10111bc495
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}
294 lines
10 KiB
C++
294 lines
10 KiB
C++
/*
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* Copyright (c) 2015 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 <list>
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#include <memory>
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#include "webrtc/api/call/audio_state.h"
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#include "webrtc/call.h"
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#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
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#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
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#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/test/mock_voice_engine.h"
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namespace {
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struct CallHelper {
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explicit CallHelper(
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
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: voice_engine_(decoder_factory) {
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webrtc::AudioState::Config audio_state_config;
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audio_state_config.voice_engine = &voice_engine_;
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audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
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EXPECT_CALL(voice_engine_, audio_device_module());
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EXPECT_CALL(voice_engine_, audio_processing());
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EXPECT_CALL(voice_engine_, audio_transport());
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webrtc::Call::Config config(&event_log_);
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config.audio_state = webrtc::AudioState::Create(audio_state_config);
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call_.reset(webrtc::Call::Create(config));
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}
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webrtc::Call* operator->() { return call_.get(); }
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webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; }
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private:
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testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
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webrtc::RtcEventLogNullImpl event_log_;
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std::unique_ptr<webrtc::Call> call_;
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};
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} // namespace
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namespace webrtc {
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TEST(CallTest, ConstructDestruct) {
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CallHelper call;
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}
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TEST(CallTest, CreateDestroy_AudioSendStream) {
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CallHelper call;
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AudioSendStream::Config config(nullptr);
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config.rtp.ssrc = 42;
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config.voe_channel_id = 123;
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AudioSendStream* stream = call->CreateAudioSendStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyAudioSendStream(stream);
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}
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TEST(CallTest, CreateDestroy_AudioReceiveStream) {
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
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CallHelper call(decoder_factory);
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AudioReceiveStream::Config config;
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config.rtp.remote_ssrc = 42;
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config.voe_channel_id = 123;
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config.decoder_factory = decoder_factory;
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AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyAudioReceiveStream(stream);
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}
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TEST(CallTest, CreateDestroy_AudioSendStreams) {
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CallHelper call;
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AudioSendStream::Config config(nullptr);
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config.voe_channel_id = 123;
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std::list<AudioSendStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.rtp.ssrc = ssrc;
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AudioSendStream* stream = call->CreateAudioSendStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyAudioSendStream(s);
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}
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streams.clear();
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}
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}
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TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
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CallHelper call(decoder_factory);
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AudioReceiveStream::Config config;
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config.voe_channel_id = 123;
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config.decoder_factory = decoder_factory;
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std::list<AudioReceiveStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.rtp.remote_ssrc = ssrc;
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AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyAudioReceiveStream(s);
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}
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streams.clear();
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}
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}
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TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
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CallHelper call(decoder_factory);
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constexpr int kRecvChannelId = 101;
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// Set up the mock to create a channel proxy which we know of, so that we can
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// add our expectations to it.
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test::MockVoEChannelProxy* recv_channel_proxy = nullptr;
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EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_))
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.WillRepeatedly(testing::Invoke([&](int channel_id) {
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test::MockVoEChannelProxy* channel_proxy =
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new testing::NiceMock<test::MockVoEChannelProxy>();
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EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
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.WillRepeatedly(testing::ReturnRef(decoder_factory));
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// If being called for the send channel, save a pointer to the channel
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// proxy for later.
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if (channel_id == kRecvChannelId) {
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EXPECT_FALSE(recv_channel_proxy);
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recv_channel_proxy = channel_proxy;
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}
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return channel_proxy;
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}));
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AudioReceiveStream::Config recv_config;
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recv_config.rtp.remote_ssrc = 42;
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recv_config.rtp.local_ssrc = 777;
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recv_config.voe_channel_id = kRecvChannelId;
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recv_config.decoder_factory = decoder_factory;
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AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
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EXPECT_NE(recv_stream, nullptr);
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EXPECT_CALL(*recv_channel_proxy, AssociateSendChannel(testing::_)).Times(1);
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AudioSendStream::Config send_config(nullptr);
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send_config.rtp.ssrc = 777;
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send_config.voe_channel_id = 123;
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AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
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EXPECT_NE(send_stream, nullptr);
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EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1);
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call->DestroyAudioSendStream(send_stream);
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EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1);
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call->DestroyAudioReceiveStream(recv_stream);
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}
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TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
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CallHelper call(decoder_factory);
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constexpr int kRecvChannelId = 101;
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// Set up the mock to create a channel proxy which we know of, so that we can
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// add our expectations to it.
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test::MockVoEChannelProxy* recv_channel_proxy = nullptr;
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EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_))
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.WillRepeatedly(testing::Invoke([&](int channel_id) {
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test::MockVoEChannelProxy* channel_proxy =
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new testing::NiceMock<test::MockVoEChannelProxy>();
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EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
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.WillRepeatedly(testing::ReturnRef(decoder_factory));
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// If being called for the send channel, save a pointer to the channel
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// proxy for later.
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if (channel_id == kRecvChannelId) {
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EXPECT_FALSE(recv_channel_proxy);
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recv_channel_proxy = channel_proxy;
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// We need to set this expectation here since the channel proxy is
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// created as a side effect of CreateAudioReceiveStream().
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EXPECT_CALL(*recv_channel_proxy,
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AssociateSendChannel(testing::_)).Times(1);
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}
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return channel_proxy;
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}));
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AudioSendStream::Config send_config(nullptr);
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send_config.rtp.ssrc = 777;
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send_config.voe_channel_id = 123;
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AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
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EXPECT_NE(send_stream, nullptr);
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AudioReceiveStream::Config recv_config;
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recv_config.rtp.remote_ssrc = 42;
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recv_config.rtp.local_ssrc = 777;
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recv_config.voe_channel_id = kRecvChannelId;
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recv_config.decoder_factory = decoder_factory;
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AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
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EXPECT_NE(recv_stream, nullptr);
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EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1);
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call->DestroyAudioReceiveStream(recv_stream);
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call->DestroyAudioSendStream(send_stream);
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}
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TEST(CallTest, CreateDestroy_FlexfecReceiveStream) {
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CallHelper call;
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FlexfecReceiveStream::Config config;
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config.flexfec_payload_type = 118;
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config.flexfec_ssrc = 38837212;
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config.protected_media_ssrcs = {27273};
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FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyFlexfecReceiveStream(stream);
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}
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TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) {
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CallHelper call;
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FlexfecReceiveStream::Config config;
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config.flexfec_payload_type = 118;
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std::list<FlexfecReceiveStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.flexfec_ssrc = ssrc;
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config.protected_media_ssrcs = {ssrc + 1};
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FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyFlexfecReceiveStream(s);
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}
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streams.clear();
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}
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}
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TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
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CallHelper call;
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FlexfecReceiveStream::Config config;
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config.flexfec_payload_type = 118;
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config.protected_media_ssrcs = {1324234};
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FlexfecReceiveStream* stream;
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std::list<FlexfecReceiveStream*> streams;
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config.flexfec_ssrc = 838383;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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config.flexfec_ssrc = 424993;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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config.flexfec_ssrc = 99383;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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config.flexfec_ssrc = 5548;
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stream = call->CreateFlexfecReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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streams.push_back(stream);
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for (auto s : streams) {
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call->DestroyFlexfecReceiveStream(s);
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}
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}
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} // namespace webrtc
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