/* * 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 #include #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/audio/audio_send_stream.h" #include "webrtc/audio/audio_state.h" #include "webrtc/audio/conversion.h" #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h" #include "webrtc/modules/congestion_controller/include/congestion_controller.h" #include "webrtc/modules/pacing/paced_sender.h" #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h" #include "webrtc/test/mock_voe_channel_proxy.h" #include "webrtc/test/mock_voice_engine.h" namespace webrtc { namespace test { namespace { using testing::_; using testing::Return; const int kChannelId = 1; const uint32_t kSsrc = 1234; const char* kCName = "foo_name"; const int kAudioLevelId = 2; const int kAbsSendTimeId = 3; const int kTransportSequenceNumberId = 4; const int kEchoDelayMedian = 254; const int kEchoDelayStdDev = -3; const int kEchoReturnLoss = -65; const int kEchoReturnLossEnhancement = 101; const unsigned int kSpeechInputLevel = 96; const CallStatistics kCallStats = { 1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123}; const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671}; const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354}; const int kTelephoneEventPayloadType = 123; const int kTelephoneEventCode = 45; const int kTelephoneEventDuration = 6789; struct ConfigHelper { ConfigHelper() : simulated_clock_(123456), stream_config_(nullptr), congestion_controller_(&simulated_clock_, &bitrate_observer_, &remote_bitrate_observer_) { using testing::Invoke; using testing::StrEq; EXPECT_CALL(voice_engine_, RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); EXPECT_CALL(voice_engine_, DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); AudioState::Config config; config.voice_engine = &voice_engine_; audio_state_ = AudioState::Create(config); EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) .WillOnce(Invoke([this](int channel_id) { EXPECT_FALSE(channel_proxy_); channel_proxy_ = new testing::StrictMock(); 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_, SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1); EXPECT_CALL(*channel_proxy_, SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1); EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber( kTransportSequenceNumberId)) .Times(1); EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects( congestion_controller_.pacer(), congestion_controller_.GetTransportFeedbackObserver(), congestion_controller_.packet_router())) .Times(1); EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()) .Times(1); EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)) .Times(1); EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()) .Times(1); return channel_proxy_; })); 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)); stream_config_.rtp.extensions.push_back( RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); stream_config_.rtp.extensions.push_back(RtpExtension( RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); } AudioSendStream::Config& config() { return stream_config_; } rtc::scoped_refptr audio_state() { return audio_state_; } MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } CongestionController* congestion_controller() { return &congestion_controller_; } void SetupMockForSendTelephoneEvent() { EXPECT_TRUE(channel_proxy_); EXPECT_CALL(*channel_proxy_, SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType)) .WillOnce(Return(true)); EXPECT_CALL(*channel_proxy_, SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration)) .WillOnce(Return(true)); } void SetupMockForGetStats() { using testing::DoAll; using testing::SetArgReferee; std::vector 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(voice_engine_, GetSendCodec(kChannelId, _)) .WillRepeatedly(DoAll(SetArgReferee<1>(kCodecInst), Return(0))); EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_)) .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0))); EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_)) .WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0))); EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _)) .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss), SetArgReferee<1>(kEchoReturnLossEnhancement), Return(0))); EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _)) .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian), SetArgReferee<1>(kEchoDelayStdDev), Return(0))); } private: SimulatedClock simulated_clock_; testing::StrictMock voice_engine_; rtc::scoped_refptr audio_state_; AudioSendStream::Config stream_config_; testing::StrictMock* channel_proxy_ = nullptr; testing::NiceMock bitrate_observer_; testing::NiceMock remote_bitrate_observer_; CongestionController congestion_controller_; }; } // namespace TEST(AudioSendStreamTest, ConfigToString) { AudioSendStream::Config config(nullptr); config.rtp.ssrc = kSsrc; config.rtp.extensions.push_back( RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); config.rtp.c_name = kCName; config.voe_channel_id = kChannelId; config.cng_payload_type = 42; EXPECT_EQ( "{rtp: {ssrc: 1234, extensions: [{uri: " "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], " "nack: {rtp_history_ms: 0}, c_name: foo_name}, voe_channel_id: 1, " "cng_payload_type: 42}", config.ToString()); } TEST(AudioSendStreamTest, ConstructDestruct) { ConfigHelper helper; internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), helper.congestion_controller()); } TEST(AudioSendStreamTest, SendTelephoneEvent) { ConfigHelper helper; internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), helper.congestion_controller()); helper.SetupMockForSendTelephoneEvent(); EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType, kTelephoneEventCode, kTelephoneEventDuration)); } TEST(AudioSendStreamTest, SetMuted) { ConfigHelper helper; internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), helper.congestion_controller()); EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true)); send_stream.SetMuted(true); } TEST(AudioSendStreamTest, GetStats) { ConfigHelper helper; internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), helper.congestion_controller()); helper.SetupMockForGetStats(); AudioSendStream::Stats stats = send_stream.GetStats(); EXPECT_EQ(kSsrc, stats.local_ssrc); EXPECT_EQ(static_cast(kCallStats.bytesSent), stats.bytes_sent); EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent); EXPECT_EQ(static_cast(kReportBlock.cumulative_num_packets_lost), stats.packets_lost); EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost); EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name); EXPECT_EQ(static_cast(kReportBlock.extended_highest_sequence_number), stats.ext_seqnum); EXPECT_EQ(static_cast(kReportBlock.interarrival_jitter / (kCodecInst.plfreq / 1000)), stats.jitter_ms); EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms); EXPECT_EQ(static_cast(kSpeechInputLevel), stats.audio_level); 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_FALSE(stats.typing_noise_detected); } TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) { ConfigHelper helper; internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), helper.congestion_controller()); helper.SetupMockForGetStats(); EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); internal::AudioState* internal_audio_state = static_cast(helper.audio_state().get()); VoiceEngineObserver* voe_observer = static_cast(internal_audio_state); voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING); EXPECT_TRUE(send_stream.GetStats().typing_noise_detected); voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING); EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); } } // namespace test } // namespace webrtc