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, 0 insertions, 146 deletions

diff --git a/webrtc/modules/video_processing/main/source/video_decimator.cc b/webrtc/modules/video_processing/main/source/video_decimator.cc
deleted file mode 100644
index 34c29c1677..0000000000
--- a/webrtc/modules/video_processing/main/source/video_decimator.cc
+++ /dev/null
@@ -1,146 +0,0 @@
-/*
- *  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/base/checks.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/video_decimator.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#define VD_MIN(a, b) ((a) < (b)) ? (a) : (b)
-
-namespace webrtc {
-
-VPMVideoDecimator::VPMVideoDecimator() {
-  Reset();
-}
-
-VPMVideoDecimator::~VPMVideoDecimator() {}
-
-void VPMVideoDecimator::Reset()  {
-  overshoot_modifier_ = 0;
-  drop_count_ = 0;
-  keep_count_ = 0;
-  target_frame_rate_ = 30;
-  incoming_frame_rate_ = 0.0f;
-  memset(incoming_frame_times_, 0, sizeof(incoming_frame_times_));
-  enable_temporal_decimation_ = true;
-}
-
-void VPMVideoDecimator::EnableTemporalDecimation(bool enable) {
-  enable_temporal_decimation_ = enable;
-}
-
-void VPMVideoDecimator::SetTargetFramerate(int frame_rate) {
-  RTC_DCHECK(frame_rate);
-  target_frame_rate_ = frame_rate;
-}
-
-bool VPMVideoDecimator::DropFrame() {
-  if (!enable_temporal_decimation_) return false;
-
-  if (incoming_frame_rate_ <= 0) return false;
-
-  const uint32_t incomingframe_rate =
-      static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
-
-  if (target_frame_rate_ == 0) return true;
-
-  bool drop = false;
-  if (incomingframe_rate > target_frame_rate_) {
-    int32_t overshoot =
-        overshoot_modifier_ + (incomingframe_rate - target_frame_rate_);
-    if (overshoot < 0) {
-      overshoot = 0;
-      overshoot_modifier_ = 0;
-    }
-
-    if (overshoot && 2 * overshoot < (int32_t) incomingframe_rate) {
-      if (drop_count_) {  // Just got here so drop to be sure.
-          drop_count_ = 0;
-          return true;
-      }
-      const uint32_t dropVar = incomingframe_rate / overshoot;
-
-      if (keep_count_ >= dropVar) {
-          drop = true;
-          overshoot_modifier_ = -((int32_t) incomingframe_rate % overshoot) / 3;
-          keep_count_ = 1;
-      } else {
-          keep_count_++;
-      }
-    } else {
-      keep_count_ = 0;
-      const uint32_t dropVar = overshoot / target_frame_rate_;
-      if (drop_count_ < dropVar) {
-          drop = true;
-          drop_count_++;
-      } else {
-          overshoot_modifier_ = overshoot % target_frame_rate_;
-          drop = false;
-          drop_count_ = 0;
-      }
-    }
-  }
-  return drop;
-}
-
-
-uint32_t VPMVideoDecimator::Decimatedframe_rate() {
-ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
-  if (!enable_temporal_decimation_) {
-    return static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
-  }
-  return VD_MIN(target_frame_rate_,
-      static_cast<uint32_t>(incoming_frame_rate_ + 0.5f));
-}
-
-uint32_t VPMVideoDecimator::Inputframe_rate() {
-  ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
-  return static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
-}
-
-void VPMVideoDecimator::UpdateIncomingframe_rate() {
-  int64_t now = TickTime::MillisecondTimestamp();
-  if (incoming_frame_times_[0] == 0) {
-    // First no shift.
-  } else {
-    // Shift.
-    for (int i = kFrameCountHistory_size - 2; i >= 0; i--) {
-        incoming_frame_times_[i+1] = incoming_frame_times_[i];
-    }
-  }
-  incoming_frame_times_[0] = now;
-  ProcessIncomingframe_rate(now);
-}
-
-void VPMVideoDecimator::ProcessIncomingframe_rate(int64_t now) {
-  int32_t num = 0;
-  int32_t nrOfFrames = 0;
-  for (num = 1; num < (kFrameCountHistory_size - 1); num++) {
-    // Don't use data older than 2sec.
-    if (incoming_frame_times_[num] <= 0 ||
-        now - incoming_frame_times_[num] > kFrameHistoryWindowMs) {
-      break;
-    } else {
-      nrOfFrames++;
-    }
-  }
-  if (num > 1) {
-    int64_t diff = now - incoming_frame_times_[num-1];
-    incoming_frame_rate_ = 1.0;
-    if (diff > 0) {
-      incoming_frame_rate_ = nrOfFrames * 1000.0f / static_cast<float>(diff);
-    }
-  } else {
-    incoming_frame_rate_ = static_cast<float>(nrOfFrames);
-  }
-}
-
-}  // namespace webrtc