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webrtc / src / / b4a1ae5299fd57be66c7cbb7a982179bb1ecfb90 / . / webrtc / video / end_to_end_tests.cc
blob: 84bd88862256972d7c3795596c2ca04737cadaf6 [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>
#include <map>
#include <sstream>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/event.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/frame_callback.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/histogram.h"
#include "webrtc/test/null_transport.h"
#include "webrtc/test/rtcp_packet_parser.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video_encoder.h"
namespace webrtc {
static const unsigned long kSilenceTimeoutMs = 2000;
class EndToEndTest : public test::CallTest {
public:
EndToEndTest() {}
virtual ~EndToEndTest() {
EXPECT_EQ(nullptr, send_stream_);
EXPECT_TRUE(receive_streams_.empty());
}
protected:
class UnusedTransport : public Transport {
private:
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
ADD_FAILURE() << "Unexpected RTP sent.";
return false;
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
ADD_FAILURE() << "Unexpected RTCP sent.";
return false;
}
};
void DecodesRetransmittedFrame(bool use_rtx, bool use_red);
void ReceivesPliAndRecovers(int rtp_history_ms);
void RespectsRtcpMode(RtcpMode rtcp_mode);
void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
void TestRtpStatePreservation(bool use_rtx);
void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare);
};
TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
CreateCalls(Call::Config(), Call::Config());
test::NullTransport transport;
CreateSendConfig(1, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateStreams();
receive_streams_[0]->Start();
receive_streams_[0]->Start();
DestroyStreams();
}
TEST_F(EndToEndTest, ReceiverCanBeStoppedTwice) {
CreateCalls(Call::Config(), Call::Config());
test::NullTransport transport;
CreateSendConfig(1, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateStreams();
receive_streams_[0]->Stop();
receive_streams_[0]->Stop();
DestroyStreams();
}
TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
static const int kWidth = 320;
static const int kHeight = 240;
// This constant is chosen to be higher than the timeout in the video_render
// module. This makes sure that frames aren't dropped if there are no other
// frames in the queue.
static const int kDelayRenderCallbackMs = 1000;
class Renderer : public VideoRenderer {
public:
Renderer() : event_(EventWrapper::Create()) {}
void RenderFrame(const VideoFrame& video_frame,
int /*time_to_render_ms*/) override {
event_->Set();
}
bool IsTextureSupported() const override { return false; }
EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
rtc::scoped_ptr<EventWrapper> event_;
} renderer;
class TestFrameCallback : public I420FrameCallback {
public:
TestFrameCallback() : event_(EventWrapper::Create()) {}
EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
private:
void FrameCallback(VideoFrame* frame) override {
SleepMs(kDelayRenderCallbackMs);
event_->Set();
}
rtc::scoped_ptr<EventWrapper> event_;
};
CreateCalls(Call::Config(), Call::Config());
test::DirectTransport sender_transport(sender_call_.get());
test::DirectTransport receiver_transport(receiver_call_.get());
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
TestFrameCallback pre_render_callback;
receive_configs_[0].pre_render_callback = &pre_render_callback;
receive_configs_[0].renderer = &renderer;
CreateStreams();
Start();
// Create frames that are smaller than the send width/height, this is done to
// check that the callbacks are done after processing video.
rtc::scoped_ptr<test::FrameGenerator> frame_generator(
test::FrameGenerator::CreateChromaGenerator(kWidth, kHeight));
send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
<< "Timed out while waiting for pre-render callback.";
EXPECT_EQ(kEventSignaled, renderer.Wait())
<< "Timed out while waiting for the frame to render.";
Stop();
sender_transport.StopSending();
receiver_transport.StopSending();
DestroyStreams();
}
TEST_F(EndToEndTest, TransmitsFirstFrame) {
class Renderer : public VideoRenderer {
public:
Renderer() : event_(EventWrapper::Create()) {}
void RenderFrame(const VideoFrame& video_frame,
int /*time_to_render_ms*/) override {
event_->Set();
}
bool IsTextureSupported() const override { return false; }
EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
rtc::scoped_ptr<EventWrapper> event_;
} renderer;
CreateCalls(Call::Config(), Call::Config());
test::DirectTransport sender_transport(sender_call_.get());
test::DirectTransport receiver_transport(receiver_call_.get());
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
receive_configs_[0].renderer = &renderer;
CreateStreams();
Start();
rtc::scoped_ptr<test::FrameGenerator> frame_generator(
test::FrameGenerator::CreateChromaGenerator(
encoder_config_.streams[0].width, encoder_config_.streams[0].height));
send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
EXPECT_EQ(kEventSignaled, renderer.Wait())
<< "Timed out while waiting for the frame to render.";
Stop();
sender_transport.StopSending();
receiver_transport.StopSending();
DestroyStreams();
}
TEST_F(EndToEndTest, SendsAndReceivesVP9) {
class VP9Observer : public test::EndToEndTest, public VideoRenderer {
public:
VP9Observer()
: EndToEndTest(2 * kDefaultTimeoutMs),
encoder_(VideoEncoder::Create(VideoEncoder::kVp9)),
decoder_(VP9Decoder::Create()),
frame_counter_(0) {}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for enough frames to be decoded.";
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = encoder_.get();
send_config->encoder_settings.payload_name = "VP9";
send_config->encoder_settings.payload_type = 124;
encoder_config->streams[0].min_bitrate_bps = 50000;
encoder_config->streams[0].target_bitrate_bps =
encoder_config->streams[0].max_bitrate_bps = 2000000;
(*receive_configs)[0].renderer = this;
(*receive_configs)[0].decoders.resize(1);
(*receive_configs)[0].decoders[0].payload_type =
send_config->encoder_settings.payload_type;
(*receive_configs)[0].decoders[0].payload_name =
send_config->encoder_settings.payload_name;
(*receive_configs)[0].decoders[0].decoder = decoder_.get();
}
void RenderFrame(const VideoFrame& video_frame,
int time_to_render_ms) override {
const int kRequiredFrames = 500;
if (++frame_counter_ == kRequiredFrames)
observation_complete_->Set();
}
bool IsTextureSupported() const override { return false; }
private:
rtc::scoped_ptr<webrtc::VideoEncoder> encoder_;
rtc::scoped_ptr<webrtc::VideoDecoder> decoder_;
int frame_counter_;
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, SendsAndReceivesH264) {
class H264Observer : public test::EndToEndTest, public VideoRenderer {
public:
H264Observer()
: EndToEndTest(2 * kDefaultTimeoutMs),
fake_encoder_(Clock::GetRealTimeClock()),
frame_counter_(0) {}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for enough frames to be decoded.";
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms =
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->encoder_settings.encoder = &fake_encoder_;
send_config->encoder_settings.payload_name = "H264";
send_config->encoder_settings.payload_type = kFakeSendPayloadType;
encoder_config->streams[0].min_bitrate_bps = 50000;
encoder_config->streams[0].target_bitrate_bps =
encoder_config->streams[0].max_bitrate_bps = 2000000;
(*receive_configs)[0].renderer = this;
(*receive_configs)[0].decoders.resize(1);
(*receive_configs)[0].decoders[0].payload_type =
send_config->encoder_settings.payload_type;
(*receive_configs)[0].decoders[0].payload_name =
send_config->encoder_settings.payload_name;
(*receive_configs)[0].decoders[0].decoder = &fake_decoder_;
}
void RenderFrame(const VideoFrame& video_frame,
int time_to_render_ms) override {
const int kRequiredFrames = 500;
if (++frame_counter_ == kRequiredFrames)
observation_complete_->Set();
}
bool IsTextureSupported() const override { return false; }
private:
test::FakeH264Decoder fake_decoder_;
test::FakeH264Encoder fake_encoder_;
int frame_counter_;
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
class SyncRtcpObserver : public test::EndToEndTest {
public:
SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
uint32_t ssrc = 0;
ssrc |= static_cast<uint32_t>(packet[4]) << 24;
ssrc |= static_cast<uint32_t>(packet[5]) << 16;
ssrc |= static_cast<uint32_t>(packet[6]) << 8;
ssrc |= static_cast<uint32_t>(packet[7]) << 0;
EXPECT_EQ(kReceiverLocalSsrc, ssrc);
observation_complete_->Set();
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for a receiver RTCP packet to be sent.";
}
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
static const int kNumberOfNacksToObserve = 2;
static const int kLossBurstSize = 2;
static const int kPacketsBetweenLossBursts = 9;
class NackObserver : public test::EndToEndTest {
public:
NackObserver()
: EndToEndTest(kLongTimeoutMs),
rtp_parser_(RtpHeaderParser::Create()),
sent_rtp_packets_(0),
packets_left_to_drop_(0),
nacks_left_(kNumberOfNacksToObserve) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
// Never drop retransmitted packets.
if (dropped_packets_.find(header.sequenceNumber) !=
dropped_packets_.end()) {
retransmitted_packets_.insert(header.sequenceNumber);
if (nacks_left_ <= 0 &&
retransmitted_packets_.size() == dropped_packets_.size()) {
observation_complete_->Set();
}
return SEND_PACKET;
}
++sent_rtp_packets_;
// Enough NACKs received, stop dropping packets.
if (nacks_left_ <= 0)
return SEND_PACKET;
// Check if it's time for a new loss burst.
if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
packets_left_to_drop_ = kLossBurstSize;
// Never drop padding packets as those won't be retransmitted.
if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
--packets_left_to_drop_;
dropped_packets_.insert(header.sequenceNumber);
return DROP_PACKET;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
if (packet_type == RTCPUtility::RTCPPacketTypes::kRtpfbNack) {
--nacks_left_;
break;
}
packet_type = parser.Iterate();
}
return SEND_PACKET;
}
void ModifyConfigs(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;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out waiting for packets to be NACKed, retransmitted and "
"rendered.";
}
rtc::CriticalSection crit_;
rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
std::set<uint16_t> dropped_packets_;
std::set<uint16_t> retransmitted_packets_;
uint64_t sent_rtp_packets_;
int packets_left_to_drop_;
int nacks_left_ GUARDED_BY(&crit_);
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, CanReceiveFec) {
class FecRenderObserver : public test::EndToEndTest, public VideoRenderer {
public:
FecRenderObserver()
: EndToEndTest(kDefaultTimeoutMs), state_(kFirstPacket) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
int encapsulated_payload_type = -1;
if (header.payloadType == kRedPayloadType) {
encapsulated_payload_type =
static_cast<int>(packet[header.headerLength]);
if (encapsulated_payload_type != kFakeSendPayloadType)
EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
} else {
EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
}
if (protected_sequence_numbers_.count(header.sequenceNumber) != 0) {
// Retransmitted packet, should not count.
protected_sequence_numbers_.erase(header.sequenceNumber);
EXPECT_GT(protected_timestamps_.count(header.timestamp), 0u);
protected_timestamps_.erase(header.timestamp);
return SEND_PACKET;
}
switch (state_) {
case kFirstPacket:
state_ = kDropEveryOtherPacketUntilFec;
break;
case kDropEveryOtherPacketUntilFec:
if (encapsulated_payload_type == kUlpfecPayloadType) {
state_ = kDropNextMediaPacket;
return SEND_PACKET;
}
if (header.sequenceNumber % 2 == 0)
return DROP_PACKET;
break;
case kDropNextMediaPacket:
if (encapsulated_payload_type == kFakeSendPayloadType) {
protected_sequence_numbers_.insert(header.sequenceNumber);
protected_timestamps_.insert(header.timestamp);
state_ = kDropEveryOtherPacketUntilFec;
return DROP_PACKET;
}
break;
}
return SEND_PACKET;
}
void RenderFrame(const VideoFrame& video_frame,
int time_to_render_ms) override {
rtc::CritScope lock(&crit_);
// Rendering frame with timestamp of packet that was dropped -> FEC
// protection worked.
if (protected_timestamps_.count(video_frame.timestamp()) != 0)
observation_complete_->Set();
}
bool IsTextureSupported() const override { return false; }
enum {
kFirstPacket,
kDropEveryOtherPacketUntilFec,
kDropNextMediaPacket,
} state_;
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// TODO(pbos): Run this test with combined NACK/FEC enabled as well.
// int rtp_history_ms = 1000;
// (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms;
// send_config->rtp.nack.rtp_history_ms = rtp_history_ms;
send_config->rtp.fec.red_payload_type = kRedPayloadType;
send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
(*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
(*receive_configs)[0].renderer = this;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out waiting for dropped frames frames to be rendered.";
}
rtc::CriticalSection crit_;
std::set<uint32_t> protected_sequence_numbers_ GUARDED_BY(crit_);
std::set<uint32_t> protected_timestamps_ GUARDED_BY(crit_);
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
// Flacky on all platforms. See webrtc:4328.
TEST_F(EndToEndTest, DISABLED_ReceivedFecPacketsNotNacked) {
class FecNackObserver : public test::EndToEndTest {
public:
explicit FecNackObserver()
: EndToEndTest(kDefaultTimeoutMs),
state_(kFirstPacket),
fec_sequence_number_(0),
has_last_sequence_number_(false),
last_sequence_number_(0) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock_(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
int encapsulated_payload_type = -1;
if (header.payloadType == kRedPayloadType) {
encapsulated_payload_type =
static_cast<int>(packet[header.headerLength]);
if (encapsulated_payload_type != kFakeSendPayloadType)
EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
} else {
EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
}
if (has_last_sequence_number_ &&
!IsNewerSequenceNumber(header.sequenceNumber,
last_sequence_number_)) {
// Drop retransmitted packets.
return DROP_PACKET;
}
last_sequence_number_ = header.sequenceNumber;
has_last_sequence_number_ = true;
bool fec_packet = encapsulated_payload_type == kUlpfecPayloadType;
switch (state_) {
case kFirstPacket:
state_ = kDropEveryOtherPacketUntilFec;
break;
case kDropEveryOtherPacketUntilFec:
if (fec_packet) {
state_ = kDropAllMediaPacketsUntilFec;
} else if (header.sequenceNumber % 2 == 0) {
return DROP_PACKET;
}
break;
case kDropAllMediaPacketsUntilFec:
if (!fec_packet)
return DROP_PACKET;
fec_sequence_number_ = header.sequenceNumber;
state_ = kVerifyFecPacketNotInNackList;
break;
case kVerifyFecPacketNotInNackList:
// Continue to drop packets. Make sure no frame can be decoded.
if (fec_packet || header.sequenceNumber % 2 == 0)
return DROP_PACKET;
break;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock_(&crit_);
if (state_ == kVerifyFecPacketNotInNackList) {
test::RtcpPacketParser rtcp_parser;
rtcp_parser.Parse(packet, length);
std::vector<uint16_t> nacks = rtcp_parser.nack_item()->last_nack_list();
if (!nacks.empty() &&
IsNewerSequenceNumber(nacks.back(), fec_sequence_number_)) {
EXPECT_TRUE(std::find(
nacks.begin(), nacks.end(), fec_sequence_number_) == nacks.end());
observation_complete_->Set();
}
}
return SEND_PACKET;
}
// TODO(holmer): Investigate why we don't send FEC packets when the bitrate
// is 10 kbps.
Call::Config GetSenderCallConfig() override {
Call::Config config;
const int kMinBitrateBps = 30000;
config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
return config;
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Configure hybrid NACK/FEC.
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.fec.red_payload_type = kRedPayloadType;
send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for FEC packets to be received.";
}
enum {
kFirstPacket,
kDropEveryOtherPacketUntilFec,
kDropAllMediaPacketsUntilFec,
kVerifyFecPacketNotInNackList,
} state_;
rtc::CriticalSection crit_;
uint16_t fec_sequence_number_ GUARDED_BY(&crit_);
bool has_last_sequence_number_;
uint16_t last_sequence_number_;
} test;
// At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
// Configure some network delay.
const int kNetworkDelayMs = 50;
FakeNetworkPipe::Config config;
config.queue_delay_ms = kNetworkDelayMs;
RunBaseTest(&test, config);
}
// This test drops second RTP packet with a marker bit set, makes sure it's
// retransmitted and renders. Retransmission SSRCs are also checked.
void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
// Must be set high enough to allow the bitrate probing to finish.
static const int kMinProbePackets = 30;
static const int kDroppedFrameNumber = kMinProbePackets + 1;
class RetransmissionObserver : public test::EndToEndTest,
public I420FrameCallback {
public:
explicit RetransmissionObserver(bool use_rtx, bool use_red)
: EndToEndTest(kDefaultTimeoutMs),
payload_type_(GetPayloadType(false, use_red)),
retransmission_ssrc_(use_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
retransmission_payload_type_(GetPayloadType(use_rtx, use_red)),
marker_bits_observed_(0),
num_packets_observed_(0),
retransmitted_timestamp_(0),
frame_retransmitted_(false) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
// We accept some padding or RTX packets in the beginning to enable
// bitrate probing.
if (num_packets_observed_++ < kMinProbePackets &&
header.payloadType != payload_type_) {
EXPECT_TRUE(retransmission_payload_type_ == header.payloadType ||
length == header.headerLength + header.paddingLength);
return SEND_PACKET;
}
if (header.timestamp == retransmitted_timestamp_) {
EXPECT_EQ(retransmission_ssrc_, header.ssrc);
EXPECT_EQ(retransmission_payload_type_, header.payloadType);
frame_retransmitted_ = true;
return SEND_PACKET;
}
EXPECT_EQ(kSendSsrcs[0], header.ssrc);
EXPECT_EQ(payload_type_, header.payloadType);
// Found the final packet of the frame to inflict loss to, drop this and
// expect a retransmission.
if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) {
retransmitted_timestamp_ = header.timestamp;
return DROP_PACKET;
}
return SEND_PACKET;
}
void FrameCallback(VideoFrame* frame) override {
rtc::CritScope lock(&crit_);
if (frame->timestamp() == retransmitted_timestamp_) {
EXPECT_TRUE(frame_retransmitted_);
observation_complete_->Set();
}
}
void ModifyConfigs(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].pre_render_callback = this;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (payload_type_ == kRedPayloadType) {
send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
}
if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
kSendRtxSsrcs[0];
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
kSendRtxPayloadType;
}
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for retransmission to render.";
}
int GetPayloadType(bool use_rtx, bool use_red) {
return use_rtx ? kSendRtxPayloadType
: (use_red ? kRedPayloadType : kFakeSendPayloadType);
}
rtc::CriticalSection crit_;
const int payload_type_;
const uint32_t retransmission_ssrc_;
const int retransmission_payload_type_;
int marker_bits_observed_;
int num_packets_observed_;
uint32_t retransmitted_timestamp_ GUARDED_BY(&crit_);
bool frame_retransmitted_;
} test(use_rtx, use_red);
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, DecodesRetransmittedFrame) {
DecodesRetransmittedFrame(false, false);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
DecodesRetransmittedFrame(true, false);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameByRed) {
DecodesRetransmittedFrame(false, true);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
DecodesRetransmittedFrame(true, true);
}
TEST_F(EndToEndTest, UsesFrameCallbacks) {
static const int kWidth = 320;
static const int kHeight = 240;
class Renderer : public VideoRenderer {
public:
Renderer() : event_(EventWrapper::Create()) {}
void RenderFrame(const VideoFrame& video_frame,
int /*time_to_render_ms*/) override {
EXPECT_EQ(0, *video_frame.buffer(kYPlane))
<< "Rendered frame should have zero luma which is applied by the "
"pre-render callback.";
event_->Set();
}
bool IsTextureSupported() const override { return false; }
EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
rtc::scoped_ptr<EventWrapper> event_;
} renderer;
class TestFrameCallback : public I420FrameCallback {
public:
TestFrameCallback(int expected_luma_byte, int next_luma_byte)
: event_(EventWrapper::Create()),
expected_luma_byte_(expected_luma_byte),
next_luma_byte_(next_luma_byte) {}
EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
private:
virtual void FrameCallback(VideoFrame* frame) {
EXPECT_EQ(kWidth, frame->width())
<< "Width not as expected, callback done before resize?";
EXPECT_EQ(kHeight, frame->height())
<< "Height not as expected, callback done before resize?";
// Previous luma specified, observed luma should be fairly close.
if (expected_luma_byte_ != -1) {
EXPECT_NEAR(expected_luma_byte_, *frame->buffer(kYPlane), 10);
}
memset(frame->buffer(kYPlane),
next_luma_byte_,
frame->allocated_size(kYPlane));
event_->Set();
}
rtc::scoped_ptr<EventWrapper> event_;
int expected_luma_byte_;
int next_luma_byte_;
};
TestFrameCallback pre_encode_callback(-1, 255); // Changes luma to 255.
TestFrameCallback pre_render_callback(255, 0); // Changes luma from 255 to 0.
CreateCalls(Call::Config(), Call::Config());
test::DirectTransport sender_transport(sender_call_.get());
test::DirectTransport receiver_transport(receiver_call_.get());
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, &sender_transport);
rtc::scoped_ptr<VideoEncoder> encoder(
VideoEncoder::Create(VideoEncoder::kVp8));
send_config_.encoder_settings.encoder = encoder.get();
send_config_.encoder_settings.payload_name = "VP8";
ASSERT_EQ(1u, encoder_config_.streams.size()) << "Test setup error.";
encoder_config_.streams[0].width = kWidth;
encoder_config_.streams[0].height = kHeight;
send_config_.pre_encode_callback = &pre_encode_callback;
CreateMatchingReceiveConfigs(&receiver_transport);
receive_configs_[0].pre_render_callback = &pre_render_callback;
receive_configs_[0].renderer = &renderer;
CreateStreams();
Start();
// Create frames that are smaller than the send width/height, this is done to
// check that the callbacks are done after processing video.
rtc::scoped_ptr<test::FrameGenerator> frame_generator(
test::FrameGenerator::CreateChromaGenerator(kWidth / 2, kHeight / 2));
send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait())
<< "Timed out while waiting for pre-encode callback.";
EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
<< "Timed out while waiting for pre-render callback.";
EXPECT_EQ(kEventSignaled, renderer.Wait())
<< "Timed out while waiting for the frame to render.";
Stop();
sender_transport.StopSending();
receiver_transport.StopSending();
DestroyStreams();
}
void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
static const int kPacketsToDrop = 1;
class PliObserver : public test::EndToEndTest, public VideoRenderer {
public:
explicit PliObserver(int rtp_history_ms)
: EndToEndTest(kLongTimeoutMs),
rtp_history_ms_(rtp_history_ms),
nack_enabled_(rtp_history_ms > 0),
highest_dropped_timestamp_(0),
frames_to_drop_(0),
received_pli_(false) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
// Drop all retransmitted packets to force a PLI.
if (header.timestamp <= highest_dropped_timestamp_)
return DROP_PACKET;
if (frames_to_drop_ > 0) {
highest_dropped_timestamp_ = header.timestamp;
--frames_to_drop_;
return DROP_PACKET;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
packet_type != RTCPUtility::RTCPPacketTypes::kInvalid;
packet_type = parser.Iterate()) {
if (!nack_enabled_)
EXPECT_NE(packet_type, RTCPUtility::RTCPPacketTypes::kRtpfbNack);
if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbPli) {
received_pli_ = true;
break;
}
}
return SEND_PACKET;
}
void RenderFrame(const VideoFrame& video_frame,
int time_to_render_ms) override {
rtc::CritScope lock(&crit_);
if (received_pli_ &&
video_frame.timestamp() > highest_dropped_timestamp_) {
observation_complete_->Set();
}
if (!received_pli_)
frames_to_drop_ = kPacketsToDrop;
}
bool IsTextureSupported() const override { return false; }
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
(*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
(*receive_configs)[0].renderer = this;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait()) << "Timed out waiting for PLI to be "
"received and a frame to be "
"rendered afterwards.";
}
rtc::CriticalSection crit_;
int rtp_history_ms_;
bool nack_enabled_;
uint32_t highest_dropped_timestamp_ GUARDED_BY(&crit_);
int frames_to_drop_ GUARDED_BY(&crit_);
bool received_pli_ GUARDED_BY(&crit_);
} test(rtp_history_ms);
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) {
ReceivesPliAndRecovers(1000);
}
TEST_F(EndToEndTest, ReceivesPliAndRecoversWithoutNack) {
ReceivesPliAndRecovers(0);
}
TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
class PacketInputObserver : public PacketReceiver {
public:
explicit PacketInputObserver(PacketReceiver* receiver)
: receiver_(receiver), delivered_packet_(EventWrapper::Create()) {}
EventTypeWrapper Wait() {
return delivered_packet_->Wait(kDefaultTimeoutMs);
}
private:
DeliveryStatus DeliverPacket(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) override {
if (RtpHeaderParser::IsRtcp(packet, length)) {
return receiver_->DeliverPacket(media_type, packet, length,
packet_time);
} else {
DeliveryStatus delivery_status =
receiver_->DeliverPacket(media_type, packet, length, packet_time);
EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
delivered_packet_->Set();
return delivery_status;
}
}
PacketReceiver* receiver_;
rtc::scoped_ptr<EventWrapper> delivered_packet_;
};
CreateCalls(Call::Config(), Call::Config());
test::DirectTransport send_transport(sender_call_.get());
test::DirectTransport receive_transport(receiver_call_.get());
PacketInputObserver input_observer(receiver_call_->Receiver());
send_transport.SetReceiver(&input_observer);
receive_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, &send_transport);
CreateMatchingReceiveConfigs(&receive_transport);
CreateStreams();
CreateFrameGeneratorCapturer();
Start();
receiver_call_->DestroyVideoReceiveStream(receive_streams_[0]);
receive_streams_.clear();
// Wait() waits for a received packet.
EXPECT_EQ(kEventSignaled, input_observer.Wait());
Stop();
DestroyStreams();
send_transport.StopSending();
receive_transport.StopSending();
}
void EndToEndTest::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 {
if (++sent_rtp_ % 3 == 0)
return DROP_PACKET;
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
++sent_rtcp_;
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
bool has_report_block = false;
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
EXPECT_NE(RTCPUtility::RTCPPacketTypes::kSr, packet_type);
if (packet_type == RTCPUtility::RTCPPacketTypes::kRr) {
has_report_block = true;
break;
}
packet_type = parser.Iterate();
}
switch (rtcp_mode_) {
case RtcpMode::kCompound:
if (!has_report_block) {
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 (!has_report_block)
observation_complete_->Set();
break;
case RtcpMode::kOff:
RTC_NOTREACHED();
break;
}
return SEND_PACKET;
}
void ModifyConfigs(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_EQ(kEventSignaled, Wait())
<< (rtcp_mode_ == RtcpMode::kCompound
? "Timed out before observing enough compound packets."
: "Timed out before receiving a non-compound RTCP packet.");
}
RtcpMode rtcp_mode_;
int sent_rtp_;
int sent_rtcp_;
} test(rtcp_mode);
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, UsesRtcpCompoundMode) {
RespectsRtcpMode(RtcpMode::kCompound);
}
TEST_F(EndToEndTest, UsesRtcpReducedSizeMode) {
RespectsRtcpMode(RtcpMode::kReducedSize);
}
// Test sets up a Call multiple senders with different resolutions and SSRCs.
// Another is set up to receive all three of these with different renderers.
class MultiStreamTest {
public:
static const size_t kNumStreams = 3;
struct CodecSettings {
uint32_t ssrc;
int width;
int height;
} codec_settings[kNumStreams];
MultiStreamTest() {
// TODO(sprang): Cleanup when msvc supports explicit initializers for array.
codec_settings[0] = {1, 640, 480};
codec_settings[1] = {2, 320, 240};
codec_settings[2] = {3, 240, 160};
}
virtual ~MultiStreamTest() {}
void RunTest() {
rtc::scoped_ptr<Call> sender_call(Call::Create(Call::Config()));
rtc::scoped_ptr<Call> receiver_call(Call::Create(Call::Config()));
rtc::scoped_ptr<test::DirectTransport> sender_transport(
CreateSendTransport(sender_call.get()));
rtc::scoped_ptr<test::DirectTransport> receiver_transport(
CreateReceiveTransport(receiver_call.get()));
sender_transport->SetReceiver(receiver_call->Receiver());
receiver_transport->SetReceiver(sender_call->Receiver());
rtc::scoped_ptr<VideoEncoder> encoders[kNumStreams];
for (size_t i = 0; i < kNumStreams; ++i)
encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8));
VideoSendStream* send_streams[kNumStreams];
VideoReceiveStream* receive_streams[kNumStreams];
test::FrameGeneratorCapturer* frame_generators[kNumStreams];
ScopedVector<VideoDecoder> allocated_decoders;
for (size_t i = 0; i < kNumStreams; ++i) {
uint32_t ssrc = codec_settings[i].ssrc;
int width = codec_settings[i].width;
int height = codec_settings[i].height;
VideoSendStream::Config send_config(sender_transport.get());
send_config.rtp.ssrcs.push_back(ssrc);
send_config.encoder_settings.encoder = encoders[i].get();
send_config.encoder_settings.payload_name = "VP8";
send_config.encoder_settings.payload_type = 124;
VideoEncoderConfig encoder_config;
encoder_config.streams = test::CreateVideoStreams(1);
VideoStream* stream = &encoder_config.streams[0];
stream->width = width;
stream->height = height;
stream->max_framerate = 5;
stream->min_bitrate_bps = stream->target_bitrate_bps =
stream->max_bitrate_bps = 100000;
UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
send_streams[i] =
sender_call->CreateVideoSendStream(send_config, encoder_config);
send_streams[i]->Start();
VideoReceiveStream::Config receive_config(receiver_transport.get());
receive_config.rtp.remote_ssrc = ssrc;
receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalSsrc;
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(send_config.encoder_settings);
allocated_decoders.push_back(decoder.decoder);
receive_config.decoders.push_back(decoder);
UpdateReceiveConfig(i, &receive_config);
receive_streams[i] =
receiver_call->CreateVideoReceiveStream(receive_config);
receive_streams[i]->Start();
frame_generators[i] = test::FrameGeneratorCapturer::Create(
send_streams[i]->Input(), width, height, 30,
Clock::GetRealTimeClock());
frame_generators[i]->Start();
}
Wait();
for (size_t i = 0; i < kNumStreams; ++i) {
frame_generators[i]->Stop();
sender_call->DestroyVideoSendStream(send_streams[i]);
receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
delete frame_generators[i];
}
sender_transport->StopSending();
receiver_transport->StopSending();
}
protected:
virtual void Wait() = 0;
// Note: frame_generator is a point-to-pointer, since the actual instance
// hasn't been created at the time of this call. Only when packets/frames
// start flowing should this be dereferenced.
virtual void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) {}
virtual void UpdateReceiveConfig(size_t stream_index,
VideoReceiveStream::Config* receive_config) {
}
virtual test::DirectTransport* CreateSendTransport(Call* sender_call) {
return new test::DirectTransport(sender_call);
}
virtual test::DirectTransport* CreateReceiveTransport(Call* receiver_call) {
return new test::DirectTransport(receiver_call);
}
};
// Each renderer verifies that it receives the expected resolution, and as soon
// as every renderer has received a frame, the test finishes.
TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
class VideoOutputObserver : public VideoRenderer {
public:
VideoOutputObserver(const MultiStreamTest::CodecSettings& settings,
uint32_t ssrc,
test::FrameGeneratorCapturer** frame_generator)
: settings_(settings),
ssrc_(ssrc),
frame_generator_(frame_generator),
done_(EventWrapper::Create()) {}
void RenderFrame(const VideoFrame& video_frame,
int time_to_render_ms) override {
EXPECT_EQ(settings_.width, video_frame.width());
EXPECT_EQ(settings_.height, video_frame.height());
(*frame_generator_)->Stop();
done_->Set();
}
uint32_t Ssrc() { return ssrc_; }
bool IsTextureSupported() const override { return false; }
EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
private:
const MultiStreamTest::CodecSettings& settings_;
const uint32_t ssrc_;
test::FrameGeneratorCapturer** const frame_generator_;
rtc::scoped_ptr<EventWrapper> done_;
};
class Tester : public MultiStreamTest {
public:
Tester() {}
virtual ~Tester() {}
protected:
void Wait() override {
for (const auto& observer : observers_) {
EXPECT_EQ(EventTypeWrapper::kEventSignaled, observer->Wait())
<< "Time out waiting for from on ssrc " << observer->Ssrc();
}
}
void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
observers_[stream_index].reset(new VideoOutputObserver(
codec_settings[stream_index], send_config->rtp.ssrcs.front(),
frame_generator));
}
void UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStream::Config* receive_config) override {
receive_config->renderer = observers_[stream_index].get();
}
private:
rtc::scoped_ptr<VideoOutputObserver> observers_[kNumStreams];
} tester;
tester.RunTest();
}
TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {
static const int kExtensionId = 5;
class RtpExtensionHeaderObserver : public test::DirectTransport {
public:
RtpExtensionHeaderObserver(Call* sender_call,
const uint32_t& first_media_ssrc,
const std::map<uint32_t, uint32_t>& ssrc_map)
: DirectTransport(sender_call),
done_(EventWrapper::Create()),
parser_(RtpHeaderParser::Create()),
first_media_ssrc_(first_media_ssrc),
rtx_to_media_ssrcs_(ssrc_map),
padding_observed_(false),
rtx_padding_observed_(false),
retransmit_observed_(false),
started_(false) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kExtensionId);
}
virtual ~RtpExtensionHeaderObserver() {}
bool SendRtp(const uint8_t* data,
size_t length,
const PacketOptions& options) override {
{
rtc::CritScope cs(&lock_);
if (IsDone())
return false;
if (started_) {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(data, length, &header));
bool drop_packet = false;
EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
EXPECT_EQ(options.packet_id,
header.extension.transportSequenceNumber);
if (!streams_observed_.empty()) {
// Unwrap packet id and verify uniqueness.
int64_t packet_id = unwrapper_.Unwrap(options.packet_id);
EXPECT_TRUE(received_packed_ids_.insert(packet_id).second);
}
// Drop (up to) every 17th packet, so we get retransmits.
// Only drop media, and not on the first stream (otherwise it will be
// hard to distinguish from padding, which is always sent on the first
// stream).
if (header.payloadType != kSendRtxPayloadType &&
header.ssrc != first_media_ssrc_ &&
header.extension.transportSequenceNumber % 17 == 0) {
dropped_seq_[header.ssrc].insert(header.sequenceNumber);
drop_packet = true;
}
size_t payload_length =
length - (header.headerLength + header.paddingLength);
if (payload_length == 0) {
padding_observed_ = true;
} else if (header.payloadType == kSendRtxPayloadType) {
uint16_t original_sequence_number =
ByteReader<uint16_t>::ReadBigEndian(&data[header.headerLength]);
uint32_t original_ssrc =
rtx_to_media_ssrcs_.find(header.ssrc)->second;
std::set<uint16_t>* seq_no_map = &dropped_seq_[original_ssrc];
auto it = seq_no_map->find(original_sequence_number);
if (it != seq_no_map->end()) {
retransmit_observed_ = true;
seq_no_map->erase(it);
} else {
rtx_padding_observed_ = true;
}
} else {
streams_observed_.insert(header.ssrc);
}
if (IsDone())
done_->Set();
if (drop_packet)
return true;
}
}
return test::DirectTransport::SendRtp(data, length, options);
}
bool IsDone() {
bool observed_types_ok =
streams_observed_.size() == MultiStreamTest::kNumStreams &&
padding_observed_ && retransmit_observed_ && rtx_padding_observed_;
if (!observed_types_ok)
return false;
// We should not have any gaps in the sequence number range.
size_t seqno_range =
*received_packed_ids_.rbegin() - *received_packed_ids_.begin() + 1;
return seqno_range == received_packed_ids_.size();
}
EventTypeWrapper Wait() {
{
// Can't be sure until this point that rtx_to_media_ssrcs_ etc have
// been initialized and are OK to read.
rtc::CritScope cs(&lock_);
started_ = true;
}
return done_->Wait(kDefaultTimeoutMs);
}
rtc::CriticalSection lock_;
rtc::scoped_ptr<EventWrapper> done_;
rtc::scoped_ptr<RtpHeaderParser> parser_;
SequenceNumberUnwrapper unwrapper_;
std::set<int64_t> received_packed_ids_;
std::set<uint32_t> streams_observed_;
std::map<uint32_t, std::set<uint16_t>> dropped_seq_;
const uint32_t& first_media_ssrc_;
const std::map<uint32_t, uint32_t>& rtx_to_media_ssrcs_;
bool padding_observed_;
bool rtx_padding_observed_;
bool retransmit_observed_;
bool started_;
};
class TransportSequenceNumberTester : public MultiStreamTest {
public:
TransportSequenceNumberTester()
: first_media_ssrc_(0), observer_(nullptr) {}
virtual ~TransportSequenceNumberTester() {}
protected:
void Wait() override {
RTC_DCHECK(observer_ != nullptr);
EXPECT_EQ(EventTypeWrapper::kEventSignaled, observer_->Wait());
}
void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
// Force some padding to be sent.
const int kPaddingBitrateBps = 50000;
int total_target_bitrate = 0;
for (const VideoStream& stream : encoder_config->streams)
total_target_bitrate += stream.target_bitrate_bps;
encoder_config->min_transmit_bitrate_bps =
total_target_bitrate + kPaddingBitrateBps;
// Configure RTX for redundant payload padding.
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[stream_index]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
rtx_to_media_ssrcs_[kSendRtxSsrcs[stream_index]] =
send_config->rtp.ssrcs[0];
if (stream_index == 0)
first_media_ssrc_ = send_config->rtp.ssrcs[0];
}
void UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStream::Config* receive_config) override {
receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
receive_config->rtp.extensions.clear();
receive_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
}
test::DirectTransport* CreateSendTransport(Call* sender_call) override {
observer_ = new RtpExtensionHeaderObserver(sender_call, first_media_ssrc_,
rtx_to_media_ssrcs_);
return observer_;
}
private:
uint32_t first_media_ssrc_;
std::map<uint32_t, uint32_t> rtx_to_media_ssrcs_;
RtpExtensionHeaderObserver* observer_;
} tester;
tester.RunTest();
}
void TransportFeedbackTest(bool feedback_enabled) {
static const int kExtensionId = 5;
class TransportFeedbackObserver : public test::DirectTransport {
public:
TransportFeedbackObserver(Call* receiver_call, rtc::Event* done_event)
: DirectTransport(receiver_call), done_(done_event) {}
virtual ~TransportFeedbackObserver() {}
bool SendRtcp(const uint8_t* data, size_t length) override {
RTCPUtility::RTCPParserV2 parser(data, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback) {
done_->Set();
break;
}
packet_type = parser.Iterate();
}
return test::DirectTransport::SendRtcp(data, length);
}
rtc::Event* done_;
};
class TransportFeedbackTester : public MultiStreamTest {
public:
explicit TransportFeedbackTester(bool feedback_enabled)
: feedback_enabled_(feedback_enabled), done_(false, false) {}
virtual ~TransportFeedbackTester() {}
protected:
void Wait() override {
const int64_t kDisabledFeedbackTimeoutMs = 5000;
EXPECT_EQ(feedback_enabled_, done_.Wait(feedback_enabled_
? test::CallTest::kDefaultTimeoutMs
: kDisabledFeedbackTimeoutMs));
}
void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
}
void UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStream::Config* receive_config) override {
receive_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
receive_config->rtp.transport_cc = feedback_enabled_;
}
test::DirectTransport* CreateReceiveTransport(
Call* receiver_call) override {
return new TransportFeedbackObserver(receiver_call, &done_);
}
private:
const bool feedback_enabled_;
rtc::Event done_;
} tester(feedback_enabled);
tester.RunTest();
}
TEST_F(EndToEndTest, ReceivesTransportFeedback) {
TransportFeedbackTest(true);
}
TEST_F(EndToEndTest, TransportFeedbackNotConfigured) {
TransportFeedbackTest(false);
}
TEST_F(EndToEndTest, ObserversEncodedFrames) {
class EncodedFrameTestObserver : public EncodedFrameObserver {
public:
EncodedFrameTestObserver()
: length_(0),
frame_type_(kEmptyFrame),
called_(EventWrapper::Create()) {}
virtual ~EncodedFrameTestObserver() {}
virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
frame_type_ = encoded_frame.frame_type_;
length_ = encoded_frame.length_;
buffer_.reset(new uint8_t[length_]);
memcpy(buffer_.get(), encoded_frame.data_, length_);
called_->Set();
}
EventTypeWrapper Wait() { return called_->Wait(kDefaultTimeoutMs); }
void ExpectEqualFrames(const EncodedFrameTestObserver& observer) {
ASSERT_EQ(length_, observer.length_)
<< "Observed frames are of different lengths.";
EXPECT_EQ(frame_type_, observer.frame_type_)
<< "Observed frames have different frame types.";
EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_))
<< "Observed encoded frames have different content.";
}
private:
rtc::scoped_ptr<uint8_t[]> buffer_;
size_t length_;
FrameType frame_type_;
rtc::scoped_ptr<EventWrapper> called_;
};
EncodedFrameTestObserver post_encode_observer;
EncodedFrameTestObserver pre_decode_observer;
CreateCalls(Call::Config(), Call::Config());
test::DirectTransport sender_transport(sender_call_.get());
test::DirectTransport receiver_transport(receiver_call_.get());
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
send_config_.post_encode_callback = &post_encode_observer;
receive_configs_[0].pre_decode_callback = &pre_decode_observer;
CreateStreams();
Start();
rtc::scoped_ptr<test::FrameGenerator> frame_generator(
test::FrameGenerator::CreateChromaGenerator(
encoder_config_.streams[0].width, encoder_config_.streams[0].height));
send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
EXPECT_EQ(kEventSignaled, post_encode_observer.Wait())
<< "Timed out while waiting for send-side encoded-frame callback.";
EXPECT_EQ(kEventSignaled, pre_decode_observer.Wait())
<< "Timed out while waiting for pre-decode encoded-frame callback.";
post_encode_observer.ExpectEqualFrames(pre_decode_observer);
Stop();
sender_transport.StopSending();
receiver_transport.StopSending();
DestroyStreams();
}
TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
class RembObserver : public test::EndToEndTest {
public:
RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
bool received_psfb = false;
bool received_remb = false;
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRemb) {
const RTCPUtility::RTCPPacket& packet = parser.Packet();
EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc);
received_psfb = true;
} else if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRembItem) {
const RTCPUtility::RTCPPacket& packet = parser.Packet();
EXPECT_GT(packet.REMBItem.BitRate, 0u);
EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u);
EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]);
received_remb = true;
}
packet_type = parser.Iterate();
}
if (received_psfb && received_remb)
observation_complete_->Set();
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for a "
"receiver RTCP REMB packet to be "
"sent.";
}
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, VerifyBandwidthStats) {
class RtcpObserver : public test::EndToEndTest {
public:
RtcpObserver()
: EndToEndTest(kDefaultTimeoutMs),
sender_call_(nullptr),
receiver_call_(nullptr),
has_seen_pacer_delay_(false) {}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
Call::Stats sender_stats = sender_call_->GetStats();
Call::Stats receiver_stats = receiver_call_->GetStats();
if (!has_seen_pacer_delay_)
has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
if (sender_stats.send_bandwidth_bps > 0 &&
receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_) {
observation_complete_->Set();
}
return SEND_PACKET;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
"non-zero bandwidth stats.";
}
private:
Call* sender_call_;
Call* receiver_call_;
bool has_seen_pacer_delay_;
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, VerifyNackStats) {
static const int kPacketNumberToDrop = 200;
class NackObserver : public test::EndToEndTest {
public:
NackObserver()
: EndToEndTest(kLongTimeoutMs),
sent_rtp_packets_(0),
dropped_rtp_packet_(0),
dropped_rtp_packet_requested_(false),
send_stream_(nullptr),
start_runtime_ms_(-1) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
if (++sent_rtp_packets_ == kPacketNumberToDrop) {
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
RTPHeader header;
EXPECT_TRUE(parser->Parse(packet, length, &header));
dropped_rtp_packet_ = header.sequenceNumber;
return DROP_PACKET;
}
VerifyStats();
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
test::RtcpPacketParser rtcp_parser;
rtcp_parser.Parse(packet, length);
std::vector<uint16_t> nacks = rtcp_parser.nack_item()->last_nack_list();
if (!nacks.empty() && std::find(
nacks.begin(), nacks.end(), dropped_rtp_packet_) != nacks.end()) {
dropped_rtp_packet_requested_ = true;
}
return SEND_PACKET;
}
void VerifyStats() EXCLUSIVE_LOCKS_REQUIRED(&crit_) {
if (!dropped_rtp_packet_requested_)
return;
int send_stream_nack_packets = 0;
int receive_stream_nack_packets = 0;
VideoSendStream::Stats stats = send_stream_->GetStats();
for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
stats.substreams.begin(); it != stats.substreams.end(); ++it) {
const VideoSendStream::StreamStats& stream_stats = it->second;
send_stream_nack_packets +=
stream_stats.rtcp_packet_type_counts.nack_packets;
}
for (size_t i = 0; i < receive_streams_.size(); ++i) {
VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
receive_stream_nack_packets +=
stats.rtcp_packet_type_counts.nack_packets;
}
if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
// NACK packet sent on receive stream and received on sent stream.
if (MinMetricRunTimePassed())
observation_complete_->Set();
}
}
bool MinMetricRunTimePassed() {
int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
if (start_runtime_ms_ == -1) {
start_runtime_ms_ = now;
return false;
}
int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
return elapsed_sec > metrics::kMinRunTimeInSeconds;
}
void ModifyConfigs(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;
}
void OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
receive_streams_ = receive_streams;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out waiting for packet to be NACKed.";
}
rtc::CriticalSection crit_;
uint64_t sent_rtp_packets_;
uint16_t dropped_rtp_packet_ GUARDED_BY(&crit_);
bool dropped_rtp_packet_requested_ GUARDED_BY(&crit_);
std::vector<VideoReceiveStream*> receive_streams_;
VideoSendStream* send_stream_;
int64_t start_runtime_ms_;
} test;
test::ClearHistograms();
RunBaseTest(&test, FakeNetworkPipe::Config());
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.UniqueNackRequestsSentInPercent"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
EXPECT_GT(test::LastHistogramSample(
"WebRTC.Video.NackPacketsSentPerMinute"), 0);
EXPECT_GT(test::LastHistogramSample(
"WebRTC.Video.NackPacketsReceivedPerMinute"), 0);
}
void EndToEndTest::VerifyHistogramStats(bool use_rtx,
bool use_red,
bool screenshare) {
class StatsObserver : public test::EndToEndTest {
public:
StatsObserver(bool use_rtx, bool use_red, bool screenshare)
: EndToEndTest(kLongTimeoutMs),
use_rtx_(use_rtx),
use_red_(use_red),
screenshare_(screenshare),
sender_call_(nullptr),
receiver_call_(nullptr),
start_runtime_ms_(-1) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (MinMetricRunTimePassed())
observation_complete_->Set();
// GetStats calls GetSendChannelRtcpStatistics
// (via VideoSendStream::GetRtt) which updates ReportBlockStats used by
// WebRTC.Video.SentPacketsLostInPercent.
// TODO(asapersson): Remove dependency on calling GetStats.
sender_call_->GetStats();
return SEND_PACKET;
}
bool MinMetricRunTimePassed() {
int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
if (start_runtime_ms_ == -1) {
start_runtime_ms_ = now;
return false;
}
int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// NACK
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
// FEC
if (use_red_) {
send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
}
// RTX
if (use_rtx_) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
kSendRtxSsrcs[0];
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
kSendRtxPayloadType;
}
encoder_config->content_type =
screenshare_ ? VideoEncoderConfig::ContentType::kScreen
: VideoEncoderConfig::ContentType::kRealtimeVideo;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out waiting for packet to be NACKed.";
}
const bool use_rtx_;
const bool use_red_;
const bool screenshare_;
Call* sender_call_;
Call* receiver_call_;
int64_t start_runtime_ms_;
} test(use_rtx, use_red, screenshare);
test::ClearHistograms();
RunBaseTest(&test, FakeNetworkPipe::Config());
// Delete the call for Call stats to be reported.
sender_call_.reset();
receiver_call_.reset();
std::string video_prefix =
screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
// Verify that stats have been updated once.
EXPECT_EQ(
1, test::NumHistogramSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
EXPECT_EQ(1,
test::NumHistogramSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.BitrateReceivedInKbps"));
EXPECT_EQ(
1, test::NumHistogramSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.PacerBitrateInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.NackPacketsReceivedPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.FirPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.FirPacketsReceivedPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PliPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PliPacketsReceivedPerMinute"));
EXPECT_EQ(
1, test::NumHistogramSamples(video_prefix + "KeyFramesSentInPermille"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.KeyFramesReceivedInPermille"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.SentPacketsLostInPercent"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.ReceivedPacketsLostInPercent"));
EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "InputWidthInPixels"));
EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "InputHeightInPixels"));
EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentWidthInPixels"));
EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentHeightInPixels"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.ReceivedWidthInPixels"));
EXPECT_EQ(1,
test::NumHistogramSamples("WebRTC.Video.ReceivedHeightInPixels"));
EXPECT_EQ(static_cast<int>(encoder_config_.streams[0].width),
test::LastHistogramSample(video_prefix + "InputWidthInPixels"));
EXPECT_EQ(static_cast<int>(encoder_config_.streams[0].height),
test::LastHistogramSample(video_prefix + "InputHeightInPixels"));
EXPECT_EQ(static_cast<int>(encoder_config_.streams[0].width),
test::LastHistogramSample(video_prefix + "SentWidthInPixels"));
EXPECT_EQ(static_cast<int>(encoder_config_.streams[0].height),
test::LastHistogramSample(video_prefix + "SentHeightInPixels"));
EXPECT_EQ(static_cast<int>(encoder_config_.streams[0].width),
test::LastHistogramSample("WebRTC.Video.ReceivedWidthInPixels"));
EXPECT_EQ(static_cast<int>(encoder_config_.streams[0].height),
test::LastHistogramSample("WebRTC.Video.ReceivedHeightInPixels"));
EXPECT_EQ(1,
test::NumHistogramSamples(video_prefix + "InputFramesPerSecond"));
EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentFramesPerSecond"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.DecodedFramesPerSecond"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.RenderFramesPerSecond"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.OnewayDelayInMs"));
EXPECT_EQ(
1, test::NumHistogramSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "EncodeTimeInMs"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.DecodeTimeInMs"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.BitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.BitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.MediaBitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.MediaBitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PaddingBitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PaddingBitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.RetransmittedBitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SendSideDelayInMs"));
EXPECT_EQ(1,
test::NumHistogramSamples(video_prefix + "SendSideDelayMaxInMs"));
int num_rtx_samples = use_rtx ? 1 : 0;
EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
"WebRTC.Video.RtxBitrateSentInKbps"));
EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
"WebRTC.Video.RtxBitrateReceivedInKbps"));
int num_red_samples = use_red ? 1 : 0;
EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
"WebRTC.Video.FecBitrateSentInKbps"));
EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
"WebRTC.Video.FecBitrateReceivedInKbps"));
EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
"WebRTC.Video.ReceivedFecPacketsInPercent"));
}
TEST_F(EndToEndTest, VerifyHistogramStatsWithRtx) {
const bool kEnabledRtx = true;
const bool kEnabledRed = false;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_F(EndToEndTest, VerifyHistogramStatsWithRed) {
const bool kEnabledRtx = false;
const bool kEnabledRed = true;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_F(EndToEndTest, VerifyHistogramStatsWithScreenshare) {
const bool kEnabledRtx = false;
const bool kEnabledRed = false;
const bool kScreenshare = true;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
static const int kNumRtcpReportPacketsToObserve = 5;
class RtcpXrObserver : public test::EndToEndTest {
public:
explicit RtcpXrObserver(bool enable_rrtr)
: EndToEndTest(kDefaultTimeoutMs),
enable_rrtr_(enable_rrtr),
sent_rtcp_sr_(0),
sent_rtcp_rr_(0),
sent_rtcp_rrtr_(0),
sent_rtcp_dlrr_(0) {}
private:
// Receive stream should send RR packets (and RRTR packets if enabled).
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
if (packet_type == RTCPUtility::RTCPPacketTypes::kRr) {
++sent_rtcp_rr_;
} else if (packet_type ==
RTCPUtility::RTCPPacketTypes::kXrReceiverReferenceTime) {
++sent_rtcp_rrtr_;
}
EXPECT_NE(packet_type, RTCPUtility::RTCPPacketTypes::kSr);
EXPECT_NE(packet_type,
RTCPUtility::RTCPPacketTypes::kXrDlrrReportBlockItem);
packet_type = parser.Iterate();
}
return SEND_PACKET;
}
// Send stream should send SR packets (and DLRR packets if enabled).
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
rtc::CritScope lock(&crit_);
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
++sent_rtcp_sr_;
} else if (packet_type ==
RTCPUtility::RTCPPacketTypes::kXrDlrrReportBlockItem) {
++sent_rtcp_dlrr_;
}
EXPECT_NE(packet_type,
RTCPUtility::RTCPPacketTypes::kXrReceiverReferenceTime);
packet_type = parser.Iterate();
}
if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) {
if (enable_rrtr_) {
EXPECT_GT(sent_rtcp_rrtr_, 0);
EXPECT_GT(sent_rtcp_dlrr_, 0);
} else {
EXPECT_EQ(0, sent_rtcp_rrtr_);
EXPECT_EQ(0, sent_rtcp_dlrr_);
}
observation_complete_->Set();
}
return SEND_PACKET;
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
(*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
enable_rrtr_;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for RTCP SR/RR packets to be sent.";
}
rtc::CriticalSection crit_;
bool enable_rrtr_;
int sent_rtcp_sr_;
int sent_rtcp_rr_ GUARDED_BY(&crit_);
int sent_rtcp_rrtr_ GUARDED_BY(&crit_);
int sent_rtcp_dlrr_;
} test(enable_rrtr);
RunBaseTest(&test, FakeNetworkPipe::Config());
}
void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
bool send_single_ssrc_first) {
class SendsSetSsrcs : public test::EndToEndTest {
public:
SendsSetSsrcs(const uint32_t* ssrcs,
size_t num_ssrcs,
bool send_single_ssrc_first)
: EndToEndTest(kDefaultTimeoutMs),
num_ssrcs_(num_ssrcs),
send_single_ssrc_first_(send_single_ssrc_first),
ssrcs_to_observe_(num_ssrcs),
expect_single_ssrc_(send_single_ssrc_first),
send_stream_(nullptr) {
for (size_t i = 0; i < num_ssrcs; ++i)
valid_ssrcs_[ssrcs[i]] = true;
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
EXPECT_TRUE(valid_ssrcs_[header.ssrc])
<< "Received unknown SSRC: " << header.ssrc;
if (!valid_ssrcs_[header.ssrc])
observation_complete_->Set();
if (!is_observed_[header.ssrc]) {
is_observed_[header.ssrc] = true;
--ssrcs_to_observe_;
if (expect_single_ssrc_) {
expect_single_ssrc_ = false;
observation_complete_->Set();
}
}
if (ssrcs_to_observe_ == 0)
observation_complete_->Set();
return SEND_PACKET;
}
size_t GetNumStreams() const override { return num_ssrcs_; }
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (num_ssrcs_ > 1) {
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
encoder_config->streams[i].min_bitrate_bps = 10000;
encoder_config->streams[i].target_bitrate_bps = 15000;
encoder_config->streams[i].max_bitrate_bps = 20000;
}
}
encoder_config_all_streams_ = *encoder_config;
if (send_single_ssrc_first_)
encoder_config->streams.resize(1);
}
void OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for "
<< (send_single_ssrc_first_ ? "first SSRC." : "SSRCs.");
if (send_single_ssrc_first_) {
// Set full simulcast and continue with the rest of the SSRCs.
send_stream_->ReconfigureVideoEncoder(encoder_config_all_streams_);
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting on additional SSRCs.";
}
}
private:
std::map<uint32_t, bool> valid_ssrcs_;
std::map<uint32_t, bool> is_observed_;
const size_t num_ssrcs_;
const bool send_single_ssrc_first_;
size_t ssrcs_to_observe_;
bool expect_single_ssrc_;
VideoSendStream* send_stream_;
VideoEncoderConfig encoder_config_all_streams_;
} test(kSendSsrcs, num_ssrcs, send_single_ssrc_first);
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, ReportsSetEncoderRates) {
class EncoderRateStatsTest : public test::EndToEndTest,
public test::FakeEncoder {
public:
EncoderRateStatsTest()
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
send_stream_(nullptr),
bitrate_kbps_(0) {}
void OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
}
int32_t SetRates(uint32_t new_target_bitrate, uint32_t framerate) override {
// Make sure not to trigger on any default zero bitrates.
if (new_target_bitrate == 0)
return 0;
rtc::CritScope lock(&crit_);
bitrate_kbps_ = new_target_bitrate;
observation_complete_->Set();
return 0;
}
void PerformTest() override {
ASSERT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for encoder SetRates() call.";
// Wait for GetStats to report a corresponding bitrate.
for (unsigned int i = 0; i < kDefaultTimeoutMs; ++i) {
VideoSendStream::Stats stats = send_stream_->GetStats();
{
rtc::CritScope lock(&crit_);
if ((stats.target_media_bitrate_bps + 500) / 1000 ==
static_cast<int>(bitrate_kbps_)) {
return;
}
}
SleepMs(1);
}
FAIL()
<< "Timed out waiting for stats reporting the currently set bitrate.";
}
private:
rtc::CriticalSection crit_;
VideoSendStream* send_stream_;
uint32_t bitrate_kbps_ GUARDED_BY(crit_);
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, GetStats) {
static const int kStartBitrateBps = 3000000;
static const int kExpectedRenderDelayMs = 20;
class StatsObserver : public test::EndToEndTest, public I420FrameCallback {
public:
StatsObserver()
: EndToEndTest(kLongTimeoutMs),
send_stream_(nullptr),
expected_send_ssrcs_(),
check_stats_event_(EventWrapper::Create()) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
check_stats_event_->Set();
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
check_stats_event_->Set();
return SEND_PACKET;
}
Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
check_stats_event_->Set();
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
check_stats_event_->Set();
return SEND_PACKET;
}
void FrameCallback(VideoFrame* video_frame) override {
// Ensure that we have at least 5ms send side delay.
int64_t render_time = video_frame->render_time_ms();
if (render_time > 0)
video_frame->set_render_time_ms(render_time - 5);
}
bool CheckReceiveStats() {
for (size_t i = 0; i < receive_streams_.size(); ++i) {
VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
EXPECT_EQ(expected_receive_ssrcs_[i], stats.ssrc);
// Make sure all fields have been populated.
// TODO(pbos): Use CompoundKey if/when we ever know that all stats are
// always filled for all receivers.
receive_stats_filled_["IncomingRate"] |=
stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
receive_stats_filled_["RenderDelayAsHighAsExpected"] |=
stats.render_delay_ms >= kExpectedRenderDelayMs;
receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
receive_stats_filled_["StatisticsUpdated"] |=
stats.rtcp_stats.cumulative_lost != 0 ||
stats.rtcp_stats.extended_max_sequence_number != 0 ||
stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
receive_stats_filled_["DataCountersUpdated"] |=
stats.rtp_stats.transmitted.payload_bytes != 0 ||
stats.rtp_stats.fec.packets != 0 ||
stats.rtp_stats.transmitted.header_bytes != 0 ||
stats.rtp_stats.transmitted.packets != 0 ||
stats.rtp_stats.transmitted.padding_bytes != 0 ||
stats.rtp_stats.retransmitted.packets != 0;
receive_stats_filled_["CodecStats"] |=
stats.target_delay_ms != 0 || stats.discarded_packets != 0;
receive_stats_filled_["FrameCounts"] |=
stats.frame_counts.key_frames != 0 ||
stats.frame_counts.delta_frames != 0;
receive_stats_filled_["CName"] |= !stats.c_name.empty();
receive_stats_filled_["RtcpPacketTypeCount"] |=
stats.rtcp_packet_type_counts.fir_packets != 0 ||
stats.rtcp_packet_type_counts.nack_packets != 0 ||
stats.rtcp_packet_type_counts.pli_packets != 0 ||
stats.rtcp_packet_type_counts.nack_requests != 0 ||
stats.rtcp_packet_type_counts.unique_nack_requests != 0;
assert(stats.current_payload_type == -1 ||
stats.current_payload_type == kFakeSendPayloadType);
receive_stats_filled_["IncomingPayloadType"] |=
stats.current_payload_type == kFakeSendPayloadType;
}
return AllStatsFilled(receive_stats_filled_);
}
bool CheckSendStats() {
RTC_DCHECK(send_stream_ != nullptr);
VideoSendStream::Stats stats = send_stream_->GetStats();
send_stats_filled_["NumStreams"] |=
stats.substreams.size() == expected_send_ssrcs_.size();
send_stats_filled_["CpuOveruseMetrics"] |=
stats.avg_encode_time_ms != 0 || stats.encode_usage_percent != 0;
for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
stats.substreams.begin();
it != stats.substreams.end(); ++it) {
EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
expected_send_ssrcs_.end());
send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
stats.input_frame_rate != 0;
const VideoSendStream::StreamStats& stream_stats = it->second;
send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
stream_stats.rtcp_stats.cumulative_lost != 0 ||
stream_stats.rtcp_stats.extended_max_sequence_number != 0 ||
stream_stats.rtcp_stats.fraction_lost != 0;
send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |=
stream_stats.rtp_stats.fec.packets != 0 ||
stream_stats.rtp_stats.transmitted.padding_bytes != 0 ||
stream_stats.rtp_stats.retransmitted.packets != 0 ||
stream_stats.rtp_stats.transmitted.packets != 0;
send_stats_filled_[CompoundKey("BitrateStatisticsObserver",
it->first)] |=
stream_stats.total_bitrate_bps != 0;
send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
stream_stats.frame_counts.delta_frames != 0 ||
stream_stats.frame_counts.key_frames != 0;
send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
stats.encode_frame_rate != 0;
send_stats_filled_[CompoundKey("Delay", it->first)] |=
stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
// TODO(pbos): Use CompoundKey when the test makes sure that all SSRCs
// report dropped packets.
send_stats_filled_["RtcpPacketTypeCount"] |=
stream_stats.rtcp_packet_type_counts.fir_packets != 0 ||
stream_stats.rtcp_packet_type_counts.nack_packets != 0 ||
stream_stats.rtcp_packet_type_counts.pli_packets != 0 ||
stream_stats.rtcp_packet_type_counts.nack_requests != 0 ||
stream_stats.rtcp_packet_type_counts.unique_nack_requests != 0;
}
return AllStatsFilled(send_stats_filled_);
}
std::string CompoundKey(const char* name, uint32_t ssrc) {
std::ostringstream oss;
oss << name << "_" << ssrc;
return oss.str();
}
bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
for (std::map<std::string, bool>::const_iterator it = stats_map.begin();
it != stats_map.end();
++it) {
if (!it->second)
return false;
}
return true;
}
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
config.bitrate_config.start_bitrate_bps = kStartBitrateBps;
return config;
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->pre_encode_callback = this; // Used to inject delay.
expected_cname_ = send_config->rtp.c_name = "SomeCName";
const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
for (size_t i = 0; i < ssrcs.size(); ++i) {
expected_send_ssrcs_.insert(ssrcs[i]);
expected_receive_ssrcs_.push_back(
(*receive_configs)[i].rtp.remote_ssrc);
(*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs;
}
}
size_t GetNumStreams() const override { return kNumSsrcs; }
void OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
receive_streams_ = receive_streams;
}
void PerformTest() override {
Clock* clock = Clock::GetRealTimeClock();
int64_t now = clock->TimeInMilliseconds();
int64_t stop_time = now + test::CallTest::kLongTimeoutMs;
bool receive_ok = false;
bool send_ok = false;
while (now < stop_time) {
if (!receive_ok)
receive_ok = CheckReceiveStats();
if (!send_ok)
send_ok = CheckSendStats();
if (receive_ok && send_ok)
return;
int64_t time_until_timout_ = stop_time - now;
if (time_until_timout_ > 0)
check_stats_event_->Wait(time_until_timout_);
now = clock->TimeInMilliseconds();
}
ADD_FAILURE() << "Timed out waiting for filled stats.";
for (std::map<std::string, bool>::const_iterator it =
receive_stats_filled_.begin();
it != receive_stats_filled_.end();
++it) {
if (!it->second) {
ADD_FAILURE() << "Missing receive stats: " << it->first;
}
}
for (std::map<std::string, bool>::const_iterator it =
send_stats_filled_.begin();
it != send_stats_filled_.end();
++it) {
if (!it->second) {
ADD_FAILURE() << "Missing send stats: " << it->first;
}
}
}
std::vector<VideoReceiveStream*> receive_streams_;
std::map<std::string, bool> receive_stats_filled_;
VideoSendStream* send_stream_;
std::map<std::string, bool> send_stats_filled_;
std::vector<uint32_t> expected_receive_ssrcs_;
std::set<uint32_t> expected_send_ssrcs_;
std::string expected_cname_;
rtc::scoped_ptr<EventWrapper> check_stats_event_;
} test;
FakeNetworkPipe::Config network_config;
network_config.loss_percent = 5;
RunBaseTest(&test, network_config);
}
TEST_F(EndToEndTest, ReceiverReferenceTimeReportEnabled) {
TestXrReceiverReferenceTimeReport(true);
}
TEST_F(EndToEndTest, ReceiverReferenceTimeReportDisabled) {
TestXrReceiverReferenceTimeReport(false);
}
TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
static const size_t kNumRtpPacketsToSend = 5;
class ReceivedRtpStatsObserver : public test::EndToEndTest {
public:
ReceivedRtpStatsObserver()
: EndToEndTest(kDefaultTimeoutMs),
receive_stream_(nullptr),
sent_rtp_(0) {}
private:
void OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
receive_stream_ = receive_streams[0];
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (sent_rtp_ >= kNumRtpPacketsToSend) {
VideoReceiveStream::Stats stats = receive_stream_->GetStats();
if (kNumRtpPacketsToSend == stats.rtp_stats.transmitted.packets) {
observation_complete_->Set();
}
return DROP_PACKET;
}
++sent_rtp_;
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while verifying number of received RTP packets.";
}
VideoReceiveStream* receive_stream_;
uint32_t sent_rtp_;
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, SendsSetSsrc) { TestSendsSetSsrcs(1, false); }
TEST_F(EndToEndTest, SendsSetSimulcastSsrcs) {
TestSendsSetSsrcs(kNumSsrcs, false);
}
TEST_F(EndToEndTest, CanSwitchToUseAllSsrcs) {
TestSendsSetSsrcs(kNumSsrcs, true);
}
TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
class ObserveRedundantPayloads: public test::EndToEndTest {
public:
ObserveRedundantPayloads()
: EndToEndTest(kDefaultTimeoutMs), ssrcs_to_observe_(kNumSsrcs) {
for (size_t i = 0; i < kNumSsrcs; ++i) {
registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
}
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
if (!registered_rtx_ssrc_[header.ssrc])
return SEND_PACKET;
EXPECT_LE(header.headerLength + header.paddingLength, length);
const bool packet_is_redundant_payload =
header.headerLength + header.paddingLength < length;
if (!packet_is_redundant_payload)
return SEND_PACKET;
if (!observed_redundant_retransmission_[header.ssrc]) {
observed_redundant_retransmission_[header.ssrc] = true;
if (--ssrcs_to_observe_ == 0)
observation_complete_->Set();
}
return SEND_PACKET;
}
size_t GetNumStreams() const override { return kNumSsrcs; }
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
encoder_config->streams[i].min_bitrate_bps = 10000;
encoder_config->streams[i].target_bitrate_bps = 15000;
encoder_config->streams[i].max_bitrate_bps = 20000;
}
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
for (size_t i = 0; i < kNumSsrcs; ++i)
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
// Significantly higher than max bitrates for all video streams -> forcing
// padding to trigger redundant padding on all RTX SSRCs.
encoder_config->min_transmit_bitrate_bps = 100000;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for redundant payloads on all SSRCs.";
}
private:
size_t ssrcs_to_observe_;
std::map<uint32_t, bool> observed_redundant_retransmission_;
std::map<uint32_t, bool> registered_rtx_ssrc_;
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
static const uint32_t kMaxSequenceNumberGap = 100;
static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
class RtpSequenceObserver : public test::RtpRtcpObserver {
public:
explicit RtpSequenceObserver(bool use_rtx)
: test::RtpRtcpObserver(kDefaultTimeoutMs),
ssrcs_to_observe_(kNumSsrcs) {
for (size_t i = 0; i < kNumSsrcs; ++i) {
configured_ssrcs_[kSendSsrcs[i]] = true;
if (use_rtx)
configured_ssrcs_[kSendRtxSsrcs[i]] = true;
}
}
void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
rtc::CritScope lock(&crit_);
ssrc_observed_.clear();
ssrcs_to_observe_ = num_expected_ssrcs;
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
const uint32_t ssrc = header.ssrc;
const uint16_t sequence_number = header.sequenceNumber;
const uint32_t timestamp = header.timestamp;
const bool only_padding =
header.headerLength + header.paddingLength == length;
EXPECT_TRUE(configured_ssrcs_[ssrc])
<< "Received SSRC that wasn't configured: " << ssrc;
std::map<uint32_t, uint16_t>::iterator it =
last_observed_sequence_number_.find(header.ssrc);
if (it == last_observed_sequence_number_.end()) {
last_observed_sequence_number_[ssrc] = sequence_number;
last_observed_timestamp_[ssrc] = timestamp;
} else {
// Verify sequence numbers are reasonably close.
uint32_t extended_sequence_number = sequence_number;
// Check for roll-over.
if (sequence_number < last_observed_sequence_number_[ssrc])
extended_sequence_number += 0xFFFFu + 1;
EXPECT_LE(
extended_sequence_number - last_observed_sequence_number_[ssrc],
kMaxSequenceNumberGap)
<< "Gap in sequence numbers ("
<< last_observed_sequence_number_[ssrc] << " -> " << sequence_number
<< ") too large for SSRC: " << ssrc << ".";
last_observed_sequence_number_[ssrc] = sequence_number;
// TODO(pbos): Remove this check if we ever have monotonically
// increasing timestamps. Right now padding packets add a delta which
// can cause reordering between padding packets and regular packets,
// hence we drop padding-only packets to not flake.
if (only_padding) {
// Verify that timestamps are reasonably close.
uint64_t extended_timestamp = timestamp;
// Check for roll-over.
if (timestamp < last_observed_timestamp_[ssrc])
extended_timestamp += static_cast<uint64_t>(0xFFFFFFFFu) + 1;
EXPECT_LE(extended_timestamp - last_observed_timestamp_[ssrc],
kMaxTimestampGap)
<< "Gap in timestamps (" << last_observed_timestamp_[ssrc]
<< " -> " << timestamp << ") too large for SSRC: " << ssrc << ".";
}
last_observed_timestamp_[ssrc] = timestamp;
}
rtc::CritScope lock(&crit_);
// 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;
}
std::map<uint32_t, uint16_t> last_observed_sequence_number_;
std::map<uint32_t, uint32_t> last_observed_timestamp_;
std::map<uint32_t, bool> configured_ssrcs_;
rtc::CriticalSection crit_;
size_t ssrcs_to_observe_ GUARDED_BY(crit_);
std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_);
} observer(use_rtx);
CreateCalls(Call::Config(), Call::Config());
test::PacketTransport send_transport(sender_call_.get(), &observer,
test::PacketTransport::kSender,
FakeNetworkPipe::Config());
test::PacketTransport receive_transport(nullptr, &observer,
test::PacketTransport::kReceiver,
FakeNetworkPipe::Config());
send_transport.SetReceiver(receiver_call_->Receiver());
receive_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(kNumSsrcs, &send_transport);
if (use_rtx) {
for (size_t i = 0; i < kNumSsrcs; ++i) {
send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
}
send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
}
// Lower bitrates so that all streams send initially.
for (size_t i = 0; i < encoder_config_.streams.size(); ++i) {
encoder_config_.streams[i].min_bitrate_bps = 10000;
encoder_config_.streams[i].target_bitrate_bps = 15000;
encoder_config_.streams[i].max_bitrate_bps = 20000;
}
// Use the same total bitrates when sending a single stream to avoid lowering
// the bitrate estimate and requiring a subsequent rampup.
VideoEncoderConfig one_stream = encoder_config_;
one_stream.streams.resize(1);
for (size_t i = 1; i < encoder_config_.streams.size(); ++i) {
one_stream.streams.front().min_bitrate_bps +=
encoder_config_.streams[i].min_bitrate_bps;
one_stream.streams.front().target_bitrate_bps +=
encoder_config_.streams[i].target_bitrate_bps;
one_stream.streams.front().max_bitrate_bps +=
encoder_config_.streams[i].max_bitrate_bps;
}
CreateMatchingReceiveConfigs(&receive_transport);
CreateStreams();
CreateFrameGeneratorCapturer();
Start();
EXPECT_EQ(kEventSignaled, 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) {
frame_generator_capturer_->Stop();
sender_call_->DestroyVideoSendStream(send_stream_);
// Re-create VideoSendStream with only one stream.
send_stream_ =
sender_call_->CreateVideoSendStream(send_config_, one_stream);
send_stream_->Start();
CreateFrameGeneratorCapturer();
frame_generator_capturer_->Start();
observer.ResetExpectedSsrcs(1);
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timed out waiting for single RTP packet.";
// Reconfigure back to use all streams.
send_stream_->ReconfigureVideoEncoder(encoder_config_);
observer.ResetExpectedSsrcs(kNumSsrcs);
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
// Reconfigure down to one stream.
send_stream_->ReconfigureVideoEncoder(one_stream);
observer.ResetExpectedSsrcs(1);
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timed out waiting for single RTP packet.";
// Reconfigure back to use all streams.
send_stream_->ReconfigureVideoEncoder(encoder_config_);
observer.ResetExpectedSsrcs(kNumSsrcs);
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
}
send_transport.StopSending();
receive_transport.StopSending();
Stop();
DestroyStreams();
}
TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpState) {
TestRtpStatePreservation(false);
}
TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
TestRtpStatePreservation(true);
}
TEST_F(EndToEndTest, RespectsNetworkState) {
// TODO(pbos): Remove accepted downtime packets etc. when signaling network
// down blocks until no more packets will be sent.
// Pacer will send from its packet list and then send required padding before
// checking paused_ again. This should be enough for one round of pacing,
// otherwise increase.
static const int kNumAcceptedDowntimeRtp = 5;
// A single RTCP may be in the pipeline.
static const int kNumAcceptedDowntimeRtcp = 1;
class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
public:
NetworkStateTest()
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
encoded_frames_(EventWrapper::Create()),
packet_event_(EventWrapper::Create()),
sender_call_(nullptr),
receiver_call_(nullptr),
sender_state_(kNetworkUp),
sender_rtp_(0),
sender_rtcp_(0),
receiver_rtcp_(0),
down_frames_(0) {}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&test_crit_);
++sender_rtp_;
packet_event_->Set();
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&test_crit_);
++sender_rtcp_;
packet_event_->Set();
return SEND_PACKET;
}
Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&test_crit_);
++receiver_rtcp_;
packet_event_->Set();
return SEND_PACKET;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, encoded_frames_->Wait(kDefaultTimeoutMs))
<< "No frames received by the encoder.";
// Wait for packets from both sender/receiver.
WaitForPacketsOrSilence(false, false);
// Sender-side network down.
sender_call_->SignalNetworkState(kNetworkDown);
{
rtc::CritScope lock(&test_crit_);
// After network goes down we shouldn't be encoding more frames.
sender_state_ = kNetworkDown;
}
// Wait for receiver-packets and no sender packets.
WaitForPacketsOrSilence(true, false);
// Receiver-side network down.
receiver_call_->SignalNetworkState(kNetworkDown);
WaitForPacketsOrSilence(true, true);
// Network back up again for both.
{
rtc::CritScope lock(&test_crit_);
// It's OK to encode frames again, as we're about to bring up the
// network.
sender_state_ = kNetworkUp;
}
sender_call_->SignalNetworkState(kNetworkUp);
receiver_call_->SignalNetworkState(kNetworkUp);
WaitForPacketsOrSilence(false, false);
}
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) override {
{
rtc::CritScope lock(&test_crit_);
if (sender_state_ == kNetworkDown) {
++down_frames_;
EXPECT_LE(down_frames_, 1)
<< "Encoding more than one frame while network is down.";
if (down_frames_ > 1)
encoded_frames_->Set();
} else {
encoded_frames_->Set();
}
}
return test::FakeEncoder::Encode(
input_image, codec_specific_info, frame_types);
}
private:
void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
int64_t initial_time_ms = clock_->TimeInMilliseconds();
int initial_sender_rtp;
int initial_sender_rtcp;
int initial_receiver_rtcp;
{
rtc::CritScope lock(&test_crit_);
initial_sender_rtp = sender_rtp_;
initial_sender_rtcp = sender_rtcp_;
initial_receiver_rtcp = receiver_rtcp_;
}
bool sender_done = false;
bool receiver_done = false;
while(!sender_done || !receiver_done) {
packet_event_->Wait(kSilenceTimeoutMs);
int64_t time_now_ms = clock_->TimeInMilliseconds();
rtc::CritScope lock(&test_crit_);
if (sender_down) {
ASSERT_LE(sender_rtp_ - initial_sender_rtp, kNumAcceptedDowntimeRtp)
<< "RTP sent during sender-side downtime.";
ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
kNumAcceptedDowntimeRtcp)
<< "RTCP sent during sender-side downtime.";
if (time_now_ms - initial_time_ms >=
static_cast<int64_t>(kSilenceTimeoutMs)) {
sender_done = true;
}
} else {
if (sender_rtp_ > initial_sender_rtp)
sender_done = true;
}
if (receiver_down) {
ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
kNumAcceptedDowntimeRtcp)
<< "RTCP sent during receiver-side downtime.";
if (time_now_ms - initial_time_ms >=
static_cast<int64_t>(kSilenceTimeoutMs)) {
receiver_done = true;
}
} else {
if (receiver_rtcp_ > initial_receiver_rtcp)
receiver_done = true;
}
}
}
rtc::CriticalSection test_crit_;
const rtc::scoped_ptr<EventWrapper> encoded_frames_;
const rtc::scoped_ptr<EventWrapper> packet_event_;
Call* sender_call_;
Call* receiver_call_;
NetworkState sender_state_ GUARDED_BY(test_crit_);
int sender_rtp_ GUARDED_BY(test_crit_);
int sender_rtcp_ GUARDED_BY(test_crit_);
int receiver_rtcp_ GUARDED_BY(test_crit_);
int down_frames_ GUARDED_BY(test_crit_);
} test;
RunBaseTest(&test, FakeNetworkPipe::Config());
}
TEST_F(EndToEndTest, CallReportsRttForSender) {
static const int kSendDelayMs = 30;
static const int kReceiveDelayMs = 70;
CreateCalls(Call::Config(), Call::Config());
FakeNetworkPipe::Config config;
config.queue_delay_ms = kSendDelayMs;
test::DirectTransport sender_transport(config, sender_call_.get());
config.queue_delay_ms = kReceiveDelayMs;
test::DirectTransport receiver_transport(config, receiver_call_.get());
sender_transport.SetReceiver(receiver_call_->Receiver());
receiver_transport.SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, &sender_transport);
CreateMatchingReceiveConfigs(&receiver_transport);
CreateStreams();
CreateFrameGeneratorCapturer();
Start();
int64_t start_time_ms = clock_->TimeInMilliseconds();
while (true) {
Call::Stats stats = sender_call_->GetStats();
ASSERT_GE(start_time_ms + kDefaultTimeoutMs,
clock_->TimeInMilliseconds())
<< "No RTT stats before timeout!";
if (stats.rtt_ms != -1) {
EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs);
break;
}
SleepMs(10);
}
Stop();
DestroyStreams();
}
TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
class UnusedEncoder : public test::FakeEncoder {
public:
UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) override {
ADD_FAILURE() << "Unexpected frame encode.";
return test::FakeEncoder::Encode(
input_image, codec_specific_info, frame_types);
}
};
CreateSenderCall(Call::Config());
sender_call_->SignalNetworkState(kNetworkDown);
UnusedTransport transport;
CreateSendConfig(1, &transport);
UnusedEncoder unused_encoder;
send_config_.encoder_settings.encoder = &unused_encoder;
CreateStreams();
CreateFrameGeneratorCapturer();
Start();
SleepMs(kSilenceTimeoutMs);
Stop();
DestroyStreams();
}
TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {
CreateCalls(Call::Config(), Call::Config());
receiver_call_->SignalNetworkState(kNetworkDown);
test::DirectTransport sender_transport(sender_call_.get());
sender_transport.SetReceiver(receiver_call_->Receiver());
CreateSendConfig(1, &sender_transport);
UnusedTransport transport;
CreateMatchingReceiveConfigs(&transport);
CreateStreams();
CreateFrameGeneratorCapturer();
Start();
SleepMs(kSilenceTimeoutMs);
Stop();
sender_transport.StopSending();
DestroyStreams();
}
void VerifyEmptyNackConfig(const NackConfig& config) {
EXPECT_EQ(0, config.rtp_history_ms)
<< "Enabling NACK requires rtcp-fb: nack negotiation.";
}
void VerifyEmptyFecConfig(const FecConfig& config) {
EXPECT_EQ(-1, config.ulpfec_payload_type)
<< "Enabling FEC requires rtpmap: ulpfec negotiation.";
EXPECT_EQ(-1, config.red_payload_type)
<< "Enabling FEC requires rtpmap: red negotiation.";
EXPECT_EQ(-1, config.red_rtx_payload_type)
<< "Enabling RTX in FEC requires rtpmap: rtx negotiation.";
}
TEST_F(EndToEndTest, VerifyDefaultSendConfigParameters) {
VideoSendStream::Config default_send_config(nullptr);
EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
<< "Enabling NACK require rtcp-fb: nack negotiation.";
EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
<< "Enabling RTX requires rtpmap: rtx negotiation.";
EXPECT_TRUE(default_send_config.rtp.extensions.empty())
<< "Enabling RTP extensions require negotiation.";
VerifyEmptyNackConfig(default_send_config.rtp.nack);
VerifyEmptyFecConfig(default_send_config.rtp.fec);
}
TEST_F(EndToEndTest, VerifyDefaultReceiveConfigParameters) {
VideoReceiveStream::Config default_receive_config(nullptr);
EXPECT_EQ(RtcpMode::kCompound, default_receive_config.rtp.rtcp_mode)
<< "Reduced-size RTCP require rtcp-rsize to be negotiated.";
EXPECT_FALSE(default_receive_config.rtp.remb)
<< "REMB require rtcp-fb: goog-remb to be negotiated.";
EXPECT_FALSE(
default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
<< "RTCP XR settings require rtcp-xr to be negotiated.";
EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
<< "Enabling RTX requires rtpmap: rtx negotiation.";
EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
<< "Enabling RTP extensions require negotiation.";
VerifyEmptyNackConfig(default_receive_config.rtp.nack);
VerifyEmptyFecConfig(default_receive_config.rtp.fec);
}
} // namespace webrtc