diff options
Diffstat (limited to 'third_party/libwebrtc/audio/audio_send_stream_unittest.cc')
| -rw-r--r-- | third_party/libwebrtc/audio/audio_send_stream_unittest.cc | 949 |
1 files changed, 949 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/audio_send_stream_unittest.cc b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc new file mode 100644 index 0000000000..a81b40cbe7 --- /dev/null +++ b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc @@ -0,0 +1,949 @@ +/* + * 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 "modules/utility/maybe_worker_thread.h" +#include "system_wrappers/include/clock.h" +#include "test/gtest.h" +#include "test/mock_audio_encoder.h" +#include "test/mock_audio_encoder_factory.h" +#include "test/scoped_key_value_config.h" +#include "test/time_controller/real_time_controller.h" + +namespace webrtc { +namespace test { +namespace { + +using ::testing::_; +using ::testing::AnyNumber; +using ::testing::Eq; +using ::testing::Field; +using ::testing::InSequence; +using ::testing::Invoke; +using ::testing::Ne; +using ::testing::NiceMock; +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 = + rtc::make_ref_counted<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) + : stream_config_(/*send_transport=*/nullptr), + audio_processing_( + use_null_audio_processing + ? nullptr + : rtc::make_ref_counted<NiceMock<MockAudioProcessing>>()), + bitrate_allocator_(&limit_observer_), + worker_queue_(field_trials, + "ConfigHelper_worker_queue", + time_controller_.GetTaskQueueFactory()), + audio_encoder_(nullptr) { + using ::testing::Invoke; + + AudioState::Config config; + config.audio_mixer = AudioMixerImpl::Create(); + config.audio_processing = audio_processing_; + config.audio_device_module = rtc::make_ref_counted<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( + time_controller_.GetClock(), stream_config_, audio_state_, + time_controller_.GetTaskQueueFactory(), &rtp_transport_, + &bitrate_allocator_, &event_log_, absl::nullopt, + std::unique_ptr<voe::ChannelSendInterface>(channel_send_), + field_trials)); + } + + 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::Uri(), + 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); + } + + 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_, GetTargetBitrate()).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_)); + } + } + + MaybeWorkerThread* worker() { return &worker_queue_; } + + test::ScopedKeyValueConfig field_trials; + + private: + RealTimeController time_controller_; + 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. + MaybeWorkerThread 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.has_dscp = true; + 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, has " + "audio_network_adaptor_config: false, has_dscp: true, " + "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " + "enable_non_sender_rtt: 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) { + 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); + } + } +} + +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(true, 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().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, nullptr); + } +} + +TEST(AudioSendStreamTest, AudioNetworkAdaptorReceivesOverhead) { + for (bool use_null_audio_processing : {false, true}) { + ConfigHelper helper(true, true, use_null_audio_processing); + helper.config().send_codec_spec = + AudioSendStream::Config::SendCodecSpec(0, kOpusFormat); + const std::string kAnaConfigString = "abcde"; + + EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _)) + .WillOnce(Invoke( + [&kAnaConfigString](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); + InSequence s; + EXPECT_CALL( + *mock_encoder, + OnReceivedOverhead(Eq(kOverheadPerPacket.bytes<size_t>()))) + .Times(2); + EXPECT_CALL(*mock_encoder, + EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _)) + .WillOnce(Return(true)); + // Note: Overhead is received AFTER ANA has been enabled. + EXPECT_CALL( + *mock_encoder, + OnReceivedOverhead(Eq(kOverheadPerPacket.bytes<size_t>()))) + .WillOnce(Return()); + *return_value = std::move(mock_encoder); + })); + EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) + .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>())); + + auto send_stream = helper.CreateAudioSendStream(); + + auto stream_config = helper.config(); + stream_config.audio_network_adaptor_config = kAnaConfigString; + + send_stream->Reconfigure(stream_config, nullptr); + } +} + +// 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) { + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) { + for (bool use_null_audio_processing : {false, true}) { + ConfigHelper helper(true, true, use_null_audio_processing); + ScopedKeyValueConfig field_trials( + helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) { + for (bool use_null_audio_processing : {false, true}) { + ConfigHelper helper(true, true, use_null_audio_processing); + ScopedKeyValueConfig field_trials( + helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +TEST(AudioSendStreamTest, SSBweWithOverhead) { + for (bool use_null_audio_processing : {false, true}) { + ConfigHelper helper(true, true, use_null_audio_processing); + ScopedKeyValueConfig field_trials(helper.field_trials, + "WebRTC-Audio-LegacyOverhead/Disabled/"); + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) { + for (bool use_null_audio_processing : {false, true}) { + ConfigHelper helper(true, true, use_null_audio_processing); + ScopedKeyValueConfig field_trials( + helper.field_trials, + "WebRTC-Audio-LegacyOverhead/Disabled/" + "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) { + for (bool use_null_audio_processing : {false, true}) { + ConfigHelper helper(true, true, use_null_audio_processing); + ScopedKeyValueConfig field_trials( + helper.field_trials, + "WebRTC-Audio-LegacyOverhead/Disabled/" + "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + } +} + +// 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(), nullptr); + } +} + +TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) { + 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::Uri(), + 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, nullptr); + } +} + +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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + + 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()->RunSynchronous( + [&] { send_stream->OnBitrateUpdated(update); }); + + 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( + rtc::make_ref_counted<MockFrameEncryptor>()); + new_config.frame_encryptor = mock_frame_encryptor_0; + EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr))) + .Times(1); + send_stream->Reconfigure(new_config, nullptr); + + // Not updating the frame encryptor shouldn't force it to reconfigure. + EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(_)).Times(0); + send_stream->Reconfigure(new_config, nullptr); + + // Updating frame encryptor to a new object should force a call to the + // proxy. + rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_1( + rtc::make_ref_counted<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, nullptr); + } +} +} // namespace test +} // namespace webrtc |