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

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
 
 #include <stddef.h>
 #include <list>
 #include <vector>
 
+#include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 #define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
 #define IP_PACKET_SIZE 1500    // we assume ethernet
 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
 
 namespace webrtc {
 namespace rtcp {
 class TransportFeedback;
 }
 
 const int kVideoPayloadTypeFrequency = 90000;
 // TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy
 // and should be fixed.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458
 const int kBogusRtpRateForAudioRtcp = 8000;
 
 // Minimum RTP header size in bytes.
 const uint8_t kRtpHeaderSize = 12;
 
 struct AudioPayload {
     uint32_t    frequency;
     size_t      channels;
     uint32_t    rate;
 };
 
 struct VideoPayload {
-    RtpVideoCodecTypes   videoCodecType;
+  RtpVideoCodecTypes videoCodecType;
+  // The H264 profile only matters if videoCodecType == kRtpVideoH264.
+  H264::Profile h264_profile;
 };
 
 union PayloadUnion {
     AudioPayload Audio;
     VideoPayload Video;
 };
 
 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
 
 enum ProtectionType {
@@ skipping 335 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_