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, 204 insertions, 0 deletions

diff --git a/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
new file mode 100644
index 0000000000..c9ec372f41
--- /dev/null
+++ b/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2013 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/video_coding/test/vcm_payload_sink_factory.h"
+
+#include <assert.h>
+
+#include <algorithm>
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+namespace rtpplayer {
+
+class VcmPayloadSinkFactory::VcmPayloadSink : public PayloadSinkInterface,
+                                              public VCMPacketRequestCallback {
+ public:
+  VcmPayloadSink(VcmPayloadSinkFactory* factory,
+                 RtpStreamInterface* stream,
+                 rtc::scoped_ptr<VideoCodingModule>* vcm,
+                 rtc::scoped_ptr<FileOutputFrameReceiver>* frame_receiver)
+      : factory_(factory), stream_(stream), vcm_(), frame_receiver_() {
+    assert(factory);
+    assert(stream);
+    assert(vcm);
+    assert(vcm->get());
+    assert(frame_receiver);
+    assert(frame_receiver->get());
+    vcm_.swap(*vcm);
+    frame_receiver_.swap(*frame_receiver);
+    vcm_->RegisterPacketRequestCallback(this);
+    vcm_->RegisterReceiveCallback(frame_receiver_.get());
+  }
+
+  virtual ~VcmPayloadSink() { factory_->Remove(this); }
+
+  // PayloadSinkInterface
+  int32_t OnReceivedPayloadData(const uint8_t* payload_data,
+                                const size_t payload_size,
+                                const WebRtcRTPHeader* rtp_header) override {
+    return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
+  }
+
+  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
+    // We currently don't handle FEC.
+    return true;
+  }
+
+  // VCMPacketRequestCallback
+  int32_t ResendPackets(const uint16_t* sequence_numbers,
+                        uint16_t length) override {
+    stream_->ResendPackets(sequence_numbers, length);
+    return 0;
+  }
+
+  int DecodeAndProcess(bool should_decode, bool decode_dual_frame) {
+    if (should_decode) {
+      if (vcm_->Decode() < 0) {
+        return -1;
+      }
+    }
+    return Process() ? 0 : -1;
+  }
+
+  bool Process() {
+    if (vcm_->TimeUntilNextProcess() <= 0) {
+      if (vcm_->Process() < 0) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  bool Decode() {
+    vcm_->Decode(10000);
+    return true;
+  }
+
+ private:
+  VcmPayloadSinkFactory* factory_;
+  RtpStreamInterface* stream_;
+  rtc::scoped_ptr<VideoCodingModule> vcm_;
+  rtc::scoped_ptr<FileOutputFrameReceiver> frame_receiver_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VcmPayloadSink);
+};
+
+VcmPayloadSinkFactory::VcmPayloadSinkFactory(
+    const std::string& base_out_filename,
+    Clock* clock,
+    bool protection_enabled,
+    VCMVideoProtection protection_method,
+    int64_t rtt_ms,
+    uint32_t render_delay_ms,
+    uint32_t min_playout_delay_ms)
+    : base_out_filename_(base_out_filename),
+      clock_(clock),
+      protection_enabled_(protection_enabled),
+      protection_method_(protection_method),
+      rtt_ms_(rtt_ms),
+      render_delay_ms_(render_delay_ms),
+      min_playout_delay_ms_(min_playout_delay_ms),
+      null_event_factory_(new NullEventFactory()),
+      crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+      sinks_() {
+  assert(clock);
+  assert(crit_sect_.get());
+}
+
+VcmPayloadSinkFactory::~VcmPayloadSinkFactory() {
+  assert(sinks_.empty());
+}
+
+PayloadSinkInterface* VcmPayloadSinkFactory::Create(
+    RtpStreamInterface* stream) {
+  assert(stream);
+  CriticalSectionScoped cs(crit_sect_.get());
+
+  rtc::scoped_ptr<VideoCodingModule> vcm(
+      VideoCodingModule::Create(clock_, null_event_factory_.get()));
+  if (vcm.get() == NULL) {
+    return NULL;
+  }
+
+  const PayloadTypes& plt = stream->payload_types();
+  for (PayloadTypesIterator it = plt.begin(); it != plt.end(); ++it) {
+    if (it->codec_type() != kVideoCodecULPFEC &&
+        it->codec_type() != kVideoCodecRED) {
+      VideoCodec codec;
+      VideoCodingModule::Codec(it->codec_type(), &codec);
+      codec.plType = it->payload_type();
+      if (vcm->RegisterReceiveCodec(&codec, 1) < 0) {
+        return NULL;
+      }
+    }
+  }
+
+  vcm->SetChannelParameters(0, 0, rtt_ms_);
+  vcm->SetVideoProtection(protection_method_, protection_enabled_);
+  vcm->SetRenderDelay(render_delay_ms_);
+  vcm->SetMinimumPlayoutDelay(min_playout_delay_ms_);
+  vcm->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack, 0);
+
+  rtc::scoped_ptr<FileOutputFrameReceiver> frame_receiver(
+      new FileOutputFrameReceiver(base_out_filename_, stream->ssrc()));
+  rtc::scoped_ptr<VcmPayloadSink> sink(
+      new VcmPayloadSink(this, stream, &vcm, &frame_receiver));
+
+  sinks_.push_back(sink.get());
+  return sink.release();
+}
+
+int VcmPayloadSinkFactory::DecodeAndProcessAll(bool decode_dual_frame) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  assert(clock_);
+  bool should_decode = (clock_->TimeInMilliseconds() % 5) == 0;
+  for (Sinks::iterator it = sinks_.begin(); it != sinks_.end(); ++it) {
+    if ((*it)->DecodeAndProcess(should_decode, decode_dual_frame) < 0) {
+      return -1;
+    }
+  }
+  return 0;
+}
+
+bool VcmPayloadSinkFactory::ProcessAll() {
+  CriticalSectionScoped cs(crit_sect_.get());
+  for (Sinks::iterator it = sinks_.begin(); it != sinks_.end(); ++it) {
+    if (!(*it)->Process()) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool VcmPayloadSinkFactory::DecodeAll() {
+  CriticalSectionScoped cs(crit_sect_.get());
+  for (Sinks::iterator it = sinks_.begin(); it != sinks_.end(); ++it) {
+    if (!(*it)->Decode()) {
+      return false;
+    }
+  }
+  return true;
+}
+
+void VcmPayloadSinkFactory::Remove(VcmPayloadSink* sink) {
+  assert(sink);
+  CriticalSectionScoped cs(crit_sect_.get());
+  Sinks::iterator it = std::find(sinks_.begin(), sinks_.end(), sink);
+  assert(it != sinks_.end());
+  sinks_.erase(it);
+}
+
+}  // namespace rtpplayer
+}  // namespace webrtc