Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 126 insertions, 0 deletions

diff --git a/webrtc/modules/video_coding/timing.h b/webrtc/modules/video_coding/timing.h
new file mode 100644
index 0000000000..a4d0cf4543
--- /dev/null
+++ b/webrtc/modules/video_coding/timing.h
@@ -0,0 +1,126 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/video_coding/codec_timer.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Clock;
+class TimestampExtrapolator;
+
+class VCMTiming {
+ public:
+  // The primary timing component should be passed
+  // if this is the dual timing component.
+  explicit VCMTiming(Clock* clock, VCMTiming* master_timing = NULL);
+  ~VCMTiming();
+
+  // Resets the timing to the initial state.
+  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);
+
+  // 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.
+  void UpdateCurrentDelay(int64_t render_time_ms,
+                          int64_t actual_decode_time_ms);
+
+  // Stops the decoder timer, should be called when the decoder returns a frame
+  // or when the decoded frame callback is called.
+  int32_t StopDecodeTimer(uint32_t time_stamp,
+                          int32_t decode_time_ms,
+                          int64_t now_ms,
+                          int64_t render_time_ms);
+
+  // Used to report that a frame is passed to decoding. Updates the timestamp
+  // filter which is used to map between timestamps and receiver system time.
+  void IncomingTimestamp(uint32_t time_stamp, int64_t last_packet_time_ms);
+  // Returns the receiver system time when the frame with timestamp
+  // frame_timestamp should be rendered, assuming that the system time currently
+  // is now_ms.
+  int64_t RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms) const;
+
+  // Returns the maximum time in ms that we can wait for a frame to become
+  // complete before we must pass it to the decoder.
+  uint32_t MaxWaitingTime(int64_t render_time_ms, int64_t now_ms) const;
+
+  // Returns the current target delay which is required delay + decode time +
+  // render delay.
+  uint32_t TargetVideoDelay() const;
+
+  // Calculates whether or not there is enough time to decode a frame given a
+  // certain amount of processing time.
+  bool EnoughTimeToDecode(uint32_t available_processing_time_ms) const;
+
+  // Return current timing information.
+  void GetTimings(int* decode_ms,
+                  int* max_decode_ms,
+                  int* current_delay_ms,
+                  int* target_delay_ms,
+                  int* jitter_buffer_ms,
+                  int* min_playout_delay_ms,
+                  int* render_delay_ms) const;
+
+  enum { kDefaultRenderDelayMs = 10 };
+  enum { kDelayMaxChangeMsPerS = 100 };
+
+ protected:
+  int32_t MaxDecodeTimeMs(FrameType frame_type = kVideoFrameDelta) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  int64_t RenderTimeMsInternal(uint32_t frame_timestamp, int64_t now_ms) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  uint32_t TargetDelayInternal() const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ private:
+  void UpdateHistograms() const;
+
+  CriticalSectionWrapper* crit_sect_;
+  Clock* const clock_;
+  bool master_ GUARDED_BY(crit_sect_);
+  TimestampExtrapolator* ts_extrapolator_ GUARDED_BY(crit_sect_);
+  VCMCodecTimer codec_timer_ GUARDED_BY(crit_sect_);
+  uint32_t render_delay_ms_ GUARDED_BY(crit_sect_);
+  uint32_t min_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_