webrtc_m130/call/bitrate_estimator_tests.cc
Artem Titov dd2eebef5e Deprecate two DirectTransport ctors and remove their direct usage.
Because DirectTransport is switched on SimulatedPacketReceiverInterface
we can't create it from some specific config in ctor, so all ctors,
that accept specific config are deprecated and you should pass concrete
implementation of underlying implememntation instead.

Bug: webrtc:9630
Change-Id: I7f241f310c993d8136b40898e55a6915924d61bd
Reviewed-on: https://webrtc-review.googlesource.com/94841
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24344}
2018-08-20 11:47:28 +00:00

324 lines
12 KiB
C++

/*
* 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 "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/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:
Callback() : done_(false, false) {}
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_.size() <= 0) {
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() {
task_queue_.SendTask([this]() {
CreateCalls();
send_transport_.reset(new test::DirectTransport(
&task_queue_,
absl::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
DefaultNetworkSimulationConfig())),
sender_call_.get(), payload_type_map_));
send_transport_->SetReceiver(receiver_call_->Receiver());
receive_transport_.reset(new test::DirectTransport(
&task_queue_,
absl::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
DefaultNetworkSimulationConfig())),
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.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.remb = true;
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
});
}
virtual void TearDown() {
task_queue_.SendTask([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_(),
fake_decoder_() {
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_.reset(test::FrameGeneratorCapturer::Create(
kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt,
kDefaultFramerate, Clock::GetRealTimeClock()));
send_stream_->SetSource(frame_generator_capturer_.get(),
DegradationPreference::MAINTAIN_FRAMERATE);
send_stream_->Start();
frame_generator_capturer_->Start();
VideoReceiveStream::Decoder decoder;
decoder.decoder = &fake_decoder_;
decoder.payload_type = test_->GetVideoSendConfig()->rtp.payload_type;
decoder.payload_name = 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_) {
frame_generator_capturer_->Stop();
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::FakeDecoder fake_decoder_;
};
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) {
task_queue_.SendTask([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) {
task_queue_.SendTask([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) {
task_queue_.SendTask([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());
task_queue_.SendTask([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) {
task_queue_.SendTask([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());
task_queue_.SendTask([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());
task_queue_.SendTask([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