Implement timing frames.

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h

 /*
  *  Copyright (c) 2016 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_SOURCE_RTP_HEADER_EXTENSIONS_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
 
 #include <stdint.h>
 #include <string>
 
 #include "webrtc/api/video/video_content_type.h"
 #include "webrtc/api/video/video_rotation.h"
+#include "webrtc/api/video/video_timing.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class AbsoluteSendTime {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
   static constexpr uint8_t kValueSizeBytes = 3;
   static constexpr const char* kUri =
@@ skipping 91 matching lines @@
       "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
 
   static bool Parse(rtc::ArrayView<const uint8_t> data,
                     VideoContentType* content_type);
   static size_t ValueSize(VideoContentType) {
     return kValueSizeBytes;
   }
   static bool Write(uint8_t* data, VideoContentType content_type);
 };
 
+class VideoTimingExtension {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionVideoTiming;
+  static constexpr uint8_t kValueSizeBytes = 12;
+  static constexpr const char* kUri =
+      "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data, VideoTiming* timing);
+  static size_t ValueSize(VideoTiming) { return kValueSizeBytes; }

[åsapersson, 2017/06/12 14:33:41]

const VideoTiming&

[ilnik, 2017/06/13 08:43:13]

Done.

+  static size_t ValueSize(uint16_t delta_ms, uint8_t idx) {

[åsapersson, 2017/06/12 14:33:41]

maybe move to line 143 above Write

[åsapersson, 2017/06/12 14:33:41]

time_delta_ms

[ilnik, 2017/06/13 08:43:13]

Done.

+    return kValueSizeBytes;
+  }
+  static bool Write(uint8_t* data, const VideoTiming& timing);
+  // Writes only single time delta to position idx.
+  static bool Write(uint8_t* data, uint16_t time_delta_ms, uint8_t idx);
+};
+
 class RtpStreamId {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
   static constexpr const char* kUri =
       "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
 
   static bool Parse(rtc::ArrayView<const uint8_t> data, StreamId* rsid);
   static size_t ValueSize(const StreamId& rsid) { return rsid.size(); }
   static bool Write(uint8_t* data, const StreamId& rsid);
 
@@ skipping 12 matching lines @@
   static size_t ValueSize(const StreamId& rsid);
   static bool Write(uint8_t* data, const StreamId& rsid);
 
   static bool Parse(rtc::ArrayView<const uint8_t> data, std::string* rsid);
   static size_t ValueSize(const std::string& rsid);
   static bool Write(uint8_t* data, const std::string& rsid);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_