blob: 1b631ae5db12daac1fdee18654d4268c093591a3 [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 <stddef.h>
#include <stdint.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_encoder_config.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/rtp_config.h"
#include "call/simulated_network.h"
#include "call/simulated_packet_receiver.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
#include "rtc_base/event.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
#include "test/rtp_rtcp_observer.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kColorSpaceExtensionId = 1,
kTransportSequenceNumberExtensionId,
};
} // namespace
class ExtendedReportsEndToEndTest : public test::CallTest {
public:
ExtendedReportsEndToEndTest() {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
}
};
class RtcpXrObserver : public test::EndToEndTest {
public:
RtcpXrObserver(bool enable_rrtr,
bool expect_target_bitrate,
bool enable_zero_target_bitrate,
VideoEncoderConfig::ContentType content_type)
: EndToEndTest(test::CallTest::kDefaultTimeout),
enable_rrtr_(enable_rrtr),
expect_target_bitrate_(expect_target_bitrate),
enable_zero_target_bitrate_(enable_zero_target_bitrate),
content_type_(content_type),
sent_rtcp_sr_(0),
sent_rtcp_rr_(0),
sent_rtcp_rrtr_(0),
sent_rtcp_target_bitrate_(false),
sent_zero_rtcp_target_bitrate_(false),
sent_rtcp_dlrr_(0),
send_simulated_network_(nullptr) {
forward_transport_config_.link_capacity_kbps = 500;
forward_transport_config_.queue_delay_ms = 0;
forward_transport_config_.loss_percent = 0;
}
private:
// Receive stream should send RR packets (and RRTR packets if enabled).
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
sent_rtcp_rr_ += parser.receiver_report()->num_packets();
EXPECT_EQ(0, parser.sender_report()->num_packets());
EXPECT_GE(1, parser.xr()->num_packets());
if (parser.xr()->num_packets() > 0) {
if (parser.xr()->rrtr())
++sent_rtcp_rrtr_;
EXPECT_FALSE(parser.xr()->dlrr());
}
return SEND_PACKET;
}
// Send stream should send SR packets (and DLRR packets if enabled).
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
if (parser.sender_ssrc() == test::CallTest::kVideoSendSsrcs[1] &&
enable_zero_target_bitrate_) {
// Reduce bandwidth restriction to disable second stream after it was
// enabled for some time.
forward_transport_config_.link_capacity_kbps = 200;
send_simulated_network_->SetConfig(forward_transport_config_);
}
sent_rtcp_sr_ += parser.sender_report()->num_packets();
EXPECT_LE(parser.xr()->num_packets(), 1);
if (parser.xr()->num_packets() > 0) {
EXPECT_FALSE(parser.xr()->rrtr());
if (parser.xr()->dlrr())
++sent_rtcp_dlrr_;
if (parser.xr()->target_bitrate()) {
sent_rtcp_target_bitrate_ = true;
auto target_bitrates =
parser.xr()->target_bitrate()->GetTargetBitrates();
if (target_bitrates.empty()) {
sent_zero_rtcp_target_bitrate_ = true;
}
for (const rtcp::TargetBitrate::BitrateItem& item : target_bitrates) {
if (item.target_bitrate_kbps == 0) {
sent_zero_rtcp_target_bitrate_ = true;
break;
}
}
}
}
if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve &&
(sent_rtcp_target_bitrate_ || !expect_target_bitrate_) &&
(sent_zero_rtcp_target_bitrate_ || !enable_zero_target_bitrate_)) {
if (enable_rrtr_) {
EXPECT_GT(sent_rtcp_rrtr_, 0);
EXPECT_GT(sent_rtcp_dlrr_, 0);
} else {
EXPECT_EQ(sent_rtcp_rrtr_, 0);
EXPECT_EQ(sent_rtcp_dlrr_, 0);
}
EXPECT_EQ(expect_target_bitrate_, sent_rtcp_target_bitrate_);
EXPECT_EQ(enable_zero_target_bitrate_, sent_zero_rtcp_target_bitrate_);
observation_complete_.Set();
}
return SEND_PACKET;
}
size_t GetNumVideoStreams() const override {
// When sending a zero target bitrate, we use two spatial layers so that
// we'll still have a layer with non-zero bitrate.
return enable_zero_target_bitrate_ ? 2 : 1;
}
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) {
auto network =
std::make_unique<SimulatedNetwork>(forward_transport_config_);
send_simulated_network_ = network.get();
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
test::CallTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network)));
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (enable_zero_target_bitrate_) {
// Configure VP8 to be able to use simulcast.
send_config->rtp.payload_name = "VP8";
encoder_config->codec_type = kVideoCodecVP8;
(*receive_configs)[0].decoders.resize(1);
(*receive_configs)[0].decoders[0].payload_type =
send_config->rtp.payload_type;
(*receive_configs)[0].decoders[0].video_format =
SdpVideoFormat(send_config->rtp.payload_name);
}
encoder_config->content_type = content_type_;
(*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
enable_rrtr_;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for RTCP SR/RR packets to be sent.";
}
static const int kNumRtcpReportPacketsToObserve = 5;
Mutex mutex_;
const bool enable_rrtr_;
const bool expect_target_bitrate_;
const bool enable_zero_target_bitrate_;
const VideoEncoderConfig::ContentType content_type_;
int sent_rtcp_sr_;
int sent_rtcp_rr_ RTC_GUARDED_BY(&mutex_);
int sent_rtcp_rrtr_ RTC_GUARDED_BY(&mutex_);
bool sent_rtcp_target_bitrate_ RTC_GUARDED_BY(&mutex_);
bool sent_zero_rtcp_target_bitrate_ RTC_GUARDED_BY(&mutex_);
int sent_rtcp_dlrr_;
BuiltInNetworkBehaviorConfig forward_transport_config_;
SimulatedNetwork* send_simulated_network_;
};
TEST_F(ExtendedReportsEndToEndTest,
TestExtendedReportsWithRrtrWithoutTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/true, /*expect_target_bitrate=*/false,
/*enable_zero_target_bitrate=*/false,
VideoEncoderConfig::ContentType::kRealtimeVideo);
RunBaseTest(&test);
}
TEST_F(ExtendedReportsEndToEndTest,
TestExtendedReportsWithoutRrtrWithoutTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/false,
/*enable_zero_target_bitrate=*/false,
VideoEncoderConfig::ContentType::kRealtimeVideo);
RunBaseTest(&test);
}
TEST_F(ExtendedReportsEndToEndTest,
TestExtendedReportsWithRrtrWithTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/true, /*expect_target_bitrate=*/true,
/*enable_zero_target_bitrate=*/false,
VideoEncoderConfig::ContentType::kScreen);
RunBaseTest(&test);
}
TEST_F(ExtendedReportsEndToEndTest,
TestExtendedReportsWithoutRrtrWithTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/true,
/*enable_zero_target_bitrate=*/false,
VideoEncoderConfig::ContentType::kScreen);
RunBaseTest(&test);
}
TEST_F(ExtendedReportsEndToEndTest,
TestExtendedReportsWithoutRrtrWithTargetBitrateExplicitlySet) {
test::ScopedKeyValueConfig field_trials(
field_trials_, "WebRTC-Target-Bitrate-Rtcp/Enabled/");
RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/true,
/*enable_zero_target_bitrate=*/false,
VideoEncoderConfig::ContentType::kRealtimeVideo);
RunBaseTest(&test);
}
TEST_F(ExtendedReportsEndToEndTest,
TestExtendedReportsCanSignalZeroTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/true,
/*enable_zero_target_bitrate=*/true,
VideoEncoderConfig::ContentType::kScreen);
RunBaseTest(&test);
}
} // namespace webrtc