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, 196 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/test/SpatialAudio.cc b/webrtc/modules/audio_coding/test/SpatialAudio.cc
new file mode 100644
index 0000000000..c9f8080826
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/SpatialAudio.cc
@@ -0,0 +1,196 @@
+/*
+ *  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 <string.h>
+
+#include <math.h>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/test/SpatialAudio.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+#define NUM_PANN_COEFFS 10
+
+SpatialAudio::SpatialAudio(int testMode)
+    : _acmLeft(AudioCodingModule::Create(1)),
+      _acmRight(AudioCodingModule::Create(2)),
+      _acmReceiver(AudioCodingModule::Create(3)),
+      _testMode(testMode) {
+}
+
+SpatialAudio::~SpatialAudio() {
+  delete _channel;
+  _inFile.Close();
+  _outFile.Close();
+}
+
+int16_t SpatialAudio::Setup() {
+  _channel = new Channel;
+
+  // Register callback for the sender side.
+  CHECK_ERROR(_acmLeft->RegisterTransportCallback(_channel));
+  CHECK_ERROR(_acmRight->RegisterTransportCallback(_channel));
+  // Register the receiver ACM in channel
+  _channel->RegisterReceiverACM(_acmReceiver.get());
+
+  uint16_t sampFreqHz = 32000;
+
+  const std::string file_name = webrtc::test::ResourcePath(
+      "audio_coding/testfile32kHz", "pcm");
+  _inFile.Open(file_name, sampFreqHz, "rb", false);
+
+  std::string output_file = webrtc::test::OutputPath()
+      + "out_spatial_autotest.pcm";
+  if (_testMode == 1) {
+    output_file = webrtc::test::OutputPath() + "testspatial_out.pcm";
+    printf("\n");
+    printf("Enter the output file [%s]: ", output_file.c_str());
+    PCMFile::ChooseFile(&output_file, MAX_FILE_NAME_LENGTH_BYTE, &sampFreqHz);
+  } else {
+    output_file = webrtc::test::OutputPath() + "testspatial_out.pcm";
+  }
+  _outFile.Open(output_file, sampFreqHz, "wb", false);
+  _outFile.SaveStereo(true);
+
+  // Register all available codes as receiving codecs.
+  CodecInst codecInst;
+  int status;
+  uint8_t num_encoders = _acmReceiver->NumberOfCodecs();
+  // Register all available codes as receiving codecs once more.
+  for (uint8_t n = 0; n < num_encoders; n++) {
+    status = _acmReceiver->Codec(n, &codecInst);
+    if (status < 0) {
+      printf("Error in Codec(), no matching codec found");
+    }
+    status = _acmReceiver->RegisterReceiveCodec(codecInst);
+    if (status < 0) {
+      printf("Error in RegisterReceiveCodec() for payload type %d",
+             codecInst.pltype);
+    }
+  }
+
+  return 0;
+}
+
+void SpatialAudio::Perform() {
+  if (_testMode == 0) {
+    printf("Running SpatialAudio Test");
+    WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+                 "---------- SpatialAudio ----------");
+  }
+
+  Setup();
+
+  CodecInst codecInst;
+  _acmLeft->Codec((uint8_t) 1, &codecInst);
+  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+  EncodeDecode();
+
+  int16_t pannCntr = 0;
+
+  double leftPanning[NUM_PANN_COEFFS] = { 1.00, 0.95, 0.90, 0.85, 0.80, 0.75,
+      0.70, 0.60, 0.55, 0.50 };
+  double rightPanning[NUM_PANN_COEFFS] = { 0.50, 0.55, 0.60, 0.70, 0.75, 0.80,
+      0.85, 0.90, 0.95, 1.00 };
+
+  while ((pannCntr + 1) < NUM_PANN_COEFFS) {
+    _acmLeft->Codec((uint8_t) 0, &codecInst);
+    codecInst.pacsize = 480;
+    CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+    CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+
+    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+    pannCntr++;
+
+    // Change codec
+    _acmLeft->Codec((uint8_t) 3, &codecInst);
+    codecInst.pacsize = 320;
+    CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+    CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+
+    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+    pannCntr++;
+    if (_testMode == 0) {
+      printf(".");
+    }
+  }
+
+  _acmLeft->Codec((uint8_t) 4, &codecInst);
+  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+  EncodeDecode();
+
+  _acmLeft->Codec((uint8_t) 0, &codecInst);
+  codecInst.pacsize = 480;
+  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+  CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+  pannCntr = NUM_PANN_COEFFS - 1;
+  while (pannCntr >= 0) {
+    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+    pannCntr--;
+    if (_testMode == 0) {
+      printf(".");
+    }
+  }
+  if (_testMode == 0) {
+    printf("Done!\n");
+  }
+}
+
+void SpatialAudio::EncodeDecode(const double leftPanning,
+                                const double rightPanning) {
+  AudioFrame audioFrame;
+  int32_t outFileSampFreq = _outFile.SamplingFrequency();
+
+  const double rightToLeftRatio = rightPanning / leftPanning;
+
+  _channel->SetIsStereo(true);
+
+  while (!_inFile.EndOfFile()) {
+    _inFile.Read10MsData(audioFrame);
+    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
+      audioFrame.data_[n] = (int16_t) floor(
+          audioFrame.data_[n] * leftPanning + 0.5);
+    }
+    CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
+
+    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
+      audioFrame.data_[n] = (int16_t) floor(
+          audioFrame.data_[n] * rightToLeftRatio + 0.5);
+    }
+    CHECK_ERROR(_acmRight->Add10MsData(audioFrame));
+
+    CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
+    _outFile.Write10MsData(audioFrame);
+  }
+  _inFile.Rewind();
+}
+
+void SpatialAudio::EncodeDecode() {
+  AudioFrame audioFrame;
+  int32_t outFileSampFreq = _outFile.SamplingFrequency();
+
+  _channel->SetIsStereo(false);
+
+  while (!_inFile.EndOfFile()) {
+    _inFile.Read10MsData(audioFrame);
+    CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
+
+    CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
+    _outFile.Write10MsData(audioFrame);
+  }
+  _inFile.Rewind();
+}
+
+}  // namespace webrtc