This is a refactoring change in preparation for enabling AudioMixer with the goal to have a small CL as possible for passing audio through the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/ enables the mixer. An object of class AudioState is shared across different webrtc audio connections. It is created in tests and in WebRTCVoiceEngine. AudioState is constructed by passing a Config struct, where one argument is scoped_refptr<AudioMixer>. Populating this field has now been mandatory. Tests and WebRTCVoiceEngine create and pass either a AudioMixerImpl. WebRTCVoiceEngine passes a real AudioMixer, which is currently unused. An alternative would have tests pass a mocked audio mixer. We chose not to do that, because we believe that tests should use the real thing unless there are reasons against it. Construction time is not an issue, because the real mixer is relatively lightweight. We couldn't find a way to test any mixer-related changes in AudioState before the mixes is connected. The next dependent CL https://codereview.webrtc.org/2436033002/ contains unit tests for mixer usage. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2469743002 Cr-Commit-Position: refs/heads/master@{#15134}
513 lines
18 KiB
C++
513 lines
18 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 "webrtc/test/call_test.h"
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#include <algorithm>
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#include "webrtc/base/checks.h"
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#include "webrtc/config.h"
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#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
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#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
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#include "webrtc/test/call_test.h"
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#include "webrtc/test/testsupport/fileutils.h"
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#include "webrtc/voice_engine/include/voe_base.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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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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fake_send_audio_device_(nullptr),
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fake_recv_audio_device_(nullptr) {}
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CallTest::~CallTest() {
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}
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void CallTest::RunBaseTest(BaseTest* 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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CreateVoiceEngines();
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AudioState::Config audio_state_config;
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audio_state_config.voice_engine = voe_send_.voice_engine;
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audio_state_config.audio_mixer = AudioMixerImpl::Create();
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send_config.audio_state = AudioState::Create(audio_state_config);
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}
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CreateSenderCall(send_config);
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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.voice_engine = voe_recv_.voice_engine;
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audio_state_config.audio_mixer = AudioMixerImpl::Create();
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recv_config.audio_state = AudioState::Create(audio_state_config);
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}
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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());
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send_transport_.reset(test->CreateSendTransport(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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} 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_, num_flexfec_streams_,
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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_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_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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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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test->PerformTest();
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send_transport_->StopSending();
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receive_transport_->StopSending();
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Stop();
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DestroyStreams();
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DestroyCalls();
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if (num_audio_streams_ > 0)
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DestroyVoiceEngines();
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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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fake_send_audio_device_->Start();
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audio_send_stream_->Start();
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EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
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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 (!audio_receive_streams_.empty()) {
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fake_recv_audio_device_->Start();
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EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
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}
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for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
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flexfec_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 (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
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flexfec_recv_stream->Stop();
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if (!audio_receive_streams_.empty()) {
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fake_recv_audio_device_->Stop();
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EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
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}
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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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fake_send_audio_device_->Stop();
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EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
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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::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_.reset(Call::Create(config));
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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::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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RTC_DCHECK(num_video_streams <= kNumSsrcs);
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RTC_DCHECK_LE(num_audio_streams, 1u);
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RTC_DCHECK_LE(num_flexfec_streams, 1u);
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RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
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if (num_video_streams > 0) {
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video_send_config_ = VideoSendStream::Config(send_transport);
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video_send_config_.encoder_settings.encoder = &fake_encoder_;
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video_send_config_.encoder_settings.payload_name = "FAKE";
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video_send_config_.encoder_settings.payload_type =
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kFakeVideoSendPayloadType;
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video_send_config_.rtp.extensions.push_back(
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RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
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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_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
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video_send_config_.rtp.extensions.push_back(RtpExtension(
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RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
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}
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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_.voe_channel_id = voe_send_.channel_id;
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audio_send_config_.rtp.ssrc = kAudioSendSsrc;
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audio_send_config_.send_codec_spec.codec_inst =
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CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
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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.flexfec_payload_type = kFlexfecPayloadType;
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video_send_config_.rtp.flexfec.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::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
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RTC_DCHECK(video_receive_configs_.empty());
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RTC_DCHECK(allocated_decoders_.empty());
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if (num_video_streams_ > 0) {
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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 = 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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video_receive_configs_.push_back(video_config.Copy());
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}
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}
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RTC_DCHECK_GE(1u, num_audio_streams_);
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if (num_audio_streams_ == 1) {
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RTC_DCHECK_LE(0, voe_send_.channel_id);
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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.voe_channel_id = voe_recv_.channel_id;
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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_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 flexfec_config;
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flexfec_config.flexfec_payload_type = kFlexfecPayloadType;
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flexfec_config.flexfec_ssrc = kFlexfecSendSsrc;
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flexfec_config.protected_media_ssrcs = {kVideoSendSsrcs[0]};
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flexfec_receive_configs_.push_back(flexfec_config);
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}
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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::kBalanced);
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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::kBalanced);
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}
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void CallTest::CreateFakeAudioDevices() {
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fake_send_audio_device_.reset(new FakeAudioDevice(
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clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
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DriftingClock::kNoDrift));
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fake_recv_audio_device_.reset(new FakeAudioDevice(
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clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
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DriftingClock::kNoDrift));
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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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RTC_DCHECK(audio_send_stream_ == nullptr);
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RTC_DCHECK(audio_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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}
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void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
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frame_generator_capturer_->SetFakeRotation(rotation);
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}
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void CallTest::CreateAudioStreams() {
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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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}
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void CallTest::DestroyStreams() {
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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_)
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receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
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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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for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
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receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
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video_receive_streams_.clear();
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allocated_decoders_.clear();
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}
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void CallTest::CreateVoiceEngines() {
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CreateFakeAudioDevices();
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voe_send_.voice_engine = VoiceEngine::Create();
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voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
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EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
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decoder_factory_));
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VoEBase::ChannelConfig config;
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config.enable_voice_pacing = true;
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voe_send_.channel_id = voe_send_.base->CreateChannel(config);
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EXPECT_GE(voe_send_.channel_id, 0);
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voe_recv_.voice_engine = VoiceEngine::Create();
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voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
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EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr,
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decoder_factory_));
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voe_recv_.channel_id = voe_recv_.base->CreateChannel();
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EXPECT_GE(voe_recv_.channel_id, 0);
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}
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void CallTest::DestroyVoiceEngines() {
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voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
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voe_recv_.channel_id = -1;
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voe_recv_.base->Release();
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voe_recv_.base = nullptr;
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voe_send_.base->DeleteChannel(voe_send_.channel_id);
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voe_send_.channel_id = -1;
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voe_send_.base->Release();
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voe_send_.base = nullptr;
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VoiceEngine::Delete(voe_send_.voice_engine);
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voe_send_.voice_engine = nullptr;
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VoiceEngine::Delete(voe_recv_.voice_engine);
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voe_recv_.voice_engine = nullptr;
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}
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const int CallTest::kDefaultWidth;
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const int CallTest::kDefaultHeight;
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const int CallTest::kDefaultFramerate;
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const int CallTest::kDefaultTimeoutMs = 30 * 1000;
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const int CallTest::kLongTimeoutMs = 120 * 1000;
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const uint8_t CallTest::kVideoSendPayloadType = 100;
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const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
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const uint8_t CallTest::kSendRtxPayloadType = 98;
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const uint8_t CallTest::kRedPayloadType = 118;
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const uint8_t CallTest::kRtxRedPayloadType = 99;
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const uint8_t CallTest::kUlpfecPayloadType = 119;
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const uint8_t CallTest::kFlexfecPayloadType = 120;
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const uint8_t CallTest::kAudioSendPayloadType = 103;
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const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
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0xBADCAFF};
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const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
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0xC0FFEF};
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|
const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
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|
const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF;
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|
const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
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|
const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
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|
const int CallTest::kNackRtpHistoryMs = 1000;
|
|
|
|
BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
|
|
}
|
|
|
|
BaseTest::~BaseTest() {
|
|
}
|
|
|
|
Call::Config BaseTest::GetSenderCallConfig() {
|
|
return Call::Config(&event_log_);
|
|
}
|
|
|
|
Call::Config BaseTest::GetReceiverCallConfig() {
|
|
return Call::Config(&event_log_);
|
|
}
|
|
|
|
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
|
|
}
|
|
|
|
test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
|
|
return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
|
|
FakeNetworkPipe::Config());
|
|
}
|
|
|
|
test::PacketTransport* BaseTest::CreateReceiveTransport() {
|
|
return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
|
|
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) {
|
|
}
|
|
|
|
SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
|
|
}
|
|
|
|
bool SendTest::ShouldCreateReceivers() const {
|
|
return false;
|
|
}
|
|
|
|
EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
|
|
}
|
|
|
|
bool EndToEndTest::ShouldCreateReceivers() const {
|
|
return true;
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|