blob: 297218205f2ccaaa25ee6423ae0da1bd30e78a9f [file] [log] [blame]
/*
* 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.
*/
#include "test/gtest.h"
#include "test/gmock.h"
#include "common_video/h264/h264_common.h"
#include "media/base/mediaconstants.h"
#include "modules/pacing/packet_router.h"
#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"
#include "modules/video_coding/timing.h"
#include "rtc_base/bytebuffer.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial_default.h"
#include "test/field_trial.h"
#include "video/rtp_video_stream_receiver.h"
using testing::_;
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) {}
MOCK_METHOD1(DoOnCompleteFrame, void(video_coding::FrameObject* frame));
MOCK_METHOD1(DoOnCompleteFrameFailNullptr,
void(video_coding::FrameObject* frame));
MOCK_METHOD1(DoOnCompleteFrameFailLength,
void(video_coding::FrameObject* frame));
MOCK_METHOD1(DoOnCompleteFrameFailBitstream,
void(video_coding::FrameObject* frame));
void OnCompleteFrame(std::unique_ptr<video_coding::FrameObject> frame) {
if (!frame) {
DoOnCompleteFrameFailNullptr(nullptr);
return;
}
EXPECT_EQ(buffer_.Length(), frame->size());
if (buffer_.Length() != frame->size()) {
DoOnCompleteFrameFailLength(frame.get());
return;
}
std::vector<uint8_t> actual_data(frame->size());
frame->GetBitstream(actual_data.data());
if (memcmp(buffer_.Data(), actual_data.data(), buffer_.Length()) != 0) {
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_;
};
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) {
auto packet = rtc::MakeUnique<RtpPacketReceived>();
packet->SetSsrc(ssrc);
packet->SetSequenceNumber(sequence_number);
return packet;
}
MATCHER_P(SamePacketAs, other, "") {
return arg.Ssrc() == other.Ssrc() &&
arg.SequenceNumber() == other.SequenceNumber();
}
} // namespace
class RtpVideoStreamReceiverTest : public testing::Test {
public:
RtpVideoStreamReceiverTest() : RtpVideoStreamReceiverTest("") {}
explicit RtpVideoStreamReceiverTest(std::string field_trials)
: override_field_trials_(field_trials),
config_(CreateConfig()),
timing_(Clock::GetRealTimeClock()),
process_thread_(ProcessThread::Create("TestThread")) {}
void SetUp() {
rtp_receive_statistics_ =
rtc::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
rtp_video_stream_receiver_ = rtc::MakeUnique<RtpVideoStreamReceiver>(
&mock_transport_, nullptr, &packet_router_, &config_,
rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
&mock_nack_sender_,
&mock_key_frame_request_sender_, &mock_on_complete_frame_callback_,
&timing_);
}
WebRtcRTPHeader GetDefaultPacket() {
WebRtcRTPHeader packet;
memset(&packet, 0, sizeof(packet));
packet.type.Video.codec = kRtpVideoH264;
return packet;
}
// TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
// code.
void AddSps(WebRtcRTPHeader* packet,
uint8_t sps_id,
std::vector<uint8_t>* data) {
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,
uint8_t sps_id,
uint8_t pps_id,
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;
}
const webrtc::test::ScopedFieldTrials override_field_trials_;
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_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_;
};
TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) {
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(_));
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
&rtp_header);
}
TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrameBitstreamError) {
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(_));
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
&rtp_header);
}
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) {
std::vector<uint8_t> sps_data;
WebRtcRTPHeader sps_packet = GetDefaultPacket();
AddSps(&sps_packet, 0, &sps_data);
sps_packet.header.sequenceNumber = 0;
sps_packet.type.Video.is_first_packet_in_frame = true;
mock_on_complete_frame_callback_.AppendExpectedBitstream(
kH264StartCode, sizeof(kH264StartCode));
mock_on_complete_frame_callback_.AppendExpectedBitstream(sps_data.data(),
sps_data.size());
rtp_video_stream_receiver_->OnReceivedPayloadData(
sps_data.data(), sps_data.size(), &sps_packet);
std::vector<uint8_t> pps_data;
WebRtcRTPHeader pps_packet = GetDefaultPacket();
AddPps(&pps_packet, 0, 1, &pps_data);
pps_packet.header.sequenceNumber = 1;
pps_packet.type.Video.is_first_packet_in_frame = true;
mock_on_complete_frame_callback_.AppendExpectedBitstream(
kH264StartCode, sizeof(kH264StartCode));
mock_on_complete_frame_callback_.AppendExpectedBitstream(pps_data.data(),
pps_data.size());
rtp_video_stream_receiver_->OnReceivedPayloadData(
pps_data.data(), pps_data.size(), &pps_packet);
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(_));
rtp_video_stream_receiver_->OnReceivedPayloadData(
idr_data.data(), idr_data.size(), &idr_packet);
}
TEST_P(RtpVideoStreamReceiverTestH264, OutOfBandFmtpSpsPps) {
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=="});
rtp_video_stream_receiver_->AddReceiveCodec(codec, codec_params);
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(_));
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
&idr_packet);
}
TEST_F(RtpVideoStreamReceiverTest, PaddingInMediaStream) {
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(_));
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
&header);
header.header.sequenceNumber = 3;
rtp_video_stream_receiver_->OnReceivedPayloadData(nullptr, 0, &header);
header.frameType = kVideoFrameDelta;
header.header.sequenceNumber = 4;
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
&header);
header.header.sequenceNumber = 6;
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
&header);
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
header.header.sequenceNumber = 5;
rtp_video_stream_receiver_->OnReceivedPayloadData(nullptr, 0, &header);
}
TEST_F(RtpVideoStreamReceiverTest, RequestKeyframeIfFirstFrameIsDelta) {
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 = kVideoFrameDelta;
rtp_header.type.Video.codec = kRtpVideoGeneric;
EXPECT_CALL(mock_key_frame_request_sender_, RequestKeyFrame());
rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
&rtp_header);
}
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);
}
#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
} // namespace webrtc