blob: 90830d12c47ae5b6d1b913aa3828aee40fec65ab [file] [log] [blame]
/*
* Copyright 2018 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 <memory>
#include "api/test/simulated_network.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/include/module_common_types_public.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/call_test.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kTransportSequenceNumberExtensionId = 1,
};
} // namespace
class RtpRtcpEndToEndTest : public test::CallTest {
protected:
void RespectsRtcpMode(RtcpMode rtcp_mode);
void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp);
};
void RtpRtcpEndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
static const int kNumCompoundRtcpPacketsToObserve = 10;
class RtcpModeObserver : public test::EndToEndTest {
public:
explicit RtcpModeObserver(RtcpMode rtcp_mode)
: EndToEndTest(kDefaultTimeoutMs),
rtcp_mode_(rtcp_mode),
sent_rtp_(0),
sent_rtcp_(0) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
if (++sent_rtp_ % 3 == 0)
return DROP_PACKET;
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
++sent_rtcp_;
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
EXPECT_EQ(0, parser.sender_report()->num_packets());
switch (rtcp_mode_) {
case RtcpMode::kCompound:
// TODO(holmer): We shouldn't send transport feedback alone if
// compound RTCP is negotiated.
if (parser.receiver_report()->num_packets() == 0 &&
parser.transport_feedback()->num_packets() == 0) {
ADD_FAILURE() << "Received RTCP packet without receiver report for "
"RtcpMode::kCompound.";
observation_complete_.Set();
}
if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
observation_complete_.Set();
break;
case RtcpMode::kReducedSize:
if (parser.receiver_report()->num_packets() == 0)
observation_complete_.Set();
break;
case RtcpMode::kOff:
RTC_DCHECK_NOTREACHED();
break;
}
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< (rtcp_mode_ == RtcpMode::kCompound
? "Timed out before observing enough compound packets."
: "Timed out before receiving a non-compound RTCP packet.");
}
RtcpMode rtcp_mode_;
Mutex mutex_;
// Must be protected since RTCP can be sent by both the process thread
// and the pacer thread.
int sent_rtp_ RTC_GUARDED_BY(&mutex_);
int sent_rtcp_ RTC_GUARDED_BY(&mutex_);
} test(rtcp_mode);
RunBaseTest(&test);
}
TEST_F(RtpRtcpEndToEndTest, UsesRtcpCompoundMode) {
RespectsRtcpMode(RtcpMode::kCompound);
}
TEST_F(RtpRtcpEndToEndTest, UsesRtcpReducedSizeMode) {
RespectsRtcpMode(RtcpMode::kReducedSize);
}
void RtpRtcpEndToEndTest::TestRtpStatePreservation(
bool use_rtx,
bool provoke_rtcpsr_before_rtp) {
// This test uses other VideoStream settings than the the default settings
// implemented in DefaultVideoStreamFactory. Therefore this test implements
// its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
// in ModifyVideoConfigs.
class VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
VideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
if (encoder_config.number_of_streams > 1) {
// Lower bitrates so that all streams send initially.
RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
streams[i].min_bitrate_bps = 10000;
streams[i].target_bitrate_bps = 15000;
streams[i].max_bitrate_bps = 20000;
}
} else {
// Use the same total bitrates when sending a single stream to avoid
// lowering
// the bitrate estimate and requiring a subsequent rampup.
streams[0].min_bitrate_bps = 3 * 10000;
streams[0].target_bitrate_bps = 3 * 15000;
streams[0].max_bitrate_bps = 3 * 20000;
}
return streams;
}
};
class RtpSequenceObserver : public test::RtpRtcpObserver {
public:
explicit RtpSequenceObserver(bool use_rtx)
: test::RtpRtcpObserver(kDefaultTimeoutMs),
ssrcs_to_observe_(kNumSimulcastStreams) {
for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
ssrc_is_rtx_[kVideoSendSsrcs[i]] = false;
if (use_rtx)
ssrc_is_rtx_[kSendRtxSsrcs[i]] = true;
}
}
void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
MutexLock lock(&mutex_);
ssrc_observed_.clear();
ssrcs_to_observe_ = num_expected_ssrcs;
}
private:
void ValidateTimestampGap(uint32_t ssrc,
uint32_t timestamp,
bool only_padding)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) {
static const int32_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
auto timestamp_it = last_observed_timestamp_.find(ssrc);
if (timestamp_it == last_observed_timestamp_.end()) {
EXPECT_FALSE(only_padding);
last_observed_timestamp_[ssrc] = timestamp;
} else {
// Verify timestamps are reasonably close.
uint32_t latest_observed = timestamp_it->second;
// Wraparound handling is unnecessary here as long as an int variable
// is used to store the result.
int32_t timestamp_gap = timestamp - latest_observed;
EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap)
<< "Gap in timestamps (" << latest_observed << " -> " << timestamp
<< ") too large for SSRC: " << ssrc << ".";
timestamp_it->second = timestamp;
}
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
const uint32_t ssrc = rtp_packet.Ssrc();
const int64_t sequence_number =
seq_numbers_unwrapper_.Unwrap(rtp_packet.SequenceNumber());
const uint32_t timestamp = rtp_packet.Timestamp();
const bool only_padding = rtp_packet.payload_size() == 0;
EXPECT_TRUE(ssrc_is_rtx_.find(ssrc) != ssrc_is_rtx_.end())
<< "Received SSRC that wasn't configured: " << ssrc;
static const int64_t kMaxSequenceNumberGap = 100;
std::list<int64_t>* seq_numbers = &last_observed_seq_numbers_[ssrc];
if (seq_numbers->empty()) {
seq_numbers->push_back(sequence_number);
} else {
// We shouldn't get replays of previous sequence numbers.
for (int64_t observed : *seq_numbers) {
EXPECT_NE(observed, sequence_number)
<< "Received sequence number " << sequence_number << " for SSRC "
<< ssrc << " 2nd time.";
}
// Verify sequence numbers are reasonably close.
int64_t latest_observed = seq_numbers->back();
int64_t sequence_number_gap = sequence_number - latest_observed;
EXPECT_LE(std::abs(sequence_number_gap), kMaxSequenceNumberGap)
<< "Gap in sequence numbers (" << latest_observed << " -> "
<< sequence_number << ") too large for SSRC: " << ssrc << ".";
seq_numbers->push_back(sequence_number);
if (seq_numbers->size() >= kMaxSequenceNumberGap) {
seq_numbers->pop_front();
}
}
if (!ssrc_is_rtx_[ssrc]) {
MutexLock lock(&mutex_);
ValidateTimestampGap(ssrc, timestamp, only_padding);
// Wait for media packets on all ssrcs.
if (!ssrc_observed_[ssrc] && !only_padding) {
ssrc_observed_[ssrc] = true;
if (--ssrcs_to_observe_ == 0)
observation_complete_.Set();
}
}
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
test::RtcpPacketParser rtcp_parser;
rtcp_parser.Parse(packet, length);
if (rtcp_parser.sender_report()->num_packets() > 0) {
uint32_t ssrc = rtcp_parser.sender_report()->sender_ssrc();
uint32_t rtcp_timestamp = rtcp_parser.sender_report()->rtp_timestamp();
MutexLock lock(&mutex_);
ValidateTimestampGap(ssrc, rtcp_timestamp, false);
}
return SEND_PACKET;
}
SequenceNumberUnwrapper seq_numbers_unwrapper_;
std::map<uint32_t, std::list<int64_t>> last_observed_seq_numbers_;
std::map<uint32_t, uint32_t> last_observed_timestamp_;
std::map<uint32_t, bool> ssrc_is_rtx_;
Mutex mutex_;
size_t ssrcs_to_observe_ RTC_GUARDED_BY(mutex_);
std::map<uint32_t, bool> ssrc_observed_ RTC_GUARDED_BY(mutex_);
} observer(use_rtx);
std::unique_ptr<test::PacketTransport> send_transport;
std::unique_ptr<test::PacketTransport> receive_transport;
VideoEncoderConfig one_stream;
SendTask(
RTC_FROM_HERE, task_queue(),
[this, &observer, &send_transport, &receive_transport, &one_stream,
use_rtx]() {
CreateCalls();
send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), &observer,
test::PacketTransport::kSender, payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())));
receive_transport = std::make_unique<test::PacketTransport>(
task_queue(), nullptr, &observer, test::PacketTransport::kReceiver,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())));
send_transport->SetReceiver(receiver_call_->Receiver());
receive_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport.get());
if (use_rtx) {
for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
}
GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
}
GetVideoEncoderConfig()->video_stream_factory =
rtc::make_ref_counted<VideoStreamFactory>();
// Use the same total bitrates when sending a single stream to avoid
// lowering the bitrate estimate and requiring a subsequent rampup.
one_stream = GetVideoEncoderConfig()->Copy();
// one_stream.streams.resize(1);
one_stream.number_of_streams = 1;
CreateMatchingReceiveConfigs(receive_transport.get());
CreateVideoStreams();
CreateFrameGeneratorCapturer(30, 1280, 720);
Start();
});
EXPECT_TRUE(observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
// Test stream resetting more than once to make sure that the state doesn't
// get set once (this could be due to using std::map::insert for instance).
for (size_t i = 0; i < 3; ++i) {
SendTask(RTC_FROM_HERE, task_queue(), [&]() {
DestroyVideoSendStreams();
// Re-create VideoSendStream with only one stream.
CreateVideoSendStream(one_stream);
GetVideoSendStream()->Start();
if (provoke_rtcpsr_before_rtp) {
// Rapid Resync Request forces sending RTCP Sender Report back.
// Using this request speeds up this test because then there is no need
// to wait for a second for periodic Sender Report.
rtcp::RapidResyncRequest force_send_sr_back_request;
rtc::Buffer packet = force_send_sr_back_request.Build();
static_cast<webrtc::test::DirectTransport*>(receive_transport.get())
->SendRtcp(packet.data(), packet.size());
}
CreateFrameGeneratorCapturer(30, 1280, 720);
});
observer.ResetExpectedSsrcs(1);
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
// Reconfigure back to use all streams.
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendStream()->ReconfigureVideoEncoder(
GetVideoEncoderConfig()->Copy());
});
observer.ResetExpectedSsrcs(kNumSimulcastStreams);
EXPECT_TRUE(observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
// Reconfigure down to one stream.
SendTask(RTC_FROM_HERE, task_queue(), [this, &one_stream]() {
GetVideoSendStream()->ReconfigureVideoEncoder(one_stream.Copy());
});
observer.ResetExpectedSsrcs(1);
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
// Reconfigure back to use all streams.
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendStream()->ReconfigureVideoEncoder(
GetVideoEncoderConfig()->Copy());
});
observer.ResetExpectedSsrcs(kNumSimulcastStreams);
EXPECT_TRUE(observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
}
SendTask(RTC_FROM_HERE, task_queue(),
[this, &send_transport, &receive_transport]() {
Stop();
DestroyStreams();
send_transport.reset();
receive_transport.reset();
DestroyCalls();
});
}
TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpState) {
TestRtpStatePreservation(false, false);
}
TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
TestRtpStatePreservation(true, false);
}
TEST_F(RtpRtcpEndToEndTest,
RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
TestRtpStatePreservation(true, true);
}
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=9648.
TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
class RtpSequenceObserver : public test::RtpRtcpObserver {
public:
RtpSequenceObserver()
: test::RtpRtcpObserver(kDefaultTimeoutMs),
num_flexfec_packets_sent_(0) {}
void ResetPacketCount() {
MutexLock lock(&mutex_);
num_flexfec_packets_sent_ = 0;
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
const uint16_t sequence_number = rtp_packet.SequenceNumber();
const uint32_t timestamp = rtp_packet.Timestamp();
const uint32_t ssrc = rtp_packet.Ssrc();
if (ssrc == kVideoSendSsrcs[0] || ssrc == kSendRtxSsrcs[0]) {
return SEND_PACKET;
}
EXPECT_EQ(kFlexfecSendSsrc, ssrc) << "Unknown SSRC sent.";
++num_flexfec_packets_sent_;
// If this is the first packet, we have nothing to compare to.
if (!last_observed_sequence_number_) {
last_observed_sequence_number_.emplace(sequence_number);
last_observed_timestamp_.emplace(timestamp);
return SEND_PACKET;
}
// Verify continuity and monotonicity of RTP sequence numbers.
EXPECT_EQ(static_cast<uint16_t>(*last_observed_sequence_number_ + 1),
sequence_number);
last_observed_sequence_number_.emplace(sequence_number);
// Timestamps should be non-decreasing...
const bool timestamp_is_same_or_newer =
timestamp == *last_observed_timestamp_ ||
IsNewerTimestamp(timestamp, *last_observed_timestamp_);
EXPECT_TRUE(timestamp_is_same_or_newer);
// ...but reasonably close in time.
const int k10SecondsInRtpTimestampBase = 10 * kVideoPayloadTypeFrequency;
EXPECT_TRUE(IsNewerTimestamp(
*last_observed_timestamp_ + k10SecondsInRtpTimestampBase, timestamp));
last_observed_timestamp_.emplace(timestamp);
// Pass test when enough packets have been let through.
if (num_flexfec_packets_sent_ >= 10) {
observation_complete_.Set();
}
return SEND_PACKET;
}
absl::optional<uint16_t> last_observed_sequence_number_
RTC_GUARDED_BY(mutex_);
absl::optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(mutex_);
size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(mutex_);
Mutex mutex_;
} observer;
static constexpr int kFrameMaxWidth = 320;
static constexpr int kFrameMaxHeight = 180;
static constexpr int kFrameRate = 15;
std::unique_ptr<test::PacketTransport> send_transport;
std::unique_ptr<test::PacketTransport> receive_transport;
test::FunctionVideoEncoderFactory encoder_factory(
[]() { return VP8Encoder::Create(); });
SendTask(RTC_FROM_HERE, task_queue(), [&]() {
CreateCalls();
BuiltInNetworkBehaviorConfig lossy_delayed_link;
lossy_delayed_link.loss_percent = 2;
lossy_delayed_link.queue_delay_ms = 50;
send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), &observer,
test::PacketTransport::kSender, payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(lossy_delayed_link)));
send_transport->SetReceiver(receiver_call_->Receiver());
BuiltInNetworkBehaviorConfig flawless_link;
receive_transport = std::make_unique<test::PacketTransport>(
task_queue(), nullptr, &observer, test::PacketTransport::kReceiver,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(flawless_link)));
receive_transport->SetReceiver(sender_call_->Receiver());
// For reduced flakyness, we use a real VP8 encoder together with NACK
// and RTX.
const int kNumVideoStreams = 1;
const int kNumFlexfecStreams = 1;
CreateSendConfig(kNumVideoStreams, 0, kNumFlexfecStreams,
send_transport.get());
GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
GetVideoSendConfig()->rtp.payload_name = "VP8";
GetVideoSendConfig()->rtp.payload_type = kVideoSendPayloadType;
GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
GetVideoEncoderConfig()->codec_type = kVideoCodecVP8;
CreateMatchingReceiveConfigs(receive_transport.get());
video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
video_receive_configs_[0]
.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
kVideoSendPayloadType;
// The matching FlexFEC receive config is not created by
// CreateMatchingReceiveConfigs since this is not a test::BaseTest.
// Set up the receive config manually instead.
FlexfecReceiveStream::Config flexfec_receive_config(
receive_transport.get());
flexfec_receive_config.payload_type =
GetVideoSendConfig()->rtp.flexfec.payload_type;
flexfec_receive_config.rtp.remote_ssrc =
GetVideoSendConfig()->rtp.flexfec.ssrc;
flexfec_receive_config.protected_media_ssrcs =
GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs;
flexfec_receive_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
flexfec_receive_config.rtp.transport_cc = true;
flexfec_receive_config.rtp.extensions.emplace_back(
RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId);
flexfec_receive_configs_.push_back(flexfec_receive_config);
CreateFlexfecStreams();
CreateVideoStreams();
// RTCP might be disabled if the network is "down".
sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
Start();
});
// Initial test.
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
SendTask(RTC_FROM_HERE, task_queue(), [this, &observer]() {
// Ensure monotonicity when the VideoSendStream is restarted.
Stop();
observer.ResetPacketCount();
Start();
});
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
SendTask(RTC_FROM_HERE, task_queue(), [this, &observer]() {
// Ensure monotonicity when the VideoSendStream is recreated.
DestroyVideoSendStreams();
observer.ResetPacketCount();
CreateVideoSendStreams();
GetVideoSendStream()->Start();
CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
});
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
// Cleanup.
SendTask(RTC_FROM_HERE, task_queue(),
[this, &send_transport, &receive_transport]() {
Stop();
DestroyStreams();
send_transport.reset();
receive_transport.reset();
DestroyCalls();
});
}
} // namespace webrtc