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, 100 deletions

diff --git a/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc b/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc
deleted file mode 100644
index 83d09ef486..0000000000
--- a/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- *  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.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, DISABLED_ON_IOS(Deflickering))
-{
-    enum { NumRuns = 30 };
-    uint32_t frameNum = 0;
-    const uint32_t frame_rate = 15;
-
-    int64_t min_runtime = 0;
-    int64_t avg_runtime = 0;
-
-    // Close automatically opened Foreman.
-    fclose(source_file_);
-    const std::string input_file =
-        webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
-    source_file_  = fopen(input_file.c_str(), "rb");
-    ASSERT_TRUE(source_file_ != NULL) <<
-        "Cannot read input file: " << input_file << "\n";
-
-    const std::string output_file =
-        webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
-    FILE* deflickerFile = fopen(output_file.c_str(), "wb");
-    ASSERT_TRUE(deflickerFile != NULL) <<
-        "Could not open output file: " << output_file << "\n";
-
-    printf("\nRun time [us / frame]:\n");
-    rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-    for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++)
-    {
-        TickTime t0;
-        TickTime t1;
-        TickInterval acc_ticks;
-        uint32_t timeStamp = 1;
-
-        frameNum = 0;
-        while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-               frame_length_)
-        {
-            frameNum++;
-            EXPECT_EQ(
-                0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-            video_frame_.set_timestamp(timeStamp);
-
-            t0 = TickTime::Now();
-            VideoProcessingModule::FrameStats stats;
-            ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-            ASSERT_EQ(0, vpm_->Deflickering(&video_frame_, &stats));
-            t1 = TickTime::Now();
-            acc_ticks += (t1 - t0);
-
-            if (run_idx == 0)
-            {
-              if (PrintVideoFrame(video_frame_, deflickerFile) < 0) {
-                return;
-              }
-            }
-            timeStamp += (90000 / frame_rate);
-        }
-        ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-        printf("%u\n", static_cast<int>(acc_ticks.Microseconds() / frameNum));
-        if (acc_ticks.Microseconds() < min_runtime || run_idx == 0)
-        {
-            min_runtime = acc_ticks.Microseconds();
-        }
-        avg_runtime += acc_ticks.Microseconds();
-
-        rewind(source_file_);
-    }
-    ASSERT_EQ(0, fclose(deflickerFile));
-    // TODO(kjellander): Add verification of deflicker output file.
-
-    printf("\nAverage run time = %d us / frame\n",
-        static_cast<int>(avg_runtime / frameNum / NumRuns));
-    printf("Min run time = %d us / frame\n\n",
-        static_cast<int>(min_runtime / frameNum));
-}
-
-}  // namespace webrtc