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rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/call/bitrate_estimator_tests.cc

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/*
* 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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/event.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/mock_voice_engine.h"
#include "webrtc/test/frame_generator_capturer.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_ GUARDED_BY(crit_sect_);
Strings expected_log_lines_ 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() : mock_voice_engine_(decoder_factory_),
receive_config_(nullptr) {}
virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
virtual void SetUp() {
AudioState::Config audio_state_config;
audio_state_config.voice_engine = &mock_voice_engine_;
Call::Config config;
config.audio_state = AudioState::Create(audio_state_config);
receiver_call_.reset(Call::Create(config));
sender_call_.reset(Call::Create(config));
send_transport_.reset(new test::DirectTransport(sender_call_.get()));
send_transport_->SetReceiver(receiver_call_->Receiver());
receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
receive_transport_->SetReceiver(sender_call_->Receiver());
video_send_config_ = VideoSendStream::Config(send_transport_.get());
video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
// Encoders will be set separately per stream.
video_send_config_.encoder_settings.encoder = nullptr;
video_send_config_.encoder_settings.payload_name = "FAKE";
video_send_config_.encoder_settings.payload_type =
kFakeVideoSendPayloadType;
video_encoder_config_.streams = test::CreateVideoStreams(1);
receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
// receive_config_.decoders will be set by every stream separately.
receive_config_.rtp.remote_ssrc = video_send_config_.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() {
std::for_each(streams_.begin(), streams_.end(),
std::mem_fun(&Stream::StopSending));
send_transport_->StopSending();
receive_transport_->StopSending();
while (!streams_.empty()) {
delete streams_.back();
streams_.pop_back();
}
receiver_call_.reset();
sender_call_.reset();
}
protected:
friend class Stream;
class Stream {
public:
Stream(BitrateEstimatorTest* test, bool receive_audio)
: test_(test),
is_sending_receiving_(false),
send_stream_(nullptr),
audio_receive_stream_(nullptr),
video_receive_stream_(nullptr),
frame_generator_capturer_(),
fake_encoder_(Clock::GetRealTimeClock()),
fake_decoder_() {
test_->video_send_config_.rtp.ssrcs[0]++;
test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
send_stream_ = test_->sender_call_->CreateVideoSendStream(
test_->video_send_config_, test_->video_encoder_config_);
RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
test_->video_encoder_config_.streams[0].height, 30,
Clock::GetRealTimeClock()));
send_stream_->Start();
frame_generator_capturer_->Start();
if (receive_audio) {
AudioReceiveStream::Config receive_config;
receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
// Bogus non-default id to prevent hitting a RTC_DCHECK when creating
// the AudioReceiveStream. Every receive stream has to correspond to
// an underlying channel id.
receive_config.voe_channel_id = 0;
receive_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
receive_config.decoder_factory = test_->decoder_factory_;
audio_receive_stream_ =
test_->receiver_call_->CreateAudioReceiveStream(receive_config);
} else {
VideoReceiveStream::Decoder decoder;
decoder.decoder = &fake_decoder_;
decoder.payload_type =
test_->video_send_config_.encoder_settings.payload_type;
decoder.payload_name =
test_->video_send_config_.encoder_settings.payload_name;
test_->receive_config_.decoders.clear();
test_->receive_config_.decoders.push_back(decoder);
test_->receive_config_.rtp.remote_ssrc =
test_->video_send_config_.rtp.ssrcs[0];
test_->receive_config_.rtp.local_ssrc++;
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_);
frame_generator_capturer_.reset(nullptr);
test_->sender_call_->DestroyVideoSendStream(send_stream_);
send_stream_ = nullptr;
if (audio_receive_stream_) {
test_->receiver_call_->DestroyAudioReceiveStream(audio_receive_stream_);
audio_receive_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_;
AudioReceiveStream* audio_receive_stream_;
VideoReceiveStream* video_receive_stream_;
std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
test::FakeEncoder fake_encoder_;
test::FakeDecoder fake_decoder_;
};
testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
LogObserver receiver_log_;
std::unique_ptr<test::DirectTransport> send_transport_;
std::unique_ptr<test::DirectTransport> receive_transport_;
std::unique_ptr<Call> sender_call_;
std::unique_ptr<Call> receiver_call_;
VideoReceiveStream::Config receive_config_;
std::vector<Stream*> streams_;
};
static const char* kAbsSendTimeLog =
"RemoteBitrateEstimatorAbsSendTime: Instantiating.";
static const char* kSingleStreamLog =
"RemoteBitrateEstimatorSingleStream: Instantiating.";
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
video_send_config_.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, false));
EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
streams_.push_back(new Stream(this, false));
EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
video_send_config_.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, false));
EXPECT_TRUE(receiver_log_.Wait());
video_send_config_.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, false));
EXPECT_TRUE(receiver_log_.Wait());
}
// This test is flaky. See webrtc:5790.
TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
video_send_config_.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, false));
EXPECT_TRUE(receiver_log_.Wait());
video_send_config_.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, false));
EXPECT_TRUE(receiver_log_.Wait());
video_send_config_.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, false));
streams_[0]->StopSending();
streams_[1]->StopSending();
EXPECT_TRUE(receiver_log_.Wait());
}
} // namespace webrtc
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