2017-01-24 04:38:27 -08:00
|
|
|
/*
|
|
|
|
|
* Copyright 2017 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.
|
|
|
|
|
*/
|
|
|
|
|
|
2017-09-15 06:47:31 +02:00
|
|
|
#include "test/gmock.h"
|
|
|
|
|
#include "test/gtest.h"
|
2017-01-24 04:38:27 -08:00
|
|
|
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
#include "absl/memory/memory.h"
|
2017-09-15 06:47:31 +02:00
|
|
|
#include "common_video/h264/h264_common.h"
|
2019-01-11 09:11:00 -08:00
|
|
|
#include "media/base/media_constants.h"
|
2017-09-15 06:47:31 +02:00
|
|
|
#include "modules/pacing/packet_router.h"
|
2018-09-25 12:54:37 +02:00
|
|
|
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
|
|
|
|
|
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
|
2017-09-15 06:47:31 +02:00
|
|
|
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
|
|
|
|
#include "modules/utility/include/process_thread.h"
|
|
|
|
|
#include "modules/video_coding/frame_object.h"
|
|
|
|
|
#include "modules/video_coding/include/video_coding_defines.h"
|
|
|
|
|
#include "modules/video_coding/packet.h"
|
|
|
|
|
#include "modules/video_coding/rtp_frame_reference_finder.h"
|
2019-01-11 09:11:00 -08:00
|
|
|
#include "rtc_base/byte_buffer.h"
|
2017-09-15 06:47:31 +02:00
|
|
|
#include "rtc_base/logging.h"
|
|
|
|
|
#include "system_wrappers/include/clock.h"
|
2018-09-28 08:51:10 +02:00
|
|
|
#include "system_wrappers/include/field_trial.h"
|
2017-09-15 06:47:31 +02:00
|
|
|
#include "test/field_trial.h"
|
|
|
|
|
#include "video/rtp_video_stream_receiver.h"
|
2017-01-24 04:38:27 -08:00
|
|
|
|
2018-09-25 12:54:37 +02:00
|
|
|
using ::testing::_;
|
|
|
|
|
using ::testing::Invoke;
|
2017-01-24 04:38:27 -08:00
|
|
|
|
|
|
|
|
namespace webrtc {
|
|
|
|
|
|
|
|
|
|
namespace {
|
|
|
|
|
|
|
|
|
|
const uint8_t kH264StartCode[] = {0x00, 0x00, 0x00, 0x01};
|
|
|
|
|
|
|
|
|
|
class MockTransport : public Transport {
|
|
|
|
|
public:
|
|
|
|
|
MOCK_METHOD3(SendRtp,
|
|
|
|
|
bool(const uint8_t* packet,
|
|
|
|
|
size_t length,
|
|
|
|
|
const PacketOptions& options));
|
|
|
|
|
MOCK_METHOD2(SendRtcp, bool(const uint8_t* packet, size_t length));
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
class MockNackSender : public NackSender {
|
|
|
|
|
public:
|
|
|
|
|
MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers));
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
class MockKeyFrameRequestSender : public KeyFrameRequestSender {
|
|
|
|
|
public:
|
|
|
|
|
MOCK_METHOD0(RequestKeyFrame, void());
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
class MockOnCompleteFrameCallback
|
|
|
|
|
: public video_coding::OnCompleteFrameCallback {
|
|
|
|
|
public:
|
|
|
|
|
MockOnCompleteFrameCallback() : buffer_(rtc::ByteBuffer::ORDER_NETWORK) {}
|
|
|
|
|
|
2018-02-22 14:35:06 +01:00
|
|
|
MOCK_METHOD1(DoOnCompleteFrame, void(video_coding::EncodedFrame* frame));
|
2017-01-24 04:38:27 -08:00
|
|
|
MOCK_METHOD1(DoOnCompleteFrameFailNullptr,
|
2018-02-22 14:35:06 +01:00
|
|
|
void(video_coding::EncodedFrame* frame));
|
2017-01-24 04:38:27 -08:00
|
|
|
MOCK_METHOD1(DoOnCompleteFrameFailLength,
|
2018-02-22 14:35:06 +01:00
|
|
|
void(video_coding::EncodedFrame* frame));
|
2017-01-24 04:38:27 -08:00
|
|
|
MOCK_METHOD1(DoOnCompleteFrameFailBitstream,
|
2018-02-22 14:35:06 +01:00
|
|
|
void(video_coding::EncodedFrame* frame));
|
|
|
|
|
void OnCompleteFrame(std::unique_ptr<video_coding::EncodedFrame> frame) {
|
2017-01-24 04:38:27 -08:00
|
|
|
if (!frame) {
|
|
|
|
|
DoOnCompleteFrameFailNullptr(nullptr);
|
|
|
|
|
return;
|
|
|
|
|
}
|
|
|
|
|
EXPECT_EQ(buffer_.Length(), frame->size());
|
|
|
|
|
if (buffer_.Length() != frame->size()) {
|
|
|
|
|
DoOnCompleteFrameFailLength(frame.get());
|
|
|
|
|
return;
|
|
|
|
|
}
|
2018-11-28 16:31:29 +01:00
|
|
|
if (frame->size() != buffer_.Length() ||
|
2019-01-11 10:21:35 +01:00
|
|
|
memcmp(buffer_.Data(), frame->data(), buffer_.Length()) != 0) {
|
2017-01-24 04:38:27 -08:00
|
|
|
DoOnCompleteFrameFailBitstream(frame.get());
|
|
|
|
|
return;
|
|
|
|
|
}
|
|
|
|
|
DoOnCompleteFrame(frame.get());
|
|
|
|
|
}
|
|
|
|
|
void AppendExpectedBitstream(const uint8_t data[], size_t size_in_bytes) {
|
|
|
|
|
// TODO(Johan): Let rtc::ByteBuffer handle uint8_t* instead of char*.
|
|
|
|
|
buffer_.WriteBytes(reinterpret_cast<const char*>(data), size_in_bytes);
|
|
|
|
|
}
|
|
|
|
|
rtc::ByteBufferWriter buffer_;
|
|
|
|
|
};
|
|
|
|
|
|
2017-08-02 07:39:07 -07:00
|
|
|
class MockRtpPacketSink : public RtpPacketSinkInterface {
|
|
|
|
|
public:
|
|
|
|
|
MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&));
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
constexpr uint32_t kSsrc = 111;
|
|
|
|
|
constexpr uint16_t kSequenceNumber = 222;
|
|
|
|
|
std::unique_ptr<RtpPacketReceived> CreateRtpPacketReceived(
|
|
|
|
|
uint32_t ssrc = kSsrc,
|
|
|
|
|
uint16_t sequence_number = kSequenceNumber) {
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
auto packet = absl::make_unique<RtpPacketReceived>();
|
2017-08-02 07:39:07 -07:00
|
|
|
packet->SetSsrc(ssrc);
|
|
|
|
|
packet->SetSequenceNumber(sequence_number);
|
|
|
|
|
return packet;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
MATCHER_P(SamePacketAs, other, "") {
|
|
|
|
|
return arg.Ssrc() == other.Ssrc() &&
|
|
|
|
|
arg.SequenceNumber() == other.SequenceNumber();
|
|
|
|
|
}
|
|
|
|
|
|
2017-01-24 04:38:27 -08:00
|
|
|
} // namespace
|
|
|
|
|
|
2017-06-09 04:01:55 -07:00
|
|
|
class RtpVideoStreamReceiverTest : public testing::Test {
|
2017-01-24 04:38:27 -08:00
|
|
|
public:
|
2017-11-02 14:28:06 +01:00
|
|
|
RtpVideoStreamReceiverTest() : RtpVideoStreamReceiverTest("") {}
|
|
|
|
|
explicit RtpVideoStreamReceiverTest(std::string field_trials)
|
|
|
|
|
: override_field_trials_(field_trials),
|
|
|
|
|
config_(CreateConfig()),
|
2017-01-24 04:38:27 -08:00
|
|
|
process_thread_(ProcessThread::Create("TestThread")) {}
|
|
|
|
|
|
|
|
|
|
void SetUp() {
|
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
|
|
|
rtp_receive_statistics_ =
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
absl::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
|
|
|
|
|
rtp_video_stream_receiver_ = absl::make_unique<RtpVideoStreamReceiver>(
|
2017-04-18 23:38:35 -07:00
|
|
|
&mock_transport_, nullptr, &packet_router_, &config_,
|
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
|
|
|
rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
|
2018-02-28 11:29:47 +01:00
|
|
|
&mock_nack_sender_, &mock_key_frame_request_sender_,
|
2018-10-17 17:27:25 -07:00
|
|
|
&mock_on_complete_frame_callback_, nullptr);
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
WebRtcRTPHeader GetDefaultPacket() {
|
2018-08-02 14:03:53 +02:00
|
|
|
WebRtcRTPHeader packet = {};
|
2018-07-02 14:41:58 +02:00
|
|
|
packet.video_header().codec = kVideoCodecH264;
|
2018-08-02 14:03:53 +02:00
|
|
|
packet.video_header().video_type_header.emplace<RTPVideoHeaderH264>();
|
2017-01-24 04:38:27 -08:00
|
|
|
return packet;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
// TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
|
|
|
|
|
// code.
|
2017-06-21 07:22:40 -07:00
|
|
|
void AddSps(WebRtcRTPHeader* packet,
|
|
|
|
|
uint8_t sps_id,
|
|
|
|
|
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);
|
2018-08-02 14:03:53 +02:00
|
|
|
auto& h264 =
|
|
|
|
|
absl::get<RTPVideoHeaderH264>(packet->video_header().video_type_header);
|
|
|
|
|
h264.nalus[h264.nalus_length++] = info;
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
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);
|
2018-08-02 14:03:53 +02:00
|
|
|
auto& h264 =
|
|
|
|
|
absl::get<RTPVideoHeaderH264>(packet->video_header().video_type_header);
|
|
|
|
|
h264.nalus[h264.nalus_length++] = info;
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
void AddIdr(WebRtcRTPHeader* packet, int pps_id) {
|
|
|
|
|
NaluInfo info;
|
|
|
|
|
info.type = H264::NaluType::kIdr;
|
|
|
|
|
info.sps_id = -1;
|
|
|
|
|
info.pps_id = pps_id;
|
2018-08-02 14:03:53 +02:00
|
|
|
auto& h264 =
|
|
|
|
|
absl::get<RTPVideoHeaderH264>(packet->video_header().video_type_header);
|
|
|
|
|
h264.nalus[h264.nalus_length++] = info;
|
2017-01-24 04:38:27 -08:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
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_;
|
|
|
|
|
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) {
|
2018-08-02 14:03:53 +02:00
|
|
|
WebRtcRTPHeader rtp_header = {};
|
2017-01-24 04:38:27 -08:00
|
|
|
const std::vector<uint8_t> data({1, 2, 3, 4});
|
|
|
|
|
rtp_header.header.sequenceNumber = 1;
|
2018-07-02 14:41:58 +02:00
|
|
|
rtp_header.video_header().is_first_packet_in_frame = true;
|
2018-09-13 11:07:48 +02:00
|
|
|
rtp_header.video_header().is_last_packet_in_frame = true;
|
2017-01-24 04:38:27 -08:00
|
|
|
rtp_header.frameType = kVideoFrameKey;
|
2018-07-02 14:41:58 +02:00
|
|
|
rtp_header.video_header().codec = kVideoCodecGeneric;
|
2017-01-24 04:38:27 -08:00
|
|
|
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
|
|
|
}
|
|
|
|
|
|
2018-01-19 14:41:41 +01:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, NoInfiniteRecursionOnEncapsulatedRedPacket) {
|
|
|
|
|
const uint8_t kRedPayloadType = 125;
|
|
|
|
|
VideoCodec codec;
|
|
|
|
|
codec.plType = kRedPayloadType;
|
|
|
|
|
rtp_video_stream_receiver_->AddReceiveCodec(codec, {});
|
|
|
|
|
const std::vector<uint8_t> data({
|
|
|
|
|
0x80, // RTP version.
|
|
|
|
|
kRedPayloadType, // Payload type.
|
|
|
|
|
0, 0, 0, 0, 0, 0, // Don't care.
|
|
|
|
|
0, 0, 0x4, 0x57, // SSRC
|
|
|
|
|
kRedPayloadType, // RED header.
|
|
|
|
|
0, 0, 0, 0, 0 // Don't care.
|
|
|
|
|
});
|
|
|
|
|
RtpPacketReceived packet;
|
|
|
|
|
EXPECT_TRUE(packet.Parse(data.data(), data.size()));
|
|
|
|
|
rtp_video_stream_receiver_->StartReceive();
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(packet);
|
|
|
|
|
}
|
|
|
|
|
|
2018-07-11 17:50:41 +02:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest,
|
|
|
|
|
DropsPacketWithRedPayloadTypeAndEmptyPayload) {
|
|
|
|
|
const uint8_t kRedPayloadType = 125;
|
|
|
|
|
config_.rtp.red_payload_type = kRedPayloadType;
|
|
|
|
|
SetUp(); // re-create rtp_video_stream_receiver with red payload type.
|
|
|
|
|
// clang-format off
|
|
|
|
|
const uint8_t data[] = {
|
|
|
|
|
0x80, // RTP version.
|
|
|
|
|
kRedPayloadType, // Payload type.
|
|
|
|
|
0, 0, 0, 0, 0, 0, // Don't care.
|
|
|
|
|
0, 0, 0x4, 0x57, // SSRC
|
|
|
|
|
// Empty rtp payload.
|
|
|
|
|
};
|
|
|
|
|
// clang-format on
|
|
|
|
|
RtpPacketReceived packet;
|
|
|
|
|
// Manually convert to CopyOnWriteBuffer to be sure capacity == size
|
|
|
|
|
// and asan bot can catch read buffer overflow.
|
|
|
|
|
EXPECT_TRUE(packet.Parse(rtc::CopyOnWriteBuffer(data)));
|
|
|
|
|
rtp_video_stream_receiver_->StartReceive();
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(packet);
|
|
|
|
|
// Expect asan doesn't find anything.
|
|
|
|
|
}
|
|
|
|
|
|
2017-06-09 04:01:55 -07:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrameBitstreamError) {
|
2018-08-02 14:03:53 +02:00
|
|
|
WebRtcRTPHeader rtp_header = {};
|
2017-01-24 04:38:27 -08:00
|
|
|
const std::vector<uint8_t> data({1, 2, 3, 4});
|
|
|
|
|
rtp_header.header.sequenceNumber = 1;
|
2018-07-02 14:41:58 +02:00
|
|
|
rtp_header.video_header().is_first_packet_in_frame = true;
|
2018-09-13 11:07:48 +02:00
|
|
|
rtp_header.video_header().is_last_packet_in_frame = true;
|
2017-01-24 04:38:27 -08:00
|
|
|
rtp_header.frameType = kVideoFrameKey;
|
2018-07-02 14:41:58 +02:00
|
|
|
rtp_header.video_header().codec = kVideoCodecGeneric;
|
2017-01-24 04:38:27 -08:00
|
|
|
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()) {}
|
|
|
|
|
};
|
|
|
|
|
|
2019-01-31 12:20:57 +01:00
|
|
|
INSTANTIATE_TEST_SUITE_P(
|
2017-11-02 14:28:06 +01:00
|
|
|
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;
|
2018-07-02 14:41:58 +02:00
|
|
|
sps_packet.video_header().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;
|
2018-07-02 14:41:58 +02:00
|
|
|
pps_packet.video_header().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.header.sequenceNumber = 2;
|
2018-09-13 11:07:48 +02:00
|
|
|
idr_packet.video_header().is_first_packet_in_frame = true;
|
|
|
|
|
idr_packet.video_header().is_last_packet_in_frame = true;
|
2017-01-24 04:38:27 -08:00
|
|
|
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;
|
2018-07-02 14:41:58 +02:00
|
|
|
idr_packet.video_header().is_first_packet_in_frame = true;
|
2017-01-24 04:38:27 -08:00
|
|
|
idr_packet.header.sequenceNumber = 2;
|
2018-07-02 14:41:58 +02:00
|
|
|
idr_packet.video_header().is_first_packet_in_frame = true;
|
2018-09-13 11:07:48 +02:00
|
|
|
idr_packet.video_header().is_last_packet_in_frame = true;
|
2017-01-24 04:38:27 -08:00
|
|
|
idr_packet.frameType = kVideoFrameKey;
|
2018-07-02 14:41:58 +02:00
|
|
|
idr_packet.video_header().codec = kVideoCodecH264;
|
2017-01-24 04:38:27 -08:00
|
|
|
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;
|
2018-07-02 14:41:58 +02:00
|
|
|
header.video_header().is_first_packet_in_frame = true;
|
2018-09-13 11:07:48 +02:00
|
|
|
header.video_header().is_last_packet_in_frame = true;
|
2017-03-21 05:45:18 -07:00
|
|
|
header.header.sequenceNumber = 2;
|
|
|
|
|
header.frameType = kVideoFrameKey;
|
2018-07-02 14:41:58 +02:00
|
|
|
header.video_header().codec = kVideoCodecGeneric;
|
2017-03-21 05:45:18 -07:00
|
|
|
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
|
|
|
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
|
|
|
|
header.header.sequenceNumber = 5;
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(nullptr, 0, &header);
|
2017-03-21 05:45:18 -07:00
|
|
|
}
|
|
|
|
|
|
2017-06-09 04:01:55 -07:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, RequestKeyframeIfFirstFrameIsDelta) {
|
2018-08-02 14:03:53 +02:00
|
|
|
WebRtcRTPHeader rtp_header = {};
|
2017-05-16 08:06:30 -07:00
|
|
|
const std::vector<uint8_t> data({1, 2, 3, 4});
|
|
|
|
|
rtp_header.header.sequenceNumber = 1;
|
2018-07-02 14:41:58 +02:00
|
|
|
rtp_header.video_header().is_first_packet_in_frame = true;
|
2018-09-13 11:07:48 +02:00
|
|
|
rtp_header.video_header().is_last_packet_in_frame = true;
|
2017-05-16 08:06:30 -07:00
|
|
|
rtp_header.frameType = kVideoFrameDelta;
|
2018-07-02 14:41:58 +02:00
|
|
|
rtp_header.video_header().codec = kVideoCodecGeneric;
|
2017-05-16 08:06:30 -07:00
|
|
|
|
|
|
|
|
EXPECT_CALL(mock_key_frame_request_sender_, RequestKeyFrame());
|
2017-06-09 04:01:55 -07:00
|
|
|
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
|
|
|
|
&rtp_header);
|
2017-05-16 08:06:30 -07:00
|
|
|
}
|
|
|
|
|
|
2017-08-02 07:39:07 -07:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, SecondarySinksGetRtpNotifications) {
|
|
|
|
|
rtp_video_stream_receiver_->StartReceive();
|
|
|
|
|
|
|
|
|
|
MockRtpPacketSink secondary_sink_1;
|
|
|
|
|
MockRtpPacketSink secondary_sink_2;
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink_1);
|
|
|
|
|
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink_2);
|
|
|
|
|
|
|
|
|
|
auto rtp_packet = CreateRtpPacketReceived();
|
|
|
|
|
EXPECT_CALL(secondary_sink_1, OnRtpPacket(SamePacketAs(*rtp_packet)));
|
|
|
|
|
EXPECT_CALL(secondary_sink_2, OnRtpPacket(SamePacketAs(*rtp_packet)));
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
|
|
|
|
|
|
|
|
|
|
// Test tear-down.
|
|
|
|
|
rtp_video_stream_receiver_->StopReceive();
|
|
|
|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink_1);
|
|
|
|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink_2);
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
TEST_F(RtpVideoStreamReceiverTest, RemovedSecondarySinksGetNoRtpNotifications) {
|
|
|
|
|
rtp_video_stream_receiver_->StartReceive();
|
|
|
|
|
|
|
|
|
|
MockRtpPacketSink secondary_sink;
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
|
|
|
|
|
rtp_video_stream_receiver_->RemoveSecondarySink(&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_->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);
|
|
|
|
|
}
|
|
|
|
|
|
2018-09-25 12:54:37 +02:00
|
|
|
TEST_F(RtpVideoStreamReceiverTest, ParseGenericDescriptorOnePacket) {
|
|
|
|
|
const std::vector<uint8_t> data = {0, 1, 2, 3, 4};
|
|
|
|
|
const int kPayloadType = 123;
|
|
|
|
|
const int kSpatialIndex = 1;
|
|
|
|
|
|
|
|
|
|
VideoCodec codec;
|
|
|
|
|
codec.plType = kPayloadType;
|
|
|
|
|
rtp_video_stream_receiver_->AddReceiveCodec(codec, {});
|
|
|
|
|
rtp_video_stream_receiver_->StartReceive();
|
|
|
|
|
|
|
|
|
|
RtpHeaderExtensionMap extension_map;
|
|
|
|
|
extension_map.Register<RtpGenericFrameDescriptorExtension>(5);
|
|
|
|
|
RtpPacketReceived rtp_packet(&extension_map);
|
|
|
|
|
|
|
|
|
|
RtpGenericFrameDescriptor generic_descriptor;
|
|
|
|
|
generic_descriptor.SetFirstPacketInSubFrame(true);
|
|
|
|
|
generic_descriptor.SetLastPacketInSubFrame(true);
|
|
|
|
|
generic_descriptor.SetFrameId(100);
|
|
|
|
|
generic_descriptor.SetSpatialLayersBitmask(1 << kSpatialIndex);
|
|
|
|
|
generic_descriptor.AddFrameDependencyDiff(90);
|
|
|
|
|
generic_descriptor.AddFrameDependencyDiff(80);
|
|
|
|
|
EXPECT_TRUE(rtp_packet.SetExtension<RtpGenericFrameDescriptorExtension>(
|
|
|
|
|
generic_descriptor));
|
|
|
|
|
|
|
|
|
|
uint8_t* payload = rtp_packet.SetPayloadSize(data.size());
|
|
|
|
|
memcpy(payload, data.data(), data.size());
|
|
|
|
|
// The first byte is the header, so we ignore the first byte of |data|.
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data() + 1,
|
|
|
|
|
data.size() - 1);
|
|
|
|
|
|
|
|
|
|
rtp_packet.SetMarker(true);
|
|
|
|
|
rtp_packet.SetPayloadType(kPayloadType);
|
|
|
|
|
rtp_packet.SetSequenceNumber(1);
|
|
|
|
|
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
|
|
|
|
|
.WillOnce(Invoke([kSpatialIndex](video_coding::EncodedFrame* frame) {
|
|
|
|
|
EXPECT_EQ(frame->num_references, 2U);
|
|
|
|
|
EXPECT_EQ(frame->references[0], frame->id.picture_id - 90);
|
|
|
|
|
EXPECT_EQ(frame->references[1], frame->id.picture_id - 80);
|
|
|
|
|
EXPECT_EQ(frame->id.spatial_layer, kSpatialIndex);
|
|
|
|
|
}));
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(rtp_packet);
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
TEST_F(RtpVideoStreamReceiverTest, ParseGenericDescriptorTwoPackets) {
|
|
|
|
|
const std::vector<uint8_t> data = {0, 1, 2, 3, 4};
|
|
|
|
|
const int kPayloadType = 123;
|
|
|
|
|
const int kSpatialIndex = 1;
|
|
|
|
|
|
|
|
|
|
VideoCodec codec;
|
|
|
|
|
codec.plType = kPayloadType;
|
|
|
|
|
rtp_video_stream_receiver_->AddReceiveCodec(codec, {});
|
|
|
|
|
rtp_video_stream_receiver_->StartReceive();
|
|
|
|
|
|
|
|
|
|
RtpHeaderExtensionMap extension_map;
|
|
|
|
|
extension_map.Register<RtpGenericFrameDescriptorExtension>(5);
|
|
|
|
|
RtpPacketReceived first_packet(&extension_map);
|
|
|
|
|
|
|
|
|
|
RtpGenericFrameDescriptor first_packet_descriptor;
|
|
|
|
|
first_packet_descriptor.SetFirstPacketInSubFrame(true);
|
|
|
|
|
first_packet_descriptor.SetLastPacketInSubFrame(false);
|
|
|
|
|
first_packet_descriptor.SetFrameId(100);
|
|
|
|
|
first_packet_descriptor.SetSpatialLayersBitmask(1 << kSpatialIndex);
|
2018-10-17 14:36:08 +02:00
|
|
|
first_packet_descriptor.SetResolution(480, 360);
|
2018-09-25 12:54:37 +02:00
|
|
|
EXPECT_TRUE(first_packet.SetExtension<RtpGenericFrameDescriptorExtension>(
|
|
|
|
|
first_packet_descriptor));
|
|
|
|
|
|
|
|
|
|
uint8_t* first_packet_payload = first_packet.SetPayloadSize(data.size());
|
|
|
|
|
memcpy(first_packet_payload, data.data(), data.size());
|
|
|
|
|
// The first byte is the header, so we ignore the first byte of |data|.
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data() + 1,
|
|
|
|
|
data.size() - 1);
|
|
|
|
|
|
|
|
|
|
first_packet.SetPayloadType(kPayloadType);
|
|
|
|
|
first_packet.SetSequenceNumber(1);
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(first_packet);
|
|
|
|
|
|
|
|
|
|
RtpPacketReceived second_packet(&extension_map);
|
|
|
|
|
RtpGenericFrameDescriptor second_packet_descriptor;
|
|
|
|
|
second_packet_descriptor.SetFirstPacketInSubFrame(false);
|
|
|
|
|
second_packet_descriptor.SetLastPacketInSubFrame(true);
|
|
|
|
|
EXPECT_TRUE(second_packet.SetExtension<RtpGenericFrameDescriptorExtension>(
|
|
|
|
|
second_packet_descriptor));
|
|
|
|
|
|
|
|
|
|
second_packet.SetMarker(true);
|
|
|
|
|
second_packet.SetPayloadType(kPayloadType);
|
|
|
|
|
second_packet.SetSequenceNumber(2);
|
|
|
|
|
|
|
|
|
|
uint8_t* second_packet_payload = second_packet.SetPayloadSize(data.size());
|
|
|
|
|
memcpy(second_packet_payload, data.data(), data.size());
|
|
|
|
|
// The first byte is the header, so we ignore the first byte of |data|.
|
|
|
|
|
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data() + 1,
|
|
|
|
|
data.size() - 1);
|
|
|
|
|
|
|
|
|
|
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
|
|
|
|
|
.WillOnce(Invoke([kSpatialIndex](video_coding::EncodedFrame* frame) {
|
2018-10-17 14:36:08 +02:00
|
|
|
EXPECT_EQ(frame->num_references, 0U);
|
2018-09-25 12:54:37 +02:00
|
|
|
EXPECT_EQ(frame->id.spatial_layer, kSpatialIndex);
|
2018-10-17 14:36:08 +02:00
|
|
|
EXPECT_EQ(frame->EncodedImage()._encodedWidth, 480u);
|
|
|
|
|
EXPECT_EQ(frame->EncodedImage()._encodedHeight, 360u);
|
2018-09-25 12:54:37 +02:00
|
|
|
}));
|
|
|
|
|
|
|
|
|
|
rtp_video_stream_receiver_->OnRtpPacket(second_packet);
|
|
|
|
|
}
|
|
|
|
|
|
2017-08-02 07:39:07 -07:00
|
|
|
#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
|
|
|
|
|
|
2018-11-16 12:35:26 +01:00
|
|
|
// Initialization of WebRtcRTPHeader is a bit convoluted, with some fields
|
|
|
|
|
// zero-initialized. RtpVideoStreamReceiver depends on proper default values for
|
|
|
|
|
// the playout delay.
|
|
|
|
|
TEST(WebRtcRTPHeader, DefaultPlayoutDelayIsUnspecified) {
|
|
|
|
|
WebRtcRTPHeader webrtc_rtp_header = {};
|
|
|
|
|
EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.min_ms, -1);
|
|
|
|
|
EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.max_ms, -1);
|
|
|
|
|
}
|
|
|
|
|
|
2017-01-24 04:38:27 -08:00
|
|
|
} // namespace webrtc
|