blob: c735e0650cb6679bcccac7e8b7470787f678c1b0 [file] [log] [blame]
/*
* Copyright (c) 2014 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 "test/call_test.h"
#include <algorithm>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/video_codecs/video_encoder_config.h"
#include "call/rtp_transport_controller_send.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/ptr_util.h"
#include "test/fake_encoder.h"
#include "test/testsupport/fileutils.h"
namespace webrtc {
namespace test {
namespace {
const int kVideoRotationRtpExtensionId = 4;
}
CallTest::CallTest()
: clock_(Clock::GetRealTimeClock()),
event_log_(RtcEventLog::CreateNull()),
sender_call_transport_controller_(nullptr),
video_send_config_(nullptr),
video_send_stream_(nullptr),
audio_send_config_(nullptr),
audio_send_stream_(nullptr),
fake_encoder_factory_([this]() {
auto encoder = rtc::MakeUnique<test::FakeEncoder>(clock_);
encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
return encoder;
}),
num_video_streams_(1),
num_audio_streams_(0),
num_flexfec_streams_(0),
audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
task_queue_("CallTestTaskQueue") {}
CallTest::~CallTest() {
task_queue_.SendTask([this]() {
fake_send_audio_device_ = nullptr;
fake_recv_audio_device_ = nullptr;
frame_generator_capturer_.reset();
});
}
void CallTest::RunBaseTest(BaseTest* test) {
task_queue_.SendTask([this, test]() {
num_video_streams_ = test->GetNumVideoStreams();
num_audio_streams_ = test->GetNumAudioStreams();
num_flexfec_streams_ = test->GetNumFlexfecStreams();
RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
Call::Config send_config(test->GetSenderCallConfig());
if (num_audio_streams_ > 0) {
CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
fake_recv_audio_device_.get());
apm_send_ = AudioProcessingBuilder().Create();
apm_recv_ = AudioProcessingBuilder().Create();
EXPECT_EQ(0, fake_send_audio_device_->Init());
EXPECT_EQ(0, fake_recv_audio_device_->Init());
AudioState::Config audio_state_config;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
audio_state_config.audio_processing = apm_send_;
audio_state_config.audio_device_module = fake_send_audio_device_;
send_config.audio_state = AudioState::Create(audio_state_config);
fake_send_audio_device_->RegisterAudioCallback(
send_config.audio_state->audio_transport());
}
CreateSenderCall(send_config);
if (sender_call_transport_controller_ != nullptr) {
test->OnRtpTransportControllerSendCreated(
sender_call_transport_controller_);
}
if (test->ShouldCreateReceivers()) {
Call::Config recv_config(test->GetReceiverCallConfig());
if (num_audio_streams_ > 0) {
AudioState::Config audio_state_config;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
audio_state_config.audio_processing = apm_recv_;
audio_state_config.audio_device_module = fake_recv_audio_device_;
recv_config.audio_state = AudioState::Create(audio_state_config);
fake_recv_audio_device_->RegisterAudioCallback(
recv_config.audio_state->audio_transport());
}
CreateReceiverCall(recv_config);
}
test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
receive_transport_.reset(test->CreateReceiveTransport(&task_queue_));
send_transport_.reset(
test->CreateSendTransport(&task_queue_, sender_call_.get()));
if (test->ShouldCreateReceivers()) {
send_transport_->SetReceiver(receiver_call_->Receiver());
receive_transport_->SetReceiver(sender_call_->Receiver());
if (num_video_streams_ > 0)
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
if (num_audio_streams_ > 0)
receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
} else {
// Sender-only call delivers to itself.
send_transport_->SetReceiver(sender_call_->Receiver());
receive_transport_->SetReceiver(nullptr);
}
CreateSendConfig(num_video_streams_, num_audio_streams_,
num_flexfec_streams_, send_transport_.get());
if (test->ShouldCreateReceivers()) {
CreateMatchingReceiveConfigs(receive_transport_.get());
}
if (num_video_streams_ > 0) {
test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
&video_encoder_config_);
}
if (num_audio_streams_ > 0) {
test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
}
if (num_flexfec_streams_ > 0) {
test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
}
if (num_flexfec_streams_ > 0) {
CreateFlexfecStreams();
test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
}
if (num_video_streams_ > 0) {
CreateVideoStreams();
test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
}
if (num_audio_streams_ > 0) {
CreateAudioStreams();
test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
}
if (num_video_streams_ > 0) {
int width = kDefaultWidth;
int height = kDefaultHeight;
int frame_rate = kDefaultFramerate;
test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
CreateFrameGeneratorCapturer(frame_rate, width, height);
test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
}
Start();
});
test->PerformTest();
task_queue_.SendTask([this, test]() {
Stop();
test->OnStreamsStopped();
DestroyStreams();
send_transport_.reset();
receive_transport_.reset();
DestroyCalls();
});
}
void CallTest::CreateCalls(const Call::Config& sender_config,
const Call::Config& receiver_config) {
CreateSenderCall(sender_config);
CreateReceiverCall(receiver_config);
}
void CallTest::CreateSenderCall(const Call::Config& config) {
std::unique_ptr<RtpTransportControllerSend> controller_send =
rtc::MakeUnique<RtpTransportControllerSend>(
Clock::GetRealTimeClock(), config.event_log,
config.network_controller_factory, config.bitrate_config);
sender_call_transport_controller_ = controller_send.get();
sender_call_.reset(Call::Create(config, std::move(controller_send)));
}
void CallTest::CreateReceiverCall(const Call::Config& config) {
receiver_call_.reset(Call::Create(config));
}
void CallTest::DestroyCalls() {
sender_call_.reset();
receiver_call_.reset();
}
void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
size_t num_video_streams,
size_t num_used_ssrcs,
Transport* send_transport) {
RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, kNumSsrcs);
*video_config = VideoSendStream::Config(send_transport);
video_config->encoder_settings.encoder_factory = &fake_encoder_factory_;
video_config->rtp.payload_name = "FAKE";
video_config->rtp.payload_type = kFakeVideoSendPayloadType;
video_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
video_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
&video_encoder_config_);
for (size_t i = 0; i < num_video_streams; ++i)
video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
video_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
}
void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport) {
RTC_DCHECK_LE(num_audio_streams, 1);
RTC_DCHECK_LE(num_flexfec_streams, 1);
if (num_audio_streams > 0) {
audio_send_config_ = AudioSendStream::Config(send_transport);
audio_send_config_.rtp.ssrc = kAudioSendSsrc;
audio_send_config_.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
audio_send_config_.encoder_factory = audio_encoder_factory_;
}
// TODO(brandtr): Update this when we support multistream protection.
if (num_flexfec_streams > 0) {
video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
}
}
void CallTest::CreateSendConfig(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport) {
if (num_video_streams > 0) {
CreateVideoSendConfig(&video_send_config_, num_video_streams, 0,
send_transport);
}
CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
send_transport);
}
std::vector<VideoReceiveStream::Config>
CallTest::CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport) {
std::vector<VideoReceiveStream::Config> result;
RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
VideoReceiveStream::Config video_config(rtcp_send_transport);
video_config.rtp.remb = false;
video_config.rtp.transport_cc = true;
video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
for (const RtpExtension& extension : video_send_config.rtp.extensions)
video_config.rtp.extensions.push_back(extension);
video_config.renderer = &fake_renderer_;
for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(video_send_config);
allocated_decoders_.push_back(
std::unique_ptr<VideoDecoder>(decoder.decoder));
video_config.decoders.clear();
video_config.decoders.push_back(decoder);
video_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
result.push_back(video_config.Copy());
}
result[0].rtp.protected_by_flexfec = (num_flexfec_streams_ == 1);
return result;
}
void CallTest::CreateMatchingAudioAndFecConfigs(
Transport* rtcp_send_transport) {
RTC_DCHECK_GE(1, num_audio_streams_);
if (num_audio_streams_ == 1) {
AudioReceiveStream::Config audio_config;
audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
audio_config.rtcp_send_transport = rtcp_send_transport;
audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
audio_config.decoder_factory = audio_decoder_factory_;
audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
audio_receive_configs_.push_back(audio_config);
}
// TODO(brandtr): Update this when we support multistream protection.
RTC_DCHECK(num_flexfec_streams_ <= 1);
if (num_flexfec_streams_ == 1) {
FlexfecReceiveStream::Config config(rtcp_send_transport);
config.payload_type = kFlexfecPayloadType;
config.remote_ssrc = kFlexfecSendSsrc;
config.protected_media_ssrcs = {kVideoSendSsrcs[0]};
config.local_ssrc = kReceiverLocalVideoSsrc;
for (const RtpExtension& extension : video_send_config_.rtp.extensions)
config.rtp_header_extensions.push_back(extension);
flexfec_receive_configs_.push_back(config);
}
}
void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
video_receive_configs_.clear();
allocated_decoders_.clear();
if (num_video_streams_ > 0) {
std::vector<VideoReceiveStream::Config> new_configs =
CreateMatchingVideoReceiveConfigs(video_send_config_,
rtcp_send_transport);
for (VideoReceiveStream::Config& config : new_configs) {
video_receive_configs_.push_back(config.Copy());
}
}
CreateMatchingAudioAndFecConfigs(rtcp_send_transport);
}
void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
float speed,
int framerate,
int width,
int height) {
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
width, height, absl::nullopt, absl::nullopt, framerate * speed, clock));
video_send_stream_->SetSource(frame_generator_capturer_.get(),
DegradationPreference::MAINTAIN_FRAMERATE);
}
void CallTest::CreateFrameGeneratorCapturer(int framerate,
int width,
int height) {
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
width, height, absl::nullopt, absl::nullopt, framerate, clock_));
video_send_stream_->SetSource(frame_generator_capturer_.get(),
DegradationPreference::MAINTAIN_FRAMERATE);
}
void CallTest::CreateFakeAudioDevices(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
fake_send_audio_device_ = TestAudioDeviceModule::CreateTestAudioDeviceModule(
std::move(capturer), nullptr, 1.f);
fake_recv_audio_device_ = TestAudioDeviceModule::CreateTestAudioDeviceModule(
nullptr, std::move(renderer), 1.f);
}
void CallTest::CreateVideoStreams() {
RTC_DCHECK(video_send_stream_ == nullptr);
RTC_DCHECK(video_receive_streams_.empty());
video_send_stream_ = sender_call_->CreateVideoSendStream(
video_send_config_.Copy(), video_encoder_config_.Copy());
for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
video_receive_configs_[i].Copy()));
}
AssociateFlexfecStreamsWithVideoStreams();
}
void CallTest::CreateAudioStreams() {
RTC_DCHECK(audio_send_stream_ == nullptr);
RTC_DCHECK(audio_receive_streams_.empty());
audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
audio_receive_streams_.push_back(
receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
}
}
void CallTest::CreateFlexfecStreams() {
for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) {
flexfec_receive_streams_.push_back(
receiver_call_->CreateFlexfecReceiveStream(
flexfec_receive_configs_[i]));
}
AssociateFlexfecStreamsWithVideoStreams();
}
void CallTest::AssociateFlexfecStreamsWithVideoStreams() {
// All FlexFEC streams protect all of the video streams.
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) {
for (VideoReceiveStream* video_recv_stream : video_receive_streams_) {
video_recv_stream->AddSecondarySink(flexfec_recv_stream);
}
}
}
void CallTest::DissociateFlexfecStreamsFromVideoStreams() {
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) {
for (VideoReceiveStream* video_recv_stream : video_receive_streams_) {
video_recv_stream->RemoveSecondarySink(flexfec_recv_stream);
}
}
}
void CallTest::Start() {
if (video_send_stream_)
video_send_stream_->Start();
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
video_recv_stream->Start();
if (audio_send_stream_) {
audio_send_stream_->Start();
}
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Start();
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Start();
}
void CallTest::Stop() {
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Stop();
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Stop();
if (audio_send_stream_) {
audio_send_stream_->Stop();
}
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
video_recv_stream->Stop();
if (video_send_stream_)
video_send_stream_->Stop();
}
void CallTest::DestroyStreams() {
DissociateFlexfecStreamsFromVideoStreams();
if (audio_send_stream_)
sender_call_->DestroyAudioSendStream(audio_send_stream_);
audio_send_stream_ = nullptr;
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
if (video_send_stream_)
sender_call_->DestroyVideoSendStream(video_send_stream_);
video_send_stream_ = nullptr;
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
video_receive_streams_.clear();
allocated_decoders_.clear();
}
void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
frame_generator_capturer_->SetFakeRotation(rotation);
}
constexpr size_t CallTest::kNumSsrcs;
const int CallTest::kDefaultWidth;
const int CallTest::kDefaultHeight;
const int CallTest::kDefaultFramerate;
const int CallTest::kDefaultTimeoutMs = 30 * 1000;
const int CallTest::kLongTimeoutMs = 120 * 1000;
const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {
0xBADCAFD, 0xBADCAFE, 0xBADCAFF, 0xBADCB00, 0xBADCB01, 0xBADCB02};
const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {
0xC0FFED, 0xC0FFEE, 0xC0FFEF, 0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF;
const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
const int CallTest::kNackRtpHistoryMs = 1000;
const uint8_t CallTest::kDefaultKeepalivePayloadType =
RtpKeepAliveConfig().payload_type;
const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
{CallTest::kVideoSendPayloadType, MediaType::VIDEO},
{CallTest::kFakeVideoSendPayloadType, MediaType::VIDEO},
{CallTest::kSendRtxPayloadType, MediaType::VIDEO},
{CallTest::kRedPayloadType, MediaType::VIDEO},
{CallTest::kRtxRedPayloadType, MediaType::VIDEO},
{CallTest::kUlpfecPayloadType, MediaType::VIDEO},
{CallTest::kFlexfecPayloadType, MediaType::VIDEO},
{CallTest::kAudioSendPayloadType, MediaType::AUDIO},
{CallTest::kDefaultKeepalivePayloadType, MediaType::ANY}};
BaseTest::BaseTest() : event_log_(RtcEventLog::CreateNull()) {}
BaseTest::BaseTest(unsigned int timeout_ms)
: RtpRtcpObserver(timeout_ms), event_log_(RtcEventLog::CreateNull()) {}
BaseTest::~BaseTest() {}
std::unique_ptr<TestAudioDeviceModule::Capturer> BaseTest::CreateCapturer() {
return TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
}
std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() {
return TestAudioDeviceModule::CreateDiscardRenderer(48000);
}
void BaseTest::OnFakeAudioDevicesCreated(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* recv_audio_device) {}
Call::Config BaseTest::GetSenderCallConfig() {
return Call::Config(event_log_.get());
}
Call::Config BaseTest::GetReceiverCallConfig() {
return Call::Config(event_log_.get());
}
void BaseTest::OnRtpTransportControllerSendCreated(
RtpTransportControllerSend* controller) {}
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
test::PacketTransport* BaseTest::CreateSendTransport(
SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) {
return new PacketTransport(
task_queue, sender_call, this, test::PacketTransport::kSender,
CallTest::payload_type_map_, FakeNetworkPipe::Config());
}
test::PacketTransport* BaseTest::CreateReceiveTransport(
SingleThreadedTaskQueueForTesting* task_queue) {
return new PacketTransport(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
CallTest::payload_type_map_, FakeNetworkPipe::Config());
}
size_t BaseTest::GetNumVideoStreams() const {
return 1;
}
size_t BaseTest::GetNumAudioStreams() const {
return 0;
}
size_t BaseTest::GetNumFlexfecStreams() const {
return 0;
}
void BaseTest::ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) {}
void BaseTest::ModifyVideoCaptureStartResolution(int* width,
int* heigt,
int* frame_rate) {}
void BaseTest::OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) {}
void BaseTest::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) {}
void BaseTest::OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStream*>& receive_streams) {}
void BaseTest::ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs) {}
void BaseTest::OnFlexfecStreamsCreated(
const std::vector<FlexfecReceiveStream*>& receive_streams) {}
void BaseTest::OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer) {}
void BaseTest::OnStreamsStopped() {}
SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {}
bool SendTest::ShouldCreateReceivers() const {
return false;
}
EndToEndTest::EndToEndTest() {}
EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {}
bool EndToEndTest::ShouldCreateReceivers() const {
return true;
}
} // namespace test
} // namespace webrtc