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330 lines
12 KiB
C++
330 lines
12 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <functional>
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#include <list>
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#include <memory>
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#include <string>
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#include "api/test/create_frame_generator.h"
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#include "call/call.h"
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#include "call/fake_network_pipe.h"
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#include "call/simulated_network.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/event.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/task_queue_for_test.h"
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#include "rtc_base/thread_annotations.h"
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#include "test/call_test.h"
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#include "test/direct_transport.h"
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#include "test/encoder_settings.h"
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#include "test/fake_decoder.h"
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#include "test/fake_encoder.h"
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#include "test/frame_generator_capturer.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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// Note: If you consider to re-use this class, think twice and instead consider
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// writing tests that don't depend on the logging system.
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class LogObserver {
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public:
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LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
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~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
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void PushExpectedLogLine(const std::string& expected_log_line) {
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callback_.PushExpectedLogLine(expected_log_line);
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}
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bool Wait() { return callback_.Wait(); }
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private:
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class Callback : public rtc::LogSink {
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public:
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void OnLogMessage(const std::string& message) override {
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MutexLock lock(&mutex_);
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// Ignore log lines that are due to missing AST extensions, these are
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// logged when we switch back from AST to TOF until the wrapping bitrate
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// estimator gives up on using AST.
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if (message.find("BitrateEstimator") != std::string::npos &&
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message.find("packet is missing") == std::string::npos) {
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received_log_lines_.push_back(message);
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}
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int num_popped = 0;
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while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
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std::string a = received_log_lines_.front();
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std::string b = expected_log_lines_.front();
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received_log_lines_.pop_front();
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expected_log_lines_.pop_front();
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num_popped++;
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EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
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}
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if (expected_log_lines_.empty()) {
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if (num_popped > 0) {
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done_.Set();
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}
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return;
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}
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}
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bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }
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void PushExpectedLogLine(const std::string& expected_log_line) {
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MutexLock lock(&mutex_);
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expected_log_lines_.push_back(expected_log_line);
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}
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private:
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typedef std::list<std::string> Strings;
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Mutex mutex_;
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Strings received_log_lines_ RTC_GUARDED_BY(mutex_);
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Strings expected_log_lines_ RTC_GUARDED_BY(mutex_);
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rtc::Event done_;
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};
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Callback callback_;
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};
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} // namespace
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static const int kTOFExtensionId = 4;
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static const int kASTExtensionId = 5;
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class BitrateEstimatorTest : public test::CallTest {
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public:
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BitrateEstimatorTest() : receive_config_(nullptr) {}
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virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
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virtual void SetUp() {
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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CreateCalls();
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send_transport_.reset(new test::DirectTransport(
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task_queue(),
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std::make_unique<FakeNetworkPipe>(
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Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
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BuiltInNetworkBehaviorConfig())),
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sender_call_.get(), payload_type_map_));
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send_transport_->SetReceiver(receiver_call_->Receiver());
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receive_transport_.reset(new test::DirectTransport(
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task_queue(),
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std::make_unique<FakeNetworkPipe>(
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Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
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BuiltInNetworkBehaviorConfig())),
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receiver_call_.get(), payload_type_map_));
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receive_transport_->SetReceiver(sender_call_->Receiver());
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VideoSendStream::Config video_send_config(send_transport_.get());
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video_send_config.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
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video_send_config.encoder_settings.encoder_factory =
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&fake_encoder_factory_;
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video_send_config.encoder_settings.bitrate_allocator_factory =
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bitrate_allocator_factory_.get();
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video_send_config.rtp.payload_name = "FAKE";
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video_send_config.rtp.payload_type = kFakeVideoSendPayloadType;
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SetVideoSendConfig(video_send_config);
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VideoEncoderConfig video_encoder_config;
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test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
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SetVideoEncoderConfig(video_encoder_config);
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receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
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// receive_config_.decoders will be set by every stream separately.
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receive_config_.rtp.remote_ssrc = GetVideoSendConfig()->rtp.ssrcs[0];
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receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
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receive_config_.rtp.extensions.push_back(
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RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
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receive_config_.rtp.extensions.push_back(
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RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
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});
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}
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virtual void TearDown() {
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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for (auto* stream : streams_) {
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stream->StopSending();
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delete stream;
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}
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streams_.clear();
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send_transport_.reset();
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receive_transport_.reset();
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DestroyCalls();
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});
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}
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protected:
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friend class Stream;
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class Stream {
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public:
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explicit Stream(BitrateEstimatorTest* test)
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: test_(test),
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is_sending_receiving_(false),
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send_stream_(nullptr),
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frame_generator_capturer_(),
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decoder_factory_(
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[]() { return std::make_unique<test::FakeDecoder>(); }) {
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test_->GetVideoSendConfig()->rtp.ssrcs[0]++;
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send_stream_ = test_->sender_call_->CreateVideoSendStream(
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test_->GetVideoSendConfig()->Copy(),
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test_->GetVideoEncoderConfig()->Copy());
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RTC_DCHECK_EQ(1, test_->GetVideoEncoderConfig()->number_of_streams);
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frame_generator_capturer_ =
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std::make_unique<test::FrameGeneratorCapturer>(
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test->clock_,
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test::CreateSquareFrameGenerator(kDefaultWidth, kDefaultHeight,
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absl::nullopt, absl::nullopt),
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kDefaultFramerate, *test->task_queue_factory_);
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frame_generator_capturer_->Init();
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send_stream_->SetSource(frame_generator_capturer_.get(),
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DegradationPreference::MAINTAIN_FRAMERATE);
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send_stream_->Start();
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VideoReceiveStream::Decoder decoder;
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test_->receive_config_.decoder_factory = &decoder_factory_;
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decoder.payload_type = test_->GetVideoSendConfig()->rtp.payload_type;
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decoder.video_format =
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SdpVideoFormat(test_->GetVideoSendConfig()->rtp.payload_name);
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test_->receive_config_.decoders.clear();
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test_->receive_config_.decoders.push_back(decoder);
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test_->receive_config_.rtp.remote_ssrc =
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test_->GetVideoSendConfig()->rtp.ssrcs[0];
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test_->receive_config_.rtp.local_ssrc++;
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test_->receive_config_.renderer = &test->fake_renderer_;
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video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
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test_->receive_config_.Copy());
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video_receive_stream_->Start();
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is_sending_receiving_ = true;
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}
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~Stream() {
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EXPECT_FALSE(is_sending_receiving_);
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test_->sender_call_->DestroyVideoSendStream(send_stream_);
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frame_generator_capturer_.reset(nullptr);
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send_stream_ = nullptr;
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if (video_receive_stream_) {
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test_->receiver_call_->DestroyVideoReceiveStream(video_receive_stream_);
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video_receive_stream_ = nullptr;
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}
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}
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void StopSending() {
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if (is_sending_receiving_) {
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send_stream_->Stop();
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if (video_receive_stream_) {
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video_receive_stream_->Stop();
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}
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is_sending_receiving_ = false;
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}
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}
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private:
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BitrateEstimatorTest* test_;
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bool is_sending_receiving_;
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VideoSendStream* send_stream_;
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VideoReceiveStream* video_receive_stream_;
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std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
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test::FunctionVideoDecoderFactory decoder_factory_;
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};
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LogObserver receiver_log_;
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std::unique_ptr<test::DirectTransport> send_transport_;
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std::unique_ptr<test::DirectTransport> receive_transport_;
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VideoReceiveStream::Config receive_config_;
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std::vector<Stream*> streams_;
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};
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static const char* kAbsSendTimeLog =
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"RemoteBitrateEstimatorAbsSendTime: Instantiating.";
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static const char* kSingleStreamLog =
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"RemoteBitrateEstimatorSingleStream: Instantiating.";
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TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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GetVideoSendConfig()->rtp.extensions.push_back(
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RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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streams_.push_back(new Stream(this));
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});
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EXPECT_TRUE(receiver_log_.Wait());
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}
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TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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GetVideoSendConfig()->rtp.extensions.push_back(
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RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
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receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
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streams_.push_back(new Stream(this));
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});
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EXPECT_TRUE(receiver_log_.Wait());
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}
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TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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GetVideoSendConfig()->rtp.extensions.push_back(
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RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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streams_.push_back(new Stream(this));
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});
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EXPECT_TRUE(receiver_log_.Wait());
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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GetVideoSendConfig()->rtp.extensions[0] =
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RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
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receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
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receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
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streams_.push_back(new Stream(this));
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});
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EXPECT_TRUE(receiver_log_.Wait());
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}
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// This test is flaky. See webrtc:5790.
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TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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GetVideoSendConfig()->rtp.extensions.push_back(
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RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
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receiver_log_.PushExpectedLogLine(kSingleStreamLog);
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streams_.push_back(new Stream(this));
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});
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EXPECT_TRUE(receiver_log_.Wait());
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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GetVideoSendConfig()->rtp.extensions[0] =
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RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
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receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
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receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
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streams_.push_back(new Stream(this));
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});
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EXPECT_TRUE(receiver_log_.Wait());
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SendTask(RTC_FROM_HERE, task_queue(), [this]() {
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GetVideoSendConfig()->rtp.extensions[0] =
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RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
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receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
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receiver_log_.PushExpectedLogLine(
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"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
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streams_.push_back(new Stream(this));
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streams_[0]->StopSending();
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streams_[1]->StopSending();
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});
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EXPECT_TRUE(receiver_log_.Wait());
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}
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} // namespace webrtc
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