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, 284 insertions, 0 deletions

diff --git a/webrtc/modules/video_coding/timing.cc b/webrtc/modules/video_coding/timing.cc
new file mode 100644
index 0000000000..08dc307524
--- /dev/null
+++ b/webrtc/modules/video_coding/timing.cc
@@ -0,0 +1,284 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/video_coding/timing.h"
+
+#include <algorithm>
+
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
+
+namespace webrtc {
+
+VCMTiming::VCMTiming(Clock* clock, VCMTiming* master_timing)
+    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+      clock_(clock),
+      master_(false),
+      ts_extrapolator_(),
+      codec_timer_(),
+      render_delay_ms_(kDefaultRenderDelayMs),
+      min_playout_delay_ms_(0),
+      jitter_delay_ms_(0),
+      current_delay_ms_(0),
+      last_decode_ms_(0),
+      prev_frame_timestamp_(0),
+      num_decoded_frames_(0),
+      num_delayed_decoded_frames_(0),
+      first_decoded_frame_ms_(-1),
+      sum_missed_render_deadline_ms_(0) {
+  if (master_timing == NULL) {
+    master_ = true;
+    ts_extrapolator_ = new TimestampExtrapolator(clock_->TimeInMilliseconds());
+  } else {
+    ts_extrapolator_ = master_timing->ts_extrapolator_;
+  }
+}
+
+VCMTiming::~VCMTiming() {
+  UpdateHistograms();
+  if (master_) {
+    delete ts_extrapolator_;
+  }
+  delete crit_sect_;
+}
+
+void VCMTiming::UpdateHistograms() const {
+  CriticalSectionScoped cs(crit_sect_);
+  if (num_decoded_frames_ == 0) {
+    return;
+  }
+  int64_t elapsed_sec =
+      (clock_->TimeInMilliseconds() - first_decoded_frame_ms_) / 1000;
+  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
+    return;
+  }
+  RTC_HISTOGRAM_COUNTS_SPARSE_100(
+      "WebRTC.Video.DecodedFramesPerSecond",
+      static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+      "WebRTC.Video.DelayedFramesToRenderer",
+      num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
+  if (num_delayed_decoded_frames_ > 0) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(
+        "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
+        sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
+  }
+}
+
+void VCMTiming::Reset() {
+  CriticalSectionScoped cs(crit_sect_);
+  ts_extrapolator_->Reset(clock_->TimeInMilliseconds());
+  codec_timer_.Reset();
+  render_delay_ms_ = kDefaultRenderDelayMs;
+  min_playout_delay_ms_ = 0;
+  jitter_delay_ms_ = 0;
+  current_delay_ms_ = 0;
+  prev_frame_timestamp_ = 0;
+}
+
+void VCMTiming::ResetDecodeTime() {
+  CriticalSectionScoped lock(crit_sect_);
+  codec_timer_.Reset();
+}
+
+void VCMTiming::set_render_delay(uint32_t render_delay_ms) {
+  CriticalSectionScoped cs(crit_sect_);
+  render_delay_ms_ = render_delay_ms;
+}
+
+void VCMTiming::set_min_playout_delay(uint32_t min_playout_delay_ms) {
+  CriticalSectionScoped cs(crit_sect_);
+  min_playout_delay_ms_ = min_playout_delay_ms;
+}
+
+void VCMTiming::SetJitterDelay(uint32_t jitter_delay_ms) {
+  CriticalSectionScoped cs(crit_sect_);
+  if (jitter_delay_ms != jitter_delay_ms_) {
+    jitter_delay_ms_ = jitter_delay_ms;
+    // When in initial state, set current delay to minimum delay.
+    if (current_delay_ms_ == 0) {
+      current_delay_ms_ = jitter_delay_ms_;
+    }
+  }
+}
+
+void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
+  CriticalSectionScoped cs(crit_sect_);
+  uint32_t target_delay_ms = TargetDelayInternal();
+
+  if (current_delay_ms_ == 0) {
+    // Not initialized, set current delay to target.
+    current_delay_ms_ = target_delay_ms;
+  } else if (target_delay_ms != current_delay_ms_) {
+    int64_t delay_diff_ms =
+        static_cast<int64_t>(target_delay_ms) - current_delay_ms_;
+    // Never change the delay with more than 100 ms every second. If we're
+    // changing the delay in too large steps we will get noticeable freezes. By
+    // limiting the change we can increase the delay in smaller steps, which
+    // will be experienced as the video is played in slow motion. When lowering
+    // the delay the video will be played at a faster pace.
+    int64_t max_change_ms = 0;
+    if (frame_timestamp < 0x0000ffff && prev_frame_timestamp_ > 0xffff0000) {
+      // wrap
+      max_change_ms = kDelayMaxChangeMsPerS *
+                      (frame_timestamp + (static_cast<int64_t>(1) << 32) -
+                       prev_frame_timestamp_) /
+                      90000;
+    } else {
+      max_change_ms = kDelayMaxChangeMsPerS *
+                      (frame_timestamp - prev_frame_timestamp_) / 90000;
+    }
+    if (max_change_ms <= 0) {
+      // Any changes less than 1 ms are truncated and
+      // will be postponed. Negative change will be due
+      // to reordering and should be ignored.
+      return;
+    }
+    delay_diff_ms = std::max(delay_diff_ms, -max_change_ms);
+    delay_diff_ms = std::min(delay_diff_ms, max_change_ms);
+
+    current_delay_ms_ = current_delay_ms_ + static_cast<int32_t>(delay_diff_ms);
+  }
+  prev_frame_timestamp_ = frame_timestamp;
+}
+
+void VCMTiming::UpdateCurrentDelay(int64_t render_time_ms,
+                                   int64_t actual_decode_time_ms) {
+  CriticalSectionScoped cs(crit_sect_);
+  uint32_t target_delay_ms = TargetDelayInternal();
+  int64_t delayed_ms = actual_decode_time_ms -
+                       (render_time_ms - MaxDecodeTimeMs() - render_delay_ms_);
+  if (delayed_ms < 0) {
+    return;
+  }
+  if (current_delay_ms_ + delayed_ms <= target_delay_ms) {
+    current_delay_ms_ += static_cast<uint32_t>(delayed_ms);
+  } else {
+    current_delay_ms_ = target_delay_ms;
+  }
+}
+
+int32_t VCMTiming::StopDecodeTimer(uint32_t time_stamp,
+                                   int32_t decode_time_ms,
+                                   int64_t now_ms,
+                                   int64_t render_time_ms) {
+  CriticalSectionScoped cs(crit_sect_);
+  codec_timer_.MaxFilter(decode_time_ms, now_ms);
+  assert(decode_time_ms >= 0);
+  last_decode_ms_ = decode_time_ms;
+
+  // Update stats.
+  ++num_decoded_frames_;
+  if (num_decoded_frames_ == 1) {
+    first_decoded_frame_ms_ = now_ms;
+  }
+  int time_until_rendering_ms = render_time_ms - render_delay_ms_ - now_ms;
+  if (time_until_rendering_ms < 0) {
+    sum_missed_render_deadline_ms_ += -time_until_rendering_ms;
+    ++num_delayed_decoded_frames_;
+  }
+  return 0;
+}
+
+void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) {
+  CriticalSectionScoped cs(crit_sect_);
+  ts_extrapolator_->Update(now_ms, time_stamp);
+}
+
+int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp,
+                                int64_t now_ms) const {
+  CriticalSectionScoped cs(crit_sect_);
+  const int64_t render_time_ms = RenderTimeMsInternal(frame_timestamp, now_ms);
+  return render_time_ms;
+}
+
+int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
+                                        int64_t now_ms) const {
+  int64_t estimated_complete_time_ms =
+      ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
+  if (estimated_complete_time_ms == -1) {
+    estimated_complete_time_ms = now_ms;
+  }
+
+  // Make sure that we have at least the playout delay.
+  uint32_t actual_delay = std::max(current_delay_ms_, min_playout_delay_ms_);
+  return estimated_complete_time_ms + actual_delay;
+}
+
+// Must be called from inside a critical section.
+int32_t VCMTiming::MaxDecodeTimeMs(
+    FrameType frame_type /*= kVideoFrameDelta*/) const {
+  const int32_t decode_time_ms = codec_timer_.RequiredDecodeTimeMs(frame_type);
+  assert(decode_time_ms >= 0);
+  return decode_time_ms;
+}
+
+uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms,
+                                   int64_t now_ms) const {
+  CriticalSectionScoped cs(crit_sect_);
+
+  const int64_t max_wait_time_ms =
+      render_time_ms - now_ms - MaxDecodeTimeMs() - render_delay_ms_;
+
+  if (max_wait_time_ms < 0) {
+    return 0;
+  }
+  return static_cast<uint32_t>(max_wait_time_ms);
+}
+
+bool VCMTiming::EnoughTimeToDecode(
+    uint32_t available_processing_time_ms) const {
+  CriticalSectionScoped cs(crit_sect_);
+  int32_t max_decode_time_ms = MaxDecodeTimeMs();
+  if (max_decode_time_ms < 0) {
+    // Haven't decoded any frames yet, try decoding one to get an estimate
+    // of the decode time.
+    return true;
+  } else if (max_decode_time_ms == 0) {
+    // Decode time is less than 1, set to 1 for now since
+    // we don't have any better precision. Count ticks later?
+    max_decode_time_ms = 1;
+  }
+  return static_cast<int32_t>(available_processing_time_ms) -
+             max_decode_time_ms >
+         0;
+}
+
+uint32_t VCMTiming::TargetVideoDelay() const {
+  CriticalSectionScoped cs(crit_sect_);
+  return TargetDelayInternal();
+}
+
+uint32_t VCMTiming::TargetDelayInternal() const {
+  return std::max(min_playout_delay_ms_,
+                  jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_);
+}
+
+void VCMTiming::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 {
+  CriticalSectionScoped cs(crit_sect_);
+  *decode_ms = last_decode_ms_;
+  *max_decode_ms = MaxDecodeTimeMs();
+  *current_delay_ms = current_delay_ms_;
+  *target_delay_ms = TargetDelayInternal();
+  *jitter_buffer_ms = jitter_delay_ms_;
+  *min_playout_delay_ms = min_playout_delay_ms_;
+  *render_delay_ms = render_delay_ms_;
+}
+
+}  // namespace webrtc