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, 411 deletions

diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
deleted file mode 100644
index 47f2880638..0000000000
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
+++ /dev/null
@@ -1,411 +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/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/test/channel_transport/udp_transport.h"
-
-#define NR_OF_SOCKET_BUFFERS 500
-
-
-bool ProcThreadFunction(void *obj)
-{
-    if (obj == NULL)
-    {
-        return false;
-    }
-    TestSenderReceiver *theObj = static_cast<TestSenderReceiver *>(obj);
-
-    return theObj->ProcLoop();
-}
-
-
-TestSenderReceiver::TestSenderReceiver (void)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_running(false),
-_payloadType(0),
-_loadGenerator(NULL),
-_isSender(false),
-_isReceiver(false),
-_sendRecCB(NULL),
-_lastBytesReceived(0),
-_lastTime(-1)
-{
-    // RTP/RTCP module
-    _rtp = RtpRtcp::CreateRtpRtcp(0, false);
-    if (!_rtp)
-    {
-        throw "Could not create RTP/RTCP module";
-        exit(1);
-    }
-
-    if (_rtp->InitReceiver() != 0)
-    {
-        throw "_rtp->InitReceiver()";
-        exit(1);
-    }
-
-    if (_rtp->InitSender() != 0)
-    {
-        throw "_rtp->InitSender()";
-        exit(1);
-    }
-
-    // SocketTransport module
-    uint8_t numberOfThreads = 1;
-    _transport = UdpTransport::Create(0, numberOfThreads);
-    if (!_transport)
-    {
-        throw "Could not create transport module";
-        exit(1);
-    }
-}
-
-TestSenderReceiver::~TestSenderReceiver (void)
-{
-
-    Stop(); // N.B. without critSect
-
-    _critSect->Enter();
-
-    if (_rtp)
-    {
-        RtpRtcp::DestroyRtpRtcp(_rtp);
-        _rtp = NULL;
-    }
-
-    if (_transport)
-    {
-        UdpTransport::Destroy(_transport);
-        _transport = NULL;
-    }
-
-    delete _critSect;
-
-}
-
-
-int32_t TestSenderReceiver::InitReceiver (const uint16_t rtpPort,
-                                          const uint16_t rtcpPort,
-                                          const int8_t payloadType /*= 127*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    // init transport
-    if (_transport->InitializeReceiveSockets(this, rtpPort/*, 0, NULL, 0, true*/) != 0)
-    {
-        throw "_transport->InitializeReceiveSockets";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingRTPCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingRTPCallback";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingDataCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingRTPCallback";
-        exit(1);
-    }
-
-    if (_rtp->SetRTCPStatus(RtcpMode::kReducedSize) != 0) {
-        throw "_rtp->SetRTCPStatus";
-        exit(1);
-    }
-
-    if (_rtp->SetTMMBRStatus(true) != 0)
-    {
-        throw "_rtp->SetTMMBRStatus";
-        exit(1);
-    }
-
-    if (_rtp->RegisterReceivePayload("I420", payloadType, 90000) != 0)
-    {
-        throw "_rtp->RegisterReceivePayload";
-        exit(1);
-    }
-
-    _isReceiver = true;
-
-    return (0);
-}
-
-
-int32_t TestSenderReceiver::Start()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _eventPtr = EventWrapper::Create();
-
-    if (_rtp->SetSendingStatus(true) != 0)
-    {
-        throw "_rtp->SetSendingStatus";
-        exit(1);
-    }
-
-    _procThread = ThreadWrapper::CreateThread(ProcThreadFunction, this,
-                                              "TestSenderReceiver");
-
-    _running = true;
-
-    if (_isReceiver)
-    {
-        if (_transport->StartReceiving(NR_OF_SOCKET_BUFFERS) != 0)
-        {
-            throw "_transport->StartReceiving";
-            exit(1);
-        }
-    }
-
-    _procThread->Start();
-    _procThread->SetPriority(kRealtimePriority);
-
-    return 0;
-
-}
-
-
-int32_t TestSenderReceiver::Stop ()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _transport->StopReceiving();
-
-    if (_procThread)
-    {
-        _running = false;
-        _eventPtr->Set();
-
-        _procThread->Stop();
-        _procThread.reset();
-
-        delete _eventPtr;
-    }
-
-    return (0);
-}
-
-
-bool TestSenderReceiver::ProcLoop(void)
-{
-
-    // process RTP/RTCP module
-    _rtp->Process();
-
-    // process SocketTransport module
-    _transport->Process();
-
-    // no critSect
-    while (_running)
-    {
-        // ask RTP/RTCP module for wait time
-        int32_t rtpWait = _rtp->TimeUntilNextProcess();
-
-        // ask SocketTransport module for wait time
-        int32_t tpWait = _transport->TimeUntilNextProcess();
-
-        int32_t minWait = (rtpWait < tpWait) ? rtpWait: tpWait;
-        minWait = (minWait > 0) ? minWait : 0;
-        // wait
-        _eventPtr->Wait(minWait);
-
-        // process RTP/RTCP module
-        _rtp->Process();
-
-        // process SocketTransport module
-        _transport->Process();
-
-    }
-
-    return true;
-}
-
-
-int32_t TestSenderReceiver::ReceiveBitrateKbps ()
-{
-    size_t bytesSent;
-    uint32_t packetsSent;
-    size_t bytesReceived;
-    uint32_t packetsReceived;
-
-    if (_rtp->DataCountersRTP(&bytesSent, &packetsSent, &bytesReceived, &packetsReceived) == 0)
-    {
-        int64_t now = TickTime::MillisecondTimestamp();
-        int32_t kbps = 0;
-        if (now > _lastTime)
-        {
-            if (_lastTime > 0)
-            {
-                // 8 * bytes / ms = kbps
-                kbps = static_cast<int32_t>(
-                    (8 * (bytesReceived - _lastBytesReceived)) / (now - _lastTime));
-            }
-            _lastTime = now;
-            _lastBytesReceived = bytesReceived;
-        }
-        return (kbps);
-    }
-
-    return (-1);
-}
-
-
-int32_t TestSenderReceiver::SetPacketTimeout(const uint32_t timeoutMS)
-{
-    return (_rtp->SetPacketTimeout(timeoutMS, 0 /* RTCP timeout */));
-}
-
-
-int32_t TestSenderReceiver::OnReceivedPayloadData(const uint8_t* payloadData,
-                                                  const size_t payloadSize,
-                                                  const webrtc::WebRtcRTPHeader* rtpHeader)
-{
-    //printf("OnReceivedPayloadData\n");
-    return (0);
-}
-
-
-void TestSenderReceiver::IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                                           const size_t rtpPacketLength,
-                                           const int8_t* fromIP,
-                                           const uint16_t fromPort)
-{
-    _rtp->IncomingPacket((uint8_t *) incomingRtpPacket, rtpPacketLength);
-}
-
-
-
-void TestSenderReceiver::IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                                            const size_t rtcpPacketLength,
-                                            const int8_t* fromIP,
-                                            const uint16_t fromPort)
-{
-    _rtp->IncomingPacket((uint8_t *) incomingRtcpPacket, rtcpPacketLength);
-}
-
-
-
-
-
-///////////////////
-
-
-int32_t TestSenderReceiver::InitSender (const uint32_t startBitrateKbps,
-                                        const int8_t* ipAddr,
-                                        const uint16_t rtpPort,
-                                        const uint16_t rtcpPort /*= 0*/,
-                                        const int8_t payloadType /*= 127*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _payloadType = payloadType;
-
-    // check load generator valid
-    if (_loadGenerator)
-    {
-        _loadGenerator->SetBitrate(startBitrateKbps);
-    }
-
-    if (_rtp->RegisterSendTransport(_transport) != 0)
-    {
-        throw "_rtp->RegisterSendTransport";
-        exit(1);
-    }
-    if (_rtp->RegisterSendPayload("I420", _payloadType, 90000) != 0)
-    {
-        throw "_rtp->RegisterSendPayload";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingVideoCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingVideoCallback";
-        exit(1);
-    }
-
-    if (_rtp->SetRTCPStatus(RtcpMode::kReducedSize) != 0) {
-        throw "_rtp->SetRTCPStatus";
-        exit(1);
-    }
-
-    if (_rtp->SetSendBitrate(startBitrateKbps*1000, 0, MAX_BITRATE_KBPS) != 0)
-    {
-        throw "_rtp->SetSendBitrate";
-        exit(1);
-    }
-
-
-    // SocketTransport
-    if (_transport->InitializeSendSockets(ipAddr, rtpPort, rtcpPort))
-    {
-        throw "_transport->InitializeSendSockets";
-        exit(1);
-    }
-
-    _isSender = true;
-
-    return (0);
-}
-
-
-
-int32_t
-TestSenderReceiver::SendOutgoingData(const uint32_t timeStamp,
-                                     const uint8_t* payloadData,
-                                     const size_t payloadSize,
-                                     const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
-{
-    return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize));
-}
-
-
-int32_t TestSenderReceiver::SetLoadGenerator(TestLoadGenerator *generator)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _loadGenerator = generator;
-    return(0);
-
-}
-
-void TestSenderReceiver::OnNetworkChanged(const int32_t id,
-                                  const uint32_t minBitrateBps,
-                                  const uint32_t maxBitrateBps,
-                                  const uint8_t fractionLost,
-                                  const uint16_t roundTripTimeMs,
-                                  const uint16_t bwEstimateKbitMin,
-                                  const uint16_t bwEstimateKbitMax)
-{
-    if (_loadGenerator)
-    {
-        _loadGenerator->SetBitrate(maxBitrateBps/1000);
-    }
-
-    if (_sendRecCB)
-    {
-        _sendRecCB->OnOnNetworkChanged(maxBitrateBps,
-            fractionLost,
-            roundTripTimeMs,
-            bwEstimateKbitMin,
-            bwEstimateKbitMax);
-    }
-}