| /* |
| * Copyright (c) 2008 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 <memory> |
| |
| #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "webrtc/call/call.h" |
| #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| #include "webrtc/media/base/fakemediaengine.h" |
| #include "webrtc/media/base/fakenetworkinterface.h" |
| #include "webrtc/media/base/fakertp.h" |
| #include "webrtc/media/base/mediaconstants.h" |
| #include "webrtc/media/engine/fakewebrtccall.h" |
| #include "webrtc/media/engine/fakewebrtcvoiceengine.h" |
| #include "webrtc/media/engine/webrtcvoiceengine.h" |
| #include "webrtc/modules/audio_device/include/mock_audio_device.h" |
| #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
| #include "webrtc/pc/channel.h" |
| #include "webrtc/rtc_base/arraysize.h" |
| #include "webrtc/rtc_base/byteorder.h" |
| #include "webrtc/rtc_base/safe_conversions.h" |
| #include "webrtc/rtc_base/scoped_ref_ptr.h" |
| #include "webrtc/test/field_trial.h" |
| #include "webrtc/test/gtest.h" |
| #include "webrtc/test/mock_audio_decoder_factory.h" |
| #include "webrtc/test/mock_audio_encoder_factory.h" |
| #include "webrtc/voice_engine/transmit_mixer.h" |
| |
| using testing::_; |
| using testing::ContainerEq; |
| using testing::Return; |
| using testing::ReturnPointee; |
| using testing::SaveArg; |
| using testing::StrictMock; |
| |
| namespace { |
| |
| constexpr uint32_t kMaxUnsignaledRecvStreams = 4; |
| |
| const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1); |
| const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1); |
| const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 32000, 2); |
| const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1); |
| const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1); |
| const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1); |
| const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1); |
| const cricket::AudioCodec |
| kTelephoneEventCodec1(106, "telephone-event", 8000, 0, 1); |
| const cricket::AudioCodec |
| kTelephoneEventCodec2(107, "telephone-event", 32000, 0, 1); |
| |
| const uint32_t kSsrc0 = 0; |
| const uint32_t kSsrc1 = 1; |
| const uint32_t kSsrcX = 0x99; |
| const uint32_t kSsrcY = 0x17; |
| const uint32_t kSsrcZ = 0x42; |
| const uint32_t kSsrcW = 0x02; |
| const uint32_t kSsrcs4[] = { 11, 200, 30, 44 }; |
| |
| constexpr int kRtpHistoryMs = 5000; |
| |
| class FakeVoEWrapper : public cricket::VoEWrapper { |
| public: |
| explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine) |
| : cricket::VoEWrapper(engine) { |
| } |
| }; |
| |
| class MockTransmitMixer : public webrtc::voe::TransmitMixer { |
| public: |
| MockTransmitMixer() = default; |
| virtual ~MockTransmitMixer() = default; |
| |
| MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable)); |
| }; |
| |
| void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) { |
| RTC_DCHECK(adm); |
| EXPECT_CALL(*adm, AddRef()).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, Release()).WillOnce(Return(0)); |
| #if !defined(WEBRTC_IOS) |
| EXPECT_CALL(*adm, Recording()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm, SetRecordingChannel(webrtc::AudioDeviceModule:: |
| ChannelType::kChannelBoth)).WillOnce(Return(0)); |
| #if defined(WEBRTC_WIN) |
| EXPECT_CALL(*adm, SetRecordingDevice( |
| testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>( |
| webrtc::AudioDeviceModule::kDefaultCommunicationDevice))) |
| .WillOnce(Return(0)); |
| #else |
| EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0)); |
| #endif // #if defined(WEBRTC_WIN) |
| EXPECT_CALL(*adm, InitMicrophone()).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, StereoRecordingIsAvailable(testing::_)).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, SetStereoRecording(false)).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, Playing()).WillOnce(Return(false)); |
| #if defined(WEBRTC_WIN) |
| EXPECT_CALL(*adm, SetPlayoutDevice( |
| testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>( |
| webrtc::AudioDeviceModule::kDefaultCommunicationDevice))) |
| .WillOnce(Return(0)); |
| #else |
| EXPECT_CALL(*adm, SetPlayoutDevice(0)).WillOnce(Return(0)); |
| #endif // #if defined(WEBRTC_WIN) |
| EXPECT_CALL(*adm, InitSpeaker()).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, StereoPlayoutIsAvailable(testing::_)).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, SetStereoPlayout(false)).WillOnce(Return(0)); |
| #endif // #if !defined(WEBRTC_IOS) |
| EXPECT_CALL(*adm, BuiltInAECIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm, BuiltInAGCIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm, SetAGC(true)).WillOnce(Return(0)); |
| } |
| } // namespace |
| |
| // Tests that our stub library "works". |
| TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) { |
| StrictMock<webrtc::test::MockAudioDeviceModule> adm; |
| AdmSetupExpectations(&adm); |
| rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm = |
| new rtc::RefCountedObject< |
| StrictMock<webrtc::test::MockAudioProcessing>>(); |
| webrtc::AudioProcessing::Config apm_config; |
| EXPECT_CALL(*apm, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config)); |
| EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config)); |
| EXPECT_CALL(*apm, SetExtraOptions(testing::_)); |
| EXPECT_CALL(*apm, Initialize()).WillOnce(Return(0)); |
| EXPECT_CALL(*apm, DetachAecDump()); |
| StrictMock<MockTransmitMixer> transmit_mixer; |
| EXPECT_CALL(transmit_mixer, EnableStereoChannelSwapping(false)); |
| cricket::FakeWebRtcVoiceEngine voe(&transmit_mixer); |
| EXPECT_FALSE(voe.IsInited()); |
| { |
| cricket::WebRtcVoiceEngine engine( |
| &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, |
| new FakeVoEWrapper(&voe)); |
| engine.Init(); |
| EXPECT_TRUE(voe.IsInited()); |
| } |
| EXPECT_FALSE(voe.IsInited()); |
| } |
| |
| class FakeAudioSink : public webrtc::AudioSinkInterface { |
| public: |
| void OnData(const Data& audio) override {} |
| }; |
| |
| class FakeAudioSource : public cricket::AudioSource { |
| void SetSink(Sink* sink) override {} |
| }; |
| |
| class WebRtcVoiceEngineTestFake : public testing::Test { |
| public: |
| WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {} |
| |
| explicit WebRtcVoiceEngineTestFake(const char* field_trials) |
| : apm_(new rtc::RefCountedObject< |
| StrictMock<webrtc::test::MockAudioProcessing>>()), |
| apm_gc_(*apm_->gain_control()), |
| apm_ec_(*apm_->echo_cancellation()), |
| apm_ns_(*apm_->noise_suppression()), |
| apm_vd_(*apm_->voice_detection()), |
| call_(webrtc::Call::Config(&event_log_)), |
| voe_(&transmit_mixer_), |
| override_field_trials_(field_trials) { |
| // AudioDeviceModule. |
| AdmSetupExpectations(&adm_); |
| // AudioProcessing. |
| EXPECT_CALL(*apm_, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config_)); |
| EXPECT_CALL(*apm_, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config_)); |
| EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); |
| EXPECT_CALL(*apm_, Initialize()).WillOnce(Return(0)); |
| EXPECT_CALL(*apm_, DetachAecDump()); |
| // Default Options. |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_vd_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(false)); |
| // Init does not overwrite default AGC config. |
| EXPECT_CALL(apm_gc_, target_level_dbfs()).WillOnce(Return(1)); |
| EXPECT_CALL(apm_gc_, compression_gain_db()).WillRepeatedly(Return(5)); |
| EXPECT_CALL(apm_gc_, is_limiter_enabled()).WillRepeatedly(Return(true)); |
| EXPECT_CALL(apm_gc_, set_target_level_dbfs(1)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, set_compression_gain_db(5)).WillRepeatedly(Return(0)); |
| EXPECT_CALL(apm_gc_, enable_limiter(true)).WillRepeatedly(Return(0)); |
| // TODO(kwiberg): We should use mock factories here, but a bunch of |
| // the tests here probe the specific set of codecs provided by the builtin |
| // factories. Those tests should probably be moved elsewhere. |
| auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); |
| auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); |
| engine_.reset(new cricket::WebRtcVoiceEngine(&adm_, encoder_factory, |
| decoder_factory, nullptr, apm_, |
| new FakeVoEWrapper(&voe_))); |
| engine_->Init(); |
| send_parameters_.codecs.push_back(kPcmuCodec); |
| recv_parameters_.codecs.push_back(kPcmuCodec); |
| // Default Options. |
| EXPECT_TRUE(IsHighPassFilterEnabled()); |
| } |
| |
| bool SetupChannel() { |
| EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); |
| channel_ = engine_->CreateChannel(&call_, cricket::MediaConfig(), |
| cricket::AudioOptions()); |
| return (channel_ != nullptr); |
| } |
| |
| bool SetupRecvStream() { |
| if (!SetupChannel()) { |
| return false; |
| } |
| return AddRecvStream(kSsrcX); |
| } |
| |
| bool SetupSendStream() { |
| if (!SetupChannel()) { |
| return false; |
| } |
| if (!channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))) { |
| return false; |
| } |
| EXPECT_CALL(*apm_, set_output_will_be_muted(false)); |
| return channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_); |
| } |
| |
| bool AddRecvStream(uint32_t ssrc) { |
| EXPECT_TRUE(channel_); |
| return channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc)); |
| } |
| |
| void SetupForMultiSendStream() { |
| EXPECT_TRUE(SetupSendStream()); |
| // Remove stream added in Setup. |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrcX)); |
| // Verify the channel does not exist. |
| EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX)); |
| } |
| |
| void DeliverPacket(const void* data, int len) { |
| rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len); |
| channel_->OnPacketReceived(&packet, rtc::PacketTime()); |
| } |
| |
| void TearDown() override { |
| delete channel_; |
| } |
| |
| const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) { |
| const auto* send_stream = call_.GetAudioSendStream(ssrc); |
| EXPECT_TRUE(send_stream); |
| return *send_stream; |
| } |
| |
| const cricket::FakeAudioReceiveStream& GetRecvStream(uint32_t ssrc) { |
| const auto* recv_stream = call_.GetAudioReceiveStream(ssrc); |
| EXPECT_TRUE(recv_stream); |
| return *recv_stream; |
| } |
| |
| const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) { |
| return GetSendStream(ssrc).GetConfig(); |
| } |
| |
| const webrtc::AudioReceiveStream::Config& GetRecvStreamConfig(uint32_t ssrc) { |
| return GetRecvStream(ssrc).GetConfig(); |
| } |
| |
| void SetSend(bool enable) { |
| ASSERT_TRUE(channel_); |
| if (enable) { |
| EXPECT_CALL(adm_, RecordingIsInitialized()).WillOnce(Return(false)); |
| EXPECT_CALL(adm_, Recording()).WillOnce(Return(false)); |
| EXPECT_CALL(adm_, InitRecording()).WillOnce(Return(0)); |
| EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); |
| } |
| channel_->SetSend(enable); |
| } |
| |
| void SetSendParameters(const cricket::AudioSendParameters& params) { |
| EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); |
| ASSERT_TRUE(channel_); |
| EXPECT_TRUE(channel_->SetSendParameters(params)); |
| } |
| |
| void SetAudioSend(uint32_t ssrc, bool enable, cricket::AudioSource* source, |
| const cricket::AudioOptions* options = nullptr) { |
| EXPECT_CALL(*apm_, set_output_will_be_muted(!enable)); |
| ASSERT_TRUE(channel_); |
| if (enable && options) { |
| EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); |
| } |
| EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source)); |
| } |
| |
| void TestInsertDtmf(uint32_t ssrc, bool caller, |
| const cricket::AudioCodec& codec) { |
| EXPECT_TRUE(SetupChannel()); |
| if (caller) { |
| // If this is a caller, local description will be applied and add the |
| // send stream. |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Test we can only InsertDtmf when the other side supports telephone-event. |
| SetSendParameters(send_parameters_); |
| SetSend(true); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111)); |
| send_parameters_.codecs.push_back(codec); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| |
| if (!caller) { |
| // If this is callee, there's no active send channel yet. |
| EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123)); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Check we fail if the ssrc is invalid. |
| EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111)); |
| |
| // Test send. |
| cricket::FakeAudioSendStream::TelephoneEvent telephone_event = |
| GetSendStream(kSsrcX).GetLatestTelephoneEvent(); |
| EXPECT_EQ(-1, telephone_event.payload_type); |
| EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123)); |
| telephone_event = GetSendStream(kSsrcX).GetLatestTelephoneEvent(); |
| EXPECT_EQ(codec.id, telephone_event.payload_type); |
| EXPECT_EQ(codec.clockrate, telephone_event.payload_frequency); |
| EXPECT_EQ(2, telephone_event.event_code); |
| EXPECT_EQ(123, telephone_event.duration_ms); |
| } |
| |
| // Test that send bandwidth is set correctly. |
| // |codec| is the codec under test. |
| // |max_bitrate| is a parameter to set to SetMaxSendBandwidth(). |
| // |expected_result| is the expected result from SetMaxSendBandwidth(). |
| // |expected_bitrate| is the expected audio bitrate afterward. |
| void TestMaxSendBandwidth(const cricket::AudioCodec& codec, |
| int max_bitrate, |
| bool expected_result, |
| int expected_bitrate) { |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| parameters.max_bandwidth_bps = max_bitrate; |
| if (expected_result) { |
| SetSendParameters(parameters); |
| } else { |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| EXPECT_EQ(expected_bitrate, GetCodecBitrate(kSsrcX)); |
| } |
| |
| // Sets the per-stream maximum bitrate limit for the specified SSRC. |
| bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) { |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrc); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| |
| parameters.encodings[0].max_bitrate_bps = rtc::Optional<int>(bitrate); |
| return channel_->SetRtpSendParameters(ssrc, parameters); |
| } |
| |
| void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) { |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(codec); |
| send_parameters.max_bandwidth_bps = bitrate; |
| SetSendParameters(send_parameters); |
| } |
| |
| void CheckSendCodecBitrate(int32_t ssrc, |
| const char expected_name[], |
| int expected_bitrate) { |
| const auto& spec = GetSendStreamConfig(ssrc).send_codec_spec; |
| EXPECT_EQ(expected_name, spec->format.name); |
| EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps); |
| } |
| |
| rtc::Optional<int> GetCodecBitrate(int32_t ssrc) { |
| return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps; |
| } |
| |
| const rtc::Optional<std::string>& GetAudioNetworkAdaptorConfig(int32_t ssrc) { |
| return GetSendStreamConfig(ssrc).audio_network_adaptor_config; |
| } |
| |
| void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec, |
| int global_max, |
| int stream_max, |
| bool expected_result, |
| int expected_codec_bitrate) { |
| // Clear the bitrate limit from the previous test case. |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcX, -1)); |
| |
| // Attempt to set the requested bitrate limits. |
| SetGlobalMaxBitrate(codec, global_max); |
| EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrcX, stream_max)); |
| |
| // Verify that reading back the parameters gives results |
| // consistent with the Set() result. |
| webrtc::RtpParameters resulting_parameters = |
| channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(1UL, resulting_parameters.encodings.size()); |
| EXPECT_EQ(expected_result ? stream_max : -1, |
| resulting_parameters.encodings[0].max_bitrate_bps); |
| |
| // Verify that the codec settings have the expected bitrate. |
| EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrcX)); |
| } |
| |
| void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps, |
| int expected_min_bitrate_bps, |
| const char* start_bitrate_kbps, |
| int expected_start_bitrate_bps, |
| const char* max_bitrate_kbps, |
| int expected_max_bitrate_bps) { |
| EXPECT_TRUE(SetupSendStream()); |
| auto& codecs = send_parameters_.codecs; |
| codecs.clear(); |
| codecs.push_back(kOpusCodec); |
| codecs[0].params[cricket::kCodecParamMinBitrate] = min_bitrate_kbps; |
| codecs[0].params[cricket::kCodecParamStartBitrate] = start_bitrate_kbps; |
| codecs[0].params[cricket::kCodecParamMaxBitrate] = max_bitrate_kbps; |
| SetSendParameters(send_parameters_); |
| |
| EXPECT_EQ(expected_min_bitrate_bps, |
| call_.GetConfig().bitrate_config.min_bitrate_bps); |
| EXPECT_EQ(expected_start_bitrate_bps, |
| call_.GetConfig().bitrate_config.start_bitrate_bps); |
| EXPECT_EQ(expected_max_bitrate_bps, |
| call_.GetConfig().bitrate_config.max_bitrate_bps); |
| } |
| |
| void TestSetSendRtpHeaderExtensions(const std::string& ext) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Ensure extensions are off by default. |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure unknown extensions won't cause an error. |
| send_parameters_.extensions.push_back( |
| webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extensions stay off with an empty list of headers. |
| send_parameters_.extensions.clear(); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extension is set properly. |
| const int id = 1; |
| send_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(1u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetSendStreamConfig(kSsrcX).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id); |
| |
| // Ensure extension is set properly on new stream. |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcY))); |
| EXPECT_NE(call_.GetAudioSendStream(kSsrcX), |
| call_.GetAudioSendStream(kSsrcY)); |
| EXPECT_EQ(1u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetSendStreamConfig(kSsrcY).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetSendStreamConfig(kSsrcY).rtp.extensions[0].id); |
| |
| // Ensure all extensions go back off with an empty list. |
| send_parameters_.codecs.push_back(kPcmuCodec); |
| send_parameters_.extensions.clear(); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); |
| } |
| |
| void TestSetRecvRtpHeaderExtensions(const std::string& ext) { |
| EXPECT_TRUE(SetupRecvStream()); |
| |
| // Ensure extensions are off by default. |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure unknown extensions won't cause an error. |
| recv_parameters_.extensions.push_back( |
| webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extensions stay off with an empty list of headers. |
| recv_parameters_.extensions.clear(); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extension is set properly. |
| const int id = 2; |
| recv_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].id); |
| |
| // Ensure extension is set properly on new stream. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_NE(call_.GetAudioReceiveStream(kSsrcX), |
| call_.GetAudioReceiveStream(kSsrcY)); |
| EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].id); |
| |
| // Ensure all extensions go back off with an empty list. |
| recv_parameters_.extensions.clear(); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size()); |
| } |
| |
| webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const { |
| webrtc::AudioSendStream::Stats stats; |
| stats.local_ssrc = 12; |
| stats.bytes_sent = 345; |
| stats.packets_sent = 678; |
| stats.packets_lost = 9012; |
| stats.fraction_lost = 34.56f; |
| stats.codec_name = "codec_name_send"; |
| stats.codec_payload_type = rtc::Optional<int>(42); |
| stats.ext_seqnum = 789; |
| stats.jitter_ms = 12; |
| stats.rtt_ms = 345; |
| stats.audio_level = 678; |
| stats.aec_quality_min = 9.01f; |
| stats.echo_delay_median_ms = 234; |
| stats.echo_delay_std_ms = 567; |
| stats.echo_return_loss = 890; |
| stats.echo_return_loss_enhancement = 1234; |
| stats.residual_echo_likelihood = 0.432f; |
| stats.residual_echo_likelihood_recent_max = 0.6f; |
| stats.ana_statistics.bitrate_action_counter = rtc::Optional<uint32_t>(321); |
| stats.ana_statistics.channel_action_counter = rtc::Optional<uint32_t>(432); |
| stats.ana_statistics.dtx_action_counter = rtc::Optional<uint32_t>(543); |
| stats.ana_statistics.fec_action_counter = rtc::Optional<uint32_t>(654); |
| stats.ana_statistics.frame_length_increase_counter = |
| rtc::Optional<uint32_t>(765); |
| stats.ana_statistics.frame_length_decrease_counter = |
| rtc::Optional<uint32_t>(876); |
| stats.ana_statistics.uplink_packet_loss_fraction = |
| rtc::Optional<float>(987.0); |
| stats.typing_noise_detected = true; |
| return stats; |
| } |
| void SetAudioSendStreamStats() { |
| for (auto* s : call_.GetAudioSendStreams()) { |
| s->SetStats(GetAudioSendStreamStats()); |
| } |
| } |
| void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info, |
| bool is_sending) { |
| const auto stats = GetAudioSendStreamStats(); |
| EXPECT_EQ(info.ssrc(), stats.local_ssrc); |
| EXPECT_EQ(info.bytes_sent, stats.bytes_sent); |
| EXPECT_EQ(info.packets_sent, stats.packets_sent); |
| EXPECT_EQ(info.packets_lost, stats.packets_lost); |
| EXPECT_EQ(info.fraction_lost, stats.fraction_lost); |
| EXPECT_EQ(info.codec_name, stats.codec_name); |
| EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); |
| EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum); |
| EXPECT_EQ(info.jitter_ms, stats.jitter_ms); |
| EXPECT_EQ(info.rtt_ms, stats.rtt_ms); |
| EXPECT_EQ(info.audio_level, stats.audio_level); |
| EXPECT_EQ(info.aec_quality_min, stats.aec_quality_min); |
| EXPECT_EQ(info.echo_delay_median_ms, stats.echo_delay_median_ms); |
| EXPECT_EQ(info.echo_delay_std_ms, stats.echo_delay_std_ms); |
| EXPECT_EQ(info.echo_return_loss, stats.echo_return_loss); |
| EXPECT_EQ(info.echo_return_loss_enhancement, |
| stats.echo_return_loss_enhancement); |
| EXPECT_EQ(info.residual_echo_likelihood, stats.residual_echo_likelihood); |
| EXPECT_EQ(info.residual_echo_likelihood_recent_max, |
| stats.residual_echo_likelihood_recent_max); |
| EXPECT_EQ(info.ana_statistics.bitrate_action_counter, |
| stats.ana_statistics.bitrate_action_counter); |
| EXPECT_EQ(info.ana_statistics.channel_action_counter, |
| stats.ana_statistics.channel_action_counter); |
| EXPECT_EQ(info.ana_statistics.dtx_action_counter, |
| stats.ana_statistics.dtx_action_counter); |
| EXPECT_EQ(info.ana_statistics.fec_action_counter, |
| stats.ana_statistics.fec_action_counter); |
| EXPECT_EQ(info.ana_statistics.frame_length_increase_counter, |
| stats.ana_statistics.frame_length_increase_counter); |
| EXPECT_EQ(info.ana_statistics.frame_length_decrease_counter, |
| stats.ana_statistics.frame_length_decrease_counter); |
| EXPECT_EQ(info.ana_statistics.uplink_packet_loss_fraction, |
| stats.ana_statistics.uplink_packet_loss_fraction); |
| EXPECT_EQ(info.typing_noise_detected, |
| stats.typing_noise_detected && is_sending); |
| } |
| |
| webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const { |
| webrtc::AudioReceiveStream::Stats stats; |
| stats.remote_ssrc = 123; |
| stats.bytes_rcvd = 456; |
| stats.packets_rcvd = 768; |
| stats.packets_lost = 101; |
| stats.fraction_lost = 23.45f; |
| stats.codec_name = "codec_name_recv"; |
| stats.codec_payload_type = rtc::Optional<int>(42); |
| stats.ext_seqnum = 678; |
| stats.jitter_ms = 901; |
| stats.jitter_buffer_ms = 234; |
| stats.jitter_buffer_preferred_ms = 567; |
| stats.delay_estimate_ms = 890; |
| stats.audio_level = 1234; |
| stats.total_samples_received = 5678901; |
| stats.concealed_samples = 234; |
| stats.expand_rate = 5.67f; |
| stats.speech_expand_rate = 8.90f; |
| stats.secondary_decoded_rate = 1.23f; |
| stats.secondary_discarded_rate = 0.12f; |
| stats.accelerate_rate = 4.56f; |
| stats.preemptive_expand_rate = 7.89f; |
| stats.decoding_calls_to_silence_generator = 12; |
| stats.decoding_calls_to_neteq = 345; |
| stats.decoding_normal = 67890; |
| stats.decoding_plc = 1234; |
| stats.decoding_cng = 5678; |
| stats.decoding_plc_cng = 9012; |
| stats.decoding_muted_output = 3456; |
| stats.capture_start_ntp_time_ms = 7890; |
| return stats; |
| } |
| void SetAudioReceiveStreamStats() { |
| for (auto* s : call_.GetAudioReceiveStreams()) { |
| s->SetStats(GetAudioReceiveStreamStats()); |
| } |
| } |
| void VerifyVoiceReceiverInfo(const cricket::VoiceReceiverInfo& info) { |
| const auto stats = GetAudioReceiveStreamStats(); |
| EXPECT_EQ(info.ssrc(), stats.remote_ssrc); |
| EXPECT_EQ(info.bytes_rcvd, stats.bytes_rcvd); |
| EXPECT_EQ(info.packets_rcvd, stats.packets_rcvd); |
| EXPECT_EQ(info.packets_lost, stats.packets_lost); |
| EXPECT_EQ(info.fraction_lost, stats.fraction_lost); |
| EXPECT_EQ(info.codec_name, stats.codec_name); |
| EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); |
| EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum); |
| EXPECT_EQ(info.jitter_ms, stats.jitter_ms); |
| EXPECT_EQ(info.jitter_buffer_ms, stats.jitter_buffer_ms); |
| EXPECT_EQ(info.jitter_buffer_preferred_ms, |
| stats.jitter_buffer_preferred_ms); |
| EXPECT_EQ(info.delay_estimate_ms, stats.delay_estimate_ms); |
| EXPECT_EQ(info.audio_level, stats.audio_level); |
| EXPECT_EQ(info.total_samples_received, stats.total_samples_received); |
| EXPECT_EQ(info.concealed_samples, stats.concealed_samples); |
| EXPECT_EQ(info.expand_rate, stats.expand_rate); |
| EXPECT_EQ(info.speech_expand_rate, stats.speech_expand_rate); |
| EXPECT_EQ(info.secondary_decoded_rate, stats.secondary_decoded_rate); |
| EXPECT_EQ(info.secondary_discarded_rate, stats.secondary_discarded_rate); |
| EXPECT_EQ(info.accelerate_rate, stats.accelerate_rate); |
| EXPECT_EQ(info.preemptive_expand_rate, stats.preemptive_expand_rate); |
| EXPECT_EQ(info.decoding_calls_to_silence_generator, |
| stats.decoding_calls_to_silence_generator); |
| EXPECT_EQ(info.decoding_calls_to_neteq, stats.decoding_calls_to_neteq); |
| EXPECT_EQ(info.decoding_normal, stats.decoding_normal); |
| EXPECT_EQ(info.decoding_plc, stats.decoding_plc); |
| EXPECT_EQ(info.decoding_cng, stats.decoding_cng); |
| EXPECT_EQ(info.decoding_plc_cng, stats.decoding_plc_cng); |
| EXPECT_EQ(info.decoding_muted_output, stats.decoding_muted_output); |
| EXPECT_EQ(info.capture_start_ntp_time_ms, stats.capture_start_ntp_time_ms); |
| } |
| void VerifyVoiceSendRecvCodecs(const cricket::VoiceMediaInfo& info) const { |
| EXPECT_EQ(send_parameters_.codecs.size(), info.send_codecs.size()); |
| for (const cricket::AudioCodec& codec : send_parameters_.codecs) { |
| ASSERT_EQ(info.send_codecs.count(codec.id), 1U); |
| EXPECT_EQ(info.send_codecs.find(codec.id)->second, |
| codec.ToCodecParameters()); |
| } |
| EXPECT_EQ(recv_parameters_.codecs.size(), info.receive_codecs.size()); |
| for (const cricket::AudioCodec& codec : recv_parameters_.codecs) { |
| ASSERT_EQ(info.receive_codecs.count(codec.id), 1U); |
| EXPECT_EQ(info.receive_codecs.find(codec.id)->second, |
| codec.ToCodecParameters()); |
| } |
| } |
| |
| bool IsHighPassFilterEnabled() { |
| return engine_->GetApmConfigForTest().high_pass_filter.enabled; |
| } |
| |
| protected: |
| StrictMock<webrtc::test::MockAudioDeviceModule> adm_; |
| rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_; |
| webrtc::test::MockGainControl& apm_gc_; |
| webrtc::test::MockEchoCancellation& apm_ec_; |
| webrtc::test::MockNoiseSuppression& apm_ns_; |
| webrtc::test::MockVoiceDetection& apm_vd_; |
| StrictMock<MockTransmitMixer> transmit_mixer_; |
| webrtc::RtcEventLogNullImpl event_log_; |
| cricket::FakeCall call_; |
| cricket::FakeWebRtcVoiceEngine voe_; |
| std::unique_ptr<cricket::WebRtcVoiceEngine> engine_; |
| cricket::VoiceMediaChannel* channel_ = nullptr; |
| cricket::AudioSendParameters send_parameters_; |
| cricket::AudioRecvParameters recv_parameters_; |
| FakeAudioSource fake_source_; |
| webrtc::AudioProcessing::Config apm_config_; |
| |
| private: |
| webrtc::test::ScopedFieldTrials override_field_trials_; |
| }; |
| |
| // Tests that we can create and destroy a channel. |
| TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) { |
| EXPECT_TRUE(SetupChannel()); |
| } |
| |
| // Test that we can add a send stream and that it has the correct defaults. |
| TEST_F(WebRtcVoiceEngineTestFake, CreateSendStream) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); |
| EXPECT_EQ(kSsrcX, config.rtp.ssrc); |
| EXPECT_EQ("", config.rtp.c_name); |
| EXPECT_EQ(0u, config.rtp.extensions.size()); |
| EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_), |
| config.send_transport); |
| } |
| |
| // Test that we can add a receive stream and that it has the correct defaults. |
| TEST_F(WebRtcVoiceEngineTestFake, CreateRecvStream) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| const webrtc::AudioReceiveStream::Config& config = |
| GetRecvStreamConfig(kSsrcX); |
| EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc); |
| EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc); |
| EXPECT_FALSE(config.rtp.transport_cc); |
| EXPECT_EQ(0u, config.rtp.extensions.size()); |
| EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_), |
| config.rtcp_send_transport); |
| EXPECT_EQ("", config.sync_group); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) { |
| const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs(); |
| bool opus_found = false; |
| for (cricket::AudioCodec codec : codecs) { |
| if (codec.name == "opus") { |
| EXPECT_TRUE(HasTransportCc(codec)); |
| opus_found = true; |
| } |
| } |
| EXPECT_TRUE(opus_found); |
| } |
| |
| // Test that we set our inbound codecs properly, including changing PT. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs[0].id = 106; // collide with existing CN 32k |
| parameters.codecs[2].id = 126; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {106, {"ISAC", 16000, 1}}, |
| {126, {"telephone-event", 8000, 1}}, |
| {107, {"telephone-event", 32000, 1}}}))); |
| } |
| |
| // Test that we fail to set an unknown inbound codec. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1)); |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| // Test that we fail if we have duplicate types in the inbound list. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = kIsacCodec.id; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| // Test that we can decode OPUS without stereo parameters. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {103, {"ISAC", 16000, 1}}, |
| {111, {"opus", 48000, 2}}}))); |
| } |
| |
| // Test that we can decode OPUS with stereo = 0. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[2].params["stereo"] = "0"; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {103, {"ISAC", 16000, 1}}, |
| {111, {"opus", 48000, 2, {{"stereo", "0"}}}}}))); |
| } |
| |
| // Test that we can decode OPUS with stereo = 1. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[2].params["stereo"] = "1"; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {103, {"ISAC", 16000, 1}}, |
| {111, {"opus", 48000, 2, {{"stereo", "1"}}}}}))); |
| } |
| |
| // Test that changes to recv codecs are applied to all streams. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs[0].id = 106; // collide with existing CN 32k |
| parameters.codecs[2].id = 126; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| for (const auto& ssrc : {kSsrcX, kSsrcY}) { |
| EXPECT_TRUE(AddRecvStream(ssrc)); |
| EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {106, {"ISAC", 16000, 1}}, |
| {126, {"telephone-event", 8000, 1}}, |
| {107, {"telephone-event", 32000, 1}}}))); |
| } |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].id = 106; // collide with existing CN 32k |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map; |
| ASSERT_EQ(1, dm.count(106)); |
| EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106)); |
| } |
| |
| // Test that we can apply the same set of codecs again while playing. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| // Remapping a payload type to a different codec should fail. |
| parameters.codecs[0] = kOpusCodec; |
| parameters.codecs[0].id = kIsacCodec.id; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we can add a codec while playing. |
| TEST_F(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| channel_->SetPlayout(true); |
| |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we accept adding the same codec with a different payload type. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847 |
| TEST_F(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| ++parameters.codecs[0].id; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Test that when autobw is enabled, bitrate is kept as the default |
| // value. autobw is enabled for the following tests because the target |
| // bitrate is <= 0. |
| |
| // ISAC, default bitrate == 32000. |
| TestMaxSendBandwidth(kIsacCodec, 0, true, 32000); |
| |
| // PCMU, default bitrate == 64000. |
| TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000); |
| |
| // opus, default bitrate == 32000 in mono. |
| TestMaxSendBandwidth(kOpusCodec, -1, true, 32000); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // ISAC, default bitrate == 32000. |
| TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000); |
| // Rates above the max (56000) should be capped. |
| TestMaxSendBandwidth(kIsacCodec, 100000, true, 32000); |
| |
| // opus, default bitrate == 64000. |
| TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000); |
| TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000); |
| // Rates above the max (510000) should be capped. |
| TestMaxSendBandwidth(kOpusCodec, 600000, true, 510000); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Test that we can only set a maximum bitrate for a fixed-rate codec |
| // if it's bigger than the fixed rate. |
| |
| // PCMU, fixed bitrate == 64000. |
| TestMaxSendBandwidth(kPcmuCodec, 0, true, 64000); |
| TestMaxSendBandwidth(kPcmuCodec, 1, false, 64000); |
| TestMaxSendBandwidth(kPcmuCodec, 128000, true, 64000); |
| TestMaxSendBandwidth(kPcmuCodec, 32000, false, 64000); |
| TestMaxSendBandwidth(kPcmuCodec, 64000, true, 64000); |
| TestMaxSendBandwidth(kPcmuCodec, 63999, false, 64000); |
| TestMaxSendBandwidth(kPcmuCodec, 64001, true, 64000); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) { |
| EXPECT_TRUE(SetupChannel()); |
| const int kDesiredBitrate = 128000; |
| cricket::AudioSendParameters parameters; |
| parameters.codecs = engine_->send_codecs(); |
| parameters.max_bandwidth_bps = kDesiredBitrate; |
| SetSendParameters(parameters); |
| |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| |
| EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX)); |
| } |
| |
| // Test that bitrate cannot be set for CBR codecs. |
| // Bitrate is ignored if it is higher than the fixed bitrate. |
| // Bitrate less then the fixed bitrate is an error. |
| TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // PCMU, default bitrate == 64000. |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| |
| send_parameters_.max_bandwidth_bps = 128000; |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| |
| send_parameters_.max_bandwidth_bps = 128; |
| EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| } |
| |
| // Test that the per-stream bitrate limit and the global |
| // bitrate limit both apply. |
| TEST_F(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // opus, default bitrate == 32000. |
| SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 32000); |
| SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000); |
| SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000); |
| SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000); |
| |
| // CBR codecs allow both maximums to exceed the bitrate. |
| SetAndExpectMaxBitrate(kPcmuCodec, 0, 0, true, 64000); |
| SetAndExpectMaxBitrate(kPcmuCodec, 64001, 0, true, 64000); |
| SetAndExpectMaxBitrate(kPcmuCodec, 0, 64001, true, 64000); |
| SetAndExpectMaxBitrate(kPcmuCodec, 64001, 64001, true, 64000); |
| |
| // CBR codecs don't allow per stream maximums to be too low. |
| SetAndExpectMaxBitrate(kPcmuCodec, 0, 63999, false, 64000); |
| SetAndExpectMaxBitrate(kPcmuCodec, 64001, 63999, false, 64000); |
| } |
| |
| // Test that an attempt to set RtpParameters for a stream that does not exist |
| // fails. |
| TEST_F(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) { |
| EXPECT_TRUE(SetupChannel()); |
| webrtc::RtpParameters nonexistent_parameters = |
| channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(0, nonexistent_parameters.encodings.size()); |
| |
| nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, |
| CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) { |
| // This test verifies that setting RtpParameters succeeds only if |
| // the structure contains exactly one encoding. |
| // TODO(skvlad): Update this test when we start supporting setting parameters |
| // for each encoding individually. |
| |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); |
| // Two or more encodings should result in failure. |
| parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters)); |
| // Zero encodings should also fail. |
| parameters.encodings.clear(); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters)); |
| } |
| |
| // Changing the SSRC through RtpParameters is not allowed. |
| TEST_F(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); |
| parameters.encodings[0].ssrc = rtc::Optional<uint32_t>(0xdeadbeef); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters)); |
| } |
| |
| // Test that a stream will not be sending if its encoding is made |
| // inactive through SetRtpSendParameters. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| // Get current parameters and change "active" to false. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_TRUE(parameters.encodings[0].active); |
| parameters.encodings[0].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters)); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Now change it back to active and verify we resume sending. |
| parameters.encodings[0].active = true; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters)); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that SetRtpSendParameters configures the correct encoding channel for |
| // each SSRC. |
| TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) { |
| SetupForMultiSendStream(); |
| // Create send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| // Configure one stream to be limited by the stream config, another to be |
| // limited by the global max, and the third one with no per-stream limit |
| // (still subject to the global limit). |
| SetGlobalMaxBitrate(kOpusCodec, 32000); |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 24000)); |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 48000)); |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1)); |
| |
| EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0])); |
| EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[1])); |
| EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2])); |
| |
| // Remove the global cap; the streams should switch to their respective |
| // maximums (or remain unchanged if there was no other limit on them.) |
| SetGlobalMaxBitrate(kOpusCodec, -1); |
| EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0])); |
| EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[1])); |
| EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2])); |
| } |
| |
| // Test that GetRtpSendParameters returns the currently configured codecs. |
| TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSendParameters(parameters); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); |
| EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); |
| } |
| |
| // Test that GetRtpSendParameters returns an SSRC. |
| TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) { |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc); |
| } |
| |
| // Test that if we set/get parameters multiple times, we get the same results. |
| TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSendParameters(parameters); |
| |
| webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrcX); |
| |
| // We should be able to set the params we just got. |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, initial_params)); |
| |
| // ... And this shouldn't change the params returned by GetRtpSendParameters. |
| webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrcX)); |
| } |
| |
| // Test that max_bitrate_bps in send stream config gets updated correctly when |
| // SetRtpSendParameters is called. |
| TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) { |
| webrtc::test::ScopedFieldTrials override_field_trials( |
| "WebRTC-Audio-SendSideBwe/Enabled/"); |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(send_parameters); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); |
| // Expect empty on parameters.encodings[0].max_bitrate_bps; |
| EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps); |
| |
| constexpr int kMaxBitrateBps = 6000; |
| rtp_parameters.encodings[0].max_bitrate_bps = |
| rtc::Optional<int>(kMaxBitrateBps); |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters)); |
| |
| const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps; |
| EXPECT_EQ(max_bitrate, kMaxBitrateBps); |
| } |
| |
| // Test that GetRtpReceiveParameters returns the currently configured codecs. |
| TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpReceiveParameters(kSsrcX); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); |
| EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); |
| } |
| |
| // Test that GetRtpReceiveParameters returns an SSRC. |
| TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) { |
| EXPECT_TRUE(SetupRecvStream()); |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpReceiveParameters(kSsrcX); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc); |
| } |
| |
| // Test that if we set/get parameters multiple times, we get the same results. |
| TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| webrtc::RtpParameters initial_params = |
| channel_->GetRtpReceiveParameters(kSsrcX); |
| |
| // We should be able to set the params we just got. |
| EXPECT_TRUE(channel_->SetRtpReceiveParameters(kSsrcX, initial_params)); |
| |
| // ... And this shouldn't change the params returned by |
| // GetRtpReceiveParameters. |
| webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrcX); |
| EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(kSsrcX)); |
| } |
| |
| // Test that GetRtpReceiveParameters returns parameters correctly when SSRCs |
| // aren't signaled. It should return an empty "RtpEncodingParameters" when |
| // configured to receive an unsignaled stream and no packets have been received |
| // yet, and start returning the SSRC once a packet has been received. |
| TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) { |
| ASSERT_TRUE(SetupChannel()); |
| // Call necessary methods to configure receiving a default stream as |
| // soon as it arrives. |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| // Call GetRtpReceiveParameters before configured to receive an unsignaled |
| // stream. Should return nothing. |
| EXPECT_EQ(webrtc::RtpParameters(), channel_->GetRtpReceiveParameters(0)); |
| |
| // Set a sink for an unsignaled stream. |
| std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink()); |
| // Value of "0" means "unsignaled stream". |
| channel_->SetRawAudioSink(0, std::move(fake_sink)); |
| |
| // Call GetRtpReceiveParameters before the SSRC is known. Value of "0" |
| // in this method means "unsignaled stream". |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpReceiveParameters(0); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); |
| |
| // Receive PCMU packet (SSRC=1). |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| // The |ssrc| member should still be unset. |
| rtp_parameters = channel_->GetRtpReceiveParameters(0); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); |
| } |
| |
| // Test that we apply codecs properly. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[0].bitrate = 22000; |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000); |
| EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that WebRtcVoiceEngine reconfigures, rather than recreates its |
| // AudioSendStream. |
| TEST_F(WebRtcVoiceEngineTestFake, DontRecreateSendStream) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[0].bitrate = 48000; |
| const int initial_num = call_.GetNumCreatedSendStreams(); |
| SetSendParameters(parameters); |
| EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); |
| // Calling SetSendCodec again with same codec which is already set. |
| // In this case media channel shouldn't send codec to VoE. |
| SetSendParameters(parameters); |
| EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); |
| } |
| |
| // TODO(ossu): Revisit if these tests need to be here, now that these kinds of |
| // tests should be available in AudioEncoderOpusTest. |
| |
| // Test that if clockrate is not 48000 for opus, we fail. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].clockrate = 50000; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channels=0 for opus, we fail. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 0; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channels=0 for opus, we fail. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channel is 1 for opus and there's no stereo, we fail. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channel is 1 for opus and stereo=0, we fail. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| parameters.codecs[0].params["stereo"] = "0"; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channel is 1 for opus and stereo=1, we fail. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that with bitrate=0 and no stereo, bitrate is 32000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 32000); |
| } |
| |
| // Test that with bitrate=0 and stereo=0, bitrate is 32000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "0"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 32000); |
| } |
| |
| // Test that with bitrate=invalid and stereo=0, bitrate is 32000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["stereo"] = "0"; |
| // bitrate that's out of the range between 6000 and 510000 will be clamped. |
| parameters.codecs[0].bitrate = 5999; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 6000); |
| |
| parameters.codecs[0].bitrate = 510001; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 510000); |
| } |
| |
| // Test that with bitrate=0 and stereo=1, bitrate is 64000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 64000); |
| } |
| |
| // Test that with bitrate=invalid and stereo=1, bitrate is 64000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["stereo"] = "1"; |
| // bitrate that's out of the range between 6000 and 510000 will be clamped. |
| parameters.codecs[0].bitrate = 5999; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 6000); |
| |
| parameters.codecs[0].bitrate = 510001; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 510000); |
| } |
| |
| // Test that with bitrate=N and stereo unset, bitrate is N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 96000; |
| SetSendParameters(parameters); |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_EQ(96000, spec.target_bitrate_bps); |
| EXPECT_EQ("opus", spec.format.name); |
| EXPECT_EQ(2, spec.format.num_channels); |
| EXPECT_EQ(48000, spec.format.clockrate_hz); |
| } |
| |
| // Test that with bitrate=N and stereo=0, bitrate is N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "0"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| // Test that with bitrate=N and without any parameters, bitrate is N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| // Test that with bitrate=N and stereo=1, bitrate is N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "1"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithBitrates) { |
| SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", |
| 200000); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithHighMaxBitrate) { |
| SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, |
| SetSendCodecsWithoutBitratesUsesCorrectDefaults) { |
| SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) { |
| SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, |
| SetMaxSendBandwidthForAudioDoesntAffectBwe) { |
| SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", |
| 200000); |
| send_parameters_.max_bandwidth_bps = 100000; |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(100000, call_.GetConfig().bitrate_config.min_bitrate_bps) |
| << "Setting max bitrate should keep previous min bitrate."; |
| EXPECT_EQ(-1, call_.GetConfig().bitrate_config.start_bitrate_bps) |
| << "Setting max bitrate should not reset start bitrate."; |
| EXPECT_EQ(200000, call_.GetConfig().bitrate_config.max_bitrate_bps); |
| } |
| |
| // Test that we can enable NACK with opus as caller. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| SetSendParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that we can enable NACK with opus as callee. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| SetSendParameters(parameters); |
| // NACK should be enabled even with no send stream. |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that we can enable NACK on receive streams. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| SetSendParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that we can disable NACK. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNack) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| SetSendParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| |
| parameters.codecs.clear(); |
| parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(parameters); |
| EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that we can disable NACK on receive streams. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| SetSendParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| |
| parameters.codecs.clear(); |
| parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(parameters); |
| EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that NACK is enabled on a new receive stream. |
| TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| SetSendParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| EXPECT_TRUE(AddRecvStream(kSsrcZ)); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcZ).rtp.nack.rtp_history_ms); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(send_parameters.codecs[0].feedback_params.params().empty()); |
| SetSendParameters(send_parameters); |
| |
| cricket::AudioRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(kIsacCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr); |
| EXPECT_FALSE( |
| call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc); |
| |
| send_parameters.codecs = engine_->send_codecs(); |
| SetSendParameters(send_parameters); |
| ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr); |
| EXPECT_TRUE( |
| call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc); |
| } |
| |
| // Test that we can switch back and forth between Opus and ISAC with CN. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| cricket::AudioSendParameters opus_parameters; |
| opus_parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(opus_parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); |
| } |
| |
| cricket::AudioSendParameters isac_parameters; |
| isac_parameters.codecs.push_back(kIsacCodec); |
| isac_parameters.codecs.push_back(kCn16000Codec); |
| isac_parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(isac_parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| } |
| |
| SetSendParameters(opus_parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); |
| } |
| } |
| |
| // Test that we handle various ways of specifying bitrate. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); // bitrate == 32000 |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| EXPECT_EQ(32000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| EXPECT_EQ(32000, spec.target_bitrate_bps); |
| } |
| parameters.codecs[0].bitrate = 28000; // bitrate == 28000 |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| EXPECT_EQ(28000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0] = kPcmuCodec; // bitrate == 64000 |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(0, spec.payload_type); |
| EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str()); |
| EXPECT_EQ(64000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(0, spec.payload_type); |
| EXPECT_STREQ("PCMU", spec.format.name.c_str()); |
| EXPECT_EQ(64000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0] = kOpusCodec; |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_STREQ("opus", spec.format.name.c_str()); |
| EXPECT_EQ(32000, spec.target_bitrate_bps); |
| } |
| } |
| |
| // Test that we fail if no codecs are specified. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that we can set send codecs even with telephone-event codec as the first |
| // one on the list. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSendParameters(parameters); |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that payload type range is limited for telephone-event codec. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].id = 0; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSendParameters(parameters); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = 128; // DTMF |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = 127; |
| SetSendParameters(parameters); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = -1; // DTMF |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we can set send codecs even with CN codec as the first |
| // one on the list. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // wideband CN |
| parameters.codecs[1].id = 96; |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(98, send_codec_spec.cng_payload_type); |
| } |
| |
| // Test that we set VAD and DTMF types correctly as caller. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| // TODO(juberti): cn 32000 |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we set VAD and DTMF types correctly as callee. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| // TODO(juberti): cn 32000 |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| SetSendParameters(parameters); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we only apply VAD if we have a CN codec that matches the |
| // send codec clockrate. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| // Set ISAC(16K) and CN(16K). VAD should be activated. |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = 97; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| } |
| // Set PCMU(8K) and CN(16K). VAD should not be activated. |
| parameters.codecs[0] = kPcmuCodec; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type); |
| } |
| // Set PCMU(8K) and CN(8K). VAD should be activated. |
| parameters.codecs[1] = kCn8000Codec; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1, send_codec_spec.format.num_channels); |
| EXPECT_EQ(13, send_codec_spec.cng_payload_type); |
| } |
| // Set ISAC(16K) and CN(8K). VAD should not be activated. |
| parameters.codecs[0] = kIsacCodec; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type); |
| } |
| } |
| |
| // Test that we perform case-insensitive matching of codec names. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs[0].name = "iSaC"; |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| } |
| |
| class WebRtcVoiceEngineWithSendSideBweTest : public WebRtcVoiceEngineTestFake { |
| public: |
| WebRtcVoiceEngineWithSendSideBweTest() |
| : WebRtcVoiceEngineTestFake("WebRTC-Audio-SendSideBwe/Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVoiceEngineWithSendSideBweTest, |
| SupportsTransportSequenceNumberHeaderExtension) { |
| cricket::RtpCapabilities capabilities = engine_->GetCapabilities(); |
| ASSERT_FALSE(capabilities.header_extensions.empty()); |
| for (const webrtc::RtpExtension& extension : capabilities.header_extensions) { |
| if (extension.uri == webrtc::RtpExtension::kTransportSequenceNumberUri) { |
| EXPECT_EQ(webrtc::RtpExtension::kTransportSequenceNumberDefaultId, |
| extension.id); |
| return; |
| } |
| } |
| FAIL() << "Transport sequence number extension not in header-extension list."; |
| } |
| |
| // Test support for audio level header extension. |
| TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); |
| } |
| TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); |
| } |
| |
| // Test support for transport sequence number header extension. |
| TEST_F(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions( |
| webrtc::RtpExtension::kTransportSequenceNumberUri); |
| } |
| TEST_F(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions( |
| webrtc::RtpExtension::kTransportSequenceNumberUri); |
| } |
| |
| // Test that we can create a channel and start sending on it. |
| TEST_F(WebRtcVoiceEngineTestFake, Send) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSendParameters(send_parameters_); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that a channel will send if and only if it has a source and is enabled |
| // for sending. |
| TEST_F(WebRtcVoiceEngineTestFake, SendStateWithAndWithoutSource) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSendParameters(send_parameters_); |
| SetAudioSend(kSsrcX, true, nullptr); |
| SetSend(true); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| SetAudioSend(kSsrcX, true, &fake_source_); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| SetAudioSend(kSsrcX, true, nullptr); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that a channel is muted/unmuted. |
| TEST_F(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSendParameters(send_parameters_); |
| EXPECT_FALSE(GetSendStream(kSsrcX).muted()); |
| SetAudioSend(kSsrcX, true, nullptr); |
| EXPECT_FALSE(GetSendStream(kSsrcX).muted()); |
| SetAudioSend(kSsrcX, false, nullptr); |
| EXPECT_TRUE(GetSendStream(kSsrcX).muted()); |
| } |
| |
| // Test that SetSendParameters() does not alter a stream's send state. |
| TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Turn on sending. |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Changing RTP header extensions will recreate the AudioSendStream. |
| send_parameters_.extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Turn off sending. |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Changing RTP header extensions will recreate the AudioSendStream. |
| send_parameters_.extensions.clear(); |
| SetSendParameters(send_parameters_); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that we can create a channel and start playing out on it. |
| TEST_F(WebRtcVoiceEngineTestFake, Playout) { |
| EXPECT_TRUE(SetupRecvStream()); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we can add and remove send streams. |
| TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Set the global state for sending. |
| SetSend(true); |
| |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| SetAudioSend(ssrc, true, &fake_source_); |
| // Verify that we are in a sending state for all the created streams. |
| EXPECT_TRUE(GetSendStream(ssrc).IsSending()); |
| } |
| EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size()); |
| |
| // Delete the send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(channel_->RemoveSendStream(ssrc)); |
| EXPECT_FALSE(call_.GetAudioSendStream(ssrc)); |
| EXPECT_FALSE(channel_->RemoveSendStream(ssrc)); |
| } |
| EXPECT_EQ(0u, call_.GetAudioSendStreams().size()); |
| } |
| |
| // Test SetSendCodecs correctly configure the codecs in all send streams. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Create send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| |
| cricket::AudioSendParameters parameters; |
| // Set ISAC(16K) and CN(16K). VAD should be activated. |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = 97; |
| SetSendParameters(parameters); |
| |
| // Verify ISAC and VAD are corrected configured on all send channels. |
| for (uint32_t ssrc : kSsrcs4) { |
| ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); |
| const auto& send_codec_spec = |
| *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| } |
| |
| // Change to PCMU(8K) and CN(16K). |
| parameters.codecs[0] = kPcmuCodec; |
| SetSendParameters(parameters); |
| for (uint32_t ssrc : kSsrcs4) { |
| ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); |
| const auto& send_codec_spec = |
| *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type); |
| } |
| } |
| |
| // Test we can SetSend on all send streams correctly. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Create the send channels and they should be a "not sending" date. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| SetAudioSend(ssrc, true, &fake_source_); |
| EXPECT_FALSE(GetSendStream(ssrc).IsSending()); |
| } |
| |
| // Set the global state for starting sending. |
| SetSend(true); |
| for (uint32_t ssrc : kSsrcs4) { |
| // Verify that we are in a sending state for all the send streams. |
| EXPECT_TRUE(GetSendStream(ssrc).IsSending()); |
| } |
| |
| // Set the global state for stopping sending. |
| SetSend(false); |
| for (uint32_t ssrc : kSsrcs4) { |
| // Verify that we are in a stop state for all the send streams. |
| EXPECT_FALSE(GetSendStream(ssrc).IsSending()); |
| } |
| } |
| |
| // Test we can set the correct statistics on all send streams. |
| TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Create send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| |
| // Create a receive stream to check that none of the send streams end up in |
| // the receive stream stats. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| |
| // We need send codec to be set to get all stats. |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| SetAudioSendStreamStats(); |
| |
| // Check stats for the added streams. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| |
| // We have added 4 send streams. We should see empty stats for all. |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); |
| for (const auto& sender : info.senders) { |
| VerifyVoiceSenderInfo(sender, false); |
| } |
| VerifyVoiceSendRecvCodecs(info); |
| |
| // We have added one receive stream. We should see empty stats. |
| EXPECT_EQ(info.receivers.size(), 1u); |
| EXPECT_EQ(info.receivers[0].ssrc(), 0); |
| } |
| |
| // Remove the kSsrcY stream. No receiver stats. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); |
| EXPECT_EQ(0u, info.receivers.size()); |
| } |
| |
| // Deliver a new packet - a default receive stream should be created and we |
| // should see stats again. |
| { |
| cricket::VoiceMediaInfo info; |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| SetAudioReceiveStreamStats(); |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); |
| EXPECT_EQ(1u, info.receivers.size()); |
| VerifyVoiceReceiverInfo(info.receivers[0]); |
| VerifyVoiceSendRecvCodecs(info); |
| } |
| } |
| |
| // Test that we can add and remove receive streams, and do proper send/playout. |
| // We can receive on multiple streams while sending one stream. |
| TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Start playout without a receive stream. |
| SetSendParameters(send_parameters_); |
| channel_->SetPlayout(true); |
| |
| // Adding another stream should enable playout on the new stream only. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Make sure only the new stream is played out. |
| EXPECT_TRUE(GetRecvStream(kSsrcY).started()); |
| |
| // Adding yet another stream should have stream 2 and 3 enabled for playout. |
| EXPECT_TRUE(AddRecvStream(kSsrcZ)); |
| EXPECT_TRUE(GetRecvStream(kSsrcY).started()); |
| EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); |
| |
| // Stop sending. |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Stop playout. |
| channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrcY).started()); |
| EXPECT_FALSE(GetRecvStream(kSsrcZ).started()); |
| |
| // Restart playout and make sure recv streams are played out. |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcY).started()); |
| EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); |
| |
| // Now remove the recv streams. |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcZ)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); |
| } |
| |
| // Test that we can create a channel configured for Codian bridges, |
| // and start sending on it. |
| TEST_F(WebRtcVoiceEngineTestFake, CodianSend) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.adjust_agc_delta = rtc::Optional<int>(-10); |
| EXPECT_CALL(apm_gc_, |
| set_target_level_dbfs(11)).Times(2).WillRepeatedly(Return(0)); |
| SetSendParameters(send_parameters_); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(adm_, |
| BuiltInAGCIsAvailable()).Times(2).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, SetAGC(true)).Times(2).WillRepeatedly(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).Times(2).WillOnce(Return(0)); |
| send_parameters_.options.tx_agc_target_dbov = rtc::Optional<uint16_t>(3); |
| send_parameters_.options.tx_agc_digital_compression_gain = |
| rtc::Optional<uint16_t>(9); |
| send_parameters_.options.tx_agc_limiter = rtc::Optional<bool>(true); |
| send_parameters_.options.auto_gain_control = rtc::Optional<bool>(true); |
| EXPECT_CALL(apm_gc_, set_target_level_dbfs(3)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, set_compression_gain_db(9)).WillRepeatedly(Return(0)); |
| EXPECT_CALL(apm_gc_, enable_limiter(true)).WillRepeatedly(Return(0)); |
| SetSendParameters(send_parameters_); |
| |
| // Check interaction with adjust_agc_delta. Both should be respected, for |
| // backwards compatibility. |
| send_parameters_.options.adjust_agc_delta = rtc::Optional<int>(-10); |
| EXPECT_CALL(apm_gc_, set_target_level_dbfs(13)).WillOnce(Return(0)); |
| SetSendParameters(send_parameters_); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(adm_, SetRecordingSampleRate(48000)).WillOnce(Return(0)); |
| EXPECT_CALL(adm_, SetPlayoutSampleRate(44100)).WillOnce(Return(0)); |
| send_parameters_.options.recording_sample_rate = |
| rtc::Optional<uint32_t>(48000); |
| send_parameters_.options.playout_sample_rate = rtc::Optional<uint32_t>(44100); |
| SetSendParameters(send_parameters_); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = rtc::Optional<bool>(true); |
| send_parameters_.options.audio_network_adaptor_config = |
| rtc::Optional<std::string>("1234"); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = rtc::Optional<bool>(true); |
| send_parameters_.options.audio_network_adaptor_config = |
| rtc::Optional<std::string>("1234"); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| cricket::AudioOptions options; |
| options.audio_network_adaptor = rtc::Optional<bool>(false); |
| SetAudioSend(kSsrcX, true, nullptr, &options); |
| EXPECT_EQ(rtc::Optional<std::string>(), GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = rtc::Optional<bool>(true); |
| send_parameters_.options.audio_network_adaptor_config = |
| rtc::Optional<std::string>("1234"); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| const int initial_num = call_.GetNumCreatedSendStreams(); |
| cricket::AudioOptions options; |
| options.audio_network_adaptor = rtc::Optional<bool>(); |
| // Unvalued |options.audio_network_adaptor|.should not reset audio network |
| // adaptor. |
| SetAudioSend(kSsrcX, true, nullptr, &options); |
| // AudioSendStream not expected to be recreated. |
| EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| class WebRtcVoiceEngineWithSendSideBweWithOverheadTest |
| : public WebRtcVoiceEngineTestFake { |
| public: |
| WebRtcVoiceEngineWithSendSideBweWithOverheadTest() |
| : WebRtcVoiceEngineTestFake( |
| "WebRTC-Audio-SendSideBwe/Enabled/WebRTC-SendSideBwe-WithOverhead/" |
| "Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest, MinAndMaxBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(parameters); |
| const int initial_num = call_.GetNumCreatedSendStreams(); |
| EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); |
| |
| // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12) |
| constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12; |
| constexpr int kOpusMaxPtimeMs = WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60; |
| constexpr int kMinOverheadBps = |
| kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs; |
| |
| constexpr int kOpusMinBitrateBps = 6000; |
| EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadBps, |
| GetSendStreamConfig(kSsrcX).min_bitrate_bps); |
| constexpr int kOpusBitrateFbBps = 32000; |
| EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadBps, |
| GetSendStreamConfig(kSsrcX).max_bitrate_bps); |
| |
| parameters.options.audio_network_adaptor = rtc::Optional<bool>(true); |
| parameters.options.audio_network_adaptor_config = |
| rtc::Optional<std::string>("1234"); |
| SetSendParameters(parameters); |
| |
| constexpr int kMinOverheadWithAnaBps = |
| kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs; |
| |
| EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadWithAnaBps, |
| GetSendStreamConfig(kSsrcX).min_bitrate_bps); |
| |
| EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadWithAnaBps, |
| GetSendStreamConfig(kSsrcX).max_bitrate_bps); |
| } |
| |
| // This test is similar to |
| // WebRtcVoiceEngineTestFake.SetRtpSendParameterUpdatesMaxBitrate but with an |
| // additional field trial. |
| TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest, |
| SetRtpSendParameterUpdatesMaxBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(send_parameters); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); |
| // Expect empty on parameters.encodings[0].max_bitrate_bps; |
| EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps); |
| |
| constexpr int kMaxBitrateBps = 6000; |
| rtp_parameters.encodings[0].max_bitrate_bps = |
| rtc::Optional<int>(kMaxBitrateBps); |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters)); |
| |
| const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps; |
| #if WEBRTC_OPUS_SUPPORT_120MS_PTIME |
| constexpr int kMinOverhead = 3333; |
| #else |
| constexpr int kMinOverhead = 6666; |
| #endif |
| EXPECT_EQ(max_bitrate, kMaxBitrateBps + kMinOverhead); |
| } |
| |
| // Test that we can set the outgoing SSRC properly. |
| // SSRC is set in SetupSendStream() by calling AddSendStream. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, GetStats) { |
| // Setup. We need send codec to be set to get all stats. |
| EXPECT_TRUE(SetupSendStream()); |
| // SetupSendStream adds a send stream with kSsrcX, so the receive |
| // stream has to use a different SSRC. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| SetAudioSendStreamStats(); |
| |
| // Check stats for the added streams. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| |
| // We have added one send stream. We should see the stats we've set. |
| EXPECT_EQ(1u, info.senders.size()); |
| VerifyVoiceSenderInfo(info.senders[0], false); |
| // We have added one receive stream. We should see empty stats. |
| EXPECT_EQ(info.receivers.size(), 1u); |
| EXPECT_EQ(info.receivers[0].ssrc(), 0); |
| } |
| |
| // Start sending - this affects some reported stats. |
| { |
| cricket::VoiceMediaInfo info; |
| SetSend(true); |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| VerifyVoiceSenderInfo(info.senders[0], true); |
| VerifyVoiceSendRecvCodecs(info); |
| } |
| |
| // Remove the kSsrcY stream. No receiver stats. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| EXPECT_EQ(1u, info.senders.size()); |
| EXPECT_EQ(0u, info.receivers.size()); |
| } |
| |
| // Deliver a new packet - a default receive stream should be created and we |
| // should see stats again. |
| { |
| cricket::VoiceMediaInfo info; |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| SetAudioReceiveStreamStats(); |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| EXPECT_EQ(1u, info.senders.size()); |
| EXPECT_EQ(1u, info.receivers.size()); |
| VerifyVoiceReceiverInfo(info.receivers[0]); |
| VerifyVoiceSendRecvCodecs(info); |
| } |
| } |
| |
| // Test that we can set the outgoing SSRC properly with multiple streams. |
| // SSRC is set in SetupSendStream() by calling AddSendStream. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); |
| } |
| |
| // Test that the local SSRC is the same on sending and receiving channels if the |
| // receive channel is created before the send channel. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); |
| EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); |
| } |
| |
| // Test that we can properly receive packets. |
| TEST_F(WebRtcVoiceEngineTestFake, Recv) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(1)); |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame, |
| sizeof(kPcmuFrame))); |
| } |
| |
| // Test that we can properly receive packets on multiple streams. |
| TEST_F(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) { |
| EXPECT_TRUE(SetupChannel()); |
| const uint32_t ssrc1 = 1; |
| const uint32_t ssrc2 = 2; |
| const uint32_t ssrc3 = 3; |
| EXPECT_TRUE(AddRecvStream(ssrc1)); |
| EXPECT_TRUE(AddRecvStream(ssrc2)); |
| EXPECT_TRUE(AddRecvStream(ssrc3)); |
| // Create packets with the right SSRCs. |
| unsigned char packets[4][sizeof(kPcmuFrame)]; |
| for (size_t i = 0; i < arraysize(packets); ++i) { |
| memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame)); |
| rtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i)); |
| } |
| |
| const cricket::FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1); |
| const cricket::FakeAudioReceiveStream& s2 = GetRecvStream(ssrc2); |
| const cricket::FakeAudioReceiveStream& s3 = GetRecvStream(ssrc3); |
| |
| EXPECT_EQ(s1.received_packets(), 0); |
| EXPECT_EQ(s2.received_packets(), 0); |
| EXPECT_EQ(s3.received_packets(), 0); |
| |
| DeliverPacket(packets[0], sizeof(packets[0])); |
| EXPECT_EQ(s1.received_packets(), 0); |
| EXPECT_EQ(s2.received_packets(), 0); |
| EXPECT_EQ(s3.received_packets(), 0); |
| |
| DeliverPacket(packets[1], sizeof(packets[1])); |
| EXPECT_EQ(s1.received_packets(), 1); |
| EXPECT_TRUE(s1.VerifyLastPacket(packets[1], sizeof(packets[1]))); |
| EXPECT_EQ(s2.received_packets(), 0); |
| EXPECT_EQ(s3.received_packets(), 0); |
| |
| DeliverPacket(packets[2], sizeof(packets[2])); |
| EXPECT_EQ(s1.received_packets(), 1); |
| EXPECT_EQ(s2.received_packets(), 1); |
| EXPECT_TRUE(s2.VerifyLastPacket(packets[2], sizeof(packets[2]))); |
| EXPECT_EQ(s3.received_packets(), 0); |
| |
| DeliverPacket(packets[3], sizeof(packets[3])); |
| EXPECT_EQ(s1.received_packets(), 1); |
| EXPECT_EQ(s2.received_packets(), 1); |
| EXPECT_EQ(s3.received_packets(), 1); |
| EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3]))); |
| |
| EXPECT_TRUE(channel_->RemoveRecvStream(ssrc3)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(ssrc2)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(ssrc1)); |
| } |
| |
| // Test that receiving on an unsignaled stream works (a stream is created). |
| TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaled) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_EQ(0, call_.GetAudioReceiveStreams().size()); |
| |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE(GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, |
| sizeof(kPcmuFrame))); |
| } |
| |
| // Test that receiving N unsignaled stream works (streams will be created), and |
| // that packets are forwarded to them all. |
| TEST_F(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) { |
| EXPECT_TRUE(SetupChannel()); |
| unsigned char packet[sizeof(kPcmuFrame)]; |
| memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| // Note that SSRC = 0 is not supported. |
| for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) { |
| rtc::SetBE32(&packet[8], ssrc); |
| DeliverPacket(packet, sizeof(packet)); |
| |
| // Verify we have one new stream for each loop iteration. |
| EXPECT_EQ(ssrc, call_.GetAudioReceiveStreams().size()); |
| EXPECT_EQ(1, GetRecvStream(ssrc).received_packets()); |
| EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); |
| } |
| |
| // Sending on the same SSRCs again should not create new streams. |
| for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) { |
| rtc::SetBE32(&packet[8], ssrc); |
| DeliverPacket(packet, sizeof(packet)); |
| |
| EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size()); |
| EXPECT_EQ(2, GetRecvStream(ssrc).received_packets()); |
| EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); |
| } |
| |
| // Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced. |
| constexpr uint32_t kAnotherSsrc = 667; |
| rtc::SetBE32(&packet[8], kAnotherSsrc); |
| DeliverPacket(packet, sizeof(packet)); |
| |
| const auto& streams = call_.GetAudioReceiveStreams(); |
| EXPECT_EQ(kMaxUnsignaledRecvStreams, streams.size()); |
| size_t i = 0; |
| for (uint32_t ssrc = 2; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc, ++i) { |
| EXPECT_EQ(ssrc, streams[i]->GetConfig().rtp.remote_ssrc); |
| EXPECT_EQ(2, streams[i]->received_packets()); |
| } |
| EXPECT_EQ(kAnotherSsrc, streams[i]->GetConfig().rtp.remote_ssrc); |
| EXPECT_EQ(1, streams[i]->received_packets()); |
| // Sanity check that we've checked all streams. |
| EXPECT_EQ(kMaxUnsignaledRecvStreams, (i + 1)); |
| } |
| |
| // Test that a default channel is created even after a signaled stream has been |
| // added, and that this stream will get any packets for unknown SSRCs. |
| TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) { |
| EXPECT_TRUE(SetupChannel()); |
| unsigned char packet[sizeof(kPcmuFrame)]; |
| memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| // Add a known stream, send packet and verify we got it. |
| const uint32_t signaled_ssrc = 1; |
| rtc::SetBE32(&packet[8], signaled_ssrc); |
| EXPECT_TRUE(AddRecvStream(signaled_ssrc)); |
| DeliverPacket(packet, sizeof(packet)); |
| EXPECT_TRUE(GetRecvStream(signaled_ssrc).VerifyLastPacket( |
| packet, sizeof(packet))); |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| |
| // Note that the first unknown SSRC cannot be 0, because we only support |
| // creating receive streams for SSRC!=0. |
| const uint32_t unsignaled_ssrc = 7011; |
| rtc::SetBE32(&packet[8], unsignaled_ssrc); |
| DeliverPacket(packet, sizeof(packet)); |
| EXPECT_TRUE(GetRecvStream(unsignaled_ssrc).VerifyLastPacket( |
| packet, sizeof(packet))); |
| EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); |
| |
| DeliverPacket(packet, sizeof(packet)); |
| EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets()); |
| |
| rtc::SetBE32(&packet[8], signaled_ssrc); |
| DeliverPacket(packet, sizeof(packet)); |
| EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets()); |
| EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); |
| } |
| |
| // Two tests to verify that adding a receive stream with the same SSRC as a |
| // previously added unsignaled stream will only recreate underlying stream |
| // objects if the stream parameters have changed. |
| TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) { |
| EXPECT_TRUE(SetupChannel()); |
| |
| // Spawn unsignaled stream with SSRC=1. |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame, |
| sizeof(kPcmuFrame))); |
| |
| // Verify that the underlying stream object in Call is not recreated when a |
| // stream with SSRC=1 is added. |
| const auto& streams = call_.GetAudioReceiveStreams(); |
| EXPECT_EQ(1, streams.size()); |
| int audio_receive_stream_id = streams.front()->id(); |
| EXPECT_TRUE(AddRecvStream(1)); |
| EXPECT_EQ(1, streams.size()); |
| EXPECT_EQ(audio_receive_stream_id, streams.front()->id()); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Recreate) { |
| EXPECT_TRUE(SetupChannel()); |
| |
| // Spawn unsignaled stream with SSRC=1. |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame, |
| sizeof(kPcmuFrame))); |
| |
| // Verify that the underlying stream object in Call *is* recreated when a |
| // stream with SSRC=1 is added, and which has changed stream parameters. |
| const auto& streams = call_.GetAudioReceiveStreams(); |
| EXPECT_EQ(1, streams.size()); |
| int audio_receive_stream_id = streams.front()->id(); |
| cricket::StreamParams stream_params; |
| stream_params.ssrcs.push_back(1); |
| stream_params.sync_label = "sync_label"; |
| EXPECT_TRUE(channel_->AddRecvStream(stream_params)); |
| EXPECT_EQ(1, streams.size()); |
| EXPECT_NE(audio_receive_stream_id, streams.front()->id()); |
| } |
| |
| // Test that we properly handle failures to add a receive stream. |
| TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamFail) { |
| EXPECT_TRUE(SetupChannel()); |
| voe_.set_fail_create_channel(true); |
| EXPECT_FALSE(AddRecvStream(2)); |
| } |
| |
| // Test that we properly handle failures to add a send stream. |
| TEST_F(WebRtcVoiceEngineTestFake, AddSendStreamFail) { |
| EXPECT_TRUE(SetupChannel()); |
| voe_.set_fail_create_channel(true); |
| EXPECT_FALSE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2))); |
| } |
| |
| // Test that AddRecvStream creates new stream. |
| TEST_F(WebRtcVoiceEngineTestFake, AddRecvStream) { |
| EXPECT_TRUE(SetupRecvStream()); |
| int channel_num = voe_.GetLastChannel(); |
| EXPECT_TRUE(AddRecvStream(1)); |
| EXPECT_NE(channel_num, voe_.GetLastChannel()); |
| } |
| |
| // Test that after adding a recv stream, we do not decode more codecs than |
| // those previously passed into SetRecvCodecs. |
| TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}}))); |
| } |
| |
| // Test that we properly clean up any streams that were added, even if |
| // not explicitly removed. |
| TEST_F(WebRtcVoiceEngineTestFake, StreamCleanup) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(AddRecvStream(1)); |
| EXPECT_TRUE(AddRecvStream(2)); |
| EXPECT_EQ(3, voe_.GetNumChannels()); // default channel + 2 added |
| delete channel_; |
| channel_ = NULL; |
| EXPECT_EQ(0, voe_.GetNumChannels()); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithZeroSsrc) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_FALSE(AddRecvStream(0)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, TestNoLeakingWhenAddRecvStreamFail) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(1)); |
| // Manually delete channel to simulate a failure. |
| int channel = voe_.GetLastChannel(); |
| EXPECT_EQ(0, voe_.DeleteChannel(channel)); |
| // Add recv stream 2 should work. |
| EXPECT_TRUE(AddRecvStream(2)); |
| int new_channel = voe_.GetLastChannel(); |
| EXPECT_NE(channel, new_channel); |
| // The last created channel is deleted too. |
| EXPECT_EQ(0, voe_.DeleteChannel(new_channel)); |
| } |
| |
| // Test the InsertDtmf on default send stream as caller. |
| TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) { |
| TestInsertDtmf(0, true, kTelephoneEventCodec1); |
| } |
| |
| // Test the InsertDtmf on default send stream as callee |
| TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) { |
| TestInsertDtmf(0, false, kTelephoneEventCodec2); |
| } |
| |
| // Test the InsertDtmf on specified send stream as caller. |
| TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) { |
| TestInsertDtmf(kSsrcX, true, kTelephoneEventCodec2); |
| } |
| |
| // Test the InsertDtmf on specified send stream as callee. |
| TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) { |
| TestInsertDtmf(kSsrcX, false, kTelephoneEventCodec1); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(adm_, |
| BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, |
| BuiltInAGCIsAvailable()).Times(4).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, |
| BuiltInNSIsAvailable()).Times(2).WillRepeatedly(Return(false)); |
| |
| EXPECT_EQ(50, voe_.GetNetEqCapacity()); |
| EXPECT_FALSE(voe_.GetNetEqFastAccelerate()); |
| |
| // Nothing set in AudioOptions, so everything should be as default. |
| send_parameters_.options = cricket::AudioOptions(); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(IsHighPassFilterEnabled()); |
| EXPECT_EQ(50, voe_.GetNetEqCapacity()); |
| EXPECT_FALSE(voe_.GetNetEqFastAccelerate()); |
| |
| // Turn echo cancellation off |
| EXPECT_CALL(apm_ec_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(false)).WillOnce(Return(0)); |
| send_parameters_.options.echo_cancellation = rtc::Optional<bool>(false); |
| SetSendParameters(send_parameters_); |
| |
| // Turn echo cancellation back on, with settings, and make sure |
| // nothing else changed. |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| send_parameters_.options.echo_cancellation = rtc::Optional<bool>(true); |
| SetSendParameters(send_parameters_); |
| |
| // Turn on delay agnostic aec and make sure nothing change w.r.t. echo |
| // control. |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(true); |
| SetSendParameters(send_parameters_); |
| |
| // Turn off echo cancellation and delay agnostic aec. |
| EXPECT_CALL(apm_ec_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(false)).WillOnce(Return(0)); |
| send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(false); |
| send_parameters_.options.extended_filter_aec = rtc::Optional<bool>(false); |
| send_parameters_.options.echo_cancellation = rtc::Optional<bool>(false); |
| SetSendParameters(send_parameters_); |
| |
| // Turning delay agnostic aec back on should also turn on echo cancellation. |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(true); |
| SetSendParameters(send_parameters_); |
| |
| // Turn off AGC |
| EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); |
| send_parameters_.options.auto_gain_control = rtc::Optional<bool>(false); |
| SetSendParameters(send_parameters_); |
| |
| // Turn AGC back on |
| EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); |
| send_parameters_.options.auto_gain_control = rtc::Optional<bool>(true); |
| send_parameters_.options.adjust_agc_delta = rtc::Optional<int>(); |
| SetSendParameters(send_parameters_); |
| |
| // Turn off other options (and stereo swapping on). |
| EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_vd_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(true)); |
| send_parameters_.options.noise_suppression = rtc::Optional<bool>(false); |
| send_parameters_.options.highpass_filter = rtc::Optional<bool>(false); |
| send_parameters_.options.typing_detection = rtc::Optional<bool>(false); |
| send_parameters_.options.stereo_swapping = rtc::Optional<bool>(true); |
| SetSendParameters(send_parameters_); |
| EXPECT_FALSE(IsHighPassFilterEnabled()); |
| |
| // Set options again to ensure it has no impact. |
| EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_vd_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(true)); |
| SetSendParameters(send_parameters_); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(adm_, |
| BuiltInAECIsAvailable()).Times(8).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, |
| BuiltInAGCIsAvailable()).Times(8).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, |
| BuiltInNSIsAvailable()).Times(8).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, |
| RecordingIsInitialized()).Times(2).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, Recording()).Times(2).WillRepeatedly(Return(false)); |
| EXPECT_CALL(adm_, InitRecording()).Times(2).WillRepeatedly(Return(0)); |
| webrtc::AudioProcessing::Config apm_config; |
| EXPECT_CALL(*apm_, GetConfig()) |
| .Times(10) |
| .WillRepeatedly(ReturnPointee(&apm_config)); |
| EXPECT_CALL(*apm_, ApplyConfig(_)) |
| .Times(10) |
| .WillRepeatedly(SaveArg<0>(&apm_config)); |
| EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(10); |
| |
| std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1( |
| static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel( |
| &call_, cricket::MediaConfig(), cricket::AudioOptions()))); |
| std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel2( |
| static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel( |
| &call_, cricket::MediaConfig(), cricket::AudioOptions()))); |
| |
| // Have to add a stream to make SetSend work. |
| cricket::StreamParams stream1; |
| stream1.ssrcs.push_back(1); |
| channel1->AddSendStream(stream1); |
| cricket::StreamParams stream2; |
| stream2.ssrcs.push_back(2); |
| channel2->AddSendStream(stream2); |
| |
| // AEC and AGC and NS |
| cricket::AudioSendParameters parameters_options_all = send_parameters_; |
| parameters_options_all.options.echo_cancellation = rtc::Optional<bool>(true); |
| parameters_options_all.options.auto_gain_control = rtc::Optional<bool>(true); |
| parameters_options_all.options.noise_suppression = rtc::Optional<bool>(true); |
| EXPECT_CALL(adm_, SetAGC(true)).Times(2).WillRepeatedly(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).Times(2).WillRepeatedly(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).Times(2).WillRepeatedly(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).Times(2).WillRepeatedly(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(true)).Times(2).WillRepeatedly(Return(0)); |
| EXPECT_TRUE(channel1->SetSendParameters(parameters_options_all)); |
| EXPECT_EQ(parameters_options_all.options, channel1->options()); |
| EXPECT_TRUE(channel2->SetSendParameters(parameters_options_all)); |
| EXPECT_EQ(parameters_options_all.options, channel2->options()); |
| |
| // unset NS |
| cricket::AudioSendParameters parameters_options_no_ns = send_parameters_; |
| parameters_options_no_ns.options.noise_suppression = |
| rtc::Optional<bool>(false); |
| EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_TRUE(channel1->SetSendParameters(parameters_options_no_ns)); |
| cricket::AudioOptions expected_options = parameters_options_all.options; |
| expected_options.echo_cancellation = rtc::Optional<bool>(true); |
| expected_options.auto_gain_control = rtc::Optional<bool>(true); |
| expected_options.noise_suppression = rtc::Optional<bool>(false); |
| EXPECT_EQ(expected_options, channel1->options()); |
| |
| // unset AGC |
| cricket::AudioSendParameters parameters_options_no_agc = send_parameters_; |
| parameters_options_no_agc.options.auto_gain_control = |
| rtc::Optional<bool>(false); |
| EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc)); |
| expected_options.echo_cancellation = rtc::Optional<bool>(true); |
| expected_options.auto_gain_control = rtc::Optional<bool>(false); |
| expected_options.noise_suppression = rtc::Optional<bool>(true); |
| EXPECT_EQ(expected_options, channel2->options()); |
| |
| EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters_options_all)); |
| |
| EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); |
| channel1->SetSend(true); |
| |
| EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); |
| channel2->SetSend(true); |
| |
| // Make sure settings take effect while we are sending. |
| cricket::AudioSendParameters parameters_options_no_agc_nor_ns = |
| send_parameters_; |
| parameters_options_no_agc_nor_ns.options.auto_gain_control = |
| rtc::Optional<bool>(false); |
| parameters_options_no_agc_nor_ns.options.noise_suppression = |
| rtc::Optional<bool>(false); |
| EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); |
| EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc_nor_ns)); |
| expected_options.echo_cancellation = rtc::Optional<bool>(true); |
| expected_options.auto_gain_control = rtc::Optional<bool>(false); |
| expected_options.noise_suppression = rtc::Optional<bool>(false); |
| EXPECT_EQ(expected_options, channel2->options()); |
| } |
| |
| // This test verifies DSCP settings are properly applied on voice media channel. |
| TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::FakeNetworkInterface network_interface; |
| cricket::MediaConfig config; |
| std::unique_ptr<cricket::VoiceMediaChannel> channel; |
| |
| webrtc::AudioProcessing::Config apm_config; |
| EXPECT_CALL(*apm_, GetConfig()) |
| .Times(3) |
| .WillRepeatedly(ReturnPointee(&apm_config)); |
| EXPECT_CALL(*apm_, ApplyConfig(_)) |
| .Times(3) |
| .WillRepeatedly(SaveArg<0>(&apm_config)); |
| EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(3); |
| |
| channel.reset( |
| engine_->CreateChannel(&call_, config, cricket::AudioOptions())); |
| channel->SetInterface(&network_interface); |
| // Default value when DSCP is disabled should be DSCP_DEFAULT. |
| EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); |
| |
| config.enable_dscp = true; |
| channel.reset( |
| engine_->CreateChannel(&call_, config, cricket::AudioOptions())); |
| channel->SetInterface(&network_interface); |
| EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp()); |
| |
| // Verify that setting the option to false resets the |
| // DiffServCodePoint. |
| config.enable_dscp = false; |
| channel.reset( |
| engine_->CreateChannel(&call_, config, cricket::AudioOptions())); |
| channel->SetInterface(&network_interface); |
| // Default value when DSCP is disabled should be DSCP_DEFAULT. |
| EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); |
| |
| channel->SetInterface(nullptr); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelId) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::WebRtcVoiceMediaChannel* media_channel = |
| static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); |
| EXPECT_EQ(-1, media_channel->GetReceiveChannelId(0)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| int channel_id = voe_.GetLastChannel(); |
| EXPECT_EQ(channel_id, media_channel->GetReceiveChannelId(kSsrcX)); |
| EXPECT_EQ(-1, media_channel->GetReceiveChannelId(kSsrcY)); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| int channel_id2 = voe_.GetLastChannel(); |
| EXPECT_EQ(channel_id2, media_channel->GetReceiveChannelId(kSsrcY)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, TestGetSendChannelId) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::WebRtcVoiceMediaChannel* media_channel = |
| static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); |
| EXPECT_EQ(-1, media_channel->GetSendChannelId(0)); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| int channel_id = voe_.GetLastChannel(); |
| EXPECT_EQ(channel_id, media_channel->GetSendChannelId(kSsrcX)); |
| EXPECT_EQ(-1, media_channel->GetSendChannelId(kSsrcY)); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcY))); |
| int channel_id2 = voe_.GetLastChannel(); |
| EXPECT_EQ(channel_id2, media_channel->GetSendChannelId(kSsrcY)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolume) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_FALSE(channel_->SetOutputVolume(kSsrcY, 0.5)); |
| cricket::StreamParams stream; |
| stream.ssrcs.push_back(kSsrcY); |
| EXPECT_TRUE(channel_->AddRecvStream(stream)); |
| EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrcY).gain()); |
| EXPECT_TRUE(channel_->SetOutputVolume(kSsrcY, 3)); |
| EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcY).gain()); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) { |
| EXPECT_TRUE(SetupChannel()); |
| |
| // Spawn an unsignaled stream by sending a packet - gain should be 1. |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrc1).gain()); |
| |
| // Should remember the volume "2" which will be set on new unsignaled streams, |
| // and also set the gain to 2 on existing unsignaled streams. |
| EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 2)); |
| EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrc1).gain()); |
| |
| // Spawn an unsignaled stream by sending a packet - gain should be 2. |
| unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; |
| memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); |
| rtc::SetBE32(&pcmuFrame2[8], kSsrcX); |
| DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); |
| EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain()); |
| |
| // Setting gain with SSRC=0 should affect all unsignaled streams. |
| EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 3)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain()); |
| } |
| EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcX).gain()); |
| |
| // Setting gain on an individual stream affects only that. |
| EXPECT_TRUE(channel_->SetOutputVolume(kSsrcX, 4)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain()); |
| } |
| EXPECT_DOUBLE_EQ(4, GetRecvStream(kSsrcX).gain()); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromSyncLabel) { |
| const uint32_t kAudioSsrc = 123; |
| const std::string kSyncLabel = "AvSyncLabel"; |
| |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kAudioSsrc); |
| sp.sync_label = kSyncLabel; |
| // Creating two channels to make sure that sync label is set properly for both |
| // the default voice channel and following ones. |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| sp.ssrcs[0] += 1; |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| |
| ASSERT_EQ(2, call_.GetAudioReceiveStreams().size()); |
| EXPECT_EQ(kSyncLabel, |
| call_.GetAudioReceiveStream(kAudioSsrc)->GetConfig().sync_group) |
| << "SyncGroup should be set based on sync_label"; |
| EXPECT_EQ(kSyncLabel, |
| call_.GetAudioReceiveStream(kAudioSsrc + 1)->GetConfig().sync_group) |
| << "SyncGroup should be set based on sync_label"; |
| } |
| |
| // TODO(solenberg): Remove, once recv streams are configured through Call. |
| // (This is then covered by TestSetRecvRtpHeaderExtensions.) |
| TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) { |
| // Test that setting the header extensions results in the expected state |
| // changes on an associated Call. |
| std::vector<uint32_t> ssrcs; |
| ssrcs.push_back(223); |
| ssrcs.push_back(224); |
| |
| EXPECT_TRUE(SetupSendStream()); |
| SetSendParameters(send_parameters_); |
| for (uint32_t ssrc : ssrcs) { |
| EXPECT_TRUE(channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| |
| EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); |
| for (uint32_t ssrc : ssrcs) { |
| const auto* s = call_.GetAudioReceiveStream(ssrc); |
| EXPECT_NE(nullptr, s); |
| EXPECT_EQ(0, s->GetConfig().rtp.extensions.size()); |
| } |
| |
| // Set up receive extensions. |
| cricket::RtpCapabilities capabilities = engine_->GetCapabilities(); |
| cricket::AudioRecvParameters recv_parameters; |
| recv_parameters.extensions = capabilities.header_extensions; |
| channel_->SetRecvParameters(recv_parameters); |
| EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); |
| for (uint32_t ssrc : ssrcs) { |
| const auto* s = call_.GetAudioReceiveStream(ssrc); |
| EXPECT_NE(nullptr, s); |
| const auto& s_exts = s->GetConfig().rtp.extensions; |
| EXPECT_EQ(capabilities.header_extensions.size(), s_exts.size()); |
| for (const auto& e_ext : capabilities.header_extensions) { |
| for (const auto& s_ext : s_exts) { |
| if (e_ext.id == s_ext.id) { |
| EXPECT_EQ(e_ext.uri, s_ext.uri); |
| } |
| } |
| } |
| } |
| |
| // Disable receive extensions. |
| channel_->SetRecvParameters(cricket::AudioRecvParameters()); |
| for (uint32_t ssrc : ssrcs) { |
| const auto* s = call_.GetAudioReceiveStream(ssrc); |
| EXPECT_NE(nullptr, s); |
| EXPECT_EQ(0, s->GetConfig().rtp.extensions.size()); |
| } |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) { |
| // Test that packets are forwarded to the Call when configured accordingly. |
| const uint32_t kAudioSsrc = 1; |
| rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| static const unsigned char kRtcp[] = { |
| 0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, |
| 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, |
| 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, |
| 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 |
| }; |
| rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp)); |
| |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::WebRtcVoiceMediaChannel* media_channel = |
| static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(media_channel->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(kAudioSsrc))); |
| |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| const cricket::FakeAudioReceiveStream* s = |
| call_.GetAudioReceiveStream(kAudioSsrc); |
| EXPECT_EQ(0, s->received_packets()); |
| channel_->OnPacketReceived(&kPcmuPacket, rtc::PacketTime()); |
| EXPECT_EQ(1, s->received_packets()); |
| channel_->OnRtcpReceived(&kRtcpPacket, rtc::PacketTime()); |
| EXPECT_EQ(2, s->received_packets()); |
| } |
| |
| // All receive channels should be associated with the first send channel, |
| // since they do not send RTCP SR. |
| TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcZ))); |
| EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); |
| EXPECT_TRUE(AddRecvStream(kSsrcW)); |
| EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcW).rtp.local_ssrc); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) { |
| EXPECT_TRUE(SetupRecvStream()); |
| EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcY))); |
| EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); |
| EXPECT_TRUE(AddRecvStream(kSsrcZ)); |
| EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcW))); |
| EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); |
| EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetRawAudioSink) { |
| EXPECT_TRUE(SetupChannel()); |
| std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink()); |
| std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink()); |
| |
| // Setting the sink before a recv stream exists should do nothing. |
| channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_1)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); |
| |
| // Now try actually setting the sink. |
| channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_2)); |
| EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); |
| |
| // Now try resetting it. |
| channel_->SetRawAudioSink(kSsrcX, nullptr); |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) { |
| EXPECT_TRUE(SetupChannel()); |
| std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink()); |
| std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink()); |
| std::unique_ptr<FakeAudioSink> fake_sink_3(new FakeAudioSink()); |
| std::unique_ptr<FakeAudioSink> fake_sink_4(new FakeAudioSink()); |
| |
| // Should be able to set a default sink even when no stream exists. |
| channel_->SetRawAudioSink(0, std::move(fake_sink_1)); |
| |
| // Spawn an unsignaled stream by sending a packet - it should be assigned the |
| // default sink. |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); |
| |
| // Try resetting the default sink. |
| channel_->SetRawAudioSink(kSsrc0, nullptr); |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); |
| |
| // Try setting the default sink while the default stream exists. |
| channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_2)); |
| EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); |
| |
| // If we remove and add a default stream, it should get the same sink. |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc1)); |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); |
| |
| // Spawn another unsignaled stream - it should be assigned the default sink |
| // and the previous unsignaled stream should lose it. |
| unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; |
| memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); |
| rtc::SetBE32(&pcmuFrame2[8], kSsrcX); |
| DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); |
| } |
| EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); |
| |
| // Reset the default sink - the second unsignaled stream should lose it. |
| channel_->SetRawAudioSink(kSsrc0, nullptr); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); |
| } |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); |
| |
| // Try setting the default sink while two streams exists. |
| channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_3)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); |
| } |
| EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); |
| |
| // Try setting the sink for the first unsignaled stream using its known SSRC. |
| channel_->SetRawAudioSink(kSsrc1, std::move(fake_sink_4)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); |
| } |
| EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_NE(GetRecvStream(kSsrc1).sink(), GetRecvStream(kSsrcX).sink()); |
| } |
| } |
| |
| // Test that, just like the video channel, the voice channel communicates the |
| // network state to the call. |
| TEST_F(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) { |
| EXPECT_TRUE(SetupChannel()); |
| |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::AUDIO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::VIDEO)); |
| |
| channel_->OnReadyToSend(false); |
| EXPECT_EQ(webrtc::kNetworkDown, |
| call_.GetNetworkState(webrtc::MediaType::AUDIO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::VIDEO)); |
| |
| channel_->OnReadyToSend(true); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::AUDIO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::VIDEO)); |
| } |
| |
| // Test that playout is still started after changing parameters |
| TEST_F(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) { |
| SetupRecvStream(); |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| |
| // Changing RTP header extensions will recreate the AudioReceiveStream. |
| cricket::AudioRecvParameters parameters; |
| parameters.extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); |
| channel_->SetRecvParameters(parameters); |
| |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Tests that the library initializes and shuts down properly. |
| TEST(WebRtcVoiceEngineTest, StartupShutdown) { |
| // If the VoiceEngine wants to gather available codecs early, that's fine but |
| // we never want it to create a decoder at this stage. |
| rtc::scoped_refptr<webrtc::AudioProcessing> apm = |
| webrtc::AudioProcessing::Create(); |
| cricket::WebRtcVoiceEngine engine( |
| nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); |
| engine.Init(); |
| webrtc::RtcEventLogNullImpl event_log; |
| std::unique_ptr<webrtc::Call> call( |
| webrtc::Call::Create(webrtc::Call::Config(&event_log))); |
| cricket::VoiceMediaChannel* channel = engine.CreateChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions()); |
| EXPECT_TRUE(channel != nullptr); |
| delete channel; |
| } |
| |
| // Tests that reference counting on the external ADM is correct. |
| TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) { |
| testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; |
| EXPECT_CALL(adm, AddRef()).Times(3).WillRepeatedly(Return(0)); |
| EXPECT_CALL(adm, Release()).Times(3).WillRepeatedly(Return(0)); |
| // Return 100ms just in case this function gets called. If we don't, |
| // we could enter a tight loop since the mock would return 0. |
| EXPECT_CALL(adm, TimeUntilNextProcess()).WillRepeatedly(Return(100)); |
| { |
| rtc::scoped_refptr<webrtc::AudioProcessing> apm = |
| webrtc::AudioProcessing::Create(); |
| cricket::WebRtcVoiceEngine engine( |
| &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); |
| engine.Init(); |
| webrtc::RtcEventLogNullImpl event_log; |
| std::unique_ptr<webrtc::Call> call( |
| webrtc::Call::Create(webrtc::Call::Config(&event_log))); |
| cricket::VoiceMediaChannel* channel = engine.CreateChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions()); |
| EXPECT_TRUE(channel != nullptr); |
| delete channel; |
| } |
| } |
| |
| // Verify the payload id of common audio codecs, including CN, ISAC, and G722. |
| TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) { |
| // TODO(ossu): Why are the payload types of codecs with non-static payload |
| // type assignments checked here? It shouldn't really matter. |
| rtc::scoped_refptr<webrtc::AudioProcessing> apm = |
| webrtc::AudioProcessing::Create(); |
| cricket::WebRtcVoiceEngine engine( |
| nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); |
| engine.Init(); |
| for (const cricket::AudioCodec& codec : engine.send_codecs()) { |
| auto is_codec = [&codec](const char* name, int clockrate = 0) { |
| return STR_CASE_CMP(codec.name.c_str(), name) == 0 && |
| (clockrate == 0 || codec.clockrate == clockrate); |
| }; |
| if (is_codec("CN", 16000)) { |
| EXPECT_EQ(105, codec.id); |
| } else if (is_codec("CN", 32000)) { |
| EXPECT_EQ(106, codec.id); |
| } else if (is_codec("ISAC", 16000)) { |
| EXPECT_EQ(103, codec.id); |
| } else if (is_codec("ISAC", 32000)) { |
| EXPECT_EQ(104, codec.id); |
| } else if (is_codec("G722", 8000)) { |
| EXPECT_EQ(9, codec.id); |
| } else if (is_codec("telephone-event", 8000)) { |
| EXPECT_EQ(126, codec.id); |
| // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned. |
| // Remove these checks once both send and receive side assigns payload types |
| // dynamically. |
| } else if (is_codec("telephone-event", 16000)) { |
| EXPECT_EQ(113, codec.id); |
| } else if (is_codec("telephone-event", 32000)) { |
| EXPECT_EQ(112, codec.id); |
| } else if (is_codec("telephone-event", 48000)) { |
| EXPECT_EQ(110, codec.id); |
| } else if (is_codec("opus")) { |
| EXPECT_EQ(111, codec.id); |
| ASSERT_TRUE(codec.params.find("minptime") != codec.params.end()); |
| EXPECT_EQ("10", codec.params.find("minptime")->second); |
| ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end()); |
| EXPECT_EQ("1", codec.params.find("useinbandfec")->second); |
| } |
| } |
| } |
| |
| // Tests that VoE supports at least 32 channels |
| TEST(WebRtcVoiceEngineTest, Has32Channels) { |
| rtc::scoped_refptr<webrtc::AudioProcessing> apm = |
| webrtc::AudioProcessing::Create(); |
| cricket::WebRtcVoiceEngine engine( |
| nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); |
| engine.Init(); |
| webrtc::RtcEventLogNullImpl event_log; |
| std::unique_ptr<webrtc::Call> call( |
| webrtc::Call::Create(webrtc::Call::Config(&event_log))); |
| |
| cricket::VoiceMediaChannel* channels[32]; |
| int num_channels = 0; |
| while (num_channels < arraysize(channels)) { |
| cricket::VoiceMediaChannel* channel = engine.CreateChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions()); |
| if (!channel) |
| break; |
| channels[num_channels++] = channel; |
| } |
| |
| int expected = arraysize(channels); |
| EXPECT_EQ(expected, num_channels); |
| |
| while (num_channels > 0) { |
| delete channels[--num_channels]; |
| } |
| } |
| |
| // Test that we set our preferred codecs properly. |
| TEST(WebRtcVoiceEngineTest, SetRecvCodecs) { |
| // TODO(ossu): I'm not sure of the intent of this test. It's either: |
| // - Check that our builtin codecs are usable by Channel. |
| // - The codecs provided by the engine is usable by Channel. |
| // It does not check that the codecs in the RecvParameters are actually |
| // what we sent in - though it's probably reasonable to expect so, if |
| // SetRecvParameters returns true. |
| // I think it will become clear once audio decoder injection is completed. |
| rtc::scoped_refptr<webrtc::AudioProcessing> apm = |
| webrtc::AudioProcessing::Create(); |
| cricket::WebRtcVoiceEngine engine( |
| nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm); |
| engine.Init(); |
| webrtc::RtcEventLogNullImpl event_log; |
| std::unique_ptr<webrtc::Call> call( |
| webrtc::Call::Create(webrtc::Call::Config(&event_log))); |
| cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(), |
| cricket::AudioOptions(), call.get()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs = engine.recv_codecs(); |
| EXPECT_TRUE(channel.SetRecvParameters(parameters)); |
| } |
| |
| TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) { |
| std::vector<webrtc::AudioCodecSpec> specs; |
| webrtc::AudioCodecSpec spec1{{"codec1", 48000, 2, {{"param1", "value1"}}}, |
| {48000, 2, 16000, 10000, 20000}}; |
| spec1.info.allow_comfort_noise = false; |
| spec1.info.supports_network_adaption = true; |
| specs.push_back(spec1); |
| webrtc::AudioCodecSpec spec2{{"codec2", 32000, 1}, {32000, 1, 32000}}; |
| spec2.info.allow_comfort_noise = false; |
| specs.push_back(spec2); |
| specs.push_back(webrtc::AudioCodecSpec{ |
| {"codec3", 16000, 1, {{"param1", "value1b"}, {"param2", "value2"}}}, |
| {16000, 1, 13300}}); |
| specs.push_back( |
| webrtc::AudioCodecSpec{{"codec4", 8000, 1}, {8000, 1, 64000}}); |
| specs.push_back( |
| webrtc::AudioCodecSpec{{"codec5", 8000, 2}, {8000, 1, 64000}}); |
| |
| rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> unused_encoder_factory = |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(); |
| rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory = |
| new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>; |
| EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders()) |
| .WillOnce(Return(specs)); |
| |
| rtc::scoped_refptr<webrtc::AudioProcessing> apm = |
| webrtc::AudioProcessing::Create(); |
| cricket::WebRtcVoiceEngine engine(nullptr, unused_encoder_factory, |
| mock_decoder_factory, nullptr, apm); |
| engine.Init(); |
| auto codecs = engine.recv_codecs(); |
| EXPECT_EQ(11, codecs.size()); |
| |
| // Rather than just ASSERTing that there are enough codecs, ensure that we can |
| // check the actual values safely, to provide better test results. |
| auto get_codec = |
| [&codecs](size_t index) -> const cricket::AudioCodec& { |
| static const cricket::AudioCodec missing_codec(0, "<missing>", 0, 0, 0); |
| if (codecs.size() > index) |
| return codecs[index]; |
| return missing_codec; |
| }; |
| |
| // Ensure the general codecs are generated first and in order. |
| for (size_t i = 0; i != specs.size(); ++i) { |
| EXPECT_EQ(specs[i].format.name, get_codec(i).name); |
| EXPECT_EQ(specs[i].format.clockrate_hz, get_codec(i).clockrate); |
| EXPECT_EQ(specs[i].format.num_channels, get_codec(i).channels); |
| EXPECT_EQ(specs[i].format.parameters, get_codec(i).params); |
| } |
| |
| // Find the index of a codec, or -1 if not found, so that we can easily check |
| // supplementary codecs are ordered after the general codecs. |
| auto find_codec = |
| [&codecs](const webrtc::SdpAudioFormat& format) -> int { |
| for (size_t i = 0; i != codecs.size(); ++i) { |
| const cricket::AudioCodec& codec = codecs[i]; |
| if (STR_CASE_CMP(codec.name.c_str(), format.name.c_str()) == 0 && |
| codec.clockrate == format.clockrate_hz && |
| codec.channels == format.num_channels) { |
| return rtc::checked_cast<int>(i); |
| } |
| } |
| return -1; |
| }; |
| |
| // Ensure all supplementary codecs are generated last. Their internal ordering |
| // is not important. |
| // Without this cast, the comparison turned unsigned and, thus, failed for -1. |
| const int num_specs = static_cast<int>(specs.size()); |
| EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs); |
| EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs); |
| EXPECT_EQ(find_codec({"cn", 32000, 1}), -1); |
| EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs); |
| EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs); |
| EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs); |
| EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs); |
| } |
