blob: 50da12bbdf298187e1213d47ee57284b7a7136b0 [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 <functional>
#include <list>
#include <memory>
#include <string>
#include "api/test/create_frame_generator.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread_annotations.h"
#include "test/call_test.h"
#include "test/direct_transport.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/fake_encoder.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
// Note: If you consider to re-use this class, think twice and instead consider
// writing tests that don't depend on the logging system.
class LogObserver {
public:
LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
void PushExpectedLogLine(const std::string& expected_log_line) {
callback_.PushExpectedLogLine(expected_log_line);
}
bool Wait() { return callback_.Wait(); }
private:
class Callback : public rtc::LogSink {
public:
void OnLogMessage(const std::string& message) override {
rtc::CritScope lock(&crit_sect_);
// Ignore log lines that are due to missing AST extensions, these are
// logged when we switch back from AST to TOF until the wrapping bitrate
// estimator gives up on using AST.
if (message.find("BitrateEstimator") != std::string::npos &&
message.find("packet is missing") == std::string::npos) {
received_log_lines_.push_back(message);
}
int num_popped = 0;
while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
std::string a = received_log_lines_.front();
std::string b = expected_log_lines_.front();
received_log_lines_.pop_front();
expected_log_lines_.pop_front();
num_popped++;
EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
}
if (expected_log_lines_.empty()) {
if (num_popped > 0) {
done_.Set();
}
return;
}
}
bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }
void PushExpectedLogLine(const std::string& expected_log_line) {
rtc::CritScope lock(&crit_sect_);
expected_log_lines_.push_back(expected_log_line);
}
private:
typedef std::list<std::string> Strings;
rtc::CriticalSection crit_sect_;
Strings received_log_lines_ RTC_GUARDED_BY(crit_sect_);
Strings expected_log_lines_ RTC_GUARDED_BY(crit_sect_);
rtc::Event done_;
};
Callback callback_;
};
} // namespace
static const int kTOFExtensionId = 4;
static const int kASTExtensionId = 5;
class BitrateEstimatorTest : public test::CallTest {
public:
BitrateEstimatorTest() : receive_config_(nullptr) {}
virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
virtual void SetUp() {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
CreateCalls();
send_transport_.reset(new test::DirectTransport(
task_queue(),
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())),
sender_call_.get(), payload_type_map_));
send_transport_->SetReceiver(receiver_call_->Receiver());
receive_transport_.reset(new test::DirectTransport(
task_queue(),
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())),
receiver_call_.get(), payload_type_map_));
receive_transport_->SetReceiver(sender_call_->Receiver());
VideoSendStream::Config video_send_config(send_transport_.get());
video_send_config.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
video_send_config.encoder_settings.encoder_factory =
&fake_encoder_factory_;
video_send_config.encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();
video_send_config.rtp.payload_name = "FAKE";
video_send_config.rtp.payload_type = kFakeVideoSendPayloadType;
SetVideoSendConfig(video_send_config);
VideoEncoderConfig video_encoder_config;
test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
SetVideoEncoderConfig(video_encoder_config);
receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
// receive_config_.decoders will be set by every stream separately.
receive_config_.rtp.remote_ssrc = GetVideoSendConfig()->rtp.ssrcs[0];
receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
});
}
virtual void TearDown() {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
for (auto* stream : streams_) {
stream->StopSending();
delete stream;
}
streams_.clear();
send_transport_.reset();
receive_transport_.reset();
DestroyCalls();
});
}
protected:
friend class Stream;
class Stream {
public:
explicit Stream(BitrateEstimatorTest* test)
: test_(test),
is_sending_receiving_(false),
send_stream_(nullptr),
frame_generator_capturer_(),
decoder_factory_(
[]() { return std::make_unique<test::FakeDecoder>(); }) {
test_->GetVideoSendConfig()->rtp.ssrcs[0]++;
send_stream_ = test_->sender_call_->CreateVideoSendStream(
test_->GetVideoSendConfig()->Copy(),
test_->GetVideoEncoderConfig()->Copy());
RTC_DCHECK_EQ(1, test_->GetVideoEncoderConfig()->number_of_streams);
frame_generator_capturer_ =
std::make_unique<test::FrameGeneratorCapturer>(
test->clock_,
test::CreateSquareFrameGenerator(kDefaultWidth, kDefaultHeight,
absl::nullopt, absl::nullopt),
kDefaultFramerate, *test->task_queue_factory_);
frame_generator_capturer_->Init();
send_stream_->SetSource(frame_generator_capturer_.get(),
DegradationPreference::MAINTAIN_FRAMERATE);
send_stream_->Start();
VideoReceiveStream::Decoder decoder;
decoder.decoder_factory = &decoder_factory_;
decoder.payload_type = test_->GetVideoSendConfig()->rtp.payload_type;
decoder.video_format =
SdpVideoFormat(test_->GetVideoSendConfig()->rtp.payload_name);
test_->receive_config_.decoders.clear();
test_->receive_config_.decoders.push_back(decoder);
test_->receive_config_.rtp.remote_ssrc =
test_->GetVideoSendConfig()->rtp.ssrcs[0];
test_->receive_config_.rtp.local_ssrc++;
test_->receive_config_.renderer = &test->fake_renderer_;
video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
test_->receive_config_.Copy());
video_receive_stream_->Start();
is_sending_receiving_ = true;
}
~Stream() {
EXPECT_FALSE(is_sending_receiving_);
test_->sender_call_->DestroyVideoSendStream(send_stream_);
frame_generator_capturer_.reset(nullptr);
send_stream_ = nullptr;
if (video_receive_stream_) {
test_->receiver_call_->DestroyVideoReceiveStream(video_receive_stream_);
video_receive_stream_ = nullptr;
}
}
void StopSending() {
if (is_sending_receiving_) {
send_stream_->Stop();
if (video_receive_stream_) {
video_receive_stream_->Stop();
}
is_sending_receiving_ = false;
}
}
private:
BitrateEstimatorTest* test_;
bool is_sending_receiving_;
VideoSendStream* send_stream_;
VideoReceiveStream* video_receive_stream_;
std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
test::FunctionVideoDecoderFactory decoder_factory_;
};
LogObserver receiver_log_;
std::unique_ptr<test::DirectTransport> send_transport_;
std::unique_ptr<test::DirectTransport> receive_transport_;
VideoReceiveStream::Config receive_config_;
std::vector<Stream*> streams_;
};
static const char* kAbsSendTimeLog =
"RemoteBitrateEstimatorAbsSendTime: Instantiating.";
static const char* kSingleStreamLog =
"RemoteBitrateEstimatorSingleStream: Instantiating.";
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
}
// This test is flaky. See webrtc:5790.
TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions[0] =
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine(
"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
streams_.push_back(new Stream(this));
streams_[0]->StopSending();
streams_[1]->StopSending();
});
EXPECT_TRUE(receiver_log_.Wait());
}
} // namespace webrtc