Implement timing frames.

webrtc/common_types.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_COMMON_TYPES_H_
 #define WEBRTC_COMMON_TYPES_H_
 
 #include <stddef.h>
 #include <string.h>
 
 #include <ostream>
 #include <string>
 #include <vector>
 
 #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/base/checks.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/typedefs.h"
 
 #if defined(_MSC_VER)
 // Disable "new behavior: elements of array will be default initialized"
 // warning. Affects OverUseDetectorOptions.
 #pragma warning(disable : 4351)
 #endif
@@ skipping 548 matching lines @@
   uint32_t maxFramerate;
 
   unsigned int qpMax;
   unsigned char numberOfSimulcastStreams;
   SimulcastStream simulcastStream[kMaxSimulcastStreams];
   SpatialLayer spatialLayers[kMaxSpatialLayers];
 
   VideoCodecMode mode;
   bool expect_encode_from_texture;
 
+  // Timing frames configuration. There is delay of delay_ms between two
+  // consequent timing frames, excluding outliers. Frame is always made a
+  // timing frame if it's at least outlier_ratio in percent of "ideal" average
+  // frame given bitrate and framerate, i.e. if it's bigger than
+  // |outlier_ratio / 100.0 * bitrate_bps / fps| in bits.

[holmer, 2017/06/19 07:48:26]

Can you comment on why we have logic like this? Because we want them not to be
sent too often? Do we want them on large frames for a reason?

[ilnik, 2017/06/19 08:41:51]

Done.

+  struct TimingFrameTriggerThresholds {
+    int64_t delay_ms;
+    uint16_t outlier_ratio_percent;
+  } timingFrameTriggerThresholds;

[holmer, 2017/06/19 07:48:26]

timing_frame_thresholds

[ilnik, 2017/06/19 08:41:51]

Done.

+
   bool operator==(const VideoCodec& other) const = delete;
   bool operator!=(const VideoCodec& other) const = delete;
 
   // Accessors for codec specific information.
   // There is a const version of each that returns a reference,
   // and a non-const version that returns a pointer, in order
   // to allow modification of the parameters.
   VideoCodecVP8* VP8();
   const VideoCodecVP8& VP8() const;
   VideoCodecVP9* VP9();
@@ skipping 155 matching lines @@
   // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
   // ts_126114v120700p.pdf
   bool hasVideoRotation;
   VideoRotation videoRotation;
 
   // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
   // a corresponding bool flag.
   bool hasVideoContentType;
   VideoContentType videoContentType;
 
+  bool hasVideoTiming;
+  VideoTiming videoTiming;

[holmer, 2017/06/19 07:48:26]

No camelCase. I see why you used it, but since we are using non_camel_case below
I think you can do that too.

[ilnik, 2017/06/19 08:41:51]

Done.

+
   PlayoutDelay playout_delay = {-1, -1};
 
   // For identification of a stream when ssrc is not signaled. See
   // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
   // TODO(danilchap): Update url from draft to release version.
   StreamId stream_id;
   StreamId repaired_stream_id;
 };
 
 struct RTPHeader {
@@ skipping 112 matching lines @@
 enum class RtcpMode { kOff, kCompound, kReducedSize };
 
 enum NetworkState {
   kNetworkUp,
   kNetworkDown,
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_