| /* |
| * 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 <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "audio/audio_send_stream.h" |
| #include "audio/audio_state.h" |
| #include "audio/conversion.h" |
| #include "call/fake_rtp_transport_controller_send.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" |
| #include "modules/audio_mixer/audio_mixer_impl.h" |
| #include "modules/audio_processing/include/mock_audio_processing.h" |
| #include "modules/congestion_controller/include/mock/mock_congestion_observer.h" |
| #include "modules/congestion_controller/include/send_side_congestion_controller.h" |
| #include "modules/pacing/mock/mock_paced_sender.h" |
| #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
| #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
| #include "rtc_base/ptr_util.h" |
| #include "rtc_base/task_queue.h" |
| #include "test/gtest.h" |
| #include "test/mock_audio_encoder.h" |
| #include "test/mock_audio_encoder_factory.h" |
| #include "test/mock_voe_channel_proxy.h" |
| #include "test/mock_voice_engine.h" |
| #include "voice_engine/transmit_mixer.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| using testing::_; |
| using testing::Eq; |
| using testing::Ne; |
| using testing::Invoke; |
| using testing::Return; |
| using testing::StrEq; |
| |
| const int kChannelId = 1; |
| const uint32_t kSsrc = 1234; |
| const char* kCName = "foo_name"; |
| const int kAudioLevelId = 2; |
| const int kTransportSequenceNumberId = 4; |
| const int kEchoDelayMedian = 254; |
| const int kEchoDelayStdDev = -3; |
| const int kEchoReturnLoss = -65; |
| const int kEchoReturnLossEnhancement = 101; |
| const float kResidualEchoLikelihood = -1.0f; |
| const int32_t kSpeechInputLevel = 96; |
| const double kTotalInputEnergy = 0.25; |
| const double kTotalInputDuration = 0.5; |
| const CallStatistics kCallStats = { |
| 1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123}; |
| const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354}; |
| const int kTelephoneEventPayloadType = 123; |
| const int kTelephoneEventPayloadFrequency = 65432; |
| const int kTelephoneEventCode = 45; |
| const int kTelephoneEventDuration = 6789; |
| const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000}; |
| constexpr int kIsacPayloadType = 103; |
| const SdpAudioFormat kIsacFormat = {"isac", 16000, 1}; |
| const SdpAudioFormat kOpusFormat = {"opus", 48000, 2}; |
| const SdpAudioFormat kG722Format = {"g722", 8000, 1}; |
| const AudioCodecSpec kCodecSpecs[] = { |
| {kIsacFormat, {16000, 1, 32000, 10000, 32000}}, |
| {kOpusFormat, {48000, 1, 32000, 6000, 510000}}, |
| {kG722Format, {16000, 1, 64000}}}; |
| |
| class MockLimitObserver : public BitrateAllocator::LimitObserver { |
| public: |
| MOCK_METHOD2(OnAllocationLimitsChanged, |
| void(uint32_t min_send_bitrate_bps, |
| uint32_t max_padding_bitrate_bps)); |
| }; |
| |
| class MockTransmitMixer : public voe::TransmitMixer { |
| public: |
| MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t()); |
| MOCK_CONST_METHOD0(GetTotalInputEnergy, double()); |
| MOCK_CONST_METHOD0(GetTotalInputDuration, double()); |
| MOCK_CONST_METHOD0(typing_noise_detected, bool()); |
| }; |
| |
| std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock( |
| int payload_type, |
| const SdpAudioFormat& format) { |
| for (const auto& spec : kCodecSpecs) { |
| if (format == spec.format) { |
| std::unique_ptr<MockAudioEncoder> encoder(new MockAudioEncoder); |
| ON_CALL(*encoder.get(), SampleRateHz()) |
| .WillByDefault(Return(spec.info.sample_rate_hz)); |
| ON_CALL(*encoder.get(), NumChannels()) |
| .WillByDefault(Return(spec.info.num_channels)); |
| ON_CALL(*encoder.get(), RtpTimestampRateHz()) |
| .WillByDefault(Return(spec.format.clockrate_hz)); |
| return encoder; |
| } |
| } |
| return nullptr; |
| } |
| |
| rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() { |
| rtc::scoped_refptr<MockAudioEncoderFactory> factory = |
| new rtc::RefCountedObject<MockAudioEncoderFactory>(); |
| ON_CALL(*factory.get(), GetSupportedEncoders()) |
| .WillByDefault(Return(std::vector<AudioCodecSpec>( |
| std::begin(kCodecSpecs), std::end(kCodecSpecs)))); |
| ON_CALL(*factory.get(), QueryAudioEncoder(_)) |
| .WillByDefault(Invoke([](const SdpAudioFormat& format) { |
| for (const auto& spec : kCodecSpecs) { |
| if (format == spec.format) { |
| return rtc::Optional<AudioCodecInfo>(spec.info); |
| } |
| } |
| return rtc::Optional<AudioCodecInfo>(); |
| })); |
| ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _)) |
| .WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format, |
| std::unique_ptr<AudioEncoder>* return_value) { |
| *return_value = SetupAudioEncoderMock(payload_type, format); |
| })); |
| return factory; |
| } |
| |
| struct ConfigHelper { |
| ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call) |
| : stream_config_(nullptr), |
| audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()), |
| simulated_clock_(123456), |
| send_side_cc_(rtc::MakeUnique<SendSideCongestionController>( |
| &simulated_clock_, |
| nullptr /* observer */, |
| &event_log_, |
| &pacer_)), |
| fake_transport_(&packet_router_, &pacer_, send_side_cc_.get()), |
| bitrate_allocator_(&limit_observer_), |
| worker_queue_("ConfigHelper_worker_queue"), |
| audio_encoder_(nullptr) { |
| using testing::Invoke; |
| |
| EXPECT_CALL(voice_engine_, audio_device_module()); |
| EXPECT_CALL(voice_engine_, audio_transport()); |
| |
| AudioState::Config config; |
| config.voice_engine = &voice_engine_; |
| config.audio_mixer = AudioMixerImpl::Create(); |
| config.audio_processing = audio_processing_; |
| audio_state_ = AudioState::Create(config); |
| |
| SetupDefaultChannelProxy(audio_bwe_enabled); |
| |
| EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) |
| .WillOnce(Invoke([this](int channel_id) { |
| return channel_proxy_; |
| })); |
| |
| SetupMockForSetupSendCodec(expect_set_encoder_call); |
| |
| // Use ISAC as default codec so as to prevent unnecessary |voice_engine_| |
| // calls from the default ctor behavior. |
| stream_config_.send_codec_spec = |
| rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
| {kIsacPayloadType, kIsacFormat}); |
| stream_config_.voe_channel_id = kChannelId; |
| stream_config_.rtp.ssrc = kSsrc; |
| stream_config_.rtp.nack.rtp_history_ms = 200; |
| stream_config_.rtp.c_name = kCName; |
| stream_config_.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
| if (audio_bwe_enabled) { |
| AddBweToConfig(&stream_config_); |
| } |
| stream_config_.encoder_factory = SetupEncoderFactoryMock(); |
| stream_config_.min_bitrate_bps = 10000; |
| stream_config_.max_bitrate_bps = 65000; |
| } |
| |
| AudioSendStream::Config& config() { return stream_config_; } |
| MockAudioEncoderFactory& mock_encoder_factory() { |
| return *static_cast<MockAudioEncoderFactory*>( |
| stream_config_.encoder_factory.get()); |
| } |
| rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } |
| MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } |
| RtpTransportControllerSendInterface* transport() { return &fake_transport_; } |
| BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; } |
| rtc::TaskQueue* worker_queue() { return &worker_queue_; } |
| RtcEventLog* event_log() { return &event_log_; } |
| MockVoiceEngine* voice_engine() { return &voice_engine_; } |
| |
| static void AddBweToConfig(AudioSendStream::Config* config) { |
| config->rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberId)); |
| config->send_codec_spec->transport_cc_enabled = true; |
| } |
| |
| void SetupDefaultChannelProxy(bool audio_bwe_enabled) { |
| channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); |
| EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _)) |
| .WillRepeatedly(Invoke( |
| [this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) { |
| *rtp_rtcp_module = &this->rtp_rtcp_; |
| *rtp_receiver = nullptr; // Not deemed necessary for tests yet. |
| })); |
| EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1); |
| EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1); |
| EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1); |
| EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1); |
| EXPECT_CALL(*channel_proxy_, |
| SetSendAudioLevelIndicationStatus(true, kAudioLevelId)) |
| .Times(1); |
| if (audio_bwe_enabled) { |
| EXPECT_CALL(*channel_proxy_, |
| EnableSendTransportSequenceNumber(kTransportSequenceNumberId)) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects( |
| &fake_transport_, Ne(nullptr))) |
| .Times(1); |
| } else { |
| EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects( |
| &fake_transport_, Eq(nullptr))) |
| .Times(1); |
| } |
| EXPECT_CALL(*channel_proxy_, SetBitrate(_, _)) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects()) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2); |
| EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1); |
| EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) |
| .Times(1); // Destructor resets the event log |
| EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_)).Times(1); |
| EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull())) |
| .Times(1); // Destructor resets the rtt stats. |
| } |
| |
| void SetupMockForSetupSendCodec(bool expect_set_encoder_call) { |
| if (expect_set_encoder_call) { |
| EXPECT_CALL(*channel_proxy_, SetEncoderForMock(_, _)) |
| .WillOnce(Invoke( |
| [this](int payload_type, std::unique_ptr<AudioEncoder>* encoder) { |
| this->audio_encoder_ = std::move(*encoder); |
| return true; |
| })); |
| } |
| } |
| |
| void SetupMockForModifyEncoder() { |
| // Let ModifyEncoder to invoke mock audio encoder. |
| EXPECT_CALL(*channel_proxy_, ModifyEncoder(_)) |
| .WillRepeatedly(Invoke( |
| [this](rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> |
| modifier) { |
| if (this->audio_encoder_) |
| modifier(&this->audio_encoder_); |
| })); |
| } |
| |
| RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; } |
| |
| void SetupMockForSendTelephoneEvent() { |
| EXPECT_TRUE(channel_proxy_); |
| EXPECT_CALL(*channel_proxy_, |
| SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType, |
| kTelephoneEventPayloadFrequency)) |
| .WillOnce(Return(true)); |
| EXPECT_CALL(*channel_proxy_, |
| SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration)) |
| .WillOnce(Return(true)); |
| } |
| |
| void SetupMockForGetStats() { |
| using testing::DoAll; |
| using testing::SetArgPointee; |
| using testing::SetArgReferee; |
| |
| std::vector<ReportBlock> report_blocks; |
| webrtc::ReportBlock block = kReportBlock; |
| report_blocks.push_back(block); // Has wrong SSRC. |
| block.source_SSRC = kSsrc; |
| report_blocks.push_back(block); // Correct block. |
| block.fraction_lost = 0; |
| report_blocks.push_back(block); // Duplicate SSRC, bad fraction_lost. |
| |
| EXPECT_TRUE(channel_proxy_); |
| EXPECT_CALL(*channel_proxy_, GetRTCPStatistics()) |
| .WillRepeatedly(Return(kCallStats)); |
| EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks()) |
| .WillRepeatedly(Return(report_blocks)); |
| EXPECT_CALL(*channel_proxy_, GetANAStatistics()) |
| .WillRepeatedly(Return(ANAStats())); |
| EXPECT_CALL(voice_engine_, transmit_mixer()) |
| .WillRepeatedly(Return(&transmit_mixer_)); |
| |
| EXPECT_CALL(transmit_mixer_, AudioLevelFullRange()) |
| .WillRepeatedly(Return(kSpeechInputLevel)); |
| EXPECT_CALL(transmit_mixer_, GetTotalInputEnergy()) |
| .WillRepeatedly(Return(kTotalInputEnergy)); |
| EXPECT_CALL(transmit_mixer_, GetTotalInputDuration()) |
| .WillRepeatedly(Return(kTotalInputDuration)); |
| EXPECT_CALL(transmit_mixer_, typing_noise_detected()) |
| .WillRepeatedly(Return(true)); |
| |
| // We have to set the instantaneous value, the average, min and max. We only |
| // care about the instantaneous value, so we set all to the same value. |
| audio_processing_stats_.echo_return_loss.Set( |
| kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss); |
| audio_processing_stats_.echo_return_loss_enhancement.Set( |
| kEchoReturnLossEnhancement, kEchoReturnLossEnhancement, |
| kEchoReturnLossEnhancement, kEchoReturnLossEnhancement); |
| audio_processing_stats_.delay_median = kEchoDelayMedian; |
| audio_processing_stats_.delay_standard_deviation = kEchoDelayStdDev; |
| |
| EXPECT_CALL(*audio_processing_, GetStatistics()) |
| .WillRepeatedly(Return(audio_processing_stats_)); |
| } |
| |
| private: |
| testing::StrictMock<MockVoiceEngine> voice_engine_; |
| rtc::scoped_refptr<AudioState> audio_state_; |
| AudioSendStream::Config stream_config_; |
| testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
| rtc::scoped_refptr<MockAudioProcessing> audio_processing_; |
| MockTransmitMixer transmit_mixer_; |
| AudioProcessing::AudioProcessingStatistics audio_processing_stats_; |
| SimulatedClock simulated_clock_; |
| PacketRouter packet_router_; |
| testing::NiceMock<MockPacedSender> pacer_; |
| std::unique_ptr<SendSideCongestionController> send_side_cc_; |
| FakeRtpTransportControllerSend fake_transport_; |
| MockRtcEventLog event_log_; |
| MockRtpRtcp rtp_rtcp_; |
| MockRtcpRttStats rtcp_rtt_stats_; |
| testing::NiceMock<MockLimitObserver> limit_observer_; |
| BitrateAllocator bitrate_allocator_; |
| // |worker_queue| is defined last to ensure all pending tasks are cancelled |
| // and deleted before any other members. |
| rtc::TaskQueue worker_queue_; |
| std::unique_ptr<AudioEncoder> audio_encoder_; |
| }; |
| } // namespace |
| |
| TEST(AudioSendStreamTest, ConfigToString) { |
| AudioSendStream::Config config(nullptr); |
| config.rtp.ssrc = kSsrc; |
| config.rtp.c_name = kCName; |
| config.voe_channel_id = kChannelId; |
| config.min_bitrate_bps = 12000; |
| config.max_bitrate_bps = 34000; |
| config.send_codec_spec = |
| rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
| {kIsacPayloadType, kIsacFormat}); |
| config.send_codec_spec->nack_enabled = true; |
| config.send_codec_spec->transport_cc_enabled = false; |
| config.send_codec_spec->cng_payload_type = rtc::Optional<int>(42); |
| config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory(); |
| config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
| EXPECT_EQ( |
| "{rtp: {ssrc: 1234, extensions: [{uri: " |
| "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " |
| "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, " |
| "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " |
| "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " |
| "cng_payload_type: 42, payload_type: 103, " |
| "format: {name: isac, clockrate_hz: 16000, num_channels: 1, " |
| "parameters: {}}}}", |
| config.ToString()); |
| } |
| |
| TEST(AudioSendStreamTest, ConstructDestruct) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| } |
| |
| TEST(AudioSendStreamTest, SendTelephoneEvent) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| helper.SetupMockForSendTelephoneEvent(); |
| EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType, |
| kTelephoneEventPayloadFrequency, kTelephoneEventCode, |
| kTelephoneEventDuration)); |
| } |
| |
| TEST(AudioSendStreamTest, SetMuted) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true)); |
| send_stream.SetMuted(true); |
| } |
| |
| TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { |
| ConfigHelper helper(true, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| } |
| |
| TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| } |
| |
| TEST(AudioSendStreamTest, GetStats) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| helper.SetupMockForGetStats(); |
| AudioSendStream::Stats stats = send_stream.GetStats(); |
| EXPECT_EQ(kSsrc, stats.local_ssrc); |
| EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent); |
| EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent); |
| EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost), |
| stats.packets_lost); |
| EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost); |
| EXPECT_EQ(std::string(kIsacCodec.plname), stats.codec_name); |
| EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number), |
| stats.ext_seqnum); |
| EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter / |
| (kIsacCodec.plfreq / 1000)), |
| stats.jitter_ms); |
| EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms); |
| EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level); |
| EXPECT_EQ(kTotalInputEnergy, stats.total_input_energy); |
| EXPECT_EQ(kTotalInputDuration, stats.total_input_duration); |
| EXPECT_EQ(-1, stats.aec_quality_min); |
| EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms); |
| EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms); |
| EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss); |
| EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement); |
| EXPECT_EQ(kResidualEchoLikelihood, stats.residual_echo_likelihood); |
| EXPECT_TRUE(stats.typing_noise_detected); |
| } |
| |
| TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) { |
| ConfigHelper helper(false, true); |
| auto stream_config = helper.config(); |
| stream_config.send_codec_spec = |
| rtc::Optional<AudioSendStream::Config::SendCodecSpec>({0, kOpusFormat}); |
| const std::string kAnaConfigString = "abcde"; |
| const std::string kAnaReconfigString = "12345"; |
| |
| stream_config.audio_network_adaptor_config = |
| rtc::Optional<std::string>(kAnaConfigString); |
| |
| EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _)) |
| .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString]( |
| int payload_type, const SdpAudioFormat& format, |
| std::unique_ptr<AudioEncoder>* return_value) { |
| auto mock_encoder = SetupAudioEncoderMock(payload_type, format); |
| EXPECT_CALL(*mock_encoder, |
| EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _)) |
| .WillOnce(Return(true)); |
| EXPECT_CALL(*mock_encoder, |
| EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString), _)) |
| .WillOnce(Return(true)); |
| *return_value = std::move(mock_encoder); |
| })); |
| |
| internal::AudioSendStream send_stream( |
| stream_config, helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| |
| stream_config.audio_network_adaptor_config = |
| rtc::Optional<std::string>(kAnaReconfigString); |
| |
| helper.SetupMockForModifyEncoder(); |
| send_stream.Reconfigure(stream_config); |
| } |
| |
| // VAD is applied when codec is mono and the CNG frequency matches the codec |
| // clock rate. |
| TEST(AudioSendStreamTest, SendCodecCanApplyVad) { |
| ConfigHelper helper(false, false); |
| auto stream_config = helper.config(); |
| stream_config.send_codec_spec = |
| rtc::Optional<AudioSendStream::Config::SendCodecSpec>({9, kG722Format}); |
| stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105); |
| using ::testing::Invoke; |
| std::unique_ptr<AudioEncoder> stolen_encoder; |
| EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _)) |
| .WillOnce( |
| Invoke([&stolen_encoder](int payload_type, |
| std::unique_ptr<AudioEncoder>* encoder) { |
| stolen_encoder = std::move(*encoder); |
| return true; |
| })); |
| |
| internal::AudioSendStream send_stream( |
| stream_config, helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| |
| // We cannot truly determine if the encoder created is an AudioEncoderCng. It |
| // is the only reasonable implementation that will return something from |
| // ReclaimContainedEncoders, though. |
| ASSERT_TRUE(stolen_encoder); |
| EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty()); |
| } |
| |
| TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| EXPECT_CALL(*helper.channel_proxy(), |
| SetBitrate(helper.config().max_bitrate_bps, _)); |
| send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50, |
| 6000); |
| } |
| |
| TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); |
| send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); |
| } |
| |
| // Test that AudioSendStream doesn't recreate the encoder unnecessarily. |
| TEST(AudioSendStreamTest, DontRecreateEncoder) { |
| ConfigHelper helper(false, false); |
| // WillOnce is (currently) the default used by ConfigHelper if asked to set an |
| // expectation for SetEncoder. Since this behavior is essential for this test |
| // to be correct, it's instead set-up manually here. Otherwise a simple change |
| // to ConfigHelper (say to WillRepeatedly) would silently make this test |
| // useless. |
| EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _)) |
| .WillOnce(Return(true)); |
| |
| auto stream_config = helper.config(); |
| stream_config.send_codec_spec = |
| rtc::Optional<AudioSendStream::Config::SendCodecSpec>({9, kG722Format}); |
| stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105); |
| internal::AudioSendStream send_stream( |
| stream_config, helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| send_stream.Reconfigure(stream_config); |
| } |
| |
| TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) { |
| ConfigHelper helper(false, true); |
| internal::AudioSendStream send_stream( |
| helper.config(), helper.audio_state(), helper.worker_queue(), |
| helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
| helper.rtcp_rtt_stats(), rtc::Optional<RtpState>()); |
| auto new_config = helper.config(); |
| ConfigHelper::AddBweToConfig(&new_config); |
| EXPECT_CALL(*helper.channel_proxy(), |
| EnableSendTransportSequenceNumber(kTransportSequenceNumberId)) |
| .Times(1); |
| { |
| ::testing::InSequence seq; |
| EXPECT_CALL(*helper.channel_proxy(), ResetSenderCongestionControlObjects()) |
| .Times(1); |
| EXPECT_CALL(*helper.channel_proxy(), RegisterSenderCongestionControlObjects( |
| helper.transport(), Ne(nullptr))) |
| .Times(1); |
| } |
| send_stream.Reconfigure(new_config); |
| } |
| |
| |
| } // namespace test |
| } // namespace webrtc |
