOAD_NAME_SIZE - 1);
   payload.typeSpecific.Video.videoCodecType =
       ConvertToRtpVideoCodecType(video_codec.codecType);
+  if (video_codec.codecType == kVideoCodecH264)
+    payload.typeSpecific.Video.h264_profile = video_codec.H264().profile;
   payload.audio = false;
   return payload;
 }
 
 bool IsPayloadTypeValid(int8_t payload_type) {
   assert(payload_type >= 0);
 
   // Sanity check.
   switch (payload_type) {
     // Reserved payload types to avoid RTCP conflicts when marker bit is set.
@@ skipping 292 matching lines @@
     const char* payload_name) const {
   rtc::CritScope cs(&crit_sect_);
   for (const auto& it : payload_type_map_) {
     if (_stricmp(it.second.name, payload_name) == 0)
       return it.first;
   }
   return -1;
 }
 
 }  // namespace webrtc