Add sender controlled playout delay limits

webrtc/modules/video_coding/timing.h

 /*
  *  Copyright (c) 2011 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 23 matching lines @@
   void Reset();
   void ResetDecodeTime();
 
   // Set the amount of time needed to render an image. Defaults to 10 ms.
   void set_render_delay(uint32_t render_delay_ms);
 
   // Set the minimum time the video must be delayed on the receiver to
   // get the desired jitter buffer level.
   void SetJitterDelay(uint32_t required_delay_ms);
 
-  // Set the minimum playout delay required to sync video with audio.
-  void set_min_playout_delay(uint32_t min_playout_delay);
+  // Set the minimum playout delay from capture to render in ms.
+  void set_min_playout_delay(uint32_t min_playout_delay_ms);

[stefan-webrtc, 2016/05/28 05:01:45]

I'd prefer these to all be ints instead since it doesn't matter whether they're
uint32_t or int.

[Irfan, 2016/06/01 08:38:33]

I cleaned up all the delay variables here alongside so that we can get rid of a
bunch of casts that were being done.

+
+  // Returns the minimum playout delay from capture to render in ms.
+  uint32_t min_playout_delay();
+
+  // Set the maximum playout delay from capture to render in ms.
+  void set_max_playout_delay(uint32_t max_playout_delay_ms);
+
+  // Returns the maximum playout delay from capture to render in ms.
+  uint32_t max_playout_delay();
 
   // Increases or decreases the current delay to get closer to the target delay.
   // Calculates how long it has been since the previous call to this function,
   // and increases/decreases the delay in proportion to the time difference.
   void UpdateCurrentDelay(uint32_t frame_timestamp);
 
   // Increases or decreases the current delay to get closer to the target delay.
   // Given the actual decode time in ms and the render time in ms for a frame,
   // this function calculates how late the frame is and increases the delay
   // accordingly.
@@ skipping 48 matching lines @@
 
  private:
   void UpdateHistograms() const;
 
   CriticalSectionWrapper* crit_sect_;
   Clock* const clock_;
   bool master_ GUARDED_BY(crit_sect_);
   TimestampExtrapolator* ts_extrapolator_ GUARDED_BY(crit_sect_);
   std::unique_ptr<VCMCodecTimer> codec_timer_ GUARDED_BY(crit_sect_);
   uint32_t render_delay_ms_ GUARDED_BY(crit_sect_);
+  // Best-effort playout delay range for frames from capture to render.
+  // The receiver tries to keep the delay between |min_playout_delay_ms_|
+  // and |max_playout_delay_ms_| taking the network jitter into account.
+  // A special case is where min_playout_delay_ms_ = max_playout_delay_ms_ = 0,
+  // in which case the receiver tries to play the frames as they arrive.
   uint32_t min_playout_delay_ms_ GUARDED_BY(crit_sect_);
+  uint32_t max_playout_delay_ms_ GUARDED_BY(crit_sect_);
   uint32_t jitter_delay_ms_ GUARDED_BY(crit_sect_);
   uint32_t current_delay_ms_ GUARDED_BY(crit_sect_);
   int last_decode_ms_ GUARDED_BY(crit_sect_);
   uint32_t prev_frame_timestamp_ GUARDED_BY(crit_sect_);
 
   // Statistics.
   size_t num_decoded_frames_ GUARDED_BY(crit_sect_);
   size_t num_delayed_decoded_frames_ GUARDED_BY(crit_sect_);
   int64_t first_decoded_frame_ms_ GUARDED_BY(crit_sect_);
   uint64_t sum_missed_render_deadline_ms_ GUARDED_BY(crit_sect_);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_TIMING_H_