2017-01-24 04:38:27 -08:00
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/*
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* Copyright 2017 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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2017-09-15 06:47:31 +02:00
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#include "test/gtest.h"
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#include "test/gmock.h"
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2017-01-24 04:38:27 -08:00
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2017-09-15 06:47:31 +02:00
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#include "common_video/h264/h264_common.h"
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#include "media/base/mediaconstants.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/video_coding/include/video_coding_defines.h"
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#include "modules/video_coding/packet.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "modules/video_coding/timing.h"
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#include "rtc_base/bytebuffer.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ptr_util.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/field_trial_default.h"
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#include "test/field_trial.h"
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#include "video/rtp_video_stream_receiver.h"
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2017-01-24 04:38:27 -08:00
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using testing::_;
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namespace webrtc {
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namespace {
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const uint8_t kH264StartCode[] = {0x00, 0x00, 0x00, 0x01};
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class MockTransport : public Transport {
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public:
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MOCK_METHOD3(SendRtp,
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bool(const uint8_t* packet,
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size_t length,
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const PacketOptions& options));
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MOCK_METHOD2(SendRtcp, bool(const uint8_t* packet, size_t length));
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};
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class MockNackSender : public NackSender {
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public:
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MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers));
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};
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class MockKeyFrameRequestSender : public KeyFrameRequestSender {
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public:
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MOCK_METHOD0(RequestKeyFrame, void());
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};
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class MockOnCompleteFrameCallback
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: public video_coding::OnCompleteFrameCallback {
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public:
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MockOnCompleteFrameCallback() : buffer_(rtc::ByteBuffer::ORDER_NETWORK) {}
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MOCK_METHOD1(DoOnCompleteFrame, void(video_coding::FrameObject* frame));
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MOCK_METHOD1(DoOnCompleteFrameFailNullptr,
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void(video_coding::FrameObject* frame));
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MOCK_METHOD1(DoOnCompleteFrameFailLength,
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void(video_coding::FrameObject* frame));
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MOCK_METHOD1(DoOnCompleteFrameFailBitstream,
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void(video_coding::FrameObject* frame));
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void OnCompleteFrame(std::unique_ptr<video_coding::FrameObject> frame) {
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if (!frame) {
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DoOnCompleteFrameFailNullptr(nullptr);
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return;
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}
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EXPECT_EQ(buffer_.Length(), frame->size());
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if (buffer_.Length() != frame->size()) {
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DoOnCompleteFrameFailLength(frame.get());
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return;
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}
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std::vector<uint8_t> actual_data(frame->size());
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frame->GetBitstream(actual_data.data());
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if (memcmp(buffer_.Data(), actual_data.data(), buffer_.Length()) != 0) {
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DoOnCompleteFrameFailBitstream(frame.get());
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return;
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}
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DoOnCompleteFrame(frame.get());
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}
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void AppendExpectedBitstream(const uint8_t data[], size_t size_in_bytes) {
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// TODO(Johan): Let rtc::ByteBuffer handle uint8_t* instead of char*.
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buffer_.WriteBytes(reinterpret_cast<const char*>(data), size_in_bytes);
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}
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rtc::ByteBufferWriter buffer_;
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};
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2017-08-02 07:39:07 -07:00
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class MockRtpPacketSink : public RtpPacketSinkInterface {
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public:
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MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&));
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};
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constexpr uint32_t kSsrc = 111;
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constexpr uint16_t kSequenceNumber = 222;
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std::unique_ptr<RtpPacketReceived> CreateRtpPacketReceived(
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uint32_t ssrc = kSsrc,
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uint16_t sequence_number = kSequenceNumber) {
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auto packet = rtc::MakeUnique<RtpPacketReceived>();
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packet->SetSsrc(ssrc);
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packet->SetSequenceNumber(sequence_number);
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return packet;
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}
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MATCHER_P(SamePacketAs, other, "") {
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return arg.Ssrc() == other.Ssrc() &&
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arg.SequenceNumber() == other.SequenceNumber();
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}
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2017-01-24 04:38:27 -08:00
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} // namespace
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2017-06-09 04:01:55 -07:00
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class RtpVideoStreamReceiverTest : public testing::Test {
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2017-01-24 04:38:27 -08:00
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public:
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2017-11-02 14:28:06 +01:00
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RtpVideoStreamReceiverTest() : RtpVideoStreamReceiverTest("") {}
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explicit RtpVideoStreamReceiverTest(std::string field_trials)
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: override_field_trials_(field_trials),
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config_(CreateConfig()),
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2017-01-24 04:38:27 -08:00
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timing_(Clock::GetRealTimeClock()),
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process_thread_(ProcessThread::Create("TestThread")) {}
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void SetUp() {
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Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
Reason for revert:
Identified a configuration problem in the video quality tests. Intend to fix and reland.
Original issue's description:
> Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
>
> Reason for revert:
> This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec
>
> Original issue's description:
> > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
> >
> > Reason for revert:
> > Intend to fix perf failures and reland.
> >
> > Original issue's description:
> > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
> > >
> > > Reason for revert:
> > > A few perf tests broken, including
> > >
> > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
> > >
> > >
> > > Original issue's description:
> > > > Use RtxReceiveStream.
> > > >
> > > > This also has the beneficial side-effect that when a media stream
> > > > which is protected by FlexFEC receives an RTX retransmission, the
> > > > retransmitted media packet is passed into the FlexFEC machinery,
> > > > which should improve its ability to recover packets via FEC.
> > > >
> > > > BUG=webrtc:7135
> > > >
> > > > Review-Url: https://codereview.webrtc.org/3008773002
> > > > Cr-Commit-Position: refs/heads/master@{#19649}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7
> > >
> > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7135
> > >
> > > Review-Url: https://codereview.webrtc.org/3010983002
> > > Cr-Commit-Position: refs/heads/master@{#19653}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b
> >
> > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/3006063002
> > Cr-Commit-Position: refs/heads/master@{#19715}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78
>
> TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3007303002
> Cr-Commit-Position: refs/heads/master@{#19744}
> Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107
TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3012963002
Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
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rtp_receive_statistics_ =
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rtc::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
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rtp_video_stream_receiver_ = rtc::MakeUnique<RtpVideoStreamReceiver>(
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2017-04-18 23:38:35 -07:00
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&mock_transport_, nullptr, &packet_router_, &config_,
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Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
Reason for revert:
Identified a configuration problem in the video quality tests. Intend to fix and reland.
Original issue's description:
> Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
>
> Reason for revert:
> This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec
>
> Original issue's description:
> > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
> >
> > Reason for revert:
> > Intend to fix perf failures and reland.
> >
> > Original issue's description:
> > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
> > >
> > > Reason for revert:
> > > A few perf tests broken, including
> > >
> > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
> > >
> > >
> > > Original issue's description:
> > > > Use RtxReceiveStream.
> > > >
> > > > This also has the beneficial side-effect that when a media stream
> > > > which is protected by FlexFEC receives an RTX retransmission, the
> > > > retransmitted media packet is passed into the FlexFEC machinery,
> > > > which should improve its ability to recover packets via FEC.
> > > >
> > > > BUG=webrtc:7135
> > > >
> > > > Review-Url: https://codereview.webrtc.org/3008773002
> > > > Cr-Commit-Position: refs/heads/master@{#19649}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7
> > >
> > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7135
> > >
> > > Review-Url: https://codereview.webrtc.org/3010983002
> > > Cr-Commit-Position: refs/heads/master@{#19653}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b
> >
> > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/3006063002
> > Cr-Commit-Position: refs/heads/master@{#19715}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78
>
> TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3007303002
> Cr-Commit-Position: refs/heads/master@{#19744}
> Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107
TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3012963002
Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
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rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
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&mock_nack_sender_,
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2017-02-22 05:30:39 -08:00
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&mock_key_frame_request_sender_, &mock_on_complete_frame_callback_,
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Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
Reason for revert:
Identified a configuration problem in the video quality tests. Intend to fix and reland.
Original issue's description:
> Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
>
> Reason for revert:
> This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec
>
> Original issue's description:
> > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
> >
> > Reason for revert:
> > Intend to fix perf failures and reland.
> >
> > Original issue's description:
> > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
> > >
> > > Reason for revert:
> > > A few perf tests broken, including
> > >
> > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
> > >
> > >
> > > Original issue's description:
> > > > Use RtxReceiveStream.
> > > >
> > > > This also has the beneficial side-effect that when a media stream
> > > > which is protected by FlexFEC receives an RTX retransmission, the
> > > > retransmitted media packet is passed into the FlexFEC machinery,
> > > > which should improve its ability to recover packets via FEC.
> > > >
> > > > BUG=webrtc:7135
> > > >
> > > > Review-Url: https://codereview.webrtc.org/3008773002
> > > > Cr-Commit-Position: refs/heads/master@{#19649}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7
> > >
> > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7135
> > >
> > > Review-Url: https://codereview.webrtc.org/3010983002
> > > Cr-Commit-Position: refs/heads/master@{#19653}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b
> >
> > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/3006063002
> > Cr-Commit-Position: refs/heads/master@{#19715}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78
>
> TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3007303002
> Cr-Commit-Position: refs/heads/master@{#19744}
> Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107
TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3012963002
Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
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&timing_);
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2017-01-24 04:38:27 -08:00
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}
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WebRtcRTPHeader GetDefaultPacket() {
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WebRtcRTPHeader packet;
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memset(&packet, 0, sizeof(packet));
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packet.type.Video.codec = kRtpVideoH264;
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return packet;
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}
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// TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
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// code.
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2017-06-21 07:22:40 -07:00
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void AddSps(WebRtcRTPHeader* packet,
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uint8_t sps_id,
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|
|
std::vector<uint8_t>* data) {
|
2017-01-24 04:38:27 -08:00
|
|
|
NaluInfo info;
|
|
|
|
|
info.type = H264::NaluType::kSps;
|
|
|
|
|
info.sps_id = sps_id;
|
|
|
|
|
info.pps_id = -1;
|
|
|
|
|
data->push_back(H264::NaluType::kSps);
|
|
|
|
|
data->push_back(sps_id);
|
|
|
|
|
packet->type.Video.codecHeader.H264
|
|
|
|
|
.nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
void AddPps(WebRtcRTPHeader* packet,
|
2017-06-21 07:22:40 -07:00
|
|
|
uint8_t sps_id,
|
|
|
|
|
uint8_t pps_id,
|
2017-01-24 04:38:27 -08:00
|
|
|
std::vector<uint8_t>* data) {
|
|
|
|
|
NaluInfo info;
|
|
|
|
|
info.type = H264::NaluType::kPps;
|
|
|
|
|
info.sps_id = sps_id;
|
|
|
|
|
info.pps_id = pps_id;
|
|
|
|
|
data->push_back(H264::NaluType::kPps);
|
|
|
|
|
data->push_back(pps_id);
|
|
|
|
|
packet->type.Video.codecHeader.H264
|
|
|
|
|
.nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
void AddIdr(WebRtcRTPHeader* packet, int pps_id) {
|
|
|
|
|
NaluInfo info;
|
|
|
|
|
info.type = H264::NaluType::kIdr;
|
|
|
|
|
info.sps_id = -1;
|
|
|
|
|
info.pps_id = pps_id;
|
|
|
|
|
packet->type.Video.codecHeader.H264
|
|
|
|
|
.nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
protected:
|
|
|
|
|
static VideoReceiveStream::Config CreateConfig() {
|
|
|
|
|
VideoReceiveStream::Config config(nullptr);
|
|
|
|
|
config.rtp.remote_ssrc = 1111;
|
|
|
|
|
config.rtp.local_ssrc = 2222;
|
|
|
|
|
return config;
|
|
|
|
|
}
|
|
|
|
|
|
2017-11-02 14:28:06 +01:00
|
|
|
const webrtc::test::ScopedFieldTrials override_field_trials_;
|
2017-01-24 04:38:27 -08:00
|
|
|
VideoReceiveStream::Config config_;
|
|
|
|
|
MockNackSender mock_nack_sender_;
|
|
|
|
|
MockKeyFrameRequestSender mock_key_frame_request_sender_;
|
|
|
|
|
MockTransport mock_transport_;
|
|
|
|
|
MockOnCompleteFrameCallback mock_on_complete_frame_callback_;
|
|
|
|
|
PacketRouter packet_router_;
|
|
|
|
|
VCMTiming timing_;
|
|
|
|
|
std::unique_ptr<ProcessThread> process_thread_;
|
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
Reason for revert:
Identified a configuration problem in the video quality tests. Intend to fix and reland.
Original issue's description:
> Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
>
> Reason for revert:
> This change appears to break ulpfec, with severe regressions, e.g., for webrtc_perf_test FullStackTest.ForemanCifPlr5Ulpfec
>
> Original issue's description:
> > Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
> >
> > Reason for revert:
> > Intend to fix perf failures and reland.
> >
> > Original issue's description:
> > > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
> > >
> > > Reason for revert:
> > > A few perf tests broken, including
> > >
> > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberRtx
> > > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
> > >
> > >
> > > Original issue's description:
> > > > Use RtxReceiveStream.
> > > >
> > > > This also has the beneficial side-effect that when a media stream
> > > > which is protected by FlexFEC receives an RTX retransmission, the
> > > > retransmitted media packet is passed into the FlexFEC machinery,
> > > > which should improve its ability to recover packets via FEC.
> > > >
> > > > BUG=webrtc:7135
> > > >
> > > > Review-Url: https://codereview.webrtc.org/3008773002
> > > > Cr-Commit-Position: refs/heads/master@{#19649}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/5c0f6c62ea3b1d2c43f8fc152961af27033475f7
> > >
> > > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7135
> > >
> > > Review-Url: https://codereview.webrtc.org/3010983002
> > > Cr-Commit-Position: refs/heads/master@{#19653}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/3c39c0137afa274d1d524b150b50304b38a2847b
> >
> > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/3006063002
> > Cr-Commit-Position: refs/heads/master@{#19715}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/35713eaf565c0fef07c8afc158d7b8fdf7ec3d78
>
> TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3007303002
> Cr-Commit-Position: refs/heads/master@{#19744}
> Committed: https://chromium.googlesource.com/external/webrtc/+/8e7eee035178a7f10e19883681b5eaa4a7523107
TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3012963002
Cr-Commit-Position: refs/heads/master@{#19765}
2017-09-11 02:32:16 -07:00
|
|
|
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
|
2017-06-09 04:01:55 -07:00
|
|
|
std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_;
|
2017-01-24 04:38:27 -08:00
|
|
|
};
|
|
|
|
|
|
2017-06-09 04:01:55 -07:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) {
|
2017-01-24 04:38:27 -08:00
|
|
|
WebRtcRTPHeader rtp_header;
|
|
|
|
|
const std::vector<uint8_t> data({1, 2, 3, 4});
|
|
|
|
|
memset(&rtp_header, 0, sizeof(rtp_header));
|
|
|
|
|
rtp_header.header.sequenceNumber = 1;
|
|
|
|
|
rtp_header.header.markerBit = 1;
|
|
|
|
|
rtp_header.type.Video.is_first_packet_in_frame = true;
|
|
|
|
|
rtp_header.frameType = kVideoFrameKey;
|
|
|
|
|
rtp_header.type.Video.codec = kRtpVideoGeneric;
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
|
|
|
|
|
data.size());
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
|
|
|
|
&rtp_header);
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
2017-06-09 04:01:55 -07:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrameBitstreamError) {
|
2017-01-24 04:38:27 -08:00
|
|
|
WebRtcRTPHeader rtp_header;
|
|
|
|
|
const std::vector<uint8_t> data({1, 2, 3, 4});
|
|
|
|
|
memset(&rtp_header, 0, sizeof(rtp_header));
|
|
|
|
|
rtp_header.header.sequenceNumber = 1;
|
|
|
|
|
rtp_header.header.markerBit = 1;
|
|
|
|
|
rtp_header.type.Video.is_first_packet_in_frame = true;
|
|
|
|
|
rtp_header.frameType = kVideoFrameKey;
|
|
|
|
|
rtp_header.type.Video.codec = kRtpVideoGeneric;
|
|
|
|
|
constexpr uint8_t expected_bitsteam[] = {1, 2, 3, 0xff};
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
|
|
|
|
expected_bitsteam, sizeof(expected_bitsteam));
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_,
|
|
|
|
|
DoOnCompleteFrameFailBitstream(_));
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
|
|
|
|
&rtp_header);
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
2017-11-02 14:28:06 +01:00
|
|
|
class RtpVideoStreamReceiverTestH264
|
|
|
|
|
: public RtpVideoStreamReceiverTest,
|
|
|
|
|
public testing::WithParamInterface<std::string> {
|
|
|
|
|
protected:
|
|
|
|
|
RtpVideoStreamReceiverTestH264() : RtpVideoStreamReceiverTest(GetParam()) {}
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
INSTANTIATE_TEST_CASE_P(
|
|
|
|
|
SpsPpsIdrIsKeyframe,
|
|
|
|
|
RtpVideoStreamReceiverTestH264,
|
|
|
|
|
::testing::Values("", "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/"));
|
|
|
|
|
|
|
|
|
|
TEST_P(RtpVideoStreamReceiverTestH264, InBandSpsPps) {
|
2017-01-24 04:38:27 -08:00
|
|
|
std::vector<uint8_t> sps_data;
|
|
|
|
|
WebRtcRTPHeader sps_packet = GetDefaultPacket();
|
|
|
|
|
AddSps(&sps_packet, 0, &sps_data);
|
|
|
|
|
sps_packet.header.sequenceNumber = 0;
|
2017-06-21 07:22:40 -07:00
|
|
|
sps_packet.type.Video.is_first_packet_in_frame = true;
|
2017-01-24 04:38:27 -08:00
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
|
|
|
|
kH264StartCode, sizeof(kH264StartCode));
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(sps_data.data(),
|
|
|
|
|
sps_data.size());
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(
|
|
|
|
|
sps_data.data(), sps_data.size(), &sps_packet);
|
2017-01-24 04:38:27 -08:00
|
|
|
|
|
|
|
|
std::vector<uint8_t> pps_data;
|
|
|
|
|
WebRtcRTPHeader pps_packet = GetDefaultPacket();
|
|
|
|
|
AddPps(&pps_packet, 0, 1, &pps_data);
|
|
|
|
|
pps_packet.header.sequenceNumber = 1;
|
2017-06-21 07:22:40 -07:00
|
|
|
pps_packet.type.Video.is_first_packet_in_frame = true;
|
2017-01-24 04:38:27 -08:00
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
|
|
|
|
kH264StartCode, sizeof(kH264StartCode));
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(pps_data.data(),
|
|
|
|
|
pps_data.size());
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(
|
|
|
|
|
pps_data.data(), pps_data.size(), &pps_packet);
|
2017-01-24 04:38:27 -08:00
|
|
|
|
|
|
|
|
std::vector<uint8_t> idr_data;
|
|
|
|
|
WebRtcRTPHeader idr_packet = GetDefaultPacket();
|
|
|
|
|
AddIdr(&idr_packet, 1);
|
|
|
|
|
idr_packet.type.Video.is_first_packet_in_frame = true;
|
|
|
|
|
idr_packet.header.sequenceNumber = 2;
|
|
|
|
|
idr_packet.header.markerBit = 1;
|
|
|
|
|
idr_packet.frameType = kVideoFrameKey;
|
|
|
|
|
idr_data.insert(idr_data.end(), {0x65, 1, 2, 3});
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
|
|
|
|
kH264StartCode, sizeof(kH264StartCode));
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(idr_data.data(),
|
|
|
|
|
idr_data.size());
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(
|
|
|
|
|
idr_data.data(), idr_data.size(), &idr_packet);
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
2017-11-02 14:28:06 +01:00
|
|
|
TEST_P(RtpVideoStreamReceiverTestH264, OutOfBandFmtpSpsPps) {
|
2017-01-24 04:38:27 -08:00
|
|
|
constexpr int kPayloadType = 99;
|
|
|
|
|
VideoCodec codec;
|
|
|
|
|
codec.plType = kPayloadType;
|
|
|
|
|
std::map<std::string, std::string> codec_params;
|
|
|
|
|
// Example parameter sets from https://tools.ietf.org/html/rfc3984#section-8.2
|
|
|
|
|
// .
|
|
|
|
|
codec_params.insert(
|
|
|
|
|
{cricket::kH264FmtpSpropParameterSets, "Z0IACpZTBYmI,aMljiA=="});
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->AddReceiveCodec(codec, codec_params);
|
2017-01-24 04:38:27 -08:00
|
|
|
const uint8_t binary_sps[] = {0x67, 0x42, 0x00, 0x0a, 0x96,
|
|
|
|
|
0x53, 0x05, 0x89, 0x88};
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
|
|
|
|
kH264StartCode, sizeof(kH264StartCode));
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_sps,
|
|
|
|
|
sizeof(binary_sps));
|
|
|
|
|
const uint8_t binary_pps[] = {0x68, 0xc9, 0x63, 0x88};
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
|
|
|
|
kH264StartCode, sizeof(kH264StartCode));
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_pps,
|
|
|
|
|
sizeof(binary_pps));
|
|
|
|
|
|
|
|
|
|
std::vector<uint8_t> data;
|
|
|
|
|
WebRtcRTPHeader idr_packet = GetDefaultPacket();
|
|
|
|
|
AddIdr(&idr_packet, 0);
|
|
|
|
|
idr_packet.header.payloadType = kPayloadType;
|
|
|
|
|
idr_packet.type.Video.is_first_packet_in_frame = true;
|
|
|
|
|
idr_packet.header.sequenceNumber = 2;
|
|
|
|
|
idr_packet.header.markerBit = 1;
|
|
|
|
|
idr_packet.type.Video.is_first_packet_in_frame = true;
|
|
|
|
|
idr_packet.frameType = kVideoFrameKey;
|
|
|
|
|
idr_packet.type.Video.codec = kRtpVideoH264;
|
|
|
|
|
data.insert(data.end(), {1, 2, 3});
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
|
|
|
|
kH264StartCode, sizeof(kH264StartCode));
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
|
|
|
|
|
data.size());
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
|
|
|
|
&idr_packet);
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
2017-06-09 04:01:55 -07:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, PaddingInMediaStream) {
|
2017-03-21 05:45:18 -07:00
|
|
|
WebRtcRTPHeader header = GetDefaultPacket();
|
|
|
|
|
std::vector<uint8_t> data;
|
|
|
|
|
data.insert(data.end(), {1, 2, 3});
|
|
|
|
|
header.header.payloadType = 99;
|
|
|
|
|
header.type.Video.is_first_packet_in_frame = true;
|
|
|
|
|
header.header.sequenceNumber = 2;
|
|
|
|
|
header.header.markerBit = true;
|
|
|
|
|
header.frameType = kVideoFrameKey;
|
|
|
|
|
header.type.Video.codec = kRtpVideoGeneric;
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
|
|
|
|
|
data.size());
|
|
|
|
|
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
|
|
|
|
&header);
|
2017-03-21 05:45:18 -07:00
|
|
|
|
|
|
|
|
header.header.sequenceNumber = 3;
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(nullptr, 0, &header);
|
2017-03-21 05:45:18 -07:00
|
|
|
|
|
|
|
|
header.frameType = kVideoFrameDelta;
|
|
|
|
|
header.header.sequenceNumber = 4;
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
|
|
|
|
&header);
|
2017-03-21 05:45:18 -07:00
|
|
|
|
|
|
|
|
header.header.sequenceNumber = 6;
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
|
|
|
|
&header);
|
2017-03-21 05:45:18 -07:00
|
|
|
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EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
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header.header.sequenceNumber = 5;
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2017-06-09 04:01:55 -07:00
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rtp_video_stream_receiver_->OnReceivedPayloadData(nullptr, 0, &header);
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2017-03-21 05:45:18 -07:00
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}
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2017-06-09 04:01:55 -07:00
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TEST_F(RtpVideoStreamReceiverTest, RequestKeyframeIfFirstFrameIsDelta) {
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2017-05-16 08:06:30 -07:00
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WebRtcRTPHeader rtp_header;
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const std::vector<uint8_t> data({1, 2, 3, 4});
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memset(&rtp_header, 0, sizeof(rtp_header));
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rtp_header.header.sequenceNumber = 1;
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rtp_header.header.markerBit = 1;
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rtp_header.type.Video.is_first_packet_in_frame = true;
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rtp_header.frameType = kVideoFrameDelta;
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rtp_header.type.Video.codec = kRtpVideoGeneric;
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EXPECT_CALL(mock_key_frame_request_sender_, RequestKeyFrame());
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2017-06-09 04:01:55 -07:00
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rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
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&rtp_header);
|
2017-05-16 08:06:30 -07:00
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}
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2017-08-02 07:39:07 -07:00
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TEST_F(RtpVideoStreamReceiverTest, SecondarySinksGetRtpNotifications) {
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rtp_video_stream_receiver_->StartReceive();
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MockRtpPacketSink secondary_sink_1;
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MockRtpPacketSink secondary_sink_2;
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rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink_1);
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rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink_2);
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auto rtp_packet = CreateRtpPacketReceived();
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EXPECT_CALL(secondary_sink_1, OnRtpPacket(SamePacketAs(*rtp_packet)));
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EXPECT_CALL(secondary_sink_2, OnRtpPacket(SamePacketAs(*rtp_packet)));
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rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
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|
|
|
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|
|
// Test tear-down.
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rtp_video_stream_receiver_->StopReceive();
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rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink_1);
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rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink_2);
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|
}
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|
|
TEST_F(RtpVideoStreamReceiverTest, RemovedSecondarySinksGetNoRtpNotifications) {
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|
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|
|
rtp_video_stream_receiver_->StartReceive();
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|
|
|
|
|
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|
|
MockRtpPacketSink secondary_sink;
|
|
|
|
|
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|
|
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
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|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
|
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|
|
|
|
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|
|
auto rtp_packet = CreateRtpPacketReceived();
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|
|
|
|
|
|
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|
|
EXPECT_CALL(secondary_sink, OnRtpPacket(_)).Times(0);
|
|
|
|
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|
|
|
rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
|
|
|
|
|
|
|
|
|
|
// Test tear-down.
|
|
|
|
|
rtp_video_stream_receiver_->StopReceive();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
TEST_F(RtpVideoStreamReceiverTest,
|
|
|
|
|
OnlyRemovedSecondarySinksExcludedFromNotifications) {
|
|
|
|
|
rtp_video_stream_receiver_->StartReceive();
|
|
|
|
|
|
|
|
|
|
MockRtpPacketSink kept_secondary_sink;
|
|
|
|
|
MockRtpPacketSink removed_secondary_sink;
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->AddSecondarySink(&kept_secondary_sink);
|
|
|
|
|
rtp_video_stream_receiver_->AddSecondarySink(&removed_secondary_sink);
|
|
|
|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&removed_secondary_sink);
|
|
|
|
|
|
|
|
|
|
auto rtp_packet = CreateRtpPacketReceived();
|
|
|
|
|
EXPECT_CALL(kept_secondary_sink, OnRtpPacket(SamePacketAs(*rtp_packet)));
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
|
|
|
|
|
|
|
|
|
|
// Test tear-down.
|
|
|
|
|
rtp_video_stream_receiver_->StopReceive();
|
|
|
|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&kept_secondary_sink);
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
TEST_F(RtpVideoStreamReceiverTest,
|
|
|
|
|
SecondariesOfNonStartedStreamGetNoNotifications) {
|
|
|
|
|
// Explicitly showing that the stream is not in the |started| state,
|
|
|
|
|
// regardless of whether streams start out |started| or |stopped|.
|
|
|
|
|
rtp_video_stream_receiver_->StopReceive();
|
|
|
|
|
|
|
|
|
|
MockRtpPacketSink secondary_sink;
|
|
|
|
|
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
|
|
|
|
|
|
|
|
|
|
auto rtp_packet = CreateRtpPacketReceived();
|
|
|
|
|
EXPECT_CALL(secondary_sink, OnRtpPacket(_)).Times(0);
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
|
|
|
|
|
|
|
|
|
|
// Test tear-down.
|
|
|
|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
|
|
|
|
|
TEST_F(RtpVideoStreamReceiverTest, RepeatedSecondarySinkDisallowed) {
|
|
|
|
|
MockRtpPacketSink secondary_sink;
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
|
|
|
|
|
EXPECT_DEATH(rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink),
|
|
|
|
|
"");
|
|
|
|
|
|
|
|
|
|
// Test tear-down.
|
|
|
|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
|
|
|
|
|
}
|
|
|
|
|
#endif
|
|
|
|
|
|
2017-01-24 04:38:27 -08:00
|
|
|
} // namespace webrtc
|