| /* |
| * 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 "audio/audio_send_stream.h" |
| |
| #include <memory> |
| #include <string> |
| #include <thread> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "api/test/mock_frame_encryptor.h" |
| #include "audio/audio_state.h" |
| #include "audio/conversion.h" |
| #include "audio/mock_voe_channel_proxy.h" |
| #include "call/test/mock_rtp_transport_controller_send.h" |
| #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" |
| #include "modules/audio_device/include/mock_audio_device.h" |
| #include "modules/audio_mixer/audio_mixer_impl.h" |
| #include "modules/audio_mixer/sine_wave_generator.h" |
| #include "modules/audio_processing/include/audio_processing_statistics.h" |
| #include "modules/audio_processing/include/mock_audio_processing.h" |
| #include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h" |
| #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "system_wrappers/include/clock.h" |
| #include "test/field_trial.h" |
| #include "test/gtest.h" |
| #include "test/mock_audio_encoder.h" |
| #include "test/mock_audio_encoder_factory.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| using ::testing::_; |
| using ::testing::AnyNumber; |
| using ::testing::Eq; |
| using ::testing::Field; |
| using ::testing::Invoke; |
| using ::testing::Ne; |
| using ::testing::Return; |
| using ::testing::StrEq; |
| |
| static const float kTolerance = 0.0001f; |
| |
| const uint32_t kSsrc = 1234; |
| const char* kCName = "foo_name"; |
| const int kAudioLevelId = 2; |
| const int kTransportSequenceNumberId = 4; |
| const int32_t kEchoDelayMedian = 254; |
| const int32_t kEchoDelayStdDev = -3; |
| const double kDivergentFilterFraction = 0.2f; |
| const double kEchoReturnLoss = -65; |
| const double kEchoReturnLossEnhancement = 101; |
| const double kResidualEchoLikelihood = -1.0f; |
| const double kResidualEchoLikelihoodMax = 23.0f; |
| const CallSendStatistics kCallStats = {112, 12, 13456, 17890}; |
| 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; |
| 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}}}; |
| |
| // TODO(dklee): This mirrors calculation in audio_send_stream.cc, which |
| // should be made more precise in the future. This can be changed when that |
| // logic is more accurate. |
| const DataSize kOverheadPerPacket = DataSize::Bytes(20 + 8 + 10 + 12); |
| const TimeDelta kMinFrameLength = TimeDelta::Millis(20); |
| const TimeDelta kMaxFrameLength = TimeDelta::Millis(120); |
| const DataRate kMinOverheadRate = kOverheadPerPacket / kMaxFrameLength; |
| const DataRate kMaxOverheadRate = kOverheadPerPacket / kMinFrameLength; |
| |
| class MockLimitObserver : public BitrateAllocator::LimitObserver { |
| public: |
| MOCK_METHOD(void, |
| OnAllocationLimitsChanged, |
| (BitrateAllocationLimits), |
| (override)); |
| }; |
| |
| 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 ::testing::NiceMock<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)); |
| ON_CALL(*encoder.get(), GetFrameLengthRange()) |
| .WillByDefault(Return(absl::optional<std::pair<TimeDelta, TimeDelta>>{ |
| {TimeDelta::Millis(20), TimeDelta::Millis(120)}})); |
| 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) -> absl::optional<AudioCodecInfo> { |
| for (const auto& spec : kCodecSpecs) { |
| if (format == spec.format) { |
| return spec.info; |
| } |
| } |
| return absl::nullopt; |
| })); |
| ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _)) |
| .WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format, |
| absl::optional<AudioCodecPairId> codec_pair_id, |
| 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, |
| bool use_null_audio_processing) |
| : clock_(1000000), |
| task_queue_factory_(CreateDefaultTaskQueueFactory()), |
| stream_config_(/*send_transport=*/nullptr), |
| audio_processing_( |
| use_null_audio_processing |
| ? nullptr |
| : new rtc::RefCountedObject<MockAudioProcessing>()), |
| bitrate_allocator_(&limit_observer_), |
| worker_queue_(task_queue_factory_->CreateTaskQueue( |
| "ConfigHelper_worker_queue", |
| TaskQueueFactory::Priority::NORMAL)), |
| audio_encoder_(nullptr) { |
| using ::testing::Invoke; |
| |
| AudioState::Config config; |
| config.audio_mixer = AudioMixerImpl::Create(); |
| config.audio_processing = audio_processing_; |
| config.audio_device_module = |
| new rtc::RefCountedObject<MockAudioDeviceModule>(); |
| audio_state_ = AudioState::Create(config); |
| |
| SetupDefaultChannelSend(audio_bwe_enabled); |
| SetupMockForSetupSendCodec(expect_set_encoder_call); |
| SetupMockForCallEncoder(); |
| |
| // Use ISAC as default codec so as to prevent unnecessary |channel_proxy_| |
| // calls from the default ctor behavior. |
| stream_config_.send_codec_spec = |
| AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat); |
| stream_config_.rtp.ssrc = kSsrc; |
| 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; |
| } |
| |
| std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() { |
| EXPECT_CALL(rtp_transport_, GetWorkerQueue()) |
| .WillRepeatedly(Return(&worker_queue_)); |
| return std::unique_ptr<internal::AudioSendStream>( |
| new internal::AudioSendStream( |
| Clock::GetRealTimeClock(), stream_config_, audio_state_, |
| task_queue_factory_.get(), &rtp_transport_, &bitrate_allocator_, |
| &event_log_, absl::nullopt, |
| std::unique_ptr<voe::ChannelSendInterface>(channel_send_))); |
| } |
| |
| AudioSendStream::Config& config() { return stream_config_; } |
| MockAudioEncoderFactory& mock_encoder_factory() { |
| return *static_cast<MockAudioEncoderFactory*>( |
| stream_config_.encoder_factory.get()); |
| } |
| MockRtpRtcpInterface* rtp_rtcp() { return &rtp_rtcp_; } |
| MockChannelSend* channel_send() { return channel_send_; } |
| RtpTransportControllerSendInterface* transport() { return &rtp_transport_; } |
| |
| static void AddBweToConfig(AudioSendStream::Config* config) { |
| config->rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); |
| config->send_codec_spec->transport_cc_enabled = true; |
| } |
| |
| void SetupDefaultChannelSend(bool audio_bwe_enabled) { |
| EXPECT_TRUE(channel_send_ == nullptr); |
| channel_send_ = new ::testing::StrictMock<MockChannelSend>(); |
| EXPECT_CALL(*channel_send_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() { |
| return &this->rtp_rtcp_; |
| })); |
| EXPECT_CALL(rtp_rtcp_, SSRC).WillRepeatedly(Return(kSsrc)); |
| EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1); |
| EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1); |
| EXPECT_CALL(*channel_send_, SetEncoderToPacketizerFrameTransformer(_)) |
| .Times(1); |
| EXPECT_CALL(rtp_rtcp_, SetExtmapAllowMixed(false)).Times(1); |
| EXPECT_CALL(*channel_send_, |
| SetSendAudioLevelIndicationStatus(true, kAudioLevelId)) |
| .Times(1); |
| EXPECT_CALL(rtp_transport_, GetBandwidthObserver()) |
| .WillRepeatedly(Return(&bandwidth_observer_)); |
| if (audio_bwe_enabled) { |
| EXPECT_CALL(rtp_rtcp_, |
| RegisterRtpHeaderExtension(TransportSequenceNumber::kUri, |
| kTransportSequenceNumberId)) |
| .Times(1); |
| EXPECT_CALL(*channel_send_, |
| RegisterSenderCongestionControlObjects( |
| &rtp_transport_, Eq(&bandwidth_observer_))) |
| .Times(1); |
| } else { |
| EXPECT_CALL(*channel_send_, RegisterSenderCongestionControlObjects( |
| &rtp_transport_, Eq(nullptr))) |
| .Times(1); |
| } |
| EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1); |
| EXPECT_CALL(rtp_rtcp_, SetRid(std::string())).Times(1); |
| } |
| |
| void SetupMockForSetupSendCodec(bool expect_set_encoder_call) { |
| if (expect_set_encoder_call) { |
| EXPECT_CALL(*channel_send_, SetEncoder) |
| .WillOnce( |
| [this](int payload_type, std::unique_ptr<AudioEncoder> encoder) { |
| this->audio_encoder_ = std::move(encoder); |
| return true; |
| }); |
| } |
| } |
| |
| void SetupMockForCallEncoder() { |
| // Let ModifyEncoder to invoke mock audio encoder. |
| EXPECT_CALL(*channel_send_, CallEncoder(_)) |
| .WillRepeatedly( |
| [this](rtc::FunctionView<void(AudioEncoder*)> modifier) { |
| if (this->audio_encoder_) |
| modifier(this->audio_encoder_.get()); |
| }); |
| } |
| |
| void SetupMockForSendTelephoneEvent() { |
| EXPECT_TRUE(channel_send_); |
| EXPECT_CALL(*channel_send_, SetSendTelephoneEventPayloadType( |
| kTelephoneEventPayloadType, |
| kTelephoneEventPayloadFrequency)); |
| EXPECT_CALL( |
| *channel_send_, |
| SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration)) |
| .WillOnce(Return(true)); |
| } |
| |
| void SetupMockForGetStats(bool use_null_audio_processing) { |
| 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_send_); |
| EXPECT_CALL(*channel_send_, GetRTCPStatistics()) |
| .WillRepeatedly(Return(kCallStats)); |
| EXPECT_CALL(*channel_send_, GetRemoteRTCPReportBlocks()) |
| .WillRepeatedly(Return(report_blocks)); |
| EXPECT_CALL(*channel_send_, GetANAStatistics()) |
| .WillRepeatedly(Return(ANAStats())); |
| EXPECT_CALL(*channel_send_, GetBitrate()).WillRepeatedly(Return(0)); |
| |
| audio_processing_stats_.echo_return_loss = kEchoReturnLoss; |
| audio_processing_stats_.echo_return_loss_enhancement = |
| kEchoReturnLossEnhancement; |
| audio_processing_stats_.delay_median_ms = kEchoDelayMedian; |
| audio_processing_stats_.delay_standard_deviation_ms = kEchoDelayStdDev; |
| audio_processing_stats_.divergent_filter_fraction = |
| kDivergentFilterFraction; |
| audio_processing_stats_.residual_echo_likelihood = kResidualEchoLikelihood; |
| audio_processing_stats_.residual_echo_likelihood_recent_max = |
| kResidualEchoLikelihoodMax; |
| if (!use_null_audio_processing) { |
| ASSERT_TRUE(audio_processing_); |
| EXPECT_CALL(*audio_processing_, GetStatistics(true)) |
| .WillRepeatedly(Return(audio_processing_stats_)); |
| } |
| } |
| |
| TaskQueueForTest* worker() { return &worker_queue_; } |
| |
| private: |
| SimulatedClock clock_; |
| std::unique_ptr<TaskQueueFactory> task_queue_factory_; |
| rtc::scoped_refptr<AudioState> audio_state_; |
| AudioSendStream::Config stream_config_; |
| ::testing::StrictMock<MockChannelSend>* channel_send_ = nullptr; |
| rtc::scoped_refptr<MockAudioProcessing> audio_processing_; |
| AudioProcessingStats audio_processing_stats_; |
| ::testing::StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_; |
| ::testing::NiceMock<MockRtcEventLog> event_log_; |
| ::testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_; |
| ::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_; |
| ::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. |
| TaskQueueForTest worker_queue_; |
| std::unique_ptr<AudioEncoder> audio_encoder_; |
| }; |
| |
| // The audio level ranges linearly [0,32767]. |
| std::unique_ptr<AudioFrame> CreateAudioFrame1kHzSineWave(int16_t audio_level, |
| int duration_ms, |
| int sample_rate_hz, |
| size_t num_channels) { |
| size_t samples_per_channel = sample_rate_hz / (1000 / duration_ms); |
| std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0); |
| std::unique_ptr<AudioFrame> audio_frame = std::make_unique<AudioFrame>(); |
| audio_frame->UpdateFrame(0 /* RTP timestamp */, &audio_data[0], |
| samples_per_channel, sample_rate_hz, |
| AudioFrame::SpeechType::kNormalSpeech, |
| AudioFrame::VADActivity::kVadUnknown, num_channels); |
| SineWaveGenerator wave_generator(1000.0, audio_level); |
| wave_generator.GenerateNextFrame(audio_frame.get()); |
| return audio_frame; |
| } |
| |
| } // namespace |
| |
| TEST(AudioSendStreamTest, ConfigToString) { |
| AudioSendStream::Config config(/*send_transport=*/nullptr); |
| config.rtp.ssrc = kSsrc; |
| config.rtp.c_name = kCName; |
| config.min_bitrate_bps = 12000; |
| config.max_bitrate_bps = 34000; |
| config.send_codec_spec = |
| 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 = 42; |
| config.send_codec_spec->red_payload_type = 43; |
| config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory(); |
| config.rtp.extmap_allow_mixed = true; |
| config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
| config.rtcp_report_interval_ms = 2500; |
| EXPECT_EQ( |
| "{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: " |
| "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], " |
| "c_name: foo_name}, rtcp_report_interval_ms: 2500, " |
| "send_transport: null, " |
| "min_bitrate_bps: 12000, max_bitrate_bps: 34000, " |
| "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " |
| "cng_payload_type: 42, red_payload_type: 43, payload_type: 103, " |
| "format: {name: isac, clockrate_hz: 16000, num_channels: 1, " |
| "parameters: {}}}}", |
| config.ToString()); |
| } |
| |
| TEST(AudioSendStreamTest, ConstructDestruct) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SendTelephoneEvent) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| helper.SetupMockForSendTelephoneEvent(); |
| EXPECT_TRUE(send_stream->SendTelephoneEvent( |
| kTelephoneEventPayloadType, kTelephoneEventPayloadFrequency, |
| kTelephoneEventCode, kTelephoneEventDuration)); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SetMuted) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| EXPECT_CALL(*helper.channel_send(), SetInputMute(true)); |
| send_stream->SetMuted(true); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { |
| ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(true, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, GetStats) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| helper.SetupMockForGetStats(use_null_audio_processing); |
| AudioSendStream::Stats stats = send_stream->GetStats(true); |
| EXPECT_EQ(kSsrc, stats.local_ssrc); |
| EXPECT_EQ(kCallStats.payload_bytes_sent, stats.payload_bytes_sent); |
| EXPECT_EQ(kCallStats.header_and_padding_bytes_sent, |
| stats.header_and_padding_bytes_sent); |
| EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent); |
| EXPECT_EQ(kReportBlock.cumulative_num_packets_lost, stats.packets_lost); |
| EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost); |
| EXPECT_EQ(kIsacFormat.name, stats.codec_name); |
| EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter / |
| (kIsacFormat.clockrate_hz / 1000)), |
| stats.jitter_ms); |
| EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms); |
| EXPECT_EQ(0, stats.audio_level); |
| EXPECT_EQ(0, stats.total_input_energy); |
| EXPECT_EQ(0, stats.total_input_duration); |
| |
| if (!use_null_audio_processing) { |
| EXPECT_EQ(kEchoDelayMedian, stats.apm_statistics.delay_median_ms); |
| EXPECT_EQ(kEchoDelayStdDev, |
| stats.apm_statistics.delay_standard_deviation_ms); |
| EXPECT_EQ(kEchoReturnLoss, stats.apm_statistics.echo_return_loss); |
| EXPECT_EQ(kEchoReturnLossEnhancement, |
| stats.apm_statistics.echo_return_loss_enhancement); |
| EXPECT_EQ(kDivergentFilterFraction, |
| stats.apm_statistics.divergent_filter_fraction); |
| EXPECT_EQ(kResidualEchoLikelihood, |
| stats.apm_statistics.residual_echo_likelihood); |
| EXPECT_EQ(kResidualEchoLikelihoodMax, |
| stats.apm_statistics.residual_echo_likelihood_recent_max); |
| EXPECT_FALSE(stats.typing_noise_detected); |
| } |
| } |
| } |
| |
| TEST(AudioSendStreamTest, GetStatsAudioLevel) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| helper.SetupMockForGetStats(use_null_audio_processing); |
| EXPECT_CALL(*helper.channel_send(), ProcessAndEncodeAudio) |
| .Times(AnyNumber()); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 1; |
| |
| constexpr int16_t kSilentAudioLevel = 0; |
| constexpr int16_t kMaxAudioLevel = 32767; // Audio level is [0,32767]. |
| constexpr int kAudioFrameDurationMs = 10; |
| |
| // Process 10 audio frames (100 ms) of silence. After this, on the next |
| // (11-th) frame, the audio level will be updated with the maximum audio |
| // level of the first 11 frames. See AudioLevel. |
| for (size_t i = 0; i < 10; ++i) { |
| send_stream->SendAudioData( |
| CreateAudioFrame1kHzSineWave(kSilentAudioLevel, kAudioFrameDurationMs, |
| kSampleRateHz, kNumChannels)); |
| } |
| AudioSendStream::Stats stats = send_stream->GetStats(); |
| EXPECT_EQ(kSilentAudioLevel, stats.audio_level); |
| EXPECT_NEAR(0.0f, stats.total_input_energy, kTolerance); |
| EXPECT_NEAR(0.1f, stats.total_input_duration, |
| kTolerance); // 100 ms = 0.1 s |
| |
| // Process 10 audio frames (100 ms) of maximum audio level. |
| // Note that AudioLevel updates the audio level every 11th frame, processing |
| // 10 frames above was needed to see a non-zero audio level here. |
| for (size_t i = 0; i < 10; ++i) { |
| send_stream->SendAudioData(CreateAudioFrame1kHzSineWave( |
| kMaxAudioLevel, kAudioFrameDurationMs, kSampleRateHz, kNumChannels)); |
| } |
| stats = send_stream->GetStats(); |
| EXPECT_EQ(kMaxAudioLevel, stats.audio_level); |
| // Energy increases by energy*duration, where energy is audio level in |
| // [0,1]. |
| EXPECT_NEAR(0.1f, stats.total_input_energy, kTolerance); // 0.1 s of max |
| EXPECT_NEAR(0.2f, stats.total_input_duration, |
| kTolerance); // 200 ms = 0.2 s |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| helper.config().send_codec_spec = |
| AudioSendStream::Config::SendCodecSpec(0, kOpusFormat); |
| const std::string kAnaConfigString = "abcde"; |
| const std::string kAnaReconfigString = "12345"; |
| |
| helper.config().rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); |
| helper.config().audio_network_adaptor_config = kAnaConfigString; |
| |
| EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _)) |
| .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString]( |
| int payload_type, const SdpAudioFormat& format, |
| absl::optional<AudioCodecPairId> codec_pair_id, |
| 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); |
| })); |
| |
| auto send_stream = helper.CreateAudioSendStream(); |
| |
| auto stream_config = helper.config(); |
| stream_config.audio_network_adaptor_config = kAnaReconfigString; |
| |
| send_stream->Reconfigure(stream_config); |
| } |
| } |
| |
| // VAD is applied when codec is mono and the CNG frequency matches the codec |
| // clock rate. |
| TEST(AudioSendStreamTest, SendCodecCanApplyVad) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, false, use_null_audio_processing); |
| helper.config().send_codec_spec = |
| AudioSendStream::Config::SendCodecSpec(9, kG722Format); |
| helper.config().send_codec_spec->cng_payload_type = 105; |
| std::unique_ptr<AudioEncoder> stolen_encoder; |
| EXPECT_CALL(*helper.channel_send(), SetEncoder) |
| .WillOnce([&stolen_encoder](int payload_type, |
| std::unique_ptr<AudioEncoder> encoder) { |
| stolen_encoder = std::move(encoder); |
| return true; |
| }); |
| EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000)); |
| |
| auto send_stream = helper.CreateAudioSendStream(); |
| |
| // 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) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| EXPECT_CALL( |
| *helper.channel_send(), |
| OnBitrateAllocation( |
| Field(&BitrateAllocationUpdate::target_bitrate, |
| Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps))))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = |
| DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000); |
| update.packet_loss_ratio = 0; |
| update.round_trip_time = TimeDelta::Millis(50); |
| update.bwe_period = TimeDelta::Millis(6000); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) { |
| ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(true, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| EXPECT_CALL( |
| *helper.channel_send(), |
| OnBitrateAllocation(Field( |
| &BitrateAllocationUpdate::target_bitrate, |
| Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000))))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = |
| DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) { |
| ScopedFieldTrials field_trials( |
| "WebRTC-Audio-SendSideBwe/Enabled/" |
| "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(true, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| EXPECT_CALL( |
| *helper.channel_send(), |
| OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate, |
| Eq(DataRate::KilobitsPerSec(6))))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = DataRate::KilobitsPerSec(1); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) { |
| ScopedFieldTrials field_trials( |
| "WebRTC-Audio-SendSideBwe/Enabled/" |
| "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(true, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| EXPECT_CALL( |
| *helper.channel_send(), |
| OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate, |
| Eq(DataRate::KilobitsPerSec(64))))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = DataRate::KilobitsPerSec(128); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SSBweWithOverhead) { |
| ScopedFieldTrials field_trials( |
| "WebRTC-Audio-SendSideBwe/Enabled/" |
| "WebRTC-SendSideBwe-WithOverhead/Enabled/" |
| "WebRTC-Audio-LegacyOverhead/Disabled/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(true, true, use_null_audio_processing); |
| EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) |
| .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>())); |
| auto send_stream = helper.CreateAudioSendStream(); |
| const DataRate bitrate = |
| DataRate::BitsPerSec(helper.config().max_bitrate_bps) + |
| kMaxOverheadRate; |
| EXPECT_CALL(*helper.channel_send(), |
| OnBitrateAllocation(Field( |
| &BitrateAllocationUpdate::target_bitrate, Eq(bitrate)))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = bitrate; |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) { |
| ScopedFieldTrials field_trials( |
| "WebRTC-Audio-SendSideBwe/Enabled/" |
| "WebRTC-SendSideBwe-WithOverhead/Enabled/" |
| "WebRTC-Audio-LegacyOverhead/Disabled/" |
| "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(true, true, use_null_audio_processing); |
| EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) |
| .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>())); |
| auto send_stream = helper.CreateAudioSendStream(); |
| const DataRate bitrate = DataRate::KilobitsPerSec(6) + kMinOverheadRate; |
| EXPECT_CALL(*helper.channel_send(), |
| OnBitrateAllocation(Field( |
| &BitrateAllocationUpdate::target_bitrate, Eq(bitrate)))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = DataRate::KilobitsPerSec(1); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) { |
| ScopedFieldTrials field_trials( |
| "WebRTC-Audio-SendSideBwe/Enabled/" |
| "WebRTC-SendSideBwe-WithOverhead/Enabled/" |
| "WebRTC-Audio-LegacyOverhead/Disabled/" |
| "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(true, true, use_null_audio_processing); |
| EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) |
| .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>())); |
| auto send_stream = helper.CreateAudioSendStream(); |
| const DataRate bitrate = DataRate::KilobitsPerSec(64) + kMaxOverheadRate; |
| EXPECT_CALL(*helper.channel_send(), |
| OnBitrateAllocation(Field( |
| &BitrateAllocationUpdate::target_bitrate, Eq(bitrate)))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = DataRate::KilobitsPerSec(128); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| |
| EXPECT_CALL(*helper.channel_send(), |
| OnBitrateAllocation(Field(&BitrateAllocationUpdate::bwe_period, |
| Eq(TimeDelta::Millis(5000))))); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = |
| DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000); |
| update.packet_loss_ratio = 0; |
| update.round_trip_time = TimeDelta::Millis(50); |
| update.bwe_period = TimeDelta::Millis(5000); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| } |
| } |
| |
| // Test that AudioSendStream doesn't recreate the encoder unnecessarily. |
| TEST(AudioSendStreamTest, DontRecreateEncoder) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, false, use_null_audio_processing); |
| // 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_send(), SetEncoder).WillOnce(Return()); |
| |
| EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000)); |
| |
| helper.config().send_codec_spec = |
| AudioSendStream::Config::SendCodecSpec(9, kG722Format); |
| helper.config().send_codec_spec->cng_payload_type = 105; |
| auto send_stream = helper.CreateAudioSendStream(); |
| send_stream->Reconfigure(helper.config()); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) { |
| ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| auto new_config = helper.config(); |
| ConfigHelper::AddBweToConfig(&new_config); |
| |
| EXPECT_CALL(*helper.rtp_rtcp(), |
| RegisterRtpHeaderExtension(TransportSequenceNumber::kUri, |
| kTransportSequenceNumberId)) |
| .Times(1); |
| { |
| ::testing::InSequence seq; |
| EXPECT_CALL(*helper.channel_send(), ResetSenderCongestionControlObjects()) |
| .Times(1); |
| EXPECT_CALL(*helper.channel_send(), |
| RegisterSenderCongestionControlObjects(helper.transport(), |
| Ne(nullptr))) |
| .Times(1); |
| } |
| |
| send_stream->Reconfigure(new_config); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, OnTransportOverheadChanged) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| auto new_config = helper.config(); |
| |
| // CallEncoder will be called on overhead change. |
| EXPECT_CALL(*helper.channel_send(), CallEncoder); |
| |
| const size_t transport_overhead_per_packet_bytes = 333; |
| send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes); |
| |
| EXPECT_EQ(transport_overhead_per_packet_bytes, |
| send_stream->TestOnlyGetPerPacketOverheadBytes()); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, DoesntCallEncoderWhenOverheadUnchanged) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| auto new_config = helper.config(); |
| |
| // CallEncoder will be called on overhead change. |
| EXPECT_CALL(*helper.channel_send(), CallEncoder); |
| const size_t transport_overhead_per_packet_bytes = 333; |
| send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes); |
| |
| // Set the same overhead again, CallEncoder should not be called again. |
| EXPECT_CALL(*helper.channel_send(), CallEncoder).Times(0); |
| send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes); |
| |
| // New overhead, call CallEncoder again |
| EXPECT_CALL(*helper.channel_send(), CallEncoder); |
| send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes + 1); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, AudioOverheadChanged) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| const size_t audio_overhead_per_packet_bytes = 555; |
| EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) |
| .WillRepeatedly(Return(audio_overhead_per_packet_bytes)); |
| auto send_stream = helper.CreateAudioSendStream(); |
| auto new_config = helper.config(); |
| |
| BitrateAllocationUpdate update; |
| update.target_bitrate = |
| DataRate::BitsPerSec(helper.config().max_bitrate_bps) + |
| kMaxOverheadRate; |
| EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| |
| EXPECT_EQ(audio_overhead_per_packet_bytes, |
| send_stream->TestOnlyGetPerPacketOverheadBytes()); |
| |
| EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) |
| .WillRepeatedly(Return(audio_overhead_per_packet_bytes + 20)); |
| EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| |
| EXPECT_EQ(audio_overhead_per_packet_bytes + 20, |
| send_stream->TestOnlyGetPerPacketOverheadBytes()); |
| } |
| } |
| |
| TEST(AudioSendStreamTest, OnAudioAndTransportOverheadChanged) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| const size_t audio_overhead_per_packet_bytes = 555; |
| EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) |
| .WillRepeatedly(Return(audio_overhead_per_packet_bytes)); |
| auto send_stream = helper.CreateAudioSendStream(); |
| auto new_config = helper.config(); |
| |
| const size_t transport_overhead_per_packet_bytes = 333; |
| send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes); |
| |
| BitrateAllocationUpdate update; |
| update.target_bitrate = |
| DataRate::BitsPerSec(helper.config().max_bitrate_bps) + |
| kMaxOverheadRate; |
| EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation); |
| helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); }, |
| RTC_FROM_HERE); |
| |
| EXPECT_EQ( |
| transport_overhead_per_packet_bytes + audio_overhead_per_packet_bytes, |
| send_stream->TestOnlyGetPerPacketOverheadBytes()); |
| } |
| } |
| |
| // Validates that reconfiguring the AudioSendStream with a Frame encryptor |
| // correctly reconfigures on the object without crashing. |
| TEST(AudioSendStreamTest, ReconfigureWithFrameEncryptor) { |
| for (bool use_null_audio_processing : {false, true}) { |
| ConfigHelper helper(false, true, use_null_audio_processing); |
| auto send_stream = helper.CreateAudioSendStream(); |
| auto new_config = helper.config(); |
| |
| rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_0( |
| new rtc::RefCountedObject<MockFrameEncryptor>()); |
| new_config.frame_encryptor = mock_frame_encryptor_0; |
| EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr))) |
| .Times(1); |
| send_stream->Reconfigure(new_config); |
| |
| // Not updating the frame encryptor shouldn't force it to reconfigure. |
| EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(_)).Times(0); |
| send_stream->Reconfigure(new_config); |
| |
| // Updating frame encryptor to a new object should force a call to the |
| // proxy. |
| rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_1( |
| new rtc::RefCountedObject<MockFrameEncryptor>()); |
| new_config.frame_encryptor = mock_frame_encryptor_1; |
| new_config.crypto_options.sframe.require_frame_encryption = true; |
| EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr))) |
| .Times(1); |
| send_stream->Reconfigure(new_config); |
| } |
| } |
| } // namespace test |
| } // namespace webrtc |
