blob: 63acb80b8d98bafa7a9366ea224369dd1eead1b4 [file] [log] [blame]
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <algorithm> // max
#include <memory>
#include <vector>
#include "absl/algorithm/container.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/encoded_image.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video_codecs/video_encoder.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/rtp_transport_controller_send.h"
#include "call/simulated_network.h"
#include "call/video_send_stream.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "modules/video_coding/codecs/vp9/include/vp9.h"
#include "rtc_base/checks.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/event.h"
#include "rtc_base/experiments/alr_experiment.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/synchronization/sequence_checker.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/unique_id_generator.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/configurable_frame_size_encoder.h"
#include "test/fake_encoder.h"
#include "test/fake_texture_frame.h"
#include "test/field_trial.h"
#include "test/frame_forwarder.h"
#include "test/frame_generator_capturer.h"
#include "test/frame_utils.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/null_transport.h"
#include "test/rtcp_packet_parser.h"
#include "test/rtp_header_parser.h"
#include "test/testsupport/perf_test.h"
#include "test/video_encoder_proxy_factory.h"
#include "video/send_statistics_proxy.h"
#include "video/transport_adapter.h"
#include "video/video_send_stream.h"
namespace webrtc {
namespace test {
class VideoSendStreamPeer {
public:
explicit VideoSendStreamPeer(webrtc::VideoSendStream* base_class_stream)
: internal_stream_(
static_cast<internal::VideoSendStream*>(base_class_stream)) {}
absl::optional<float> GetPacingFactorOverride() const {
return internal_stream_->GetPacingFactorOverride();
}
private:
internal::VideoSendStream const* const internal_stream_;
};
} // namespace test
namespace {
enum : int { // The first valid value is 1.
kAbsSendTimeExtensionId = 1,
kTimestampOffsetExtensionId,
kTransportSequenceNumberExtensionId,
kVideoContentTypeExtensionId,
kVideoRotationExtensionId,
kVideoTimingExtensionId,
};
constexpr int64_t kRtcpIntervalMs = 1000;
enum VideoFormat {
kGeneric,
kVP8,
};
VideoFrame CreateVideoFrame(int width, int height, int64_t timestamp_ms) {
return webrtc::VideoFrame::Builder()
.set_video_frame_buffer(I420Buffer::Create(width, height))
.set_rotation(webrtc::kVideoRotation_0)
.set_timestamp_ms(timestamp_ms)
.build();
}
} // namespace
class VideoSendStreamTest : public test::CallTest {
public:
VideoSendStreamTest() {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
}
protected:
void TestNackRetransmission(uint32_t retransmit_ssrc,
uint8_t retransmit_payload_type);
void TestPacketFragmentationSize(VideoFormat format, bool with_fec);
void TestVp9NonFlexMode(uint8_t num_temporal_layers,
uint8_t num_spatial_layers);
void TestRequestSourceRotateVideo(bool support_orientation_ext);
};
TEST_F(VideoSendStreamTest, CanStartStartedStream) {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
CreateSenderCall();
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateVideoStreams();
GetVideoSendStream()->Start();
GetVideoSendStream()->Start();
DestroyStreams();
DestroyCalls();
});
}
TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
CreateSenderCall();
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateVideoStreams();
GetVideoSendStream()->Stop();
GetVideoSendStream()->Stop();
DestroyStreams();
DestroyCalls();
});
}
TEST_F(VideoSendStreamTest, SupportsCName) {
static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
class CNameObserver : public test::SendTest {
public:
CNameObserver() : SendTest(kDefaultTimeoutMs) {}
private:
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
if (parser.sdes()->num_packets() > 0) {
EXPECT_EQ(1u, parser.sdes()->chunks().size());
EXPECT_EQ(kCName, parser.sdes()->chunks()[0].cname);
observation_complete_.Set();
}
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.c_name = kCName;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP with CNAME.";
}
} test;
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
class AbsoluteSendTimeObserver : public test::SendTest {
public:
AbsoluteSendTimeObserver() : SendTest(kDefaultTimeoutMs) {
extensions_.Register<AbsoluteSendTime>(kAbsSendTimeExtensionId);
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
uint32_t abs_send_time = 0;
EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>());
EXPECT_TRUE(rtp_packet.GetExtension<AbsoluteSendTime>(&abs_send_time));
if (abs_send_time != 0) {
// Wait for at least one packet with a non-zero send time. The send time
// is a 16-bit value derived from the system clock, and it is valid
// for a packet to have a zero send time. To tell that from an
// unpopulated value we'll wait for a packet with non-zero send time.
observation_complete_.Set();
} else {
RTC_LOG(LS_WARNING)
<< "Got a packet with zero absoluteSendTime, waiting"
" for another packet...";
}
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
}
private:
RtpHeaderExtensionMap extensions_;
} test;
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
static const int kEncodeDelayMs = 5;
class TransmissionTimeOffsetObserver : public test::SendTest {
public:
TransmissionTimeOffsetObserver()
: SendTest(kDefaultTimeoutMs), encoder_factory_([]() {
return std::make_unique<test::DelayedEncoder>(
Clock::GetRealTimeClock(), kEncodeDelayMs);
}) {
extensions_.Register<TransmissionOffset>(kTimestampOffsetExtensionId);
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
int32_t toffset = 0;
EXPECT_TRUE(rtp_packet.GetExtension<TransmissionOffset>(&toffset));
EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>());
EXPECT_GT(toffset, 0);
observation_complete_.Set();
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
}
test::FunctionVideoEncoderFactory encoder_factory_;
RtpHeaderExtensionMap extensions_;
} test;
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
class TransportWideSequenceNumberObserver : public test::SendTest {
public:
TransportWideSequenceNumberObserver()
: SendTest(kDefaultTimeoutMs), encoder_factory_([]() {
return std::make_unique<test::FakeEncoder>(
Clock::GetRealTimeClock());
}) {
extensions_.Register<TransportSequenceNumber>(kExtensionId);
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
EXPECT_TRUE(rtp_packet.HasExtension<TransportSequenceNumber>());
EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>());
EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>());
observation_complete_.Set();
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
}
test::FunctionVideoEncoderFactory encoder_factory_;
RtpHeaderExtensionMap extensions_;
} test;
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsVideoRotation) {
class VideoRotationObserver : public test::SendTest {
public:
VideoRotationObserver() : SendTest(kDefaultTimeoutMs) {
extensions_.Register<VideoOrientation>(kVideoRotationExtensionId);
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
// Only the last packet of the frame is required to have the extension.
if (!rtp_packet.Marker())
return SEND_PACKET;
EXPECT_EQ(rtp_packet.GetExtension<VideoOrientation>(), kVideoRotation_90);
observation_complete_.Set();
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
}
private:
RtpHeaderExtensionMap extensions_;
} test;
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsVideoContentType) {
class VideoContentTypeObserver : public test::SendTest {
public:
VideoContentTypeObserver()
: SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
extensions_.Register<VideoContentTypeExtension>(
kVideoContentTypeExtensionId);
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
// Only the last packet of the key-frame must have extension.
if (!rtp_packet.Marker() || first_frame_sent_)
return SEND_PACKET;
// First marker bit seen means that the first frame is sent.
first_frame_sent_ = true;
VideoContentType type;
EXPECT_TRUE(rtp_packet.GetExtension<VideoContentTypeExtension>(&type));
EXPECT_TRUE(videocontenttypehelpers::IsScreenshare(type));
observation_complete_.Set();
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
}
private:
bool first_frame_sent_;
RtpHeaderExtensionMap extensions_;
} test;
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsVideoTimingFrames) {
class VideoTimingObserver : public test::SendTest {
public:
VideoTimingObserver()
: SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
extensions_.Register<VideoTimingExtension>(kVideoTimingExtensionId);
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
// Only the last packet of the frame must have extension.
// Also don't check packets of the second frame if they happen to get
// through before the test terminates.
if (!rtp_packet.Marker() || first_frame_sent_)
return SEND_PACKET;
EXPECT_TRUE(rtp_packet.HasExtension<VideoTimingExtension>());
observation_complete_.Set();
first_frame_sent_ = true;
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kVideoTimingUri, kVideoTimingExtensionId));
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for timing frames.";
}
private:
RtpHeaderExtensionMap extensions_;
bool first_frame_sent_;
} test;
RunBaseTest(&test);
}
class FakeReceiveStatistics : public ReceiveStatisticsProvider {
public:
FakeReceiveStatistics(uint32_t send_ssrc,
uint32_t last_sequence_number,
uint32_t cumulative_lost,
uint8_t fraction_lost) {
stat_.SetMediaSsrc(send_ssrc);
stat_.SetExtHighestSeqNum(last_sequence_number);
stat_.SetCumulativeLost(cumulative_lost);
stat_.SetFractionLost(fraction_lost);
}
std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override {
EXPECT_GE(max_blocks, 1u);
return {stat_};
}
private:
rtcp::ReportBlock stat_;
};
class UlpfecObserver : public test::EndToEndTest {
public:
// Some of the test cases are expected to time out.
// Use a shorter timeout window than the default one for those.
static constexpr int kReducedTimeoutMs = 10000;
UlpfecObserver(bool header_extensions_enabled,
bool use_nack,
bool expect_red,
bool expect_ulpfec,
const std::string& codec,
VideoEncoderFactory* encoder_factory)
: EndToEndTest(expect_ulpfec ? VideoSendStreamTest::kDefaultTimeoutMs
: kReducedTimeoutMs),
encoder_factory_(encoder_factory),
payload_name_(codec),
use_nack_(use_nack),
expect_red_(expect_red),
expect_ulpfec_(expect_ulpfec),
sent_media_(false),
sent_ulpfec_(false),
header_extensions_enabled_(header_extensions_enabled) {
extensions_.Register<AbsoluteSendTime>(kAbsSendTimeExtensionId);
extensions_.Register<TransportSequenceNumber>(
kTransportSequenceNumberExtensionId);
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
int encapsulated_payload_type = -1;
if (rtp_packet.PayloadType() == VideoSendStreamTest::kRedPayloadType) {
EXPECT_TRUE(expect_red_);
encapsulated_payload_type = rtp_packet.payload()[0];
if (encapsulated_payload_type !=
VideoSendStreamTest::kFakeVideoSendPayloadType) {
EXPECT_EQ(VideoSendStreamTest::kUlpfecPayloadType,
encapsulated_payload_type);
}
} else {
EXPECT_EQ(VideoSendStreamTest::kFakeVideoSendPayloadType,
rtp_packet.PayloadType());
if (rtp_packet.payload_size() > 0) {
// Not padding-only, media received outside of RED.
EXPECT_FALSE(expect_red_);
sent_media_ = true;
}
}
if (header_extensions_enabled_) {
uint32_t abs_send_time;
EXPECT_TRUE(rtp_packet.GetExtension<AbsoluteSendTime>(&abs_send_time));
uint16_t transport_seq_num;
EXPECT_TRUE(
rtp_packet.GetExtension<TransportSequenceNumber>(&transport_seq_num));
if (!first_packet_) {
uint32_t kHalf24BitsSpace = 0xFFFFFF / 2;
if (abs_send_time <= kHalf24BitsSpace &&
prev_abs_send_time_ > kHalf24BitsSpace) {
// 24 bits wrap.
EXPECT_GT(prev_abs_send_time_, abs_send_time);
} else {
EXPECT_GE(abs_send_time, prev_abs_send_time_);
}
uint16_t seq_num_diff = transport_seq_num - prev_transport_seq_num_;
EXPECT_EQ(1, seq_num_diff);
}
first_packet_ = false;
prev_abs_send_time_ = abs_send_time;
prev_transport_seq_num_ = transport_seq_num;
}
if (encapsulated_payload_type != -1) {
if (encapsulated_payload_type ==
VideoSendStreamTest::kUlpfecPayloadType) {
EXPECT_TRUE(expect_ulpfec_);
sent_ulpfec_ = true;
} else {
sent_media_ = true;
}
}
if (sent_media_ && sent_ulpfec_) {
observation_complete_.Set();
}
return SEND_PACKET;
}
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) override {
// At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
// Configure some network delay.
const int kNetworkDelayMs = 100;
BuiltInNetworkBehaviorConfig config;
config.loss_percent = 5;
config.queue_delay_ms = kNetworkDelayMs;
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
VideoSendStreamTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(config)));
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (use_nack_) {
send_config->rtp.nack.rtp_history_ms =
(*receive_configs)[0].rtp.nack.rtp_history_ms =
VideoSendStreamTest::kNackRtpHistoryMs;
}
send_config->encoder_settings.encoder_factory = encoder_factory_;
send_config->rtp.payload_name = payload_name_;
send_config->rtp.ulpfec.red_payload_type =
VideoSendStreamTest::kRedPayloadType;
send_config->rtp.ulpfec.ulpfec_payload_type =
VideoSendStreamTest::kUlpfecPayloadType;
if (!header_extensions_enabled_) {
send_config->rtp.extensions.clear();
} else {
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
}
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
(*receive_configs)[0].rtp.red_payload_type =
send_config->rtp.ulpfec.red_payload_type;
(*receive_configs)[0].rtp.ulpfec_payload_type =
send_config->rtp.ulpfec.ulpfec_payload_type;
}
void PerformTest() override {
EXPECT_EQ(expect_ulpfec_, Wait())
<< "Timed out waiting for ULPFEC and/or media packets.";
}
VideoEncoderFactory* encoder_factory_;
RtpHeaderExtensionMap extensions_;
const std::string payload_name_;
const bool use_nack_;
const bool expect_red_;
const bool expect_ulpfec_;
bool sent_media_;
bool sent_ulpfec_;
const bool header_extensions_enabled_;
bool first_packet_ = true;
uint32_t prev_abs_send_time_ = 0;
uint16_t prev_transport_seq_num_ = 0;
};
TEST_F(VideoSendStreamTest, SupportsUlpfecWithExtensions) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
UlpfecObserver test(true, false, true, true, "VP8", &encoder_factory);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsUlpfecWithoutExtensions) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
UlpfecObserver test(false, false, true, true, "VP8", &encoder_factory);
RunBaseTest(&test);
}
class VideoSendStreamWithoutUlpfecTest : public test::CallTest {
protected:
VideoSendStreamWithoutUlpfecTest()
: field_trial_("WebRTC-DisableUlpFecExperiment/Enabled/") {}
test::ScopedFieldTrials field_trial_;
};
TEST_F(VideoSendStreamWithoutUlpfecTest, NoUlpfecIfDisabledThroughFieldTrial) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
UlpfecObserver test(false, false, false, false, "VP8", &encoder_factory);
RunBaseTest(&test);
}
// The FEC scheme used is not efficient for H264, so we should not use RED/FEC
// since we'll still have to re-request FEC packets, effectively wasting
// bandwidth since the receiver has to wait for FEC retransmissions to determine
// that the received state is actually decodable.
TEST_F(VideoSendStreamTest, DoesNotUtilizeUlpfecForH264WithNackEnabled) {
test::FunctionVideoEncoderFactory encoder_factory([]() {
return std::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
});
UlpfecObserver test(false, true, false, false, "H264", &encoder_factory);
RunBaseTest(&test);
}
// Without retransmissions FEC for H264 is fine.
TEST_F(VideoSendStreamTest, DoesUtilizeUlpfecForH264WithoutNackEnabled) {
test::FunctionVideoEncoderFactory encoder_factory([]() {
return std::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
});
UlpfecObserver test(false, false, true, true, "H264", &encoder_factory);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, DoesUtilizeUlpfecForVp8WithNackEnabled) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
UlpfecObserver test(false, true, true, true, "VP8", &encoder_factory);
RunBaseTest(&test);
}
#if defined(RTC_ENABLE_VP9)
TEST_F(VideoSendStreamTest, DoesUtilizeUlpfecForVp9WithNackEnabled) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP9Encoder::Create(); });
UlpfecObserver test(false, true, true, true, "VP9", &encoder_factory);
RunBaseTest(&test);
}
#endif // defined(RTC_ENABLE_VP9)
TEST_F(VideoSendStreamTest, SupportsUlpfecWithMultithreadedH264) {
std::unique_ptr<TaskQueueFactory> task_queue_factory =
CreateDefaultTaskQueueFactory();
test::FunctionVideoEncoderFactory encoder_factory([&]() {
return std::make_unique<test::MultithreadedFakeH264Encoder>(
Clock::GetRealTimeClock(), task_queue_factory.get());
});
UlpfecObserver test(false, false, true, true, "H264", &encoder_factory);
RunBaseTest(&test);
}
// TODO(brandtr): Move these FlexFEC tests when we have created
// FlexfecSendStream.
class FlexfecObserver : public test::EndToEndTest {
public:
FlexfecObserver(bool header_extensions_enabled,
bool use_nack,
const std::string& codec,
VideoEncoderFactory* encoder_factory,
size_t num_video_streams)
: EndToEndTest(VideoSendStreamTest::kDefaultTimeoutMs),
encoder_factory_(encoder_factory),
payload_name_(codec),
use_nack_(use_nack),
sent_media_(false),
sent_flexfec_(false),
header_extensions_enabled_(header_extensions_enabled),
num_video_streams_(num_video_streams) {
extensions_.Register<AbsoluteSendTime>(kAbsSendTimeExtensionId);
extensions_.Register<TransmissionOffset>(kTimestampOffsetExtensionId);
extensions_.Register<TransportSequenceNumber>(
kTransportSequenceNumberExtensionId);
}
size_t GetNumFlexfecStreams() const override { return 1; }
size_t GetNumVideoStreams() const override { return num_video_streams_; }
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
if (rtp_packet.PayloadType() == VideoSendStreamTest::kFlexfecPayloadType) {
EXPECT_EQ(VideoSendStreamTest::kFlexfecSendSsrc, rtp_packet.Ssrc());
sent_flexfec_ = true;
} else {
EXPECT_EQ(VideoSendStreamTest::kFakeVideoSendPayloadType,
rtp_packet.PayloadType());
EXPECT_THAT(::testing::make_tuple(VideoSendStreamTest::kVideoSendSsrcs,
num_video_streams_),
::testing::Contains(rtp_packet.Ssrc()));
sent_media_ = true;
}
if (header_extensions_enabled_) {
EXPECT_TRUE(rtp_packet.HasExtension<AbsoluteSendTime>());
EXPECT_TRUE(rtp_packet.HasExtension<TransmissionOffset>());
EXPECT_TRUE(rtp_packet.HasExtension<TransportSequenceNumber>());
}
if (sent_media_ && sent_flexfec_) {
observation_complete_.Set();
}
return SEND_PACKET;
}
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) override {
// At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
// Therefore we need some network delay.
const int kNetworkDelayMs = 100;
BuiltInNetworkBehaviorConfig config;
config.loss_percent = 5;
config.queue_delay_ms = kNetworkDelayMs;
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
VideoSendStreamTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(config)));
}
std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
TaskQueueBase* task_queue) override {
// We need the RTT to be >200 ms to send FEC and the network delay for the
// send transport is 100 ms, so add 100 ms (but no loss) on the return link.
BuiltInNetworkBehaviorConfig config;
config.loss_percent = 0;
config.queue_delay_ms = 100;
return std::make_unique<test::PacketTransport>(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
VideoSendStreamTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(config)));
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (use_nack_) {
send_config->rtp.nack.rtp_history_ms =
(*receive_configs)[0].rtp.nack.rtp_history_ms =
VideoSendStreamTest::kNackRtpHistoryMs;
}
send_config->encoder_settings.encoder_factory = encoder_factory_;
send_config->rtp.payload_name = payload_name_;
if (header_extensions_enabled_) {
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
} else {
send_config->rtp.extensions.clear();
}
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out waiting for FlexFEC and/or media packets.";
}
VideoEncoderFactory* encoder_factory_;
RtpHeaderExtensionMap extensions_;
const std::string payload_name_;
const bool use_nack_;
bool sent_media_;
bool sent_flexfec_;
const bool header_extensions_enabled_;
const size_t num_video_streams_;
};
TEST_F(VideoSendStreamTest, SupportsFlexfecVp8) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
FlexfecObserver test(false, false, "VP8", &encoder_factory, 1);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsFlexfecSimulcastVp8) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
FlexfecObserver test(false, false, "VP8", &encoder_factory, 2);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackVp8) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
FlexfecObserver test(false, true, "VP8", &encoder_factory, 1);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsFlexfecWithRtpExtensionsVp8) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
FlexfecObserver test(true, false, "VP8", &encoder_factory, 1);
RunBaseTest(&test);
}
#if defined(RTC_ENABLE_VP9)
TEST_F(VideoSendStreamTest, SupportsFlexfecVp9) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP9Encoder::Create(); });
FlexfecObserver test(false, false, "VP9", &encoder_factory, 1);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackVp9) {
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP9Encoder::Create(); });
FlexfecObserver test(false, true, "VP9", &encoder_factory, 1);
RunBaseTest(&test);
}
#endif // defined(RTC_ENABLE_VP9)
TEST_F(VideoSendStreamTest, SupportsFlexfecH264) {
test::FunctionVideoEncoderFactory encoder_factory([]() {
return std::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
});
FlexfecObserver test(false, false, "H264", &encoder_factory, 1);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackH264) {
test::FunctionVideoEncoderFactory encoder_factory([]() {
return std::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
});
FlexfecObserver test(false, true, "H264", &encoder_factory, 1);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, SupportsFlexfecWithMultithreadedH264) {
std::unique_ptr<TaskQueueFactory> task_queue_factory =
CreateDefaultTaskQueueFactory();
test::FunctionVideoEncoderFactory encoder_factory([&]() {
return std::make_unique<test::MultithreadedFakeH264Encoder>(
Clock::GetRealTimeClock(), task_queue_factory.get());
});
FlexfecObserver test(false, false, "H264", &encoder_factory, 1);
RunBaseTest(&test);
}
void VideoSendStreamTest::TestNackRetransmission(
uint32_t retransmit_ssrc,
uint8_t retransmit_payload_type) {
class NackObserver : public test::SendTest {
public:
explicit NackObserver(uint32_t retransmit_ssrc,
uint8_t retransmit_payload_type)
: SendTest(kDefaultTimeoutMs),
send_count_(0),
retransmit_count_(0),
retransmit_ssrc_(retransmit_ssrc),
retransmit_payload_type_(retransmit_payload_type) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
// NACK packets two times at some arbitrary points.
const int kNackedPacketsAtOnceCount = 3;
const int kRetransmitTarget = kNackedPacketsAtOnceCount * 2;
// Skip padding packets because they will never be retransmitted.
if (rtp_packet.payload_size() == 0) {
return SEND_PACKET;
}
++send_count_;
// NACK packets at arbitrary points.
if (send_count_ == 5 || send_count_ == 25) {
nacked_sequence_numbers_.insert(
nacked_sequence_numbers_.end(),
non_padding_sequence_numbers_.end() - kNackedPacketsAtOnceCount,
non_padding_sequence_numbers_.end());
RtpRtcp::Configuration config;
config.clock = Clock::GetRealTimeClock();
config.outgoing_transport = transport_adapter_.get();
config.rtcp_report_interval_ms = kRtcpIntervalMs;
config.local_media_ssrc = kReceiverLocalVideoSsrc;
RTCPSender rtcp_sender(config);
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
EXPECT_EQ(0, rtcp_sender.SendRTCP(
feedback_state, kRtcpNack,
static_cast<int>(nacked_sequence_numbers_.size()),
&nacked_sequence_numbers_.front()));
}
uint16_t sequence_number = rtp_packet.SequenceNumber();
if (rtp_packet.Ssrc() == retransmit_ssrc_ &&
retransmit_ssrc_ != kVideoSendSsrcs[0]) {
// Not kVideoSendSsrcs[0], assume correct RTX packet. Extract sequence
// number.
const uint8_t* rtx_header = rtp_packet.payload().data();
sequence_number = (rtx_header[0] << 8) + rtx_header[1];
}
auto found = absl::c_find(nacked_sequence_numbers_, sequence_number);
if (found != nacked_sequence_numbers_.end()) {
nacked_sequence_numbers_.erase(found);
if (++retransmit_count_ == kRetransmitTarget) {
EXPECT_EQ(retransmit_ssrc_, rtp_packet.Ssrc());
EXPECT_EQ(retransmit_payload_type_, rtp_packet.PayloadType());
observation_complete_.Set();
}
} else {
non_padding_sequence_numbers_.push_back(sequence_number);
}
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
transport_adapter_.reset(
new internal::TransportAdapter(send_config->send_transport));
transport_adapter_->Enable();
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.rtx.payload_type = retransmit_payload_type_;
if (retransmit_ssrc_ != kVideoSendSsrcs[0])
send_config->rtp.rtx.ssrcs.push_back(retransmit_ssrc_);
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for NACK retransmission.";
}
std::unique_ptr<internal::TransportAdapter> transport_adapter_;
int send_count_;
int retransmit_count_;
const uint32_t retransmit_ssrc_;
const uint8_t retransmit_payload_type_;
std::vector<uint16_t> nacked_sequence_numbers_;
std::vector<uint16_t> non_padding_sequence_numbers_;
} test(retransmit_ssrc, retransmit_payload_type);
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, RetransmitsNack) {
// Normal NACKs should use the send SSRC.
TestNackRetransmission(kVideoSendSsrcs[0], kFakeVideoSendPayloadType);
}
TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
// NACKs over RTX should use a separate SSRC.
TestNackRetransmission(kSendRtxSsrcs[0], kSendRtxPayloadType);
}
void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
bool with_fec) {
// Use a fake encoder to output a frame of every size in the range [90, 290],
// for each size making sure that the exact number of payload bytes received
// is correct and that packets are fragmented to respect max packet size.
static const size_t kMaxPacketSize = 128;
static const size_t start = 90;
static const size_t stop = 290;
// Observer that verifies that the expected number of packets and bytes
// arrive for each frame size, from start_size to stop_size.
class FrameFragmentationTest : public test::SendTest {
public:
FrameFragmentationTest(size_t max_packet_size,
size_t start_size,
size_t stop_size,
bool test_generic_packetization,
bool use_fec)
: SendTest(kLongTimeoutMs),
encoder_(stop),
encoder_factory_(&encoder_),
max_packet_size_(max_packet_size),
stop_size_(stop_size),
test_generic_packetization_(test_generic_packetization),
use_fec_(use_fec),
packet_count_(0),
packets_lost_(0),
last_packet_count_(0),
last_packets_lost_(0),
accumulated_size_(0),
accumulated_payload_(0),
fec_packet_received_(false),
current_size_rtp_(start_size),
current_size_frame_(static_cast<int>(start_size)) {
// Fragmentation required, this test doesn't make sense without it.
encoder_.SetFrameSize(start_size);
RTC_DCHECK_GT(stop_size, max_packet_size);
if (!test_generic_packetization_)
encoder_.SetCodecType(kVideoCodecVP8);
}
private:
Action OnSendRtp(const uint8_t* packet, size_t size) override {
size_t length = size;
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
EXPECT_LE(length, max_packet_size_);
if (use_fec_ && rtp_packet.payload_size() > 0) {
uint8_t payload_type = rtp_packet.payload()[0];
bool is_fec = rtp_packet.PayloadType() == kRedPayloadType &&
payload_type == kUlpfecPayloadType;
if (is_fec) {
fec_packet_received_ = true;
return SEND_PACKET;
}
}
accumulated_size_ += length;
if (use_fec_)
TriggerLossReport(rtp_packet);
if (test_generic_packetization_) {
size_t overhead = rtp_packet.headers_size() + rtp_packet.padding_size();
// Only remove payload header and RED header if the packet actually
// contains payload.
if (length > overhead) {
overhead += (1 /* Generic header */);
if (use_fec_)
overhead += 1; // RED for FEC header.
}
EXPECT_GE(length, overhead);
accumulated_payload_ += length - overhead;
}
// Marker bit set indicates last packet of a frame.
if (rtp_packet.Marker()) {
if (use_fec_ && accumulated_payload_ == current_size_rtp_ - 1) {
// With FEC enabled, frame size is incremented asynchronously, so
// "old" frames one byte too small may arrive. Accept, but don't
// increase expected frame size.
accumulated_size_ = 0;
accumulated_payload_ = 0;
return SEND_PACKET;
}
EXPECT_GE(accumulated_size_, current_size_rtp_);
if (test_generic_packetization_) {
EXPECT_EQ(current_size_rtp_, accumulated_payload_);
}
// Last packet of frame; reset counters.
accumulated_size_ = 0;
accumulated_payload_ = 0;
if (current_size_rtp_ == stop_size_) {
// Done! (Don't increase size again, might arrive more @ stop_size).
observation_complete_.Set();
} else {
// Increase next expected frame size. If testing with FEC, make sure
// a FEC packet has been received for this frame size before
// proceeding, to make sure that redundancy packets don't exceed
// size limit.
if (!use_fec_) {
++current_size_rtp_;
} else if (fec_packet_received_) {
fec_packet_received_ = false;
++current_size_rtp_;
rtc::CritScope lock(&mutex_);
++current_size_frame_;
}
}
}
return SEND_PACKET;
}
void TriggerLossReport(const RtpPacket& rtp_packet) {
// Send lossy receive reports to trigger FEC enabling.
const int kLossPercent = 5;
if (++packet_count_ % (100 / kLossPercent) == 0) {
packets_lost_++;
int loss_delta = packets_lost_ - last_packets_lost_;
int packets_delta = packet_count_ - last_packet_count_;
last_packet_count_ = packet_count_;
last_packets_lost_ = packets_lost_;
uint8_t loss_ratio =
static_cast<uint8_t>(loss_delta * 255 / packets_delta);
FakeReceiveStatistics lossy_receive_stats(
kVideoSendSsrcs[0], rtp_packet.SequenceNumber(),
packets_lost_, // Cumulative lost.
loss_ratio); // Loss percent.
RtpRtcp::Configuration config;
config.clock = Clock::GetRealTimeClock();
config.receive_statistics = &lossy_receive_stats;
config.outgoing_transport = transport_adapter_.get();
config.rtcp_report_interval_ms = kRtcpIntervalMs;
config.local_media_ssrc = kVideoSendSsrcs[0];
RTCPSender rtcp_sender(config);
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
}
}
void UpdateConfiguration() {
rtc::CritScope lock(&mutex_);
// Increase frame size for next encoded frame, in the context of the
// encoder thread.
if (!use_fec_ && current_size_frame_ < static_cast<int32_t>(stop_size_)) {
++current_size_frame_;
}
encoder_.SetFrameSize(static_cast<size_t>(current_size_frame_));
}
void ModifySenderBitrateConfig(
BitrateConstraints* bitrate_config) override {
const int kMinBitrateBps = 300000;
bitrate_config->min_bitrate_bps = kMinBitrateBps;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
transport_adapter_.reset(
new internal::TransportAdapter(send_config->send_transport));
transport_adapter_->Enable();
if (use_fec_) {
send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
}
if (!test_generic_packetization_)
send_config->rtp.payload_name = "VP8";
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->rtp.max_packet_size = kMaxPacketSize;
encoder_.RegisterPostEncodeCallback([this]() { UpdateConfiguration(); });
// Make sure there is at least one extension header, to make the RTP
// header larger than the base length of 12 bytes.
EXPECT_FALSE(send_config->rtp.extensions.empty());
// Setup screen content disables frame dropping which makes this easier.
EXPECT_EQ(1u, encoder_config->simulcast_layers.size());
encoder_config->simulcast_layers[0].num_temporal_layers = 2;
encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while observing incoming RTP packets.";
}
std::unique_ptr<internal::TransportAdapter> transport_adapter_;
test::ConfigurableFrameSizeEncoder encoder_;
test::VideoEncoderProxyFactory encoder_factory_;
const size_t max_packet_size_;
const size_t stop_size_;
const bool test_generic_packetization_;
const bool use_fec_;
uint32_t packet_count_;
uint32_t packets_lost_;
uint32_t last_packet_count_;
uint32_t last_packets_lost_;
size_t accumulated_size_;
size_t accumulated_payload_;
bool fec_packet_received_;
size_t current_size_rtp_;
rtc::CriticalSection mutex_;
int current_size_frame_ RTC_GUARDED_BY(mutex_);
};
// Don't auto increment if FEC is used; continue sending frame size until
// a FEC packet has been received.
FrameFragmentationTest test(kMaxPacketSize, start, stop, format == kGeneric,
with_fec);
RunBaseTest(&test);
}
// TODO(sprang): Is there any way of speeding up these tests?
TEST_F(VideoSendStreamTest, FragmentsGenericAccordingToMaxPacketSize) {
TestPacketFragmentationSize(kGeneric, false);
}
TEST_F(VideoSendStreamTest, FragmentsGenericAccordingToMaxPacketSizeWithFec) {
TestPacketFragmentationSize(kGeneric, true);
}
TEST_F(VideoSendStreamTest, FragmentsVp8AccordingToMaxPacketSize) {
TestPacketFragmentationSize(kVP8, false);
}
TEST_F(VideoSendStreamTest, FragmentsVp8AccordingToMaxPacketSizeWithFec) {
TestPacketFragmentationSize(kVP8, true);
}
// The test will go through a number of phases.
// 1. Start sending packets.
// 2. As soon as the RTP stream has been detected, signal a low REMB value to
// suspend the stream.
// 3. Wait until |kSuspendTimeFrames| have been captured without seeing any RTP
// packets.
// 4. Signal a high REMB and then wait for the RTP stream to start again.
// When the stream is detected again, and the stats show that the stream
// is no longer suspended, the test ends.
TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
static const int kSuspendTimeFrames = 60; // Suspend for 2 seconds @ 30 fps.
class RembObserver : public test::SendTest {
public:
class CaptureObserver : public rtc::VideoSinkInterface<VideoFrame> {
public:
explicit CaptureObserver(RembObserver* remb_observer)
: remb_observer_(remb_observer) {}
void OnFrame(const VideoFrame&) {
rtc::CritScope lock(&remb_observer_->crit_);
if (remb_observer_->test_state_ == kDuringSuspend &&
++remb_observer_->suspended_frame_count_ > kSuspendTimeFrames) {
VideoSendStream::Stats stats = remb_observer_->stream_->GetStats();
EXPECT_TRUE(stats.suspended);
remb_observer_->SendRtcpFeedback(remb_observer_->high_remb_bps_);
remb_observer_->test_state_ = kWaitingForPacket;
}
}
private:
RembObserver* const remb_observer_;
};
RembObserver()
: SendTest(kDefaultTimeoutMs),
clock_(Clock::GetRealTimeClock()),
capture_observer_(this),
stream_(nullptr),
test_state_(kBeforeSuspend),
rtp_count_(0),
last_sequence_number_(0),
suspended_frame_count_(0),
low_remb_bps_(0),
high_remb_bps_(0) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
++rtp_count_;
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
last_sequence_number_ = rtp_packet.SequenceNumber();
if (test_state_ == kBeforeSuspend) {
// The stream has started. Try to suspend it.
SendRtcpFeedback(low_remb_bps_);
test_state_ = kDuringSuspend;
} else if (test_state_ == kDuringSuspend) {
if (rtp_packet.padding_size() == 0) {
// Received non-padding packet during suspension period. Reset the
// counter.
suspended_frame_count_ = 0;
}
SendRtcpFeedback(0); // REMB is only sent if value is > 0.
} else if (test_state_ == kWaitingForPacket) {
if (rtp_packet.padding_size() == 0) {
// Non-padding packet observed. Test is almost complete. Will just
// have to wait for the stats to change.
test_state_ = kWaitingForStats;
}
SendRtcpFeedback(0); // REMB is only sent if value is > 0.
} else if (test_state_ == kWaitingForStats) {
VideoSendStream::Stats stats = stream_->GetStats();
if (stats.suspended == false) {
// Stats flipped to false. Test is complete.
observation_complete_.Set();
}
SendRtcpFeedback(0); // REMB is only sent if value is > 0.
}
return SEND_PACKET;
}
void set_low_remb_bps(int value) {
rtc::CritScope lock(&crit_);
low_remb_bps_ = value;
}
void set_high_remb_bps(int value) {
rtc::CritScope lock(&crit_);
high_remb_bps_ = value;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
stream_ = send_stream;
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_capturer->AddOrUpdateSink(&capture_observer_,
rtc::VideoSinkWants());
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
transport_adapter_.reset(
new internal::TransportAdapter(send_config->send_transport));
transport_adapter_->Enable();
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->suspend_below_min_bitrate = true;
int min_bitrate_bps =
test::DefaultVideoStreamFactory::kDefaultMinBitratePerStream[0];
set_low_remb_bps(min_bitrate_bps - 10000);
int threshold_window = std::max(min_bitrate_bps / 10, 20000);
ASSERT_GT(encoder_config->max_bitrate_bps,
min_bitrate_bps + threshold_window + 5000);
set_high_remb_bps(min_bitrate_bps + threshold_window + 5000);
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out during suspend-below-min-bitrate test.";
}
enum TestState {
kBeforeSuspend,
kDuringSuspend,
kWaitingForPacket,
kWaitingForStats
};
virtual void SendRtcpFeedback(int remb_value)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0],
last_sequence_number_, rtp_count_, 0);
RtpRtcp::Configuration config;
config.clock = clock_;
config.receive_statistics = &receive_stats;
config.outgoing_transport = transport_adapter_.get();
config.rtcp_report_interval_ms = kRtcpIntervalMs;
config.local_media_ssrc = kVideoSendSsrcs[0];
RTCPSender rtcp_sender(config);
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
if (remb_value > 0) {
rtcp_sender.SetRemb(remb_value, std::vector<uint32_t>());
}
RTCPSender::FeedbackState feedback_state;
EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
}
std::unique_ptr<internal::TransportAdapter> transport_adapter_;
Clock* const clock_;
CaptureObserver capture_observer_;
VideoSendStream* stream_;
rtc::CriticalSection crit_;
TestState test_state_ RTC_GUARDED_BY(crit_);
int rtp_count_ RTC_GUARDED_BY(crit_);
int last_sequence_number_ RTC_GUARDED_BY(crit_);
int suspended_frame_count_ RTC_GUARDED_BY(crit_);
int low_remb_bps_ RTC_GUARDED_BY(crit_);
int high_remb_bps_ RTC_GUARDED_BY(crit_);
} test;
RunBaseTest(&test);
}
// This test that padding stops being send after a while if the Camera stops
// producing video frames and that padding resumes if the camera restarts.
TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
class NoPaddingWhenVideoIsMuted : public test::SendTest {
public:
NoPaddingWhenVideoIsMuted()
: SendTest(kDefaultTimeoutMs),
clock_(Clock::GetRealTimeClock()),
capturer_(nullptr) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
last_packet_time_ms_ = clock_->TimeInMilliseconds();
RtpPacket rtp_packet;
rtp_packet.Parse(packet, length);
const bool only_padding = rtp_packet.payload_size() == 0;
if (test_state_ == kBeforeStopCapture) {
// Packets are flowing, stop camera.
capturer_->Stop();
test_state_ = kWaitingForPadding;
} else if (test_state_ == kWaitingForPadding && only_padding) {
// We're still getting padding, after stopping camera.
test_state_ = kWaitingForNoPackets;
} else if (test_state_ == kWaitingForMediaAfterCameraRestart &&
!only_padding) {
// Media packets are flowing again, stop camera a second time.
capturer_->Stop();
test_state_ = kWaitingForPaddingAfterCameraStopsAgain;
} else if (test_state_ == kWaitingForPaddingAfterCameraStopsAgain &&
only_padding) {
// Padding is still flowing, test ok.
observation_complete_.Set();
}
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
const int kNoPacketsThresholdMs = 2000;
if (test_state_ == kWaitingForNoPackets &&
(last_packet_time_ms_ &&
clock_->TimeInMilliseconds() - last_packet_time_ms_.value() >
kNoPacketsThresholdMs)) {
// No packets seen for |kNoPacketsThresholdMs|, restart camera.
capturer_->Start();
test_state_ = kWaitingForMediaAfterCameraRestart;
}
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Make sure padding is sent if encoder is not producing media.
encoder_config->min_transmit_bitrate_bps = 50000;
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
rtc::CritScope lock(&crit_);
capturer_ = frame_generator_capturer;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for RTP packets to stop being sent.";
}
enum TestState {
kBeforeStopCapture,
kWaitingForPadding,
kWaitingForNoPackets,
kWaitingForMediaAfterCameraRestart,
kWaitingForPaddingAfterCameraStopsAgain
};
TestState test_state_ = kBeforeStopCapture;
Clock* const clock_;
rtc::CriticalSection crit_;
absl::optional<int64_t> last_packet_time_ms_ RTC_GUARDED_BY(crit_);
test::FrameGeneratorCapturer* capturer_ RTC_GUARDED_BY(crit_);
} test;
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) {
const int kCapacityKbps = 10000; // 10 Mbps
class PaddingIsPrimarilyRetransmissions : public test::EndToEndTest {
public:
PaddingIsPrimarilyRetransmissions()
: EndToEndTest(kDefaultTimeoutMs),
clock_(Clock::GetRealTimeClock()),
padding_length_(0),
total_length_(0),
call_(nullptr) {}
private:
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
call_ = sender_call;
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RtpPacket rtp_packet;
rtp_packet.Parse(packet, length);
padding_length_ += rtp_packet.padding_size();
total_length_ += length;
return SEND_PACKET;
}
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) override {
const int kNetworkDelayMs = 50;
BuiltInNetworkBehaviorConfig config;
config.loss_percent = 10;
config.link_capacity_kbps = kCapacityKbps;
config.queue_delay_ms = kNetworkDelayMs;
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(config)));
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Turn on RTX.
send_config->rtp.rtx.payload_type = kFakeVideoSendPayloadType;
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
}
void PerformTest() override {
// TODO(isheriff): Some platforms do not ramp up as expected to full
// capacity due to packet scheduling delays. Fix that before getting
// rid of this.
SleepMs(5000);
{
rtc::CritScope lock(&crit_);
// Expect padding to be a small percentage of total bytes sent.
EXPECT_LT(padding_length_, .1 * total_length_);
}
}
rtc::CriticalSection crit_;
Clock* const clock_;
size_t padding_length_ RTC_GUARDED_BY(crit_);
size_t total_length_ RTC_GUARDED_BY(crit_);
Call* call_;
} test;
RunBaseTest(&test);
}
// This test first observes "high" bitrate use at which point it sends a REMB to
// indicate that it should be lowered significantly. The test then observes that
// the bitrate observed is sinking well below the min-transmit-bitrate threshold
// to verify that the min-transmit bitrate respects incoming REMB.
//
// Note that the test starts at "high" bitrate and does not ramp up to "higher"
// bitrate since no receiver block or remb is sent in the initial phase.
TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
static const int kMinTransmitBitrateBps = 400000;
static const int kHighBitrateBps = 150000;
static const int kRembBitrateBps = 80000;
static const int kRembRespectedBitrateBps = 100000;
class BitrateObserver : public test::SendTest {
public:
explicit BitrateObserver(TaskQueueBase* task_queue)
: SendTest(kDefaultTimeoutMs),
task_queue_(task_queue),
retranmission_rate_limiter_(Clock::GetRealTimeClock(), 1000),
stream_(nullptr),
bitrate_capped_(false) {}
~BitrateObserver() override {
// Make sure we free |rtp_rtcp_| in the same context as we constructed it.
SendTask(RTC_FROM_HERE, task_queue_, [this]() { rtp_rtcp_ = nullptr; });
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (RtpHeaderParser::IsRtcp(packet, length))
return DROP_PACKET;
RtpPacket rtp_packet;
if (!rtp_packet.Parse(packet, length))
return DROP_PACKET;
RTC_DCHECK(stream_);
VideoSendStream::Stats stats = stream_->GetStats();
if (!stats.substreams.empty()) {
EXPECT_EQ(1u, stats.substreams.size());
int total_bitrate_bps =
stats.substreams.begin()->second.total_bitrate_bps;
test::PrintResult("bitrate_stats_", "min_transmit_bitrate_low_remb",
"bitrate_bps", static_cast<size_t>(total_bitrate_bps),
"bps", false);
if (total_bitrate_bps > kHighBitrateBps) {
rtp_rtcp_->SetRemb(kRembBitrateBps, {rtp_packet.Ssrc()});
rtp_rtcp_->Process();
bitrate_capped_ = true;
} else if (bitrate_capped_ &&
total_bitrate_bps < kRembRespectedBitrateBps) {
observation_complete_.Set();
}
}
// Packets don't have to be delivered since the test is the receiver.
return DROP_PACKET;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
stream_ = send_stream;
RtpRtcp::Configuration config;
config.clock = Clock::GetRealTimeClock();
config.outgoing_transport = feedback_transport_.get();
config.retransmission_rate_limiter = &retranmission_rate_limiter_;
rtp_rtcp_ = RtpRtcp::Create(config);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
feedback_transport_.reset(
new internal::TransportAdapter(send_config->send_transport));
feedback_transport_->Enable();
encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timeout while waiting for low bitrate stats after REMB.";
}
TaskQueueBase* const task_queue_;
std::unique_ptr<RtpRtcp> rtp_rtcp_;
std::unique_ptr<internal::TransportAdapter> feedback_transport_;
RateLimiter retranmission_rate_limiter_;
VideoSendStream* stream_;
bool bitrate_capped_;
} test(task_queue());
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
static const int kStartBitrateBps = 300000;
static const int kNewMaxBitrateBps = 1234567;
static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
class ChangingNetworkRouteTest : public test::EndToEndTest {
public:
explicit ChangingNetworkRouteTest(TaskQueueBase* task_queue)
: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
task_queue_(task_queue),
call_(nullptr) {
module_process_thread_.Detach();
task_queue_thread_.Detach();
extensions_.Register<TransportSequenceNumber>(kExtensionId);
}
~ChangingNetworkRouteTest() {
// Block until all already posted tasks run to avoid 'use after free'
// when such task accesses |this|.
SendTask(RTC_FROM_HERE, task_queue_, [] {});
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
RTC_DCHECK(!call_);
call_ = sender_call;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kExtensionId));
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
(*receive_configs)[0].rtp.transport_cc = true;
}
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kExtensionId));
(*receive_configs)[0].rtp.extensions.clear();
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
(*receive_configs)[0].rtp.transport_cc = true;
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTC_DCHECK_RUN_ON(&module_process_thread_);
task_queue_->PostTask(ToQueuedTask([this]() {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
if (!call_)
return;
Call::Stats stats = call_->GetStats();
if (stats.send_bandwidth_bps > kStartBitrateBps)
observation_complete_.Set();
}));
return SEND_PACKET;
}
void OnStreamsStopped() override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
call_ = nullptr;
}
void PerformTest() override {
rtc::NetworkRoute new_route;
new_route.connected = true;
new_route.local = rtc::RouteEndpoint::CreateWithNetworkId(10);
new_route.remote = rtc::RouteEndpoint::CreateWithNetworkId(20);
BitrateConstraints bitrate_config;
SendTask(RTC_FROM_HERE, task_queue_,
[this, &new_route, &bitrate_config]() {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
call_->GetTransportControllerSend()->OnNetworkRouteChanged(
"transport", new_route);
bitrate_config.start_bitrate_bps = kStartBitrateBps;
call_->GetTransportControllerSend()->SetSdpBitrateParameters(
bitrate_config);
});
EXPECT_TRUE(Wait())
<< "Timed out while waiting for start bitrate to be exceeded.";
SendTask(
RTC_FROM_HERE, task_queue_, [this, &new_route, &bitrate_config]() {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
bitrate_config.start_bitrate_bps = -1;
bitrate_config.max_bitrate_bps = kNewMaxBitrateBps;
call_->GetTransportControllerSend()->SetSdpBitrateParameters(
bitrate_config);
// TODO(holmer): We should set the last sent packet id here and
// verify that we correctly ignore any packet loss reported prior to
// that id.
new_route.local = rtc::RouteEndpoint::CreateWithNetworkId(
new_route.local.network_id() + 1);
call_->GetTransportControllerSend()->OnNetworkRouteChanged(
"transport", new_route);
EXPECT_GE(call_->GetStats().send_bandwidth_bps, kStartBitrateBps);
});
}
private:
webrtc::SequenceChecker module_process_thread_;
webrtc::SequenceChecker task_queue_thread_;
TaskQueueBase* const task_queue_;
RtpHeaderExtensionMap extensions_;
Call* call_ RTC_GUARDED_BY(task_queue_thread_);
} test(task_queue());
RunBaseTest(&test);
}
// Test that if specified, relay cap is lifted on transition to direct
// connection.
TEST_F(VideoSendStreamTest, RelayToDirectRoute) {
static const int kStartBitrateBps = 300000;
static const int kRelayBandwidthCapBps = 800000;
static const int kMinPacketsToSend = 100;
webrtc::test::ScopedFieldTrials field_trials(
std::string(field_trial::GetFieldTrialString()) +
"WebRTC-Bwe-NetworkRouteConstraints/relay_cap:" +
std::to_string(kRelayBandwidthCapBps) + "bps/");
class RelayToDirectRouteTest : public test::EndToEndTest {
public:
explicit RelayToDirectRouteTest(TaskQueueBase* task_queue)
: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
task_queue_(task_queue),
call_(nullptr),
packets_sent_(0),
relayed_phase_(true) {
module_process_thread_.Detach();
task_queue_thread_.Detach();
}
~RelayToDirectRouteTest() {
// Block until all already posted tasks run to avoid 'use after free'
// when such task accesses |this|.
SendTask(RTC_FROM_HERE, task_queue_, [] {});
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
RTC_DCHECK(!call_);
call_ = sender_call;
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTC_DCHECK_RUN_ON(&module_process_thread_);
task_queue_->PostTask(ToQueuedTask([this]() {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
if (!call_)
return;
bool had_time_to_exceed_cap_in_relayed_phase =
relayed_phase_ && ++packets_sent_ > kMinPacketsToSend;
bool did_exceed_cap =
call_->GetStats().send_bandwidth_bps > kRelayBandwidthCapBps;
if (did_exceed_cap || had_time_to_exceed_cap_in_relayed_phase)
observation_complete_.Set();
}));
return SEND_PACKET;
}
void OnStreamsStopped() override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
call_ = nullptr;
}
void PerformTest() override {
rtc::NetworkRoute route;
route.connected = true;
route.local = rtc::RouteEndpoint::CreateWithNetworkId(10);
route.remote = rtc::RouteEndpoint::CreateWithNetworkId(20);
SendTask(RTC_FROM_HERE, task_queue_, [this, &route]() {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
relayed_phase_ = true;
route.remote = route.remote.CreateWithTurn(true);
call_->GetTransportControllerSend()->OnNetworkRouteChanged("transport",
route);
BitrateConstraints bitrate_config;
bitrate_config.start_bitrate_bps = kStartBitrateBps;
call_->GetTransportControllerSend()->SetSdpBitrateParameters(
bitrate_config);
});
EXPECT_TRUE(Wait())
<< "Timeout waiting for sufficient packets sent count.";
SendTask(RTC_FROM_HERE, task_queue_, [this, &route]() {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
EXPECT_LE(call_->GetStats().send_bandwidth_bps, kRelayBandwidthCapBps);
route.remote = route.remote.CreateWithTurn(false);
call_->GetTransportControllerSend()->OnNetworkRouteChanged("transport",
route);
relayed_phase_ = false;
observation_complete_.Reset();
});
EXPECT_TRUE(Wait())
<< "Timeout while waiting for bandwidth to outgrow relay cap.";
}
private:
webrtc::SequenceChecker module_process_thread_;
webrtc::SequenceChecker task_queue_thread_;
TaskQueueBase* const task_queue_;
Call* call_ RTC_GUARDED_BY(task_queue_thread_);
int packets_sent_ RTC_GUARDED_BY(task_queue_thread_);
bool relayed_phase_ RTC_GUARDED_BY(task_queue_thread_);
} test(task_queue());
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, ChangingTransportOverhead) {
class ChangingTransportOverheadTest : public test::EndToEndTest {
public:
explicit ChangingTransportOverheadTest(TaskQueueBase* task_queue)
: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
task_queue_(task_queue),
call_(nullptr),
packets_sent_(0),
transport_overhead_(0) {}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
call_ = sender_call;
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
EXPECT_LE(length, kMaxRtpPacketSize);
rtc::CritScope cs(&lock_);
if (++packets_sent_ < 100)
return SEND_PACKET;
observation_complete_.Set();
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.max_packet_size = kMaxRtpPacketSize;
}
void PerformTest() override {
SendTask(RTC_FROM_HERE, task_queue_, [this]() {
transport_overhead_ = 100;
call_->GetTransportControllerSend()->OnTransportOverheadChanged(
transport_overhead_);
});
EXPECT_TRUE(Wait());
{
rtc::CritScope cs(&lock_);
packets_sent_ = 0;
}
SendTask(RTC_FROM_HERE, task_queue_, [this]() {
transport_overhead_ = 500;
call_->GetTransportControllerSend()->OnTransportOverheadChanged(
transport_overhead_);
});
EXPECT_TRUE(Wait());
}
private:
TaskQueueBase* const task_queue_;
Call* call_;
rtc::CriticalSection lock_;
int packets_sent_ RTC_GUARDED_BY(lock_);
int transport_overhead_;
const size_t kMaxRtpPacketSize = 1000;
} test(task_queue());
RunBaseTest(&test);
}
// Test class takes takes as argument a switch selecting if type switch should
// occur and a function pointer to reset the send stream. This is necessary
// since you cannot change the content type of a VideoSendStream, you need to
// recreate it. Stopping and recreating the stream can only be done on the main
// thread and in the context of VideoSendStreamTest (not BaseTest).
template <typename T>
class MaxPaddingSetTest : public test::SendTest {
public:
static const uint32_t kMinTransmitBitrateBps = 400000;
static const uint32_t kActualEncodeBitrateBps = 40000;
static const uint32_t kMinPacketsToSend = 50;
MaxPaddingSetTest(bool test_switch_content_type,
T* stream_reset_fun,
TaskQueueBase* task_queue)
: SendTest(test::CallTest::kDefaultTimeoutMs),
running_without_padding_(test_switch_content_type),
stream_resetter_(stream_reset_fun),
task_queue_(task_queue) {
RTC_DCHECK(stream_resetter_);
module_process_thread_.Detach();
task_queue_thread_.Detach();
}
~MaxPaddingSetTest() {
// Block until all already posted tasks run to avoid 'use after free'
// when such task accesses |this|.
SendTask(RTC_FROM_HERE, task_queue_, [] {});
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
if (running_without_padding_) {
encoder_config->min_transmit_bitrate_bps = 0;
encoder_config->content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
} else {
encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
}
send_stream_config_ = send_config->Copy();
encoder_config_ = encoder_config->Copy();
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
RTC_DCHECK(task_queue_->IsCurrent());
RTC_DCHECK(!call_);
RTC_DCHECK(sender_call);
call_ = sender_call;
}
// Called on the pacer thread.
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTC_DCHECK_RUN_ON(&module_process_thread_);
// Check the stats on the correct thread and signal the 'complete' flag
// once we detect that we're done.
task_queue_->PostTask(ToQueuedTask([this]() {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
// In case we get a callback during teardown.
// When this happens, OnStreamsStopped() has been called already,
// |call_| is null and the streams are being torn down.
if (!call_)
return;
++packets_sent_;
Call::Stats stats = call_->GetStats();
if (running_without_padding_) {
EXPECT_EQ(0, stats.max_padding_bitrate_bps);
// Wait until at least kMinPacketsToSend frames have been encoded, so
// that we have reliable data.
if (packets_sent_ < kMinPacketsToSend)
return;
// We've sent kMinPacketsToSend packets with default configuration,
// switch to enabling screen content and setting min transmit bitrate.
// Note that we need to recreate the stream if changing content type.
packets_sent_ = 0;
encoder_config_.min_transmit_bitrate_bps = kMinTransmitBitrateBps;
encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
running_without_padding_ = false;
(*stream_resetter_)(send_stream_config_, encoder_config_);
} else {
// Make sure the pacer has been configured with a min transmit bitrate.
if (stats.max_padding_bitrate_bps > 0) {
observation_complete_.Set();
}
}
}));
return SEND_PACKET;
}
// Called on |task_queue_|
void OnStreamsStopped() override {
RTC_DCHECK_RUN_ON(&task_queue_thread_);
RTC_DCHECK(task_queue_->IsCurrent());
call_ = nullptr;
}
void PerformTest() override {
ASSERT_TRUE(Wait()) << "Timed out waiting for a valid padding bitrate.";
}
private:
webrtc::SequenceChecker task_queue_thread_;
Call* call_ RTC_GUARDED_BY(task_queue_thread_) = nullptr;
VideoSendStream::Config send_stream_config_{nullptr};
VideoEncoderConfig encoder_config_;
webrtc::SequenceChecker module_process_thread_;
uint32_t packets_sent_ RTC_GUARDED_BY(task_queue_thread_) = 0;
bool running_without_padding_ RTC_GUARDED_BY(task_queue_thread_);
T* const stream_resetter_;
TaskQueueBase* const task_queue_;
};
TEST_F(VideoSendStreamTest, RespectsMinTransmitBitrate) {
auto reset_fun = [](const VideoSendStream::Config& send_stream_config,
const VideoEncoderConfig& encoder_config) {};
MaxPaddingSetTest<decltype(reset_fun)> test(false, &reset_fun, task_queue());
RunBaseTest(&test);
}
TEST_F(VideoSendStreamTest, RespectsMinTransmitBitrateAfterContentSwitch) {
// Function for removing and recreating the send stream with a new config.
auto reset_fun = [this](const VideoSendStream::Config& send_stream_config,
const VideoEncoderConfig& encoder_config) {
RTC_DCHECK(task_queue()->IsCurrent());
Stop();
DestroyVideoSendStreams();
SetVideoSendConfig(send_stream_config);
SetVideoEncoderConfig(encoder_config);
CreateVideoSendStreams();
SetVideoDegradation(DegradationPreference::MAINTAIN_RESOLUTION);
Start();
};
MaxPaddingSetTest<decltype(reset_fun)> test(true, &reset_fun, task_queue());
RunBaseTest(&test);
}
// This test verifies that new frame sizes reconfigures encoders even though not
// (yet) sending. The purpose of this is to permit encoding as quickly as
// possible once we start sending. Likely the frames being input are from the
// same source that will be sent later, which just means that we're ready
// earlier.
TEST_F(VideoSendStreamTest,
EncoderReconfigureOnResolutionChangeWhenNotSending) {
class EncoderObserver : public test::FakeEncoder {
public:
EncoderObserver()
: FakeEncoder(Clock::GetRealTimeClock()),
number_of_initializations_(0),
last_initialized_frame_width_(0),
last_initialized_frame_height_(0) {}
void WaitForResolution(int width, int height) {
{
rtc::CritScope lock(&crit_);
if (last_initialized_frame_width_ == width &&
last_initialized_frame_height_ == height) {
return;
}
}
EXPECT_TRUE(
init_encode_called_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
{
rtc::CritScope lock(&crit_);
EXPECT_EQ(width, last_initialized_frame_width_);
EXPECT_EQ(height, last_initialized_frame_height_);
}
}
private:
int32_t InitEncode(const VideoCodec* config,
const Settings& settings) override {
rtc::CritScope lock(&crit_);
last_initialized_frame_width_ = config->width;
last_initialized_frame_height_ = config->height;
++number_of_initializations_;
init_encode_called_.Set();
return FakeEncoder::InitEncode(config, settings);
}
int32_t Encode(const VideoFrame& input_image,
const std::vector<VideoFrameType>* frame_types) override {
ADD_FAILURE()
<< "Unexpected Encode call since the send stream is not started";
return 0;
}
rtc::CriticalSection crit_;
rtc::Event init_encode_called_;
size_t number_of_initializations_ RTC_GUARDED_BY(&crit_);
int last_initialized_frame_width_ RTC_GUARDED_BY(&crit_);
int last_initialized_frame_height_ RTC_GUARDED_BY(&crit_);
};
test::NullTransport transport;
EncoderObserver encoder;
test::VideoEncoderProxyFactory encoder_factory(&encoder);
SendTask(RTC_FROM_HERE, task_queue(), [this, &transport, &encoder_factory]() {
CreateSenderCall();
CreateSendConfig(1, 0, 0, &transport);
GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
frame_generator_capturer_->Start();
});
encoder.WaitForResolution(kDefaultWidth, kDefaultHeight);
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
frame_generator_capturer_->ChangeResolution(kDefaultWidth * 2,
kDefaultHeight * 2);
});
encoder.WaitForResolution(kDefaultWidth * 2, kDefaultHeight * 2);
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
DestroyStreams();
DestroyCalls();
});
}
TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
class StartBitrateObserver : public test::FakeEncoder {
public:
StartBitrateObserver()
: FakeEncoder(Clock::GetRealTimeClock()), start_bitrate_kbps_(0) {}
int32_t InitEncode(const VideoCodec* config,
const Settings& settings) override {
rtc::CritScope lock(&crit_);
start_bitrate_kbps_ = config->startBitrate;
start_bitrate_changed_.Set();
return FakeEncoder::InitEncode(config, settings);
}
void SetRates(const RateControlParameters& parameters) override {
rtc::CritScope lock(&crit_);
start_bitrate_kbps_ = parameters.bitrate.get_sum_kbps();
start_bitrate_changed_.Set();
FakeEncoder::SetRates(parameters);
}
int GetStartBitrateKbps() const {
rtc::CritScope lock(&crit_);
return start_bitrate_kbps_;
}
bool WaitForStartBitrate() {
return start_bitrate_changed_.Wait(
VideoSendStreamTest::kDefaultTimeoutMs);
}
private:
rtc::CriticalSection crit_;
rtc::Event start_bitrate_changed_;
int start_bitrate_kbps_ RTC_GUARDED_BY(crit_);
};
CreateSenderCall();
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
BitrateConstraints bitrate_config;
bitrate_config.start_bitrate_bps =
2 * GetVideoEncoderConfig()->max_bitrate_bps;
sender_call_->GetTransportControllerSend()->SetSdpBitrateParameters(
bitrate_config);
StartBitrateObserver encoder;
test::VideoEncoderProxyFactory encoder_factory(&encoder);
// Since this test does not use a capturer, set |internal_source| = true.
// Encoder configuration is otherwise updated on the next video frame.
encoder_factory.SetHasInternalSource(true);
GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
CreateVideoStreams();
EXPECT_TRUE(encoder.WaitForStartBitrate());
EXPECT_EQ(GetVideoEncoderConfig()->max_bitrate_bps / 1000,
encoder.GetStartBitrateKbps());
GetVideoEncoderConfig()->max_bitrate_bps =
2 * bitrate_config.start_bitrate_bps;
GetVideoSendStream()->ReconfigureVideoEncoder(
GetVideoEncoderConfig()->Copy());
// New bitrate should be reconfigured above the previous max. As there's no
// network connection this shouldn't be flaky, as no bitrate should've been
// reported in between.
EXPECT_TRUE(encoder.WaitForStartBitrate());
EXPECT_EQ(bitrate_config.start_bitrate_bps / 1000,
encoder.GetStartBitrateKbps());
DestroyStreams();
}
class StartStopBitrateObserver : public test::FakeEncoder {
public:
StartStopBitrateObserver() : FakeEncoder(Clock::GetRealTimeClock()) {}
int32_t InitEncode(const VideoCodec* config,
const Settings& settings) override {
rtc::CritScope lock(&crit_);
encoder_init_.Set();
return FakeEncoder::InitEncode(config, settings);
}
void SetRates(const RateControlParameters& parameters) override {
rtc::CritScope lock(&crit_);
bitrate_kbps_ = parameters.bitrate.get_sum_kbps();
bitrate_changed_.Set();
FakeEncoder::SetRates(parameters);
}
bool WaitForEncoderInit() {
return encoder_init_.Wait(VideoSendStreamTest::kDefaultTimeoutMs);
}
bool WaitBitrateChanged(bool non_zero) {
do {
absl::optional<int> bitrate_kbps;
{
rtc::CritScope lock(&crit_);
bitrate_kbps = bitrate_kbps_;
}
if (!bitrate_kbps)
continue;
if ((non_zero && *bitrate_kbps > 0) ||
(!non_zero && *bitrate_kbps == 0)) {
return true;
}
} while (bitrate_changed_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
return false;
}
private:
rtc::CriticalSection crit_;
rtc::Event encoder_init_;
rtc::Event bitrate_changed_;
absl::optional<int> bitrate_kbps_ RTC_GUARDED_BY(crit_);
};
// This test that if the encoder use an internal source, VideoEncoder::SetRates
// will be called with zero bitrate during initialization and that
// VideoSendStream::Stop also triggers VideoEncoder::SetRates Start to be called
// with zero bitrate.
TEST_F(VideoSendStreamTest, VideoSendStreamStopSetEncoderRateToZero) {
test::NullTransport transport;
StartStopBitrateObserver encoder;
test::VideoEncoderProxyFactory encoder_factory(&encoder);
encoder_factory.SetHasInternalSource(true);
test::FrameForwarder forwarder;
SendTask(RTC_FROM_HERE, task_queue(),
[this, &transport, &encoder_factory, &forwarder]() {
CreateSenderCall();
CreateSendConfig(1, 0, 0, &transport);
sender_call_->SignalChannelNetworkState(MediaType::VIDEO,
kNetworkUp);
GetVideoSendConfig()->encoder_settings.encoder_factory =
&encoder_factory;
CreateVideoStreams();
// Inject a frame, to force encoder creation.
GetVideoSendStream()->Start();
GetVideoSendStream()->SetSource(&forwarder,
DegradationPreference::DISABLED);
forwarder.IncomingCapturedFrame(CreateVideoFrame(640, 480, 4));
});
EXPECT_TRUE(encoder.WaitForEncoderInit());
SendTask(RTC_FROM_HERE, task_queue(),
[this]() { GetVideoSendStream()->Start(); });
EXPECT_TRUE(encoder.WaitBitrateChanged(true));
SendTask(RTC_FROM_HERE, task_queue(),
[this]() { GetVideoSendStream()->Stop(); });
EXPECT_TRUE(encoder.WaitBitrateChanged(false));
SendTask(RTC_FROM_HERE, task_queue(),
[this]() { GetVideoSendStream()->Start(); });
EXPECT_TRUE(encoder.WaitBitrateChanged(true));
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
DestroyStreams();
DestroyCalls();
});
}
// Tests that when the encoder uses an internal source, the VideoEncoder will
// be updated with a new bitrate when turning the VideoSendStream on/off with
// VideoSendStream::UpdateActiveSimulcastLayers, and when the VideoStreamEncoder
// is reconfigured with new active layers.
TEST_F(VideoSendStreamTest, VideoSendStreamUpdateActiveSimulcastLayers) {
test::NullTransport transport;
StartStopBitrateObserver encoder;
test::VideoEncoderProxyFactory encoder_factory(&encoder);
encoder_factory.SetHasInternalSource(true);
test::FrameForwarder forwarder;
SendTask(RTC_FROM_HERE, task_queue(),
[this, &transport, &encoder_factory, &forwarder]() {
CreateSenderCall();
// Create two simulcast streams.
CreateSendConfig(2, 0, 0, &transport);
sender_call_->SignalChannelNetworkState(MediaType::VIDEO,
kNetworkUp);
GetVideoSendConfig()->encoder_settings.encoder_factory =
&encoder_factory;
CreateVideoStreams();
// Inject a frame, to force encoder creation.
GetVideoSendStream()->Start();
GetVideoSendStream()->SetSource(&forwarder,
DegradationPreference::DISABLED);
forwarder.IncomingCapturedFrame(CreateVideoFrame(640, 480, 4));
});
EXPECT_TRUE(encoder.WaitForEncoderInit());
// When we turn on the simulcast layers it will update the BitrateAllocator,
// which in turn updates the VideoEncoder's bitrate.
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendStream()->UpdateActiveSimulcastLayers({true, true});
});
EXPECT_TRUE(encoder.WaitBitrateChanged(true));
GetVideoEncoderConfig()->simulcast_layers[0].active = true;
GetVideoEncoderConfig()->simulcast_layers[1].active = false;
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendStream()->ReconfigureVideoEncoder(
GetVideoEncoderConfig()->Copy());
});
// TODO(bugs.webrtc.org/8807): Currently we require a hard reconfiguration to
// update the VideoBitrateAllocator and BitrateAllocator of which layers are
// active. Once the change is made for a "soft" reconfiguration we can remove
// the expecation for an encoder init. We can also test that bitrate changes
// when just updating individual active layers, which should change the
// bitrate set to the video encoder.
EXPECT_TRUE(encoder.WaitForEncoderInit());
EXPECT_TRUE(encoder.WaitBitrateChanged(true));
// Turning off both simulcast layers should trigger a bitrate change of 0.
GetVideoEncoderConfig()->simulcast_layers[0].active = false;
GetVideoEncoderConfig()->simulcast_layers[1].active = false;
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendStream()->UpdateActiveSimulcastLayers({false, false});
});
EXPECT_TRUE(encoder.WaitBitrateChanged(false));
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
DestroyStreams();
DestroyCalls();
});
}
TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
class EncoderStateObserver : public test::SendTest, public VideoEncoder {
public:
explicit EncoderStateObserver(TaskQueueBase* task_queue)
: SendTest(kDefaultTimeoutMs),
task_queue_(task_queue),
stream_(nullptr),
initialized_(false),
callback_registered_(false),
num_releases_(0),
released_(false),
encoder_factory_(this) {}
bool IsReleased() {
rtc::CritScope lock(&crit_);
return released_;
}
bool IsReadyForEncode() {
rtc::CritScope lock(&crit_);
return initialized_ && callback_registered_;
}
size_t num_releases() {
rtc::CritScope lock(&crit_);
return num_releases_;
}
private:
void SetFecControllerOverride(
FecControllerOverride* fec_controller_override) override {
// Ignored.
}
int32_t InitEncode(const VideoCodec* codecSettings,
const Settings& settings) override {
rtc::CritScope lock(&crit_);
EXPECT_FALSE(initialized_);
initialized_ = true;
released_ = false;
return 0;
}
int32_t Encode(const VideoFrame& inputImage,
const std::vector<VideoFrameType>* frame_types) override {
EXPECT_TRUE(IsReadyForEncode());
observation_complete_.Set();
return 0;
}
int32_t RegisterEncodeCompleteCallback(
EncodedImageCallback* callback) override {
rtc::CritScope lock(&crit_);
EXPECT_TRUE(initialized_);
callback_registered_ = true;
return 0;
}
int32_t Release() override {
rtc::CritScope lock(&crit_);
EXPECT_TRUE(IsReadyForEncode());
EXPECT_FALSE(released_);
initialized_ = false;
callback_registered_ = false;
released_ = true;
++num_releases_;
return 0;
}
void SetRates(const RateControlParameters& parameters) override {
EXPECT_TRUE(IsReadyForEncode());
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
stream_ = send_stream;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
encoder_config_ = encoder_config->Copy();
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for Encode.";
SendTask(RTC_FROM_HERE, task_queue_, [this]() {
EXPECT_EQ(0u, num_releases());
stream_->ReconfigureVideoEncoder(std::move(encoder_config_));
EXPECT_EQ(0u, num_releases());
stream_->Stop();
// Encoder should not be released before destroying the VideoSendStream.
EXPECT_FALSE(IsReleased());
EXPECT_TRUE(IsReadyForEncode());
stream_->Start();
});
// Sanity check, make sure we still encode frames with this encoder.
EXPECT_TRUE(Wait()) << "Timed out while waiting for Encode.";
}
TaskQueueBase* const task_queue_;
rtc::CriticalSection crit_;
VideoSendStream* stream_;
bool initialized_ RTC_GUARDED_BY(crit_);
bool callback_registered_ RTC_GUARDED_BY(crit_);
size_t num_releases_ RTC_GUARDED_BY(crit_);
bool released_ RTC_GUARDED_BY(crit_);
test::VideoEncoderProxyFactory encoder_factory_;
VideoEncoderConfig encoder_config_;
} test_encoder(task_queue());
RunBaseTest(&test_encoder);
EXPECT_TRUE(test_encoder.IsReleased());
EXPECT_EQ(1u, test_encoder.num_releases());
}
static const size_t kVideoCodecConfigObserverNumberOfTemporalLayers = 3;
template <typename T>
class VideoCodecConfigObserver : public test::SendTest,
public test::FakeEncoder {
public:
VideoCodecConfigObserver(VideoCodecType video_codec_type,
const char* codec_name)
: SendTest(VideoSendStreamTest::kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
video_codec_type_(video_codec_type),
codec_name_(codec_name),
num_initializations_(0),
stream_(nullptr),
encoder_factory_(this) {
InitCodecSpecifics();
}
private:
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->rtp.payload_name = codec_name_;
encoder_config->codec_type = video_codec_type_;
encoder_config->encoder_specific_settings = GetEncoderSpecificSettings();
EXPECT_EQ(1u, encoder_config->simulcast_layers.size());
encoder_config->simulcast_layers[0].num_temporal_layers =
kVideoCodecConfigObserverNumberOfTemporalLayers;
encoder_config_ = encoder_config->Copy();
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
stream_ = send_stream;
}
int32_t InitEncode(const VideoCodec* config,
const Settings& settings) override {
EXPECT_EQ(video_codec_type_, config->codecType);
VerifyCodecSpecifics(*config);
++num_initializations_;
init_encode_event_.Set();
return FakeEncoder::InitEncode(config, settings);
}
void InitCodecSpecifics();
void VerifyCodecSpecifics(const VideoCodec& config) const;
rtc::scoped_refptr<VideoEncoderConfig::EncoderSpecificSettings>
GetEncoderSpecificSettings() const;
void PerformTest() override {
EXPECT_TRUE(
init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
ASSERT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
// Change encoder settings to actually trigger reconfiguration.
encoder_settings_.frameDroppingOn = !encoder_settings_.frameDroppingOn;
encoder_config_.encoder_specific_settings = GetEncoderSpecificSettings();
stream_->ReconfigureVideoEncoder(std::move(encoder_config_));