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, 303 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/test/utility.cc b/webrtc/modules/audio_coding/test/utility.cc
new file mode 100644
index 0000000000..89368bce51
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/utility.cc
@@ -0,0 +1,303 @@
+/*
+ *  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 "utility.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+
+#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
+
+namespace webrtc {
+
+ACMTestTimer::ACMTestTimer()
+    : _msec(0),
+      _sec(0),
+      _min(0),
+      _hour(0) {
+  return;
+}
+
+ACMTestTimer::~ACMTestTimer() {
+  return;
+}
+
+void ACMTestTimer::Reset() {
+  _msec = 0;
+  _sec = 0;
+  _min = 0;
+  _hour = 0;
+  return;
+}
+void ACMTestTimer::Tick10ms() {
+  _msec += 10;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::Tick1ms() {
+  _msec++;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::Tick100ms() {
+  _msec += 100;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::Tick1sec() {
+  _sec++;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::CurrentTimeHMS(char* currTime) {
+  sprintf(currTime, "%4lu:%02u:%06.3f", _hour, _min,
+          (double) _sec + (double) _msec / 1000.);
+  return;
+}
+
+void ACMTestTimer::CurrentTime(unsigned long& h, unsigned char& m,
+                               unsigned char& s, unsigned short& ms) {
+  h = _hour;
+  m = _min;
+  s = _sec;
+  ms = _msec;
+  return;
+}
+
+void ACMTestTimer::Adjust() {
+  unsigned int n;
+  if (_msec >= 1000) {
+    n = _msec / 1000;
+    _msec -= (1000 * n);
+    _sec += n;
+  }
+  if (_sec >= 60) {
+    n = _sec / 60;
+    _sec -= (n * 60);
+    _min += n;
+  }
+  if (_min >= 60) {
+    n = _min / 60;
+    _min -= (n * 60);
+    _hour += n;
+  }
+}
+
+int16_t ChooseCodec(CodecInst& codecInst) {
+
+  PrintCodecs();
+  //AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
+  uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+  int8_t codecID;
+  bool outOfRange = false;
+  char myStr[15] = "";
+  do {
+    printf("\nChoose a codec [0]: ");
+    EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+    codecID = atoi(myStr);
+    if ((codecID < 0) || (codecID >= noCodec)) {
+      printf("\nOut of range.\n");
+      outOfRange = true;
+    }
+  } while (outOfRange);
+
+  CHECK_ERROR(AudioCodingModule::Codec((uint8_t )codecID, &codecInst));
+  return 0;
+}
+
+void PrintCodecs() {
+  uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+
+  CodecInst codecInst;
+  printf("No  Name                [Hz]    [bps]\n");
+  for (uint8_t codecCntr = 0; codecCntr < noCodec; codecCntr++) {
+    AudioCodingModule::Codec(codecCntr, &codecInst);
+    printf("%2d- %-18s %5d   %6d\n", codecCntr, codecInst.plname,
+           codecInst.plfreq, codecInst.rate);
+  }
+
+}
+
+CircularBuffer::CircularBuffer(uint32_t len)
+    : _buff(NULL),
+      _idx(0),
+      _buffIsFull(false),
+      _calcAvg(false),
+      _calcVar(false),
+      _sum(0),
+      _sumSqr(0) {
+  _buff = new double[len];
+  if (_buff == NULL) {
+    _buffLen = 0;
+  } else {
+    for (uint32_t n = 0; n < len; n++) {
+      _buff[n] = 0;
+    }
+    _buffLen = len;
+  }
+}
+
+CircularBuffer::~CircularBuffer() {
+  if (_buff != NULL) {
+    delete[] _buff;
+    _buff = NULL;
+  }
+}
+
+void CircularBuffer::Update(const double newVal) {
+  assert(_buffLen > 0);
+
+  // store the value that is going to be overwritten
+  double oldVal = _buff[_idx];
+  // record the new value
+  _buff[_idx] = newVal;
+  // increment the index, to point to where we would
+  // write next
+  _idx++;
+  // it is a circular buffer, if we are at the end
+  // we have to cycle to the beginning
+  if (_idx >= _buffLen) {
+    // flag that the buffer is filled up.
+    _buffIsFull = true;
+    _idx = 0;
+  }
+
+  // Update
+
+  if (_calcAvg) {
+    // for the average we have to update
+    // the sum
+    _sum += (newVal - oldVal);
+  }
+
+  if (_calcVar) {
+    // to calculate variance we have to update
+    // the sum of squares
+    _sumSqr += (double) (newVal - oldVal) * (double) (newVal + oldVal);
+  }
+}
+
+void CircularBuffer::SetArithMean(bool enable) {
+  assert(_buffLen > 0);
+
+  if (enable && !_calcAvg) {
+    uint32_t lim;
+    if (_buffIsFull) {
+      lim = _buffLen;
+    } else {
+      lim = _idx;
+    }
+    _sum = 0;
+    for (uint32_t n = 0; n < lim; n++) {
+      _sum += _buff[n];
+    }
+  }
+  _calcAvg = enable;
+}
+
+void CircularBuffer::SetVariance(bool enable) {
+  assert(_buffLen > 0);
+
+  if (enable && !_calcVar) {
+    uint32_t lim;
+    if (_buffIsFull) {
+      lim = _buffLen;
+    } else {
+      lim = _idx;
+    }
+    _sumSqr = 0;
+    for (uint32_t n = 0; n < lim; n++) {
+      _sumSqr += _buff[n] * _buff[n];
+    }
+  }
+  _calcAvg = enable;
+}
+
+int16_t CircularBuffer::ArithMean(double& mean) {
+  assert(_buffLen > 0);
+
+  if (_buffIsFull) {
+
+    mean = _sum / (double) _buffLen;
+    return 0;
+  } else {
+    if (_idx > 0) {
+      mean = _sum / (double) _idx;
+      return 0;
+    } else {
+      return -1;
+    }
+
+  }
+}
+
+int16_t CircularBuffer::Variance(double& var) {
+  assert(_buffLen > 0);
+
+  if (_buffIsFull) {
+    var = _sumSqr / (double) _buffLen;
+    return 0;
+  } else {
+    if (_idx > 0) {
+      var = _sumSqr / (double) _idx;
+      return 0;
+    } else {
+      return -1;
+    }
+  }
+}
+
+bool FixedPayloadTypeCodec(const char* payloadName) {
+  char fixPayloadTypeCodecs[NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE][32] = { "PCMU",
+      "PCMA", "GSM", "G723", "DVI4", "LPC", "PCMA", "G722", "QCELP", "CN",
+      "MPA", "G728", "G729" };
+
+  for (int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++) {
+    if (!STR_CASE_CMP(payloadName, fixPayloadTypeCodecs[n])) {
+      return true;
+    }
+  }
+  return false;
+}
+
+void VADCallback::Reset() {
+  memset(_numFrameTypes, 0, sizeof(_numFrameTypes));
+}
+
+VADCallback::VADCallback() {
+  memset(_numFrameTypes, 0, sizeof(_numFrameTypes));
+}
+
+void VADCallback::PrintFrameTypes() {
+  printf("kEmptyFrame......... %d\n", _numFrameTypes[kEmptyFrame]);
+  printf("kAudioFrameSpeech... %d\n", _numFrameTypes[kAudioFrameSpeech]);
+  printf("kAudioFrameCN....... %d\n", _numFrameTypes[kAudioFrameCN]);
+  printf("kVideoFrameKey...... %d\n", _numFrameTypes[kVideoFrameKey]);
+  printf("kVideoFrameDelta.... %d\n", _numFrameTypes[kVideoFrameDelta]);
+}
+
+int32_t VADCallback::InFrameType(FrameType frame_type) {
+  _numFrameTypes[frame_type]++;
+  return 0;
+}
+
+}  // namespace webrtc