| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <list> |
| #include <map> |
| #include <memory> |
| #include <utility> |
| |
| #include "webrtc/api/test/mock_audio_mixer.h" |
| #include "webrtc/call/audio_state.h" |
| #include "webrtc/call/call.h" |
| #include "webrtc/call/fake_rtp_transport_controller_send.h" |
| #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| #include "webrtc/modules/audio_device/include/mock_audio_device.h" |
| #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
| #include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h" |
| #include "webrtc/modules/pacing/mock/mock_paced_sender.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/rtc_base/ptr_util.h" |
| #include "webrtc/test/fake_encoder.h" |
| #include "webrtc/test/gtest.h" |
| #include "webrtc/test/mock_audio_decoder_factory.h" |
| #include "webrtc/test/mock_transport.h" |
| #include "webrtc/test/mock_voice_engine.h" |
| |
| namespace { |
| |
| struct CallHelper { |
| explicit CallHelper( |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) |
| : voice_engine_(decoder_factory) { |
| webrtc::AudioState::Config audio_state_config; |
| audio_state_config.voice_engine = &voice_engine_; |
| audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
| audio_state_config.audio_processing = webrtc::AudioProcessing::Create(); |
| EXPECT_CALL(voice_engine_, audio_device_module()); |
| EXPECT_CALL(voice_engine_, audio_transport()); |
| webrtc::Call::Config config(&event_log_); |
| config.audio_state = webrtc::AudioState::Create(audio_state_config); |
| call_.reset(webrtc::Call::Create(config)); |
| } |
| |
| webrtc::Call* operator->() { return call_.get(); } |
| webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; } |
| |
| private: |
| testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; |
| webrtc::RtcEventLogNullImpl event_log_; |
| std::unique_ptr<webrtc::Call> call_; |
| }; |
| } // namespace |
| |
| namespace webrtc { |
| |
| TEST(CallTest, ConstructDestruct) { |
| CallHelper call; |
| } |
| |
| TEST(CallTest, CreateDestroy_AudioSendStream) { |
| CallHelper call; |
| AudioSendStream::Config config(nullptr); |
| config.rtp.ssrc = 42; |
| config.voe_channel_id = 123; |
| AudioSendStream* stream = call->CreateAudioSendStream(config); |
| EXPECT_NE(stream, nullptr); |
| call->DestroyAudioSendStream(stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_AudioReceiveStream) { |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); |
| CallHelper call(decoder_factory); |
| AudioReceiveStream::Config config; |
| config.rtp.remote_ssrc = 42; |
| config.voe_channel_id = 123; |
| config.decoder_factory = decoder_factory; |
| AudioReceiveStream* stream = call->CreateAudioReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| call->DestroyAudioReceiveStream(stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_AudioSendStreams) { |
| CallHelper call; |
| AudioSendStream::Config config(nullptr); |
| config.voe_channel_id = 123; |
| std::list<AudioSendStream*> streams; |
| for (int i = 0; i < 2; ++i) { |
| for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { |
| config.rtp.ssrc = ssrc; |
| AudioSendStream* stream = call->CreateAudioSendStream(config); |
| EXPECT_NE(stream, nullptr); |
| if (ssrc & 1) { |
| streams.push_back(stream); |
| } else { |
| streams.push_front(stream); |
| } |
| } |
| for (auto s : streams) { |
| call->DestroyAudioSendStream(s); |
| } |
| streams.clear(); |
| } |
| } |
| |
| TEST(CallTest, CreateDestroy_AudioReceiveStreams) { |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); |
| CallHelper call(decoder_factory); |
| AudioReceiveStream::Config config; |
| config.voe_channel_id = 123; |
| config.decoder_factory = decoder_factory; |
| std::list<AudioReceiveStream*> streams; |
| for (int i = 0; i < 2; ++i) { |
| for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { |
| config.rtp.remote_ssrc = ssrc; |
| AudioReceiveStream* stream = call->CreateAudioReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| if (ssrc & 1) { |
| streams.push_back(stream); |
| } else { |
| streams.push_front(stream); |
| } |
| } |
| for (auto s : streams) { |
| call->DestroyAudioReceiveStream(s); |
| } |
| streams.clear(); |
| } |
| } |
| |
| TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) { |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); |
| CallHelper call(decoder_factory); |
| ::testing::NiceMock<MockRtpRtcp> mock_rtp_rtcp; |
| |
| constexpr int kRecvChannelId = 101; |
| |
| // Set up the mock to create a channel proxy which we know of, so that we can |
| // add our expectations to it. |
| test::MockVoEChannelProxy* recv_channel_proxy = nullptr; |
| EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_)) |
| .WillRepeatedly(testing::Invoke([&](int channel_id) { |
| test::MockVoEChannelProxy* channel_proxy = |
| new testing::NiceMock<test::MockVoEChannelProxy>(); |
| EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory()) |
| .WillRepeatedly(testing::ReturnRef(decoder_factory)); |
| EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_)) |
| .WillRepeatedly(testing::Invoke( |
| [](const std::map<int, SdpAudioFormat>& codecs) { |
| EXPECT_THAT(codecs, testing::IsEmpty()); |
| })); |
| EXPECT_CALL(*channel_proxy, GetRtpRtcp(testing::_, testing::_)) |
| .WillRepeatedly(testing::SetArgPointee<0>(&mock_rtp_rtcp)); |
| // If being called for the send channel, save a pointer to the channel |
| // proxy for later. |
| if (channel_id == kRecvChannelId) { |
| EXPECT_FALSE(recv_channel_proxy); |
| recv_channel_proxy = channel_proxy; |
| } |
| return channel_proxy; |
| })); |
| |
| AudioReceiveStream::Config recv_config; |
| recv_config.rtp.remote_ssrc = 42; |
| recv_config.rtp.local_ssrc = 777; |
| recv_config.voe_channel_id = kRecvChannelId; |
| recv_config.decoder_factory = decoder_factory; |
| AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config); |
| EXPECT_NE(recv_stream, nullptr); |
| |
| EXPECT_CALL(*recv_channel_proxy, AssociateSendChannel(testing::_)).Times(1); |
| AudioSendStream::Config send_config(nullptr); |
| send_config.rtp.ssrc = 777; |
| send_config.voe_channel_id = 123; |
| AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); |
| EXPECT_NE(send_stream, nullptr); |
| |
| EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1); |
| call->DestroyAudioSendStream(send_stream); |
| |
| EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1); |
| call->DestroyAudioReceiveStream(recv_stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) { |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory( |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>); |
| CallHelper call(decoder_factory); |
| ::testing::NiceMock<MockRtpRtcp> mock_rtp_rtcp; |
| |
| constexpr int kRecvChannelId = 101; |
| |
| // Set up the mock to create a channel proxy which we know of, so that we can |
| // add our expectations to it. |
| test::MockVoEChannelProxy* recv_channel_proxy = nullptr; |
| EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_)) |
| .WillRepeatedly(testing::Invoke([&](int channel_id) { |
| test::MockVoEChannelProxy* channel_proxy = |
| new testing::NiceMock<test::MockVoEChannelProxy>(); |
| EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory()) |
| .WillRepeatedly(testing::ReturnRef(decoder_factory)); |
| EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_)) |
| .WillRepeatedly(testing::Invoke( |
| [](const std::map<int, SdpAudioFormat>& codecs) { |
| EXPECT_THAT(codecs, testing::IsEmpty()); |
| })); |
| EXPECT_CALL(*channel_proxy, GetRtpRtcp(testing::_, testing::_)) |
| .WillRepeatedly(testing::SetArgPointee<0>(&mock_rtp_rtcp)); |
| // If being called for the send channel, save a pointer to the channel |
| // proxy for later. |
| if (channel_id == kRecvChannelId) { |
| EXPECT_FALSE(recv_channel_proxy); |
| recv_channel_proxy = channel_proxy; |
| // We need to set this expectation here since the channel proxy is |
| // created as a side effect of CreateAudioReceiveStream(). |
| EXPECT_CALL(*recv_channel_proxy, |
| AssociateSendChannel(testing::_)).Times(1); |
| } |
| return channel_proxy; |
| })); |
| |
| AudioSendStream::Config send_config(nullptr); |
| send_config.rtp.ssrc = 777; |
| send_config.voe_channel_id = 123; |
| AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); |
| EXPECT_NE(send_stream, nullptr); |
| |
| AudioReceiveStream::Config recv_config; |
| recv_config.rtp.remote_ssrc = 42; |
| recv_config.rtp.local_ssrc = 777; |
| recv_config.voe_channel_id = kRecvChannelId; |
| recv_config.decoder_factory = decoder_factory; |
| AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config); |
| EXPECT_NE(recv_stream, nullptr); |
| |
| EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1); |
| call->DestroyAudioReceiveStream(recv_stream); |
| |
| call->DestroyAudioSendStream(send_stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_FlexfecReceiveStream) { |
| CallHelper call; |
| MockTransport rtcp_send_transport; |
| FlexfecReceiveStream::Config config(&rtcp_send_transport); |
| config.payload_type = 118; |
| config.remote_ssrc = 38837212; |
| config.protected_media_ssrcs = {27273}; |
| |
| FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| call->DestroyFlexfecReceiveStream(stream); |
| } |
| |
| TEST(CallTest, CreateDestroy_FlexfecReceiveStreams) { |
| CallHelper call; |
| MockTransport rtcp_send_transport; |
| FlexfecReceiveStream::Config config(&rtcp_send_transport); |
| config.payload_type = 118; |
| std::list<FlexfecReceiveStream*> streams; |
| |
| for (int i = 0; i < 2; ++i) { |
| for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { |
| config.remote_ssrc = ssrc; |
| config.protected_media_ssrcs = {ssrc + 1}; |
| FlexfecReceiveStream* stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| if (ssrc & 1) { |
| streams.push_back(stream); |
| } else { |
| streams.push_front(stream); |
| } |
| } |
| for (auto s : streams) { |
| call->DestroyFlexfecReceiveStream(s); |
| } |
| streams.clear(); |
| } |
| } |
| |
| TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) { |
| CallHelper call; |
| MockTransport rtcp_send_transport; |
| FlexfecReceiveStream::Config config(&rtcp_send_transport); |
| config.payload_type = 118; |
| config.protected_media_ssrcs = {1324234}; |
| FlexfecReceiveStream* stream; |
| std::list<FlexfecReceiveStream*> streams; |
| |
| config.remote_ssrc = 838383; |
| stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| streams.push_back(stream); |
| |
| config.remote_ssrc = 424993; |
| stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| streams.push_back(stream); |
| |
| config.remote_ssrc = 99383; |
| stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| streams.push_back(stream); |
| |
| config.remote_ssrc = 5548; |
| stream = call->CreateFlexfecReceiveStream(config); |
| EXPECT_NE(stream, nullptr); |
| streams.push_back(stream); |
| |
| for (auto s : streams) { |
| call->DestroyFlexfecReceiveStream(s); |
| } |
| } |
| |
| namespace { |
| struct CallBitrateHelper { |
| CallBitrateHelper() : CallBitrateHelper(Call::Config::BitrateConfig()) {} |
| |
| explicit CallBitrateHelper(const Call::Config::BitrateConfig& bitrate_config) |
| : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &pacer_) { |
| Call::Config config(&event_log_); |
| config.bitrate_config = bitrate_config; |
| call_.reset( |
| Call::Create(config, rtc::MakeUnique<FakeRtpTransportControllerSend>( |
| &packet_router_, &pacer_, &mock_cc_))); |
| } |
| |
| webrtc::Call* operator->() { return call_.get(); } |
| testing::NiceMock<test::MockSendSideCongestionController>& mock_cc() { |
| return mock_cc_; |
| } |
| |
| private: |
| webrtc::RtcEventLogNullImpl event_log_; |
| PacketRouter packet_router_; |
| testing::NiceMock<MockPacedSender> pacer_; |
| testing::NiceMock<test::MockSendSideCongestionController> mock_cc_; |
| std::unique_ptr<Call> call_; |
| }; |
| } // namespace |
| |
| TEST(CallBitrateTest, SetBitrateConfigWithValidConfigCallsSetBweBitrates) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 1; |
| bitrate_config.start_bitrate_bps = 2; |
| bitrate_config.max_bitrate_bps = 3; |
| |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(1, 2, 3)); |
| call->SetBitrateConfig(bitrate_config); |
| } |
| |
| TEST(CallBitrateTest, SetBitrateConfigWithDifferentMinCallsSetBweBitrates) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 10; |
| bitrate_config.start_bitrate_bps = 20; |
| bitrate_config.max_bitrate_bps = 30; |
| call->SetBitrateConfig(bitrate_config); |
| |
| bitrate_config.min_bitrate_bps = 11; |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(11, -1, 30)); |
| call->SetBitrateConfig(bitrate_config); |
| } |
| |
| TEST(CallBitrateTest, SetBitrateConfigWithDifferentStartCallsSetBweBitrates) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 10; |
| bitrate_config.start_bitrate_bps = 20; |
| bitrate_config.max_bitrate_bps = 30; |
| call->SetBitrateConfig(bitrate_config); |
| |
| bitrate_config.start_bitrate_bps = 21; |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(10, 21, 30)); |
| call->SetBitrateConfig(bitrate_config); |
| } |
| |
| TEST(CallBitrateTest, SetBitrateConfigWithDifferentMaxCallsSetBweBitrates) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 10; |
| bitrate_config.start_bitrate_bps = 20; |
| bitrate_config.max_bitrate_bps = 30; |
| call->SetBitrateConfig(bitrate_config); |
| |
| bitrate_config.max_bitrate_bps = 31; |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(10, -1, 31)); |
| call->SetBitrateConfig(bitrate_config); |
| } |
| |
| TEST(CallBitrateTest, SetBitrateConfigWithSameConfigElidesSecondCall) { |
| CallBitrateHelper call; |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 1; |
| bitrate_config.start_bitrate_bps = 2; |
| bitrate_config.max_bitrate_bps = 3; |
| |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(1, 2, 3)).Times(1); |
| call->SetBitrateConfig(bitrate_config); |
| call->SetBitrateConfig(bitrate_config); |
| } |
| |
| TEST(CallBitrateTest, |
| SetBitrateConfigWithSameMinMaxAndNegativeStartElidesSecondCall) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 1; |
| bitrate_config.start_bitrate_bps = 2; |
| bitrate_config.max_bitrate_bps = 3; |
| |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(1, 2, 3)).Times(1); |
| call->SetBitrateConfig(bitrate_config); |
| |
| bitrate_config.start_bitrate_bps = -1; |
| call->SetBitrateConfig(bitrate_config); |
| } |
| |
| TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) { |
| constexpr uint32_t kSSRC = 12345; |
| testing::NiceMock<test::MockAudioDeviceModule> mock_adm; |
| // Reply with a 10ms timer every time TimeUntilNextProcess is called to |
| // avoid entering a tight loop on the process thread. |
| EXPECT_CALL(mock_adm, TimeUntilNextProcess()) |
| .WillRepeatedly(testing::Return(10)); |
| rtc::scoped_refptr<test::MockAudioMixer> mock_mixer( |
| new rtc::RefCountedObject<test::MockAudioMixer>); |
| |
| // There's similar functionality in cricket::VoEWrapper but it's not reachable |
| // from here. Since we're working on removing VoE interfaces, I doubt it's |
| // worth making VoEWrapper more easily available. |
| struct ScopedVoiceEngine { |
| ScopedVoiceEngine() |
| : voe(VoiceEngine::Create()), |
| base(VoEBase::GetInterface(voe)) {} |
| ~ScopedVoiceEngine() { |
| base->Release(); |
| EXPECT_TRUE(VoiceEngine::Delete(voe)); |
| } |
| |
| VoiceEngine* voe; |
| VoEBase* base; |
| }; |
| ScopedVoiceEngine voice_engine; |
| |
| AudioState::Config audio_state_config; |
| audio_state_config.voice_engine = voice_engine.voe; |
| audio_state_config.audio_mixer = mock_mixer; |
| audio_state_config.audio_processing = AudioProcessing::Create(); |
| voice_engine.base->Init(&mock_adm, audio_state_config.audio_processing.get()); |
| auto audio_state = AudioState::Create(audio_state_config); |
| |
| RtcEventLogNullImpl event_log; |
| Call::Config call_config(&event_log); |
| call_config.audio_state = audio_state; |
| std::unique_ptr<Call> call(Call::Create(call_config)); |
| |
| auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) { |
| AudioSendStream::Config config(nullptr); |
| config.rtp.ssrc = ssrc; |
| config.voe_channel_id = voice_engine.base->CreateChannel(); |
| AudioSendStream* stream = call->CreateAudioSendStream(config); |
| VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine.voe); |
| auto channel_proxy = voe_impl->GetChannelProxy(config.voe_channel_id); |
| RtpRtcp* rtp_rtcp = nullptr; |
| RtpReceiver* rtp_receiver = nullptr; // Unused but required for call. |
| channel_proxy->GetRtpRtcp(&rtp_rtcp, &rtp_receiver); |
| const RtpState rtp_state = rtp_rtcp->GetRtpState(); |
| call->DestroyAudioSendStream(stream); |
| voice_engine.base->DeleteChannel(config.voe_channel_id); |
| return rtp_state; |
| }; |
| |
| const RtpState rtp_state1 = create_stream_and_get_rtp_state(kSSRC); |
| const RtpState rtp_state2 = create_stream_and_get_rtp_state(kSSRC); |
| |
| EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number); |
| EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp); |
| EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp); |
| EXPECT_EQ(rtp_state1.capture_time_ms, rtp_state2.capture_time_ms); |
| EXPECT_EQ(rtp_state1.last_timestamp_time_ms, |
| rtp_state2.last_timestamp_time_ms); |
| EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent); |
| } |
| TEST(CallBitrateTest, BiggerMaskMinUsed) { |
| CallBitrateHelper call; |
| Call::Config::BitrateConfigMask mask; |
| mask.min_bitrate_bps = rtc::Optional<int>(1234); |
| |
| EXPECT_CALL(call.mock_cc(), |
| SetBweBitrates(*mask.min_bitrate_bps, testing::_, testing::_)); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| TEST(CallBitrateTest, BiggerConfigMinUsed) { |
| CallBitrateHelper call; |
| Call::Config::BitrateConfigMask mask; |
| mask.min_bitrate_bps = rtc::Optional<int>(1000); |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, testing::_, testing::_)); |
| call->SetBitrateConfigMask(mask); |
| |
| Call::Config::BitrateConfig config; |
| config.min_bitrate_bps = 1234; |
| |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(1234, testing::_, testing::_)); |
| call->SetBitrateConfig(config); |
| } |
| |
| // The last call to set start should be used. |
| TEST(CallBitrateTest, LatestStartMaskPreferred) { |
| CallBitrateHelper call; |
| Call::Config::BitrateConfigMask mask; |
| mask.start_bitrate_bps = rtc::Optional<int>(1300); |
| |
| EXPECT_CALL(call.mock_cc(), |
| SetBweBitrates(testing::_, *mask.start_bitrate_bps, testing::_)); |
| call->SetBitrateConfigMask(mask); |
| |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.start_bitrate_bps = 1200; |
| |
| EXPECT_CALL( |
| call.mock_cc(), |
| SetBweBitrates(testing::_, bitrate_config.start_bitrate_bps, testing::_)); |
| call->SetBitrateConfig(bitrate_config); |
| } |
| |
| TEST(CallBitrateTest, SmallerMaskMaxUsed) { |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.max_bitrate_bps = bitrate_config.start_bitrate_bps + 2000; |
| CallBitrateHelper call(bitrate_config); |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.max_bitrate_bps = |
| rtc::Optional<int>(bitrate_config.start_bitrate_bps + 1000); |
| |
| EXPECT_CALL(call.mock_cc(), |
| SetBweBitrates(testing::_, testing::_, *mask.max_bitrate_bps)); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| TEST(CallBitrateTest, SmallerConfigMaxUsed) { |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.max_bitrate_bps = bitrate_config.start_bitrate_bps + 1000; |
| CallBitrateHelper call(bitrate_config); |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.max_bitrate_bps = |
| rtc::Optional<int>(bitrate_config.start_bitrate_bps + 2000); |
| |
| // Expect no calls because nothing changes |
| EXPECT_CALL(call.mock_cc(), |
| SetBweBitrates(testing::_, testing::_, testing::_)) |
| .Times(0); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| TEST(CallBitrateTest, MaskStartLessThanConfigMinClamped) { |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 2000; |
| CallBitrateHelper call(bitrate_config); |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.start_bitrate_bps = rtc::Optional<int>(1000); |
| |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(2000, 2000, testing::_)); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| TEST(CallBitrateTest, MaskStartGreaterThanConfigMaxClamped) { |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.start_bitrate_bps = 2000; |
| CallBitrateHelper call(bitrate_config); |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.max_bitrate_bps = rtc::Optional<int>(1000); |
| |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, -1, 1000)); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| TEST(CallBitrateTest, MaskMinGreaterThanConfigMaxClamped) { |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.min_bitrate_bps = 2000; |
| CallBitrateHelper call(bitrate_config); |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.max_bitrate_bps = rtc::Optional<int>(1000); |
| |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, testing::_, 1000)); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| TEST(CallBitrateTest, SettingMaskStartForcesUpdate) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.start_bitrate_bps = rtc::Optional<int>(1000); |
| |
| // SetBweBitrates should be called twice with the same params since |
| // start_bitrate_bps is set. |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, 1000, testing::_)) |
| .Times(2); |
| call->SetBitrateConfigMask(mask); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| TEST(CallBitrateTest, SetBitrateConfigWithNoChangesDoesNotCallSetBweBitrates) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfig config1; |
| config1.min_bitrate_bps = 0; |
| config1.start_bitrate_bps = 1000; |
| config1.max_bitrate_bps = -1; |
| |
| Call::Config::BitrateConfig config2; |
| config2.min_bitrate_bps = 0; |
| config2.start_bitrate_bps = -1; |
| config2.max_bitrate_bps = -1; |
| |
| // The second call should not call SetBweBitrates because it doesn't |
| // change any values. |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, 1000, -1)); |
| call->SetBitrateConfig(config1); |
| call->SetBitrateConfig(config2); |
| } |
| |
| // If SetBitrateConfig changes the max, but not the effective max, |
| // SetBweBitrates shouldn't be called, to avoid unnecessary encoder |
| // reconfigurations. |
| TEST(CallBitrateTest, SetBweBitratesNotCalledWhenEffectiveMaxUnchanged) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfig config; |
| config.min_bitrate_bps = 0; |
| config.start_bitrate_bps = -1; |
| config.max_bitrate_bps = 2000; |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, testing::_, 2000)); |
| call->SetBitrateConfig(config); |
| |
| // Reduce effective max to 1000 with the mask. |
| Call::Config::BitrateConfigMask mask; |
| mask.max_bitrate_bps = rtc::Optional<int>(1000); |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, testing::_, 1000)); |
| call->SetBitrateConfigMask(mask); |
| |
| // This leaves the effective max unchanged, so SetBweBitrates shouldn't be |
| // called again. |
| config.max_bitrate_bps = 1000; |
| call->SetBitrateConfig(config); |
| } |
| |
| // When the "start bitrate" mask is removed, SetBweBitrates shouldn't be called |
| // again, since nothing's changing. |
| TEST(CallBitrateTest, SetBweBitratesNotCalledWhenStartMaskRemoved) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.start_bitrate_bps = rtc::Optional<int>(1000); |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, 1000, -1)); |
| call->SetBitrateConfigMask(mask); |
| |
| mask.start_bitrate_bps.reset(); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| // Test that if SetBitrateConfig is called after SetBitrateConfigMask applies a |
| // "start" value, the SetBitrateConfig call won't apply that start value a |
| // second time. |
| TEST(CallBitrateTest, SetBitrateConfigAfterSetBitrateConfigMaskWithStart) { |
| CallBitrateHelper call; |
| |
| Call::Config::BitrateConfigMask mask; |
| mask.start_bitrate_bps = rtc::Optional<int>(1000); |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, 1000, -1)); |
| call->SetBitrateConfigMask(mask); |
| |
| Call::Config::BitrateConfig config; |
| config.min_bitrate_bps = 0; |
| config.start_bitrate_bps = -1; |
| config.max_bitrate_bps = 5000; |
| // The start value isn't changing, so SetBweBitrates should be called with |
| // -1. |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, -1, 5000)); |
| call->SetBitrateConfig(config); |
| } |
| |
| TEST(CallBitrateTest, SetBweBitratesNotCalledWhenClampedMinUnchanged) { |
| Call::Config::BitrateConfig bitrate_config; |
| bitrate_config.start_bitrate_bps = 500; |
| bitrate_config.max_bitrate_bps = 1000; |
| CallBitrateHelper call(bitrate_config); |
| |
| // Set min to 2000; it is clamped to the max (1000). |
| Call::Config::BitrateConfigMask mask; |
| mask.min_bitrate_bps = rtc::Optional<int>(2000); |
| EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000)); |
| call->SetBitrateConfigMask(mask); |
| |
| // Set min to 3000; the clamped value stays the same so nothing happens. |
| mask.min_bitrate_bps = rtc::Optional<int>(3000); |
| call->SetBitrateConfigMask(mask); |
| } |
| |
| } // namespace webrtc |