blob: 3ee0e2b7ac5a7b3f4dd79d51cf6840f1a0967290 [file] [log] [blame]
/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "absl/memory/memory.h"
#include "api/test/simulated_network.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_bitrate_allocation.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/rate_limiter.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/fake_encoder.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
#include "test/rtp_rtcp_observer.h"
#include "test/video_encoder_proxy_factory.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kAbsSendTimeExtensionId = 1,
};
} // namespace
class BandwidthEndToEndTest : public test::CallTest {
public:
BandwidthEndToEndTest() = default;
};
TEST_F(BandwidthEndToEndTest, ReceiveStreamSendsRemb) {
class RembObserver : public test::EndToEndTest {
public:
RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
(*receive_configs)[0].rtp.remb = true;
(*receive_configs)[0].rtp.transport_cc = false;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
if (parser.remb()->num_packets() > 0) {
EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc());
EXPECT_LT(0U, parser.remb()->bitrate_bps());
EXPECT_EQ(1U, parser.remb()->ssrcs().size());
EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]);
observation_complete_.Set();
}
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
"receiver RTCP REMB packet to be "
"sent.";
}
} test;
RunBaseTest(&test);
}
class BandwidthStatsTest : public test::EndToEndTest {
public:
explicit BandwidthStatsTest(bool send_side_bwe)
: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
sender_call_(nullptr),
receiver_call_(nullptr),
has_seen_pacer_delay_(false),
send_side_bwe_(send_side_bwe) {}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (!send_side_bwe_) {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
(*receive_configs)[0].rtp.remb = true;
(*receive_configs)[0].rtp.transport_cc = 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 && has_seen_pacer_delay_) {
if (send_side_bwe_ || receiver_stats.recv_bandwidth_bps > 0)
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_TRUE(Wait()) << "Timed out while waiting for "
"non-zero bandwidth stats.";
}
private:
Call* sender_call_;
Call* receiver_call_;
bool has_seen_pacer_delay_;
const bool send_side_bwe_;
};
TEST_F(BandwidthEndToEndTest, VerifySendSideBweStats) {
BandwidthStatsTest test(true);
RunBaseTest(&test);
}
TEST_F(BandwidthEndToEndTest, VerifyRecvSideBweStats) {
BandwidthStatsTest test(false);
RunBaseTest(&test);
}
// Verifies that it's possible to limit the send BWE by sending a REMB.
// This is verified by allowing the send BWE to ramp-up to >1000 kbps,
// then have the test generate a REMB of 500 kbps and verify that the send BWE
// is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the
// test verifies that the send BWE ramps back up to exactly 1000 kbps.
TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
class BweObserver : public test::EndToEndTest {
public:
BweObserver()
: EndToEndTest(kDefaultTimeoutMs),
sender_call_(nullptr),
clock_(Clock::GetRealTimeClock()),
sender_ssrc_(0),
remb_bitrate_bps_(1000000),
receive_transport_(nullptr),
poller_thread_(&BitrateStatsPollingThread,
this,
"BitrateStatsPollingThread"),
state_(kWaitForFirstRampUp),
retransmission_rate_limiter_(clock_, 1000) {}
~BweObserver() {}
test::PacketTransport* CreateReceiveTransport(
test::SingleThreadedTaskQueueForTesting* task_queue) override {
receive_transport_ = new test::PacketTransport(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
payload_type_map_,
absl::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())));
return receive_transport_;
}
void ModifySenderBitrateConfig(
BitrateConstraints* bitrate_config) override {
// Set a high start bitrate to reduce the test completion time.
bitrate_config->start_bitrate_bps = remb_bitrate_bps_;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
sender_ssrc_ = send_config->rtp.ssrcs[0];
encoder_config->max_bitrate_bps = 2000000;
ASSERT_EQ(1u, receive_configs->size());
RtpRtcp::Configuration config;
config.receiver_only = true;
config.clock = clock_;
config.outgoing_transport = receive_transport_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_rtcp_ = RtpRtcp::Create(config);
rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
}
static void BitrateStatsPollingThread(void* obj) {
static_cast<BweObserver*>(obj)->PollStats();
}
void PollStats() {
do {
if (sender_call_) {
Call::Stats stats = sender_call_->GetStats();
switch (state_) {
case kWaitForFirstRampUp:
if (stats.send_bandwidth_bps >= remb_bitrate_bps_) {
state_ = kWaitForRemb;
remb_bitrate_bps_ /= 2;
rtp_rtcp_->SetRemb(
remb_bitrate_bps_,
std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
rtp_rtcp_->SendRTCP(kRtcpRr);
}
break;
case kWaitForRemb:
if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
state_ = kWaitForSecondRampUp;
remb_bitrate_bps_ *= 2;
rtp_rtcp_->SetRemb(
remb_bitrate_bps_,
std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
rtp_rtcp_->SendRTCP(kRtcpRr);
}
break;
case kWaitForSecondRampUp:
if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
observation_complete_.Set();
}
break;
}
}
} while (!stop_event_.Wait(1000));
}
void PerformTest() override {
poller_thread_.Start();
EXPECT_TRUE(Wait())
<< "Timed out while waiting for bitrate to change according to REMB.";
stop_event_.Set();
poller_thread_.Stop();
}
private:
enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };
Call* sender_call_;
Clock* const clock_;
uint32_t sender_ssrc_;
int remb_bitrate_bps_;
std::unique_ptr<RtpRtcp> rtp_rtcp_;
test::PacketTransport* receive_transport_;
rtc::Event stop_event_;
rtc::PlatformThread poller_thread_;
TestState state_;
RateLimiter retransmission_rate_limiter_;
} test;
RunBaseTest(&test);
}
TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) {
// If these fields trial are on, we get lower bitrates than expected by this
// test, due to the packetization overhead and encoder pushback.
webrtc::test::ScopedFieldTrials field_trials(
std::string(field_trial::GetFieldTrialString()) +
"WebRTC-SubtractPacketizationOverhead/Disabled/"
"WebRTC-VideoRateControl/bitrate_adjuster:false/");
class EncoderRateStatsTest : public test::EndToEndTest,
public test::FakeEncoder {
public:
explicit EncoderRateStatsTest(
test::SingleThreadedTaskQueueForTesting* task_queue)
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
task_queue_(task_queue),
send_stream_(nullptr),
encoder_factory_(this),
bitrate_allocator_factory_(
CreateBuiltinVideoBitrateAllocatorFactory()),
bitrate_kbps_(0) {}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();
RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
}
int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
uint32_t framerate) override {
// Make sure not to trigger on any default zero bitrates.
if (rate_allocation.get_sum_bps() == 0)
return 0;
rtc::CritScope lock(&crit_);
bitrate_kbps_ = rate_allocation.get_sum_kbps();
observation_complete_.Set();
return 0;
}
void PerformTest() override {
ASSERT_TRUE(Wait())
<< "Timed out while waiting for encoder SetRates() call.";
task_queue_->SendTask([this]() {
WaitForEncoderTargetBitrateMatchStats();
send_stream_->Stop();
WaitForStatsReportZeroTargetBitrate();
send_stream_->Start();
WaitForEncoderTargetBitrateMatchStats();
});
}
void WaitForEncoderTargetBitrateMatchStats() {
for (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.";
}
void WaitForStatsReportZeroTargetBitrate() {
for (int i = 0; i < kDefaultTimeoutMs; ++i) {
if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
return;
}
SleepMs(1);
}
FAIL() << "Timed out waiting for stats reporting zero bitrate.";
}
private:
test::SingleThreadedTaskQueueForTesting* const task_queue_;
rtc::CriticalSection crit_;
VideoSendStream* send_stream_;
test::VideoEncoderProxyFactory encoder_factory_;
std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
} test(&task_queue_);
RunBaseTest(&test);
}
} // namespace webrtc