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webrtc/test/call_test.cc
Per K 217b384c1b Remove rtp header extension from config of Call audio and video receivers 
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.

Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
2023-01-31 11:58:43 +00:00

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/*
* 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 <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/create_frame_generator.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "call/fake_network_pipe.h"
#include "call/packet_receiver.h"
#include "call/simulated_network.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/fake_encoder.h"
#include "test/rtp_rtcp_observer.h"
#include "test/testsupport/file_utils.h"
#include "video/config/video_encoder_config.h"
namespace webrtc {
namespace test {
CallTest::CallTest()
: clock_(Clock::GetRealTimeClock()),
task_queue_factory_(CreateDefaultTaskQueueFactory()),
send_event_log_(std::make_unique<RtcEventLogNull>()),
recv_event_log_(std::make_unique<RtcEventLogNull>()),
audio_send_config_(/*send_transport=*/nullptr),
audio_send_stream_(nullptr),
frame_generator_capturer_(nullptr),
fake_encoder_factory_([this]() {
std::unique_ptr<FakeEncoder> fake_encoder;
if (video_encoder_configs_[0].codec_type == kVideoCodecVP8) {
fake_encoder = std::make_unique<FakeVp8Encoder>(clock_);
} else {
fake_encoder = std::make_unique<FakeEncoder>(clock_);
}
fake_encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
return fake_encoder;
}),
fake_decoder_factory_([]() { return std::make_unique<FakeDecoder>(); }),
bitrate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()),
num_video_streams_(1),
num_audio_streams_(0),
num_flexfec_streams_(0),
audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
task_queue_(task_queue_factory_->CreateTaskQueue(
"CallTestTaskQueue",
TaskQueueFactory::Priority::NORMAL)) {}
CallTest::~CallTest() = default;
void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
for (const RtpExtension& registered_extension : rtp_extensions_) {
if (registered_extension.id == extension.id) {
ASSERT_EQ(registered_extension.uri, extension.uri)
<< "Different URIs associated with ID " << extension.id << ".";
ASSERT_EQ(registered_extension.encrypt, extension.encrypt)
<< "Encryption mismatch associated with ID " << extension.id << ".";
return;
} else { // Different IDs.
// Different IDs referring to the same extension probably indicate
// a mistake in the test.
ASSERT_FALSE(registered_extension.uri == extension.uri &&
registered_extension.encrypt == extension.encrypt)
<< "URI " << extension.uri
<< (extension.encrypt ? " with " : " without ")
<< "encryption already registered with a different "
"ID ("
<< extension.id << " vs. " << registered_extension.id << ").";
}
}
rtp_extensions_.push_back(extension);
}
void CallTest::RunBaseTest(BaseTest* test) {
SendTask(task_queue(), [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(send_event_log_.get());
test->ModifySenderBitrateConfig(&send_config.bitrate_config);
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 (test->ShouldCreateReceivers()) {
Call::Config recv_config(recv_event_log_.get());
test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
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());
CreateReceiveTransport(test->GetReceiveTransportConfig(), test);
CreateSendTransport(test->GetSendTransportConfig(), test);
test->OnTransportCreated(send_transport_.get(), send_simulated_network_,
receive_transport_.get(),
receive_simulated_network_);
if (test->ShouldCreateReceivers()) {
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();
}
if (num_video_streams_ > 0) {
test->ModifyVideoConfigs(GetVideoSendConfig(), &video_receive_configs_,
GetVideoEncoderConfig());
}
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(GetVideoSendStream(), 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);
test->ModifyVideoDegradationPreference(&degradation_preference_);
CreateFrameGeneratorCapturer(frame_rate, width, height);
test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_);
}
Start();
});
test->PerformTest();
SendTask(task_queue(), [this, test]() {
Stop();
test->OnStreamsStopped();
DestroyStreams();
send_transport_.reset();
receive_transport_.reset();
frame_generator_capturer_ = nullptr;
DestroyCalls();
fake_send_audio_device_ = nullptr;
fake_recv_audio_device_ = nullptr;
});
}
void CallTest::CreateCalls() {
CreateCalls(Call::Config(send_event_log_.get()),
Call::Config(recv_event_log_.get()));
}
void CallTest::CreateCalls(const Call::Config& sender_config,
const Call::Config& receiver_config) {
CreateSenderCall(sender_config);
CreateReceiverCall(receiver_config);
}
void CallTest::CreateSenderCall() {
CreateSenderCall(Call::Config(send_event_log_.get()));
}
void CallTest::CreateSenderCall(const Call::Config& config) {
auto sender_config = config;
sender_config.task_queue_factory = task_queue_factory_.get();
sender_config.network_state_predictor_factory =
network_state_predictor_factory_.get();
sender_config.network_controller_factory = network_controller_factory_.get();
sender_config.trials = &field_trials_;
sender_call_.reset(Call::Create(sender_config));
}
void CallTest::CreateReceiverCall(const Call::Config& config) {
auto receiver_config = config;
receiver_config.task_queue_factory = task_queue_factory_.get();
receiver_config.trials = &field_trials_;
receiver_call_.reset(Call::Create(receiver_config));
}
void CallTest::DestroyCalls() {
send_transport_.reset();
receive_transport_.reset();
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->encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();
video_config->rtp.payload_name = "FAKE";
video_config->rtp.payload_type = kFakeVideoSendPayloadType;
video_config->rtp.extmap_allow_mixed = true;
AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kAbsSendTimeUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kTimestampOffsetUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri00,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kDependencyDescriptorUri,
&video_config->rtp.extensions);
if (video_encoder_configs_.empty()) {
video_encoder_configs_.emplace_back();
FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
&video_encoder_configs_.back());
}
for (size_t i = 0; i < num_video_streams; ++i)
video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
AddRtpExtensionByUri(RtpExtension::kVideoRotationUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kColorSpaceUri,
&video_config->rtp.extensions);
}
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) {
AudioSendStream::Config audio_send_config(send_transport);
audio_send_config.rtp.ssrc = kAudioSendSsrc;
AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
&audio_send_config.rtp.extensions);
audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
audio_send_config.min_bitrate_bps = 6000;
audio_send_config.max_bitrate_bps = 60000;
audio_send_config.encoder_factory = audio_encoder_factory_;
SetAudioConfig(audio_send_config);
}
// TODO(brandtr): Update this when we support multistream protection.
if (num_flexfec_streams > 0) {
SetSendFecConfig({kVideoSendSsrcs[0]});
}
}
void CallTest::SetAudioConfig(const AudioSendStream::Config& config) {
audio_send_config_ = config;
}
void CallTest::SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs) {
GetVideoSendConfig()->rtp.flexfec.payload_type = kFlexfecPayloadType;
GetVideoSendConfig()->rtp.flexfec.ssrc = kFlexfecSendSsrc;
GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs;
}
void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) {
send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
}
void CallTest::SetReceiveUlpFecConfig(
VideoReceiveStreamInterface::Config* receive_config) {
receive_config->rtp.red_payload_type = kRedPayloadType;
receive_config->rtp.ulpfec_payload_type = kUlpfecPayloadType;
receive_config->rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
kRedPayloadType;
}
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) {
video_send_configs_.clear();
video_send_configs_.emplace_back(nullptr);
CreateVideoSendConfig(&video_send_configs_.back(), num_video_streams, 0,
send_transport);
}
CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
send_transport);
}
void CallTest::CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport) {
CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport,
&fake_decoder_factory_, absl::nullopt,
false, 0);
}
void CallTest::CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
absl::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms) {
AddMatchingVideoReceiveConfigs(
&video_receive_configs_, video_send_config, rtcp_send_transport,
decoder_factory, decode_sub_stream, receiver_reference_time_report,
rtp_history_ms);
}
void CallTest::AddMatchingVideoReceiveConfigs(
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
absl::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms) {
RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
VideoReceiveStreamInterface::Config default_config(rtcp_send_transport);
default_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
default_config.rtp.nack.rtp_history_ms = rtp_history_ms;
// Enable RTT calculation so NTP time estimator will work.
default_config.rtp.rtcp_xr.receiver_reference_time_report =
receiver_reference_time_report;
default_config.renderer = &fake_renderer_;
for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
VideoReceiveStreamInterface::Config video_recv_config(
default_config.Copy());
video_recv_config.decoders.clear();
if (!video_send_config.rtp.rtx.ssrcs.empty()) {
video_recv_config.rtp.rtx_ssrc = video_send_config.rtp.rtx.ssrcs[i];
video_recv_config.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
video_send_config.rtp.payload_type;
}
video_recv_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
VideoReceiveStreamInterface::Decoder decoder;
decoder.payload_type = video_send_config.rtp.payload_type;
decoder.video_format = SdpVideoFormat(video_send_config.rtp.payload_name);
// Force fake decoders on non-selected simulcast streams.
if (!decode_sub_stream || i == *decode_sub_stream) {
video_recv_config.decoder_factory = decoder_factory;
} else {
video_recv_config.decoder_factory = &fake_decoder_factory_;
}
video_recv_config.decoders.push_back(decoder);
receive_configs->emplace_back(std::move(video_recv_config));
}
}
void CallTest::CreateMatchingAudioAndFecConfigs(
Transport* rtcp_send_transport) {
RTC_DCHECK_GE(1, num_audio_streams_);
if (num_audio_streams_ == 1) {
CreateMatchingAudioConfigs(rtcp_send_transport, "");
}
// TODO(brandtr): Update this when we support multistream protection.
RTC_DCHECK(num_flexfec_streams_ <= 1);
if (num_flexfec_streams_ == 1) {
CreateMatchingFecConfig(rtcp_send_transport, *GetVideoSendConfig());
}
}
void CallTest::CreateMatchingAudioConfigs(Transport* transport,
std::string sync_group) {
audio_receive_configs_.push_back(CreateMatchingAudioConfig(
audio_send_config_, audio_decoder_factory_, transport, sync_group));
}
AudioReceiveStreamInterface::Config CallTest::CreateMatchingAudioConfig(
const AudioSendStream::Config& send_config,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
Transport* transport,
std::string sync_group) {
AudioReceiveStreamInterface::Config audio_config;
audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
audio_config.rtcp_send_transport = transport;
audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
audio_config.decoder_factory = audio_decoder_factory;
audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
audio_config.sync_group = sync_group;
return audio_config;
}
void CallTest::CreateMatchingFecConfig(
Transport* transport,
const VideoSendStream::Config& send_config) {
FlexfecReceiveStream::Config config(transport);
config.payload_type = send_config.rtp.flexfec.payload_type;
config.rtp.remote_ssrc = send_config.rtp.flexfec.ssrc;
config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
if (!video_receive_configs_.empty()) {
video_receive_configs_[0].rtp.protected_by_flexfec = true;
video_receive_configs_[0].rtp.packet_sink_ = this;
}
flexfec_receive_configs_.push_back(config);
}
void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
video_receive_configs_.clear();
for (VideoSendStream::Config& video_send_config : video_send_configs_) {
CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport);
}
CreateMatchingAudioAndFecConfigs(rtcp_send_transport);
}
void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
float speed,
int framerate,
int width,
int height) {
video_sources_.clear();
auto frame_generator_capturer =
std::make_unique<test::FrameGeneratorCapturer>(
clock,
test::CreateSquareFrameGenerator(width, height, absl::nullopt,
absl::nullopt),
framerate * speed, *task_queue_factory_);
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
video_sources_.push_back(std::move(frame_generator_capturer));
ConnectVideoSourcesToStreams();
}
void CallTest::CreateFrameGeneratorCapturer(int framerate,
int width,
int height) {
video_sources_.clear();
auto frame_generator_capturer =
std::make_unique<test::FrameGeneratorCapturer>(
clock_,
test::CreateSquareFrameGenerator(width, height, absl::nullopt,
absl::nullopt),
framerate, *task_queue_factory_);
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
video_sources_.push_back(std::move(frame_generator_capturer));
ConnectVideoSourcesToStreams();
}
void CallTest::CreateFakeAudioDevices(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
fake_send_audio_device_ = TestAudioDeviceModule::Create(
task_queue_factory_.get(), std::move(capturer), nullptr, 1.f);
fake_recv_audio_device_ = TestAudioDeviceModule::Create(
task_queue_factory_.get(), nullptr, std::move(renderer), 1.f);
}
void CallTest::CreateVideoStreams() {
RTC_DCHECK(video_receive_streams_.empty());
CreateVideoSendStreams();
for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
video_receive_configs_[i].Copy()));
}
}
void CallTest::CreateVideoSendStreams() {
RTC_DCHECK(video_send_streams_.empty());
// We currently only support testing external fec controllers with a single
// VideoSendStream.
if (fec_controller_factory_.get()) {
RTC_DCHECK_LE(video_send_configs_.size(), 1);
}
// TODO(http://crbug/818127):
// Remove this workaround when ALR is not screenshare-specific.
std::list<size_t> streams_creation_order;
for (size_t i = 0; i < video_send_configs_.size(); ++i) {
// If dual streams are created, add the screenshare stream last.
if (video_encoder_configs_[i].content_type ==
VideoEncoderConfig::ContentType::kScreen) {
streams_creation_order.push_back(i);
} else {
streams_creation_order.push_front(i);
}
}
video_send_streams_.resize(video_send_configs_.size(), nullptr);
for (size_t i : streams_creation_order) {
if (fec_controller_factory_.get()) {
video_send_streams_[i] = sender_call_->CreateVideoSendStream(
video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
fec_controller_factory_->CreateFecController());
} else {
video_send_streams_[i] = sender_call_->CreateVideoSendStream(
video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy());
}
}
}
void CallTest::CreateVideoSendStream(const VideoEncoderConfig& encoder_config) {
RTC_DCHECK(video_send_streams_.empty());
video_send_streams_.push_back(sender_call_->CreateVideoSendStream(
GetVideoSendConfig()->Copy(), encoder_config.Copy()));
}
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]));
}
}
void CallTest::CreateSendTransport(const BuiltInNetworkBehaviorConfig& config,
RtpRtcpObserver* observer) {
PacketReceiver* receiver =
receiver_call_ ? receiver_call_->Receiver() : nullptr;
auto network = std::make_unique<SimulatedNetwork>(config);
send_simulated_network_ = network.get();
send_transport_ = std::make_unique<PacketTransport>(
task_queue(), sender_call_.get(), observer,
test::PacketTransport::kSender, payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network), receiver),
rtp_extensions_, rtp_extensions_);
}
void CallTest::CreateReceiveTransport(
const BuiltInNetworkBehaviorConfig& config,
RtpRtcpObserver* observer) {
auto network = std::make_unique<SimulatedNetwork>(config);
receive_simulated_network_ = network.get();
receive_transport_ = std::make_unique<PacketTransport>(
task_queue(), nullptr, observer, test::PacketTransport::kReceiver,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network),
sender_call_->Receiver()),
rtp_extensions_, rtp_extensions_);
}
void CallTest::ConnectVideoSourcesToStreams() {
for (size_t i = 0; i < video_sources_.size(); ++i)
video_send_streams_[i]->SetSource(video_sources_[i].get(),
degradation_preference_);
}
void CallTest::Start() {
StartVideoStreams();
if (audio_send_stream_) {
audio_send_stream_->Start();
}
for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Start();
}
void CallTest::StartVideoStreams() {
for (size_t i = 0; i < video_send_streams_.size(); ++i) {
std::vector<bool> active_rtp_streams(
video_send_configs_[i].rtp.ssrcs.size(), true);
video_send_streams_[i]->StartPerRtpStream(active_rtp_streams);
}
for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
video_recv_stream->Start();
}
void CallTest::Stop() {
for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Stop();
if (audio_send_stream_) {
audio_send_stream_->Stop();
}
StopVideoStreams();
}
void CallTest::StopVideoStreams() {
for (VideoSendStream* video_send_stream : video_send_streams_)
video_send_stream->Stop();
for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
video_recv_stream->Stop();
}
void CallTest::DestroyStreams() {
if (audio_send_stream_)
sender_call_->DestroyAudioSendStream(audio_send_stream_);
audio_send_stream_ = nullptr;
for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
DestroyVideoSendStreams();
for (VideoReceiveStreamInterface* 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();
video_sources_.clear();
}
void CallTest::DestroyVideoSendStreams() {
for (VideoSendStream* video_send_stream : video_send_streams_)
sender_call_->DestroyVideoSendStream(video_send_stream);
video_send_streams_.clear();
}
void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
frame_generator_capturer_->SetFakeRotation(rotation);
}
void CallTest::SetVideoDegradation(DegradationPreference preference) {
GetVideoSendStream()->SetSource(frame_generator_capturer_, preference);
}
VideoSendStream::Config* CallTest::GetVideoSendConfig() {
return &video_send_configs_[0];
}
void CallTest::SetVideoSendConfig(const VideoSendStream::Config& config) {
video_send_configs_.clear();
video_send_configs_.push_back(config.Copy());
}
VideoEncoderConfig* CallTest::GetVideoEncoderConfig() {
return &video_encoder_configs_[0];
}
void CallTest::SetVideoEncoderConfig(const VideoEncoderConfig& config) {
video_encoder_configs_.clear();
video_encoder_configs_.push_back(config.Copy());
}
VideoSendStream* CallTest::GetVideoSendStream() {
return video_send_streams_[0];
}
FlexfecReceiveStream::Config* CallTest::GetFlexFecConfig() {
return &flexfec_receive_configs_[0];
}
void CallTest::OnRtpPacket(const RtpPacketReceived& packet) {
// All FlexFEC streams protect all of the video streams.
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
flexfec_recv_stream->OnRtpPacket(packet);
}
absl::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
const std::string& uri) const {
for (const auto& extension : rtp_extensions_) {
if (extension.uri == uri) {
return extension;
}
}
return absl::nullopt;
}
void CallTest::AddRtpExtensionByUri(
const std::string& uri,
std::vector<RtpExtension>* extensions) const {
const absl::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
if (extension) {
extensions->push_back(*extension);
}
}
constexpr size_t CallTest::kNumSsrcs;
const int CallTest::kDefaultWidth;
const int CallTest::kDefaultHeight;
const int CallTest::kDefaultFramerate;
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 std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
{CallTest::kVideoSendPayloadType, MediaType::VIDEO},
{CallTest::kFakeVideoSendPayloadType, MediaType::VIDEO},
{CallTest::kSendRtxPayloadType, MediaType::VIDEO},
{CallTest::kPayloadTypeVP8, MediaType::VIDEO},
{CallTest::kPayloadTypeVP9, MediaType::VIDEO},
{CallTest::kPayloadTypeH264, MediaType::VIDEO},
{CallTest::kPayloadTypeGeneric, MediaType::VIDEO},
{CallTest::kRedPayloadType, MediaType::VIDEO},
{CallTest::kRtxRedPayloadType, MediaType::VIDEO},
{CallTest::kUlpfecPayloadType, MediaType::VIDEO},
{CallTest::kFlexfecPayloadType, MediaType::VIDEO},
{CallTest::kAudioSendPayloadType, MediaType::AUDIO}};
BaseTest::BaseTest() {}
BaseTest::BaseTest(TimeDelta timeout) : RtpRtcpObserver(timeout) {}
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) {}
void BaseTest::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {}
void BaseTest::ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) {
}
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
void BaseTest::OnTransportCreated(PacketTransport* to_receiver,
SimulatedNetworkInterface* sender_network,
PacketTransport* to_sender,
SimulatedNetworkInterface* receiver_network) {
}
BuiltInNetworkBehaviorConfig BaseTest::GetSendTransportConfig() const {
return BuiltInNetworkBehaviorConfig();
}
BuiltInNetworkBehaviorConfig BaseTest::GetReceiveTransportConfig() const {
return BuiltInNetworkBehaviorConfig();
}
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<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) {}
void BaseTest::ModifyVideoCaptureStartResolution(int* width,
int* heigt,
int* frame_rate) {}
void BaseTest::ModifyVideoDegradationPreference(
DegradationPreference* degradation_preference) {}
void BaseTest::OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>& receive_streams) {}
void BaseTest::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {}
void BaseTest::OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStreamInterface*>& 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(TimeDelta timeout) : BaseTest(timeout) {}
bool SendTest::ShouldCreateReceivers() const {
return false;
}
EndToEndTest::EndToEndTest() {}
EndToEndTest::EndToEndTest(TimeDelta timeout) : BaseTest(timeout) {}
bool EndToEndTest::ShouldCreateReceivers() const {
return true;
}
} // namespace test
} // namespace webrtc
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