Since rtp transport controller send owns the congestion controller it also should own the bitrate configuration logic, this way it can initialize the send side congestion controller with the bitrate configuration. Bug: webrtc:8415 Change-Id: Ifaa16139ca477cb1c80bf4aa24f17652af997553 Reviewed-on: https://webrtc-review.googlesource.com/54303 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22127}
605 lines
22 KiB
C++
605 lines
22 KiB
C++
/*
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* Copyright (c) 2014 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 "test/call_test.h"
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#include <algorithm>
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "call/rtp_transport_controller_send.h"
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#include "call/video_config.h"
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#include "modules/audio_mixer/audio_mixer_impl.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/ptr_util.h"
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#include "test/testsupport/fileutils.h"
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namespace webrtc {
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namespace test {
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namespace {
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const int kVideoRotationRtpExtensionId = 4;
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}
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CallTest::CallTest()
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: clock_(Clock::GetRealTimeClock()),
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event_log_(RtcEventLog::CreateNull()),
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sender_call_transport_controller_(nullptr),
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video_send_config_(nullptr),
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video_send_stream_(nullptr),
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audio_send_config_(nullptr),
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audio_send_stream_(nullptr),
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fake_encoder_(clock_),
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num_video_streams_(1),
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num_audio_streams_(0),
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num_flexfec_streams_(0),
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decoder_factory_(CreateBuiltinAudioDecoderFactory()),
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encoder_factory_(CreateBuiltinAudioEncoderFactory()),
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task_queue_("CallTestTaskQueue") {}
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CallTest::~CallTest() {
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task_queue_.SendTask([this]() {
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fake_send_audio_device_ = nullptr;
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fake_recv_audio_device_ = nullptr;
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frame_generator_capturer_.reset();
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});
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}
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void CallTest::RunBaseTest(BaseTest* test) {
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task_queue_.SendTask([this, test]() {
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num_video_streams_ = test->GetNumVideoStreams();
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num_audio_streams_ = test->GetNumAudioStreams();
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num_flexfec_streams_ = test->GetNumFlexfecStreams();
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RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
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Call::Config send_config(test->GetSenderCallConfig());
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if (num_audio_streams_ > 0) {
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CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
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test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
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fake_recv_audio_device_.get());
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apm_send_ = AudioProcessingBuilder().Create();
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apm_recv_ = AudioProcessingBuilder().Create();
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EXPECT_EQ(0, fake_send_audio_device_->Init());
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EXPECT_EQ(0, fake_recv_audio_device_->Init());
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AudioState::Config audio_state_config;
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audio_state_config.audio_mixer = AudioMixerImpl::Create();
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audio_state_config.audio_processing = apm_send_;
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audio_state_config.audio_device_module = fake_send_audio_device_;
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send_config.audio_state = AudioState::Create(audio_state_config);
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fake_send_audio_device_->RegisterAudioCallback(
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send_config.audio_state->audio_transport());
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}
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CreateSenderCall(send_config);
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if (sender_call_transport_controller_ != nullptr) {
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test->OnRtpTransportControllerSendCreated(
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sender_call_transport_controller_);
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}
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if (test->ShouldCreateReceivers()) {
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Call::Config recv_config(test->GetReceiverCallConfig());
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if (num_audio_streams_ > 0) {
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AudioState::Config audio_state_config;
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audio_state_config.audio_mixer = AudioMixerImpl::Create();
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audio_state_config.audio_processing = apm_recv_;
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audio_state_config.audio_device_module = fake_recv_audio_device_;
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recv_config.audio_state = AudioState::Create(audio_state_config);
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fake_recv_audio_device_->RegisterAudioCallback(
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recv_config.audio_state->audio_transport()); }
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CreateReceiverCall(recv_config);
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}
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test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
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receive_transport_.reset(test->CreateReceiveTransport(&task_queue_));
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send_transport_.reset(
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test->CreateSendTransport(&task_queue_, sender_call_.get()));
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if (test->ShouldCreateReceivers()) {
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send_transport_->SetReceiver(receiver_call_->Receiver());
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receive_transport_->SetReceiver(sender_call_->Receiver());
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if (num_video_streams_ > 0)
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receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
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if (num_audio_streams_ > 0)
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receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
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} else {
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// Sender-only call delivers to itself.
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send_transport_->SetReceiver(sender_call_->Receiver());
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receive_transport_->SetReceiver(nullptr);
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}
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CreateSendConfig(num_video_streams_, num_audio_streams_,
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num_flexfec_streams_, send_transport_.get());
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if (test->ShouldCreateReceivers()) {
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CreateMatchingReceiveConfigs(receive_transport_.get());
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}
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if (num_video_streams_ > 0) {
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test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
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&video_encoder_config_);
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}
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if (num_audio_streams_ > 0) {
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test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
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}
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if (num_flexfec_streams_ > 0) {
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test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
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}
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if (num_flexfec_streams_ > 0) {
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CreateFlexfecStreams();
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test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
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}
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if (num_video_streams_ > 0) {
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CreateVideoStreams();
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test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
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}
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if (num_audio_streams_ > 0) {
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CreateAudioStreams();
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test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
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}
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if (num_video_streams_ > 0) {
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int width = kDefaultWidth;
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int height = kDefaultHeight;
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int frame_rate = kDefaultFramerate;
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test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
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CreateFrameGeneratorCapturer(frame_rate, width, height);
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test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
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}
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Start();
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});
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test->PerformTest();
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task_queue_.SendTask([this, test]() {
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Stop();
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test->OnStreamsStopped();
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DestroyStreams();
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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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void CallTest::CreateCalls(const Call::Config& sender_config,
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const Call::Config& receiver_config) {
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CreateSenderCall(sender_config);
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CreateReceiverCall(receiver_config);
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}
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void CallTest::CreateSenderCall(const Call::Config& config) {
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sender_call_transport_controller_ = new RtpTransportControllerSend(
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Clock::GetRealTimeClock(), config.event_log, config.bitrate_config);
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sender_call_.reset(
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Call::Create(config, std::unique_ptr<RtpTransportControllerSend>(
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sender_call_transport_controller_)));
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}
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void CallTest::CreateReceiverCall(const Call::Config& config) {
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receiver_call_.reset(Call::Create(config));
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}
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void CallTest::DestroyCalls() {
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sender_call_.reset();
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receiver_call_.reset();
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}
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void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
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size_t num_video_streams,
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size_t num_used_ssrcs,
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Transport* send_transport) {
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RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, kNumSsrcs);
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*video_config = VideoSendStream::Config(send_transport);
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video_config->encoder_settings.encoder = &fake_encoder_;
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video_config->encoder_settings.payload_name = "FAKE";
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video_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
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video_config->rtp.extensions.push_back(
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RtpExtension(RtpExtension::kTransportSequenceNumberUri,
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kTransportSequenceNumberExtensionId));
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video_config->rtp.extensions.push_back(RtpExtension(
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RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
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FillEncoderConfiguration(num_video_streams, &video_encoder_config_);
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for (size_t i = 0; i < num_video_streams; ++i)
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video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
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video_config->rtp.extensions.push_back(RtpExtension(
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RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
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}
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void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport) {
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RTC_DCHECK_LE(num_audio_streams, 1);
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RTC_DCHECK_LE(num_flexfec_streams, 1);
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if (num_audio_streams > 0) {
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audio_send_config_ = AudioSendStream::Config(send_transport);
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audio_send_config_.rtp.ssrc = kAudioSendSsrc;
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audio_send_config_.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
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kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
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audio_send_config_.encoder_factory = encoder_factory_;
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}
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// TODO(brandtr): Update this when we support multistream protection.
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if (num_flexfec_streams > 0) {
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video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
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video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
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video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
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}
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}
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void CallTest::CreateSendConfig(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport) {
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if (num_video_streams > 0) {
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CreateVideoSendConfig(&video_send_config_, num_video_streams, 0,
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send_transport);
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}
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CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
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send_transport);
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}
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std::vector<VideoReceiveStream::Config>
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CallTest::CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport) {
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std::vector<VideoReceiveStream::Config> result;
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RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
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VideoReceiveStream::Config video_config(rtcp_send_transport);
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video_config.rtp.remb = false;
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video_config.rtp.transport_cc = true;
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video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
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for (const RtpExtension& extension : video_send_config.rtp.extensions)
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video_config.rtp.extensions.push_back(extension);
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video_config.renderer = &fake_renderer_;
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for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
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VideoReceiveStream::Decoder decoder =
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test::CreateMatchingDecoder(video_send_config.encoder_settings);
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allocated_decoders_.push_back(
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std::unique_ptr<VideoDecoder>(decoder.decoder));
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video_config.decoders.clear();
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video_config.decoders.push_back(decoder);
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video_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
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result.push_back(video_config.Copy());
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}
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result[0].rtp.protected_by_flexfec = (num_flexfec_streams_ == 1);
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return result;
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}
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void CallTest::CreateMatchingAudioAndFecConfigs(
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Transport* rtcp_send_transport) {
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RTC_DCHECK_GE(1, num_audio_streams_);
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if (num_audio_streams_ == 1) {
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AudioReceiveStream::Config audio_config;
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audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
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audio_config.rtcp_send_transport = rtcp_send_transport;
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audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
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audio_config.decoder_factory = decoder_factory_;
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audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
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audio_receive_configs_.push_back(audio_config);
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}
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// TODO(brandtr): Update this when we support multistream protection.
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RTC_DCHECK(num_flexfec_streams_ <= 1);
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if (num_flexfec_streams_ == 1) {
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FlexfecReceiveStream::Config config(rtcp_send_transport);
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config.payload_type = kFlexfecPayloadType;
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config.remote_ssrc = kFlexfecSendSsrc;
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config.protected_media_ssrcs = {kVideoSendSsrcs[0]};
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config.local_ssrc = kReceiverLocalVideoSsrc;
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for (const RtpExtension& extension : video_send_config_.rtp.extensions)
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config.rtp_header_extensions.push_back(extension);
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flexfec_receive_configs_.push_back(config);
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}
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}
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void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
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video_receive_configs_.clear();
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allocated_decoders_.clear();
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if (num_video_streams_ > 0) {
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std::vector<VideoReceiveStream::Config> new_configs =
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CreateMatchingVideoReceiveConfigs(video_send_config_,
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rtcp_send_transport);
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for (VideoReceiveStream::Config& config : new_configs) {
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video_receive_configs_.push_back(config.Copy());
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}
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}
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CreateMatchingAudioAndFecConfigs(rtcp_send_transport);
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}
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void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
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float speed,
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int framerate,
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int width,
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int height) {
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frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
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width, height, framerate * speed, clock));
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video_send_stream_->SetSource(
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frame_generator_capturer_.get(),
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VideoSendStream::DegradationPreference::kMaintainFramerate);
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}
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void CallTest::CreateFrameGeneratorCapturer(int framerate,
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int width,
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int height) {
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frame_generator_capturer_.reset(
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test::FrameGeneratorCapturer::Create(width, height, framerate, clock_));
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video_send_stream_->SetSource(
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frame_generator_capturer_.get(),
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VideoSendStream::DegradationPreference::kMaintainFramerate);
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}
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void CallTest::CreateFakeAudioDevices(
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std::unique_ptr<FakeAudioDevice::Capturer> capturer,
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std::unique_ptr<FakeAudioDevice::Renderer> renderer) {
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fake_send_audio_device_ = new rtc::RefCountedObject<FakeAudioDevice>(
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std::move(capturer), nullptr, 1.f);
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fake_recv_audio_device_ = new rtc::RefCountedObject<FakeAudioDevice>(
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nullptr, std::move(renderer), 1.f);
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}
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void CallTest::CreateVideoStreams() {
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RTC_DCHECK(video_send_stream_ == nullptr);
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RTC_DCHECK(video_receive_streams_.empty());
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video_send_stream_ = sender_call_->CreateVideoSendStream(
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video_send_config_.Copy(), video_encoder_config_.Copy());
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for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
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video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
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video_receive_configs_[i].Copy()));
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}
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AssociateFlexfecStreamsWithVideoStreams();
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}
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void CallTest::CreateAudioStreams() {
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RTC_DCHECK(audio_send_stream_ == nullptr);
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RTC_DCHECK(audio_receive_streams_.empty());
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audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
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for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
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audio_receive_streams_.push_back(
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receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
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}
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}
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void CallTest::CreateFlexfecStreams() {
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for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) {
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flexfec_receive_streams_.push_back(
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receiver_call_->CreateFlexfecReceiveStream(
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flexfec_receive_configs_[i]));
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}
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AssociateFlexfecStreamsWithVideoStreams();
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}
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void CallTest::AssociateFlexfecStreamsWithVideoStreams() {
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// All FlexFEC streams protect all of the video streams.
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for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) {
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for (VideoReceiveStream* video_recv_stream : video_receive_streams_) {
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video_recv_stream->AddSecondarySink(flexfec_recv_stream);
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}
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}
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}
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void CallTest::DissociateFlexfecStreamsFromVideoStreams() {
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for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) {
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for (VideoReceiveStream* video_recv_stream : video_receive_streams_) {
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video_recv_stream->RemoveSecondarySink(flexfec_recv_stream);
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}
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}
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}
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void CallTest::Start() {
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if (video_send_stream_)
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video_send_stream_->Start();
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for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
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video_recv_stream->Start();
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if (audio_send_stream_) {
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audio_send_stream_->Start();
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}
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for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
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audio_recv_stream->Start();
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if (frame_generator_capturer_.get() != NULL)
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frame_generator_capturer_->Start();
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}
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void CallTest::Stop() {
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if (frame_generator_capturer_.get() != NULL)
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frame_generator_capturer_->Stop();
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for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
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audio_recv_stream->Stop();
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if (audio_send_stream_) {
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audio_send_stream_->Stop();
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}
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for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
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video_recv_stream->Stop();
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if (video_send_stream_)
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video_send_stream_->Stop();
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}
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void CallTest::DestroyStreams() {
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DissociateFlexfecStreamsFromVideoStreams();
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if (audio_send_stream_)
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sender_call_->DestroyAudioSendStream(audio_send_stream_);
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audio_send_stream_ = nullptr;
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for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
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receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
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if (video_send_stream_)
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sender_call_->DestroyVideoSendStream(video_send_stream_);
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video_send_stream_ = nullptr;
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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 uint8_t CallTest::kVideoSendPayloadType = 100;
|
|
const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
|
|
const uint8_t CallTest::kSendRtxPayloadType = 98;
|
|
const uint8_t CallTest::kRedPayloadType = 118;
|
|
const uint8_t CallTest::kRtxRedPayloadType = 99;
|
|
const uint8_t CallTest::kUlpfecPayloadType = 119;
|
|
const uint8_t CallTest::kFlexfecPayloadType = 120;
|
|
const uint8_t CallTest::kAudioSendPayloadType = 103;
|
|
const uint8_t CallTest::kPayloadTypeH264 = 122;
|
|
const uint8_t CallTest::kPayloadTypeVP8 = 123;
|
|
const uint8_t CallTest::kPayloadTypeVP9 = 124;
|
|
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<FakeAudioDevice::Capturer> BaseTest::CreateCapturer() {
|
|
return FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000);
|
|
}
|
|
|
|
std::unique_ptr<FakeAudioDevice::Renderer> BaseTest::CreateRenderer() {
|
|
return FakeAudioDevice::CreateDiscardRenderer(48000);
|
|
}
|
|
|
|
void BaseTest::OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
|
|
FakeAudioDevice* 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
|