| /* |
| * 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 "media/engine/webrtc_voice_engine.h" |
| |
| #include <cstddef> |
| #include <cstdint> |
| #include <cstring> |
| #include <map> |
| #include <memory> |
| #include <optional> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/strings/match.h" |
| #include "api/audio/audio_processing.h" |
| #include "api/audio/builtin_audio_processing_builder.h" |
| #include "api/audio_codecs/audio_codec_pair_id.h" |
| #include "api/audio_codecs/audio_format.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/audio_options.h" |
| #include "api/call/audio_sink.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/environment/environment.h" |
| #include "api/environment/environment_factory.h" |
| #include "api/make_ref_counted.h" |
| #include "api/media_types.h" |
| #include "api/priority.h" |
| #include "api/ref_count.h" |
| #include "api/rtc_error.h" |
| #include "api/rtp_headers.h" |
| #include "api/rtp_parameters.h" |
| #include "api/scoped_refptr.h" |
| #include "api/transport/bitrate_settings.h" |
| #include "api/transport/rtp/rtp_source.h" |
| #include "call/audio_receive_stream.h" |
| #include "call/audio_send_stream.h" |
| #include "call/audio_state.h" |
| #include "call/call.h" |
| #include "call/call_config.h" |
| #include "media/base/codec.h" |
| #include "media/base/fake_media_engine.h" |
| #include "media/base/fake_network_interface.h" |
| #include "media/base/fake_rtp.h" |
| #include "media/base/media_channel.h" |
| #include "media/base/media_config.h" |
| #include "media/base/media_constants.h" |
| #include "media/base/media_engine.h" |
| #include "media/base/stream_params.h" |
| #include "media/engine/fake_webrtc_call.h" |
| #include "modules/audio_device/include/mock_audio_device.h" |
| #include "modules/audio_mixer/audio_mixer_impl.h" |
| #include "modules/audio_processing/include/mock_audio_processing.h" |
| #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "rtc_base/arraysize.h" |
| #include "rtc_base/byte_order.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/dscp.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/thread.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| #include "test/mock_audio_decoder_factory.h" |
| #include "test/mock_audio_encoder_factory.h" |
| #include "test/scoped_key_value_config.h" |
| |
| namespace { |
| using ::testing::_; |
| using ::testing::ContainerEq; |
| using ::testing::Contains; |
| using ::testing::Field; |
| using ::testing::IsEmpty; |
| using ::testing::Return; |
| using ::testing::ReturnPointee; |
| using ::testing::SaveArg; |
| using ::testing::StrictMock; |
| using ::testing::UnorderedElementsAreArray; |
| using ::webrtc::AudioProcessing; |
| using ::webrtc::BitrateConstraints; |
| using ::webrtc::BuiltinAudioProcessingBuilder; |
| using ::webrtc::Call; |
| using ::webrtc::CallConfig; |
| using ::webrtc::CreateEnvironment; |
| using ::webrtc::Environment; |
| using ::webrtc::scoped_refptr; |
| |
| constexpr uint32_t kMaxUnsignaledRecvStreams = 4; |
| |
| const cricket::Codec kPcmuCodec = cricket::CreateAudioCodec(0, "PCMU", 8000, 1); |
| const cricket::Codec kOpusCodec = |
| cricket::CreateAudioCodec(111, "opus", 48000, 2); |
| const cricket::Codec kG722CodecVoE = |
| cricket::CreateAudioCodec(9, "G722", 16000, 1); |
| const cricket::Codec kG722CodecSdp = |
| cricket::CreateAudioCodec(9, "G722", 8000, 1); |
| const cricket::Codec kCn8000Codec = |
| cricket::CreateAudioCodec(13, "CN", 8000, 1); |
| const cricket::Codec kCn16000Codec = |
| cricket::CreateAudioCodec(105, "CN", 16000, 1); |
| const cricket::Codec kRed48000Codec = |
| cricket::CreateAudioCodec(112, "RED", 48000, 2); |
| const cricket::Codec kTelephoneEventCodec1 = |
| cricket::CreateAudioCodec(106, "telephone-event", 8000, 1); |
| const cricket::Codec kTelephoneEventCodec2 = |
| cricket::CreateAudioCodec(107, "telephone-event", 32000, 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; |
| |
| constexpr webrtc::AudioProcessing::Config::GainController1::Mode |
| kDefaultAgcMode = |
| #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) |
| webrtc::AudioProcessing::Config::GainController1::kFixedDigital; |
| #else |
| webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog; |
| #endif |
| |
| constexpr webrtc::AudioProcessing::Config::NoiseSuppression::Level |
| kDefaultNsLevel = |
| webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh; |
| |
| void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) { |
| RTC_DCHECK(adm); |
| |
| // Setup. |
| EXPECT_CALL(*adm, Init()).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, RegisterAudioCallback(_)).WillOnce(Return(0)); |
| #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)); |
| #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, BuiltInAECIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm, BuiltInAGCIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false)); |
| |
| // Teardown. |
| EXPECT_CALL(*adm, StopPlayout()).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, StopRecording()).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, RegisterAudioCallback(nullptr)).WillOnce(Return(0)); |
| EXPECT_CALL(*adm, Terminate()).WillOnce(Return(0)); |
| } |
| |
| std::vector<cricket::Codec> AddIdToCodecs( |
| webrtc::PayloadTypePicker& pt_mapper, |
| std::vector<cricket::Codec>&& codecs_in) { |
| std::vector<cricket::Codec> codecs = std::move(codecs_in); |
| for (cricket::Codec& codec : codecs) { |
| if (codec.id == cricket::Codec::kIdNotSet) { |
| auto id_or_error = pt_mapper.SuggestMapping(codec, nullptr); |
| EXPECT_TRUE(id_or_error.ok()); |
| if (id_or_error.ok()) { |
| codec.id = id_or_error.value(); |
| } |
| } |
| } |
| return codecs; |
| } |
| |
| std::vector<cricket::Codec> ReceiveCodecsWithId( |
| cricket::WebRtcVoiceEngine& engine) { |
| webrtc::PayloadTypePicker pt_mapper; |
| std::vector<cricket::Codec> codecs = engine.recv_codecs(); |
| return AddIdToCodecs(pt_mapper, std::move(codecs)); |
| } |
| |
| } // namespace |
| |
| // Tests that our stub library "works". |
| TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) { |
| Environment env = CreateEnvironment(); |
| for (bool use_null_apm : {false, true}) { |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateStrict(); |
| AdmSetupExpectations(adm.get()); |
| rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm = |
| use_null_apm ? nullptr |
| : rtc::make_ref_counted< |
| StrictMock<webrtc::test::MockAudioProcessing>>(); |
| |
| webrtc::AudioProcessing::Config apm_config; |
| if (!use_null_apm) { |
| EXPECT_CALL(*apm, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config)); |
| EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config)); |
| EXPECT_CALL(*apm, DetachAecDump()); |
| } |
| { |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, |
| nullptr, env.field_trials()); |
| engine.Init(); |
| } |
| } |
| } |
| |
| 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::TestWithParam<bool> { |
| public: |
| WebRtcVoiceEngineTestFake() |
| : use_null_apm_(GetParam()), |
| env_(CreateEnvironment(&field_trials_)), |
| adm_(webrtc::test::MockAudioDeviceModule::CreateStrict()), |
| apm_(use_null_apm_ |
| ? nullptr |
| : rtc::make_ref_counted< |
| StrictMock<webrtc::test::MockAudioProcessing>>()), |
| call_(env_) { |
| // AudioDeviceModule. |
| AdmSetupExpectations(adm_.get()); |
| |
| if (!use_null_apm_) { |
| // AudioProcessing. |
| EXPECT_CALL(*apm_, GetConfig()) |
| .WillRepeatedly(ReturnPointee(&apm_config_)); |
| EXPECT_CALL(*apm_, ApplyConfig(_)) |
| .WillRepeatedly(SaveArg<0>(&apm_config_)); |
| EXPECT_CALL(*apm_, DetachAecDump()); |
| } |
| |
| // Default Options. |
| // 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_ = std::make_unique<cricket::WebRtcVoiceEngine>( |
| &env_.task_queue_factory(), adm_.get(), encoder_factory, |
| decoder_factory, nullptr, apm_, nullptr, env_.field_trials()); |
| engine_->Init(); |
| send_parameters_.codecs.push_back(kPcmuCodec); |
| recv_parameters_.codecs.push_back(kPcmuCodec); |
| |
| if (!use_null_apm_) { |
| // Default Options. |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_TRUE(IsHighPassFilterEnabled()); |
| EXPECT_TRUE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| VerifyGainControlEnabledCorrectly(); |
| VerifyGainControlDefaultSettings(); |
| } |
| } |
| |
| bool SetupChannel() { |
| send_channel_ = engine_->CreateSendChannel( |
| &call_, cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); |
| receive_channel_ = engine_->CreateReceiveChannel( |
| &call_, cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); |
| send_channel_->SetSsrcListChangedCallback( |
| [receive_channel = |
| receive_channel_.get()](const std::set<uint32_t>& choices) { |
| receive_channel->ChooseReceiverReportSsrc(choices); |
| }); |
| send_channel_->SetSendCodecChangedCallback( |
| [receive_channel = receive_channel_.get(), |
| send_channel = send_channel_.get()]() { |
| receive_channel->SetReceiveNackEnabled( |
| send_channel->SendCodecHasNack()); |
| receive_channel->SetReceiveNonSenderRttEnabled( |
| send_channel->SenderNonSenderRttEnabled()); |
| }); |
| return true; |
| } |
| |
| bool SetupRecvStream() { |
| if (!SetupChannel()) { |
| return false; |
| } |
| return AddRecvStream(kSsrcX); |
| } |
| |
| bool SetupSendStream() { |
| return SetupSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)); |
| } |
| |
| bool SetupSendStream(const cricket::StreamParams& sp) { |
| if (!SetupChannel()) { |
| return false; |
| } |
| if (!send_channel_->AddSendStream(sp)) { |
| return false; |
| } |
| if (!use_null_apm_) { |
| EXPECT_CALL(*apm_, set_output_will_be_muted(false)); |
| } |
| return send_channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_); |
| } |
| |
| bool AddRecvStream(uint32_t ssrc) { |
| EXPECT_TRUE(receive_channel_); |
| return receive_channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(ssrc)); |
| } |
| |
| void SetupForMultiSendStream() { |
| EXPECT_TRUE(SetupSendStream()); |
| // Remove stream added in Setup. |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); |
| EXPECT_TRUE(send_channel_->RemoveSendStream(kSsrcX)); |
| // Verify the channel does not exist. |
| EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX)); |
| } |
| |
| void DeliverPacket(const void* data, int len) { |
| webrtc::RtpPacketReceived packet; |
| packet.Parse(reinterpret_cast<const uint8_t*>(data), len); |
| receive_channel_->OnPacketReceived(packet); |
| rtc::Thread::Current()->ProcessMessages(0); |
| } |
| |
| 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::AudioReceiveStreamInterface::Config& GetRecvStreamConfig( |
| uint32_t ssrc) { |
| return GetRecvStream(ssrc).GetConfig(); |
| } |
| |
| void SetSend(bool enable) { |
| ASSERT_TRUE(send_channel_); |
| if (enable) { |
| EXPECT_CALL(*adm_, RecordingIsInitialized()) |
| .Times(::testing::AtMost(1)) |
| .WillOnce(Return(false)); |
| EXPECT_CALL(*adm_, Recording()) |
| .Times(::testing::AtMost(1)) |
| .WillOnce(Return(false)); |
| EXPECT_CALL(*adm_, InitRecording()) |
| .Times(::testing::AtMost(1)) |
| .WillOnce(Return(0)); |
| } |
| send_channel_->SetSend(enable); |
| } |
| |
| void SetSenderParameters(const cricket::AudioSenderParameter& params) { |
| ASSERT_TRUE(send_channel_); |
| EXPECT_TRUE(send_channel_->SetSenderParameters(params)); |
| } |
| |
| void SetAudioSend(uint32_t ssrc, |
| bool enable, |
| cricket::AudioSource* source, |
| const cricket::AudioOptions* options = nullptr) { |
| ASSERT_TRUE(send_channel_); |
| if (!use_null_apm_) { |
| EXPECT_CALL(*apm_, set_output_will_be_muted(!enable)); |
| } |
| EXPECT_TRUE(send_channel_->SetAudioSend(ssrc, enable, options, source)); |
| } |
| |
| void TestInsertDtmf(uint32_t ssrc, bool caller, const cricket::Codec& codec) { |
| EXPECT_TRUE(SetupChannel()); |
| if (caller) { |
| // If this is a caller, local description will be applied and add the |
| // send stream. |
| EXPECT_TRUE(send_channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Test we can only InsertDtmf when the other side supports telephone-event. |
| SetSenderParameters(send_parameters_); |
| SetSend(true); |
| EXPECT_FALSE(send_channel_->CanInsertDtmf()); |
| EXPECT_FALSE(send_channel_->InsertDtmf(ssrc, 1, 111)); |
| send_parameters_.codecs.push_back(codec); |
| SetSenderParameters(send_parameters_); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| |
| if (!caller) { |
| // If this is callee, there's no active send channel yet. |
| EXPECT_FALSE(send_channel_->InsertDtmf(ssrc, 2, 123)); |
| EXPECT_TRUE(send_channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Check we fail if the ssrc is invalid. |
| EXPECT_FALSE(send_channel_->InsertDtmf(-1, 1, 111)); |
| |
| // Test send. |
| cricket::FakeAudioSendStream::TelephoneEvent telephone_event = |
| GetSendStream(kSsrcX).GetLatestTelephoneEvent(); |
| EXPECT_EQ(-1, telephone_event.payload_type); |
| EXPECT_TRUE(send_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); |
| } |
| |
| void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) { |
| // For a caller, the answer will be applied in set remote description |
| // where SetSenderParameters() is called. |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(send_channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| send_parameters_.extmap_allow_mixed = extmap_allow_mixed; |
| SetSenderParameters(send_parameters_); |
| const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); |
| EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed); |
| } |
| |
| void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) { |
| // For a callee, the answer will be applied in set local description |
| // where SetExtmapAllowMixed() and AddSendStream() are called. |
| EXPECT_TRUE(SetupChannel()); |
| send_channel_->SetExtmapAllowMixed(extmap_allow_mixed); |
| EXPECT_TRUE(send_channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| |
| const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); |
| EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed); |
| } |
| |
| // 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::Codec& codec, |
| int max_bitrate, |
| bool expected_result, |
| int expected_bitrate) { |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(codec); |
| parameters.max_bandwidth_bps = max_bitrate; |
| if (expected_result) { |
| SetSenderParameters(parameters); |
| } else { |
| EXPECT_FALSE(send_channel_->SetSenderParameters(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 = |
| send_channel_->GetRtpSendParameters(ssrc); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| |
| parameters.encodings[0].max_bitrate_bps = bitrate; |
| return send_channel_->SetRtpSendParameters(ssrc, parameters).ok(); |
| } |
| |
| void SetGlobalMaxBitrate(const cricket::Codec& codec, int bitrate) { |
| cricket::AudioSenderParameter send_parameters; |
| send_parameters.codecs.push_back(codec); |
| send_parameters.max_bandwidth_bps = bitrate; |
| SetSenderParameters(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); |
| } |
| |
| std::optional<int> GetCodecBitrate(int32_t ssrc) { |
| auto spec = GetSendStreamConfig(ssrc).send_codec_spec; |
| if (!spec.has_value()) { |
| return std::nullopt; |
| } |
| return spec->target_bitrate_bps; |
| } |
| |
| int GetMaxBitrate(int32_t ssrc) { |
| return GetSendStreamConfig(ssrc).max_bitrate_bps; |
| } |
| |
| const std::optional<std::string>& GetAudioNetworkAdaptorConfig(int32_t ssrc) { |
| return GetSendStreamConfig(ssrc).audio_network_adaptor_config; |
| } |
| |
| void SetAndExpectMaxBitrate(const cricket::Codec& 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 = |
| send_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)); |
| EXPECT_EQ(expected_codec_bitrate, GetMaxBitrate(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; |
| EXPECT_CALL(*call_.GetMockTransportControllerSend(), |
| SetSdpBitrateParameters( |
| AllOf(Field(&BitrateConstraints::min_bitrate_bps, |
| expected_min_bitrate_bps), |
| Field(&BitrateConstraints::start_bitrate_bps, |
| expected_start_bitrate_bps), |
| Field(&BitrateConstraints::max_bitrate_bps, |
| expected_max_bitrate_bps)))); |
| |
| SetSenderParameters(send_parameters_); |
| } |
| |
| 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)); |
| SetSenderParameters(send_parameters_); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extensions stay off with an empty list of headers. |
| send_parameters_.extensions.clear(); |
| SetSenderParameters(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)); |
| SetSenderParameters(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(send_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(); |
| SetSenderParameters(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_THAT( |
| receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, |
| IsEmpty()); |
| |
| // Ensure unknown extensions won't cause an error. |
| recv_parameters_.extensions.push_back( |
| webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); |
| EXPECT_THAT( |
| receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, |
| IsEmpty()); |
| |
| // Ensure extensions stay off with an empty list of headers. |
| recv_parameters_.extensions.clear(); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); |
| EXPECT_THAT( |
| receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, |
| IsEmpty()); |
| |
| // Ensure extension is set properly. |
| const int id = 2; |
| recv_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); |
| EXPECT_EQ( |
| receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, |
| recv_parameters_.extensions); |
| |
| // Ensure extension is set properly on new stream. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_EQ( |
| receive_channel_->GetRtpReceiverParameters(kSsrcY).header_extensions, |
| recv_parameters_.extensions); |
| |
| // Ensure all extensions go back off with an empty list. |
| recv_parameters_.extensions.clear(); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); |
| EXPECT_THAT( |
| receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, |
| IsEmpty()); |
| EXPECT_THAT( |
| receive_channel_->GetRtpReceiverParameters(kSsrcY).header_extensions, |
| IsEmpty()); |
| } |
| |
| webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const { |
| webrtc::AudioSendStream::Stats stats; |
| stats.local_ssrc = 12; |
| stats.payload_bytes_sent = 345; |
| stats.header_and_padding_bytes_sent = 56; |
| stats.packets_sent = 678; |
| stats.packets_lost = 9012; |
| stats.fraction_lost = 34.56f; |
| stats.codec_name = "codec_name_send"; |
| stats.codec_payload_type = 0; |
| stats.jitter_ms = 12; |
| stats.rtt_ms = 345; |
| stats.audio_level = 678; |
| stats.apm_statistics.delay_median_ms = 234; |
| stats.apm_statistics.delay_standard_deviation_ms = 567; |
| stats.apm_statistics.echo_return_loss = 890; |
| stats.apm_statistics.echo_return_loss_enhancement = 1234; |
| stats.apm_statistics.residual_echo_likelihood = 0.432f; |
| stats.apm_statistics.residual_echo_likelihood_recent_max = 0.6f; |
| stats.ana_statistics.bitrate_action_counter = 321; |
| stats.ana_statistics.channel_action_counter = 432; |
| stats.ana_statistics.dtx_action_counter = 543; |
| stats.ana_statistics.fec_action_counter = 654; |
| stats.ana_statistics.frame_length_increase_counter = 765; |
| stats.ana_statistics.frame_length_decrease_counter = 876; |
| stats.ana_statistics.uplink_packet_loss_fraction = 987.0; |
| 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.payload_bytes_sent, stats.payload_bytes_sent); |
| EXPECT_EQ(info.header_and_padding_bytes_sent, |
| stats.header_and_padding_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.jitter_ms, stats.jitter_ms); |
| EXPECT_EQ(info.rtt_ms, stats.rtt_ms); |
| EXPECT_EQ(info.audio_level, stats.audio_level); |
| EXPECT_EQ(info.apm_statistics.delay_median_ms, |
| stats.apm_statistics.delay_median_ms); |
| EXPECT_EQ(info.apm_statistics.delay_standard_deviation_ms, |
| stats.apm_statistics.delay_standard_deviation_ms); |
| EXPECT_EQ(info.apm_statistics.echo_return_loss, |
| stats.apm_statistics.echo_return_loss); |
| EXPECT_EQ(info.apm_statistics.echo_return_loss_enhancement, |
| stats.apm_statistics.echo_return_loss_enhancement); |
| EXPECT_EQ(info.apm_statistics.residual_echo_likelihood, |
| stats.apm_statistics.residual_echo_likelihood); |
| EXPECT_EQ(info.apm_statistics.residual_echo_likelihood_recent_max, |
| stats.apm_statistics.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); |
| } |
| |
| webrtc::AudioReceiveStreamInterface::Stats GetAudioReceiveStreamStats() |
| const { |
| webrtc::AudioReceiveStreamInterface::Stats stats; |
| stats.remote_ssrc = 123; |
| stats.payload_bytes_received = 456; |
| stats.header_and_padding_bytes_received = 67; |
| stats.packets_received = 768; |
| stats.packets_lost = 101; |
| stats.codec_name = "codec_name_recv"; |
| stats.codec_payload_type = 0; |
| 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.concealment_events = 12; |
| stats.jitter_buffer_delay_seconds = 34; |
| stats.jitter_buffer_emitted_count = 77; |
| stats.total_processing_delay_seconds = 0.123; |
| 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_codec_plc = 1236; |
| 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.payload_bytes_received, stats.payload_bytes_received); |
| EXPECT_EQ(info.header_and_padding_bytes_received, |
| stats.header_and_padding_bytes_received); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_received), |
| stats.packets_received); |
| EXPECT_EQ(info.packets_lost, stats.packets_lost); |
| EXPECT_EQ(info.codec_name, stats.codec_name); |
| EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_ms), stats.jitter_ms); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_ms), |
| stats.jitter_buffer_ms); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_preferred_ms), |
| stats.jitter_buffer_preferred_ms); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(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.concealment_events, stats.concealment_events); |
| EXPECT_EQ(info.jitter_buffer_delay_seconds, |
| stats.jitter_buffer_delay_seconds); |
| EXPECT_EQ(info.jitter_buffer_emitted_count, |
| stats.jitter_buffer_emitted_count); |
| EXPECT_EQ(info.total_processing_delay_seconds, |
| stats.total_processing_delay_seconds); |
| 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_codec_plc, stats.decoding_codec_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::VoiceMediaSendInfo& send_info, |
| const cricket::VoiceMediaReceiveInfo& receive_info) const { |
| EXPECT_EQ(send_parameters_.codecs.size(), send_info.send_codecs.size()); |
| for (const cricket::Codec& codec : send_parameters_.codecs) { |
| ASSERT_EQ(send_info.send_codecs.count(codec.id), 1U); |
| EXPECT_EQ(send_info.send_codecs.find(codec.id)->second, |
| codec.ToCodecParameters()); |
| } |
| EXPECT_EQ(recv_parameters_.codecs.size(), |
| receive_info.receive_codecs.size()); |
| for (const cricket::Codec& codec : recv_parameters_.codecs) { |
| ASSERT_EQ(receive_info.receive_codecs.count(codec.id), 1U); |
| EXPECT_EQ(receive_info.receive_codecs.find(codec.id)->second, |
| codec.ToCodecParameters()); |
| } |
| } |
| |
| void VerifyGainControlEnabledCorrectly() { |
| EXPECT_TRUE(apm_config_.gain_controller1.enabled); |
| EXPECT_EQ(kDefaultAgcMode, apm_config_.gain_controller1.mode); |
| } |
| |
| void VerifyGainControlDefaultSettings() { |
| EXPECT_EQ(3, apm_config_.gain_controller1.target_level_dbfs); |
| EXPECT_EQ(9, apm_config_.gain_controller1.compression_gain_db); |
| EXPECT_TRUE(apm_config_.gain_controller1.enable_limiter); |
| } |
| |
| void VerifyEchoCancellationSettings(bool enabled) { |
| constexpr bool kDefaultUseAecm = |
| #if defined(WEBRTC_ANDROID) |
| true; |
| #else |
| false; |
| #endif |
| EXPECT_EQ(apm_config_.echo_canceller.enabled, enabled); |
| EXPECT_EQ(apm_config_.echo_canceller.mobile_mode, kDefaultUseAecm); |
| } |
| |
| bool IsHighPassFilterEnabled() { |
| return apm_config_.high_pass_filter.enabled; |
| } |
| |
| cricket::WebRtcVoiceSendChannel* SendImplFromPointer( |
| cricket::VoiceMediaSendChannelInterface* channel) { |
| return static_cast<cricket::WebRtcVoiceSendChannel*>(channel); |
| } |
| |
| cricket::WebRtcVoiceSendChannel* SendImpl() { |
| return SendImplFromPointer(send_channel_.get()); |
| } |
| cricket::WebRtcVoiceReceiveChannel* ReceiveImpl() { |
| return static_cast<cricket::WebRtcVoiceReceiveChannel*>( |
| receive_channel_.get()); |
| } |
| std::vector<cricket::Codec> SendCodecsWithId() { |
| std::vector<cricket::Codec> codecs = engine_->send_codecs(); |
| return AddIdToCodecs(pt_mapper_, std::move(codecs)); |
| } |
| |
| protected: |
| rtc::AutoThread main_thread_; |
| const bool use_null_apm_; |
| webrtc::test::ScopedKeyValueConfig field_trials_; |
| const Environment env_; |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm_; |
| rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_; |
| cricket::FakeCall call_; |
| FakeAudioSource fake_source_; |
| std::unique_ptr<cricket::WebRtcVoiceEngine> engine_; |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel_; |
| std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> receive_channel_; |
| cricket::AudioSenderParameter send_parameters_; |
| cricket::AudioReceiverParameters recv_parameters_; |
| webrtc::AudioProcessing::Config apm_config_; |
| webrtc::PayloadTypePicker pt_mapper_; |
| }; |
| |
| INSTANTIATE_TEST_SUITE_P(TestBothWithAndWithoutNullApm, |
| WebRtcVoiceEngineTestFake, |
| ::testing::Values(false, true)); |
| |
| // Tests that we can create and destroy a channel. |
| TEST_P(WebRtcVoiceEngineTestFake, CreateMediaChannel) { |
| EXPECT_TRUE(SetupChannel()); |
| } |
| |
| // Test that we can add a send stream and that it has the correct defaults. |
| TEST_P(WebRtcVoiceEngineTestFake, CreateSendStream) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(send_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(SendImpl()->transport(), config.send_transport); |
| } |
| |
| // Test that we can add a receive stream and that it has the correct defaults. |
| TEST_P(WebRtcVoiceEngineTestFake, CreateRecvStream) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| const webrtc::AudioReceiveStreamInterface::Config& config = |
| GetRecvStreamConfig(kSsrcX); |
| EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc); |
| EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc); |
| EXPECT_EQ(ReceiveImpl()->transport(), config.rtcp_send_transport); |
| EXPECT_EQ("", config.sync_group); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) { |
| const std::vector<cricket::Codec>& codecs = engine_->send_codecs(); |
| bool opus_found = false; |
| for (const cricket::Codec& 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_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| 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(receive_channel_->SetReceiverParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {106, {"OPUS", 48000, 2}}, |
| {126, {"telephone-event", 8000, 1}}, |
| {107, {"telephone-event", 32000, 1}}}))); |
| } |
| |
| // Test that we fail to set an unknown inbound codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(cricket::CreateAudioCodec(127, "XYZ", 32000, 1)); |
| EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)); |
| } |
| |
| // Test that we fail if we have duplicate types in the inbound list. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = kOpusCodec.id; |
| EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)); |
| } |
| |
| // Test that we can decode OPUS without stereo parameters. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, {111, {"opus", 48000, 2}}}))); |
| } |
| |
| // Test that we can decode OPUS with stereo = 0. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[1].params["stereo"] = "0"; |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {111, {"opus", 48000, 2, {{"stereo", "0"}}}}}))); |
| } |
| |
| // Test that we can decode OPUS with stereo = 1. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[1].params["stereo"] = "1"; |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {111, {"opus", 48000, 2, {{"stereo", "1"}}}}}))); |
| } |
| |
| // Test that changes to recv codecs are applied to all streams. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| 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(receive_channel_->SetReceiverParameters(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, {"OPUS", 48000, 2}}, |
| {126, {"telephone-event", 8000, 1}}, |
| {107, {"telephone-event", 32000, 1}}}))); |
| } |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].id = 106; // collide with existing CN 32k |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| |
| const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map; |
| ASSERT_EQ(1u, dm.count(106)); |
| EXPECT_EQ(webrtc::SdpAudioFormat("opus", 48000, 2), dm.at(106)); |
| } |
| |
| // Test that we can apply the same set of codecs again while playing. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| receive_channel_->SetPlayout(true); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| |
| // Remapping a payload type to a different codec should fail. |
| parameters.codecs[0] = kOpusCodec; |
| parameters.codecs[0].id = kPcmuCodec.id; |
| EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we can add a codec while playing. |
| TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| receive_channel_->SetPlayout(true); |
| |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(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_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| |
| ++parameters.codecs[0].id; |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| } |
| |
| // Test that we do allow setting Opus/Red by default. |
| TEST_P(WebRtcVoiceEngineTestFake, RecvRedDefault) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kRed48000Codec); |
| parameters.codecs[1].params[""] = "111/111"; |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{111, {"opus", 48000, 2}}, |
| {112, {"red", 48000, 2, {{"", "111/111"}}}}}))); |
| } |
| |
| TEST_P(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. |
| |
| // PCMU, default bitrate == 64000. |
| TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000); |
| |
| // opus, default bitrate == 32000 in mono. |
| TestMaxSendBandwidth(kOpusCodec, -1, true, 32000); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // 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_P(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_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) { |
| EXPECT_TRUE(SetupChannel()); |
| const int kDesiredBitrate = 128000; |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs = SendCodecsWithId(); |
| parameters.max_bandwidth_bps = kDesiredBitrate; |
| SetSenderParameters(parameters); |
| |
| EXPECT_TRUE(send_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_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // PCMU, default bitrate == 64000. |
| SetSenderParameters(send_parameters_); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| |
| send_parameters_.max_bandwidth_bps = 128000; |
| SetSenderParameters(send_parameters_); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| |
| send_parameters_.max_bandwidth_bps = 128; |
| EXPECT_FALSE(send_channel_->SetSenderParameters(send_parameters_)); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| } |
| |
| // Test that the per-stream bitrate limit and the global |
| // bitrate limit both apply. |
| TEST_P(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_P(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) { |
| EXPECT_TRUE(SetupChannel()); |
| webrtc::RtpParameters nonexistent_parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(0u, nonexistent_parameters.encodings.size()); |
| |
| nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE( |
| send_channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters).ok()); |
| } |
| |
| TEST_P(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 = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| // Two or more encodings should result in failure. |
| parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| // Zero encodings should also fail. |
| parameters.encodings.clear(); |
| EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| } |
| |
| // Changing the SSRC through RtpParameters is not allowed. |
| TEST_P(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| parameters.encodings[0].ssrc = 0xdeadbeef; |
| EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| } |
| |
| // Test that a stream will not be sending if its encoding is made |
| // inactive through SetRtpSendParameters. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| // Get current parameters and change "active" to false. |
| webrtc::RtpParameters parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_TRUE(parameters.encodings[0].active); |
| parameters.encodings[0].active = false; |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Now change it back to active and verify we resume sending. |
| // This should occur even when other parameters are updated. |
| parameters.encodings[0].active = true; |
| parameters.encodings[0].max_bitrate_bps = std::optional<int>(6000); |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersAdaptivePtime) { |
| EXPECT_TRUE(SetupSendStream()); |
| // Get current parameters and change "adaptive_ptime" to true. |
| webrtc::RtpParameters parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_FALSE(parameters.encodings[0].adaptive_ptime); |
| parameters.encodings[0].adaptive_ptime = true; |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX)); |
| EXPECT_EQ(16000, GetSendStreamConfig(kSsrcX).min_bitrate_bps); |
| |
| parameters.encodings[0].adaptive_ptime = false; |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| EXPECT_FALSE(GetAudioNetworkAdaptorConfig(kSsrcX)); |
| EXPECT_EQ(32000, GetSendStreamConfig(kSsrcX).min_bitrate_bps); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, |
| DisablingAdaptivePtimeDoesNotRemoveAudioNetworkAdaptorFromOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = true; |
| send_parameters_.options.audio_network_adaptor_config = {"1234"}; |
| SetSenderParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| |
| webrtc::RtpParameters parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| parameters.encodings[0].adaptive_ptime = false; |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, AdaptivePtimeFieldTrial) { |
| webrtc::test::ScopedKeyValueConfig override_field_trials( |
| field_trials_, "WebRTC-Audio-AdaptivePtime/enabled:true/"); |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| // Test that SetRtpSendParameters configures the correct encoding channel for |
| // each SSRC. |
| TEST_P(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) { |
| SetupForMultiSendStream(); |
| // Create send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(send_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_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSenderParameters(parameters); |
| |
| webrtc::RtpParameters rtp_parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(kOpusCodec.ToCodecParameters(), rtp_parameters.codecs[0]); |
| EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); |
| } |
| |
| // Test that GetRtpSendParameters returns the currently configured RTCP CNAME. |
| TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) { |
| cricket::StreamParams params = cricket::StreamParams::CreateLegacy(kSsrcX); |
| params.cname = "rtcpcname"; |
| EXPECT_TRUE(SetupSendStream(params)); |
| |
| webrtc::RtpParameters rtp_parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, |
| DetectRtpSendParameterHeaderExtensionsChange) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| webrtc::RtpParameters rtp_parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| rtp_parameters.header_extensions.emplace_back(); |
| |
| EXPECT_NE(0u, rtp_parameters.header_extensions.size()); |
| |
| webrtc::RTCError result = |
| send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters); |
| EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type()); |
| } |
| |
| // Test that GetRtpSendParameters returns an SSRC. |
| TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) { |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters rtp_parameters = |
| send_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_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSenderParameters(parameters); |
| |
| webrtc::RtpParameters initial_params = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| |
| // We should be able to set the params we just got. |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, initial_params).ok()); |
| |
| // ... And this shouldn't change the params returned by GetRtpSendParameters. |
| webrtc::RtpParameters new_params = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(initial_params, send_channel_->GetRtpSendParameters(kSsrcX)); |
| } |
| |
| // Test that we remove the codec from RTP parameters if it's not negotiated |
| // anymore. |
| TEST_P(WebRtcVoiceEngineTestFake, |
| SetSendParametersRemovesSelectedCodecFromRtpParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSenderParameters(parameters); |
| |
| webrtc::RtpParameters initial_params = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| |
| webrtc::RtpCodec opus_rtp_codec; |
| opus_rtp_codec.name = "opus"; |
| opus_rtp_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| opus_rtp_codec.num_channels = 2; |
| opus_rtp_codec.clock_rate = 48000; |
| initial_params.encodings[0].codec = opus_rtp_codec; |
| |
| // We should be able to set the params with the opus codec that has been |
| // negotiated. |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, initial_params).ok()); |
| |
| parameters.codecs.clear(); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSenderParameters(parameters); |
| |
| // Since Opus is no longer negotiated, the RTP parameters should not have a |
| // forced codec anymore. |
| webrtc::RtpParameters new_params = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(new_params.encodings[0].codec, std::nullopt); |
| } |
| |
| // Test that max_bitrate_bps in send stream config gets updated correctly when |
| // SetRtpSendParameters is called. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter send_parameters; |
| send_parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(send_parameters); |
| |
| webrtc::RtpParameters rtp_parameters = |
| send_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 = kMaxBitrateBps; |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); |
| |
| const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps; |
| EXPECT_EQ(max_bitrate, kMaxBitrateBps); |
| } |
| |
| // Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to |
| // a value <= 0, setting the parameters returns false. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterInvalidBitratePriority) { |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters rtp_parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(1UL, rtp_parameters.encodings.size()); |
| EXPECT_EQ(webrtc::kDefaultBitratePriority, |
| rtp_parameters.encodings[0].bitrate_priority); |
| |
| rtp_parameters.encodings[0].bitrate_priority = 0; |
| EXPECT_FALSE( |
| send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); |
| rtp_parameters.encodings[0].bitrate_priority = -1.0; |
| EXPECT_FALSE( |
| send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); |
| } |
| |
| // Test that the bitrate_priority in the send stream config gets updated when |
| // SetRtpSendParameters is set for the VoiceMediaChannel. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) { |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters rtp_parameters = |
| send_channel_->GetRtpSendParameters(kSsrcX); |
| |
| EXPECT_EQ(1UL, rtp_parameters.encodings.size()); |
| EXPECT_EQ(webrtc::kDefaultBitratePriority, |
| rtp_parameters.encodings[0].bitrate_priority); |
| double new_bitrate_priority = 2.0; |
| rtp_parameters.encodings[0].bitrate_priority = new_bitrate_priority; |
| EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); |
| |
| // The priority should get set for both the audio channel's rtp parameters |
| // and the audio send stream's audio config. |
| EXPECT_EQ(new_bitrate_priority, send_channel_->GetRtpSendParameters(kSsrcX) |
| .encodings[0] |
| .bitrate_priority); |
| EXPECT_EQ(new_bitrate_priority, GetSendStreamConfig(kSsrcX).bitrate_priority); |
| } |
| |
| // Test that GetRtpReceiverParameters returns the currently configured codecs. |
| TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| |
| webrtc::RtpParameters rtp_parameters = |
| receive_channel_->GetRtpReceiverParameters(kSsrcX); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(kOpusCodec.ToCodecParameters(), rtp_parameters.codecs[0]); |
| EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); |
| } |
| |
| // Test that GetRtpReceiverParameters returns an SSRC. |
| TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) { |
| EXPECT_TRUE(SetupRecvStream()); |
| webrtc::RtpParameters rtp_parameters = |
| receive_channel_->GetRtpReceiverParameters(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_P(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| |
| webrtc::RtpParameters initial_params = |
| receive_channel_->GetRtpReceiverParameters(kSsrcX); |
| |
| // ... And this shouldn't change the params returned by |
| // GetRtpReceiverParameters. |
| webrtc::RtpParameters new_params = |
| receive_channel_->GetRtpReceiverParameters(kSsrcX); |
| EXPECT_EQ(initial_params, receive_channel_->GetRtpReceiverParameters(kSsrcX)); |
| } |
| |
| // Test that GetRtpReceiverParameters 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_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) { |
| ASSERT_TRUE(SetupChannel()); |
| // Call necessary methods to configure receiving a default stream as |
| // soon as it arrives. |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| |
| // Call GetDefaultRtpReceiveParameters before configured to receive an |
| // unsignaled stream. Should return nothing. |
| EXPECT_EQ(webrtc::RtpParameters(), |
| receive_channel_->GetDefaultRtpReceiveParameters()); |
| |
| // Set a sink for an unsignaled stream. |
| std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink()); |
| receive_channel_->SetDefaultRawAudioSink(std::move(fake_sink)); |
| |
| // Call GetDefaultRtpReceiveParameters before the SSRC is known. |
| webrtc::RtpParameters rtp_parameters = |
| receive_channel_->GetDefaultRtpReceiveParameters(); |
| 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 = receive_channel_->GetDefaultRtpReceiveParameters(); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) { |
| ASSERT_TRUE(SetupChannel()); |
| cricket::AudioReceiverParameters parameters = recv_parameters_; |
| parameters.extensions.push_back( |
| RtpExtension(RtpExtension::kAudioLevelUri, /*id=*/1)); |
| ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| webrtc::RtpHeaderExtensionMap extension_map(parameters.extensions); |
| webrtc::RtpPacketReceived reference_packet(&extension_map); |
| constexpr uint8_t kAudioLevel = 123; |
| reference_packet.SetExtension<webrtc::AudioLevelExtension>( |
| webrtc::AudioLevel(/*voice_activity=*/true, kAudioLevel)); |
| // Create a packet without the extension map but with the same content. |
| webrtc::RtpPacketReceived received_packet; |
| ASSERT_TRUE(received_packet.Parse(reference_packet.Buffer())); |
| |
| receive_channel_->OnPacketReceived(received_packet); |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| webrtc::AudioLevel audio_level; |
| EXPECT_TRUE(call_.last_received_rtp_packet() |
| .GetExtension<webrtc::AudioLevelExtension>(&audio_level)); |
| EXPECT_EQ(audio_level.level(), kAudioLevel); |
| } |
| |
| // Test that we apply codecs properly. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[0].bitrate = 22000; |
| SetSenderParameters(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("OPUS", send_codec_spec.format.name.c_str()); |
| EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000); |
| EXPECT_EQ(std::nullopt, send_codec_spec.cng_payload_type); |
| EXPECT_FALSE(send_channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we use Opus/Red by default when it is |
| // listed as the first codec and there is an fmtp line. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRed) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kRed48000Codec); |
| parameters.codecs[0].params[""] = "111/111"; |
| parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(112, send_codec_spec.red_payload_type); |
| } |
| |
| // Test that we do not use Opus/Red by default when it is |
| // listed as the first codec but there is no fmtp line. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedNoFmtp) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kRed48000Codec); |
| parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type); |
| } |
| |
| // Test that we do not use Opus/Red by default. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kRed48000Codec); |
| parameters.codecs[1].params[""] = "111/111"; |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type); |
| } |
| |
| // Test that the RED fmtp line must match the payload type. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpMismatch) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kRed48000Codec); |
| parameters.codecs[0].params[""] = "8/8"; |
| parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type); |
| } |
| |
| // Test that the RED fmtp line must show 2..32 payloads. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kRed48000Codec); |
| parameters.codecs[0].params[""] = "111"; |
| parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type); |
| for (int i = 1; i < 32; i++) { |
| parameters.codecs[0].params[""] += "/111"; |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec2 = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, send_codec_spec2.payload_type); |
| EXPECT_STRCASEEQ("opus", send_codec_spec2.format.name.c_str()); |
| EXPECT_EQ(112, send_codec_spec2.red_payload_type); |
| } |
| parameters.codecs[0].params[""] += "/111"; |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec3 = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, send_codec_spec3.payload_type); |
| EXPECT_STRCASEEQ("opus", send_codec_spec3.format.name.c_str()); |
| EXPECT_EQ(std::nullopt, send_codec_spec3.red_payload_type); |
| } |
| |
| // Test that WebRtcVoiceEngine reconfigures, rather than recreates its |
| // AudioSendStream. |
| TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| 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(); |
| SetSenderParameters(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. |
| SetSenderParameters(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 do not have a send codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].clockrate = 50000; |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt); |
| } |
| |
| // Test that if channels=0 for opus, we do not have a send codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 0; |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt); |
| } |
| |
| // Test that if channels=0 for opus, we do not have a send codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt); |
| } |
| |
| // Test that if channel is 1 for opus and there's no stereo, we do not have a |
| // send codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt); |
| } |
| |
| // Test that if channel is 1 for opus and stereo=0, we do not have a send codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| parameters.codecs[0].params["stereo"] = "0"; |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt); |
| } |
| |
| // Test that if channel is 1 for opus and stereo=1, we do not have a send codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt); |
| } |
| |
| // Test that with bitrate=0 and no stereo, bitrate is 32000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 32000); |
| } |
| |
| // Test that with bitrate=0 and stereo=0, bitrate is 32000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "0"; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 32000); |
| } |
| |
| // Test that with bitrate=invalid and stereo=0, bitrate is 32000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter 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; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 6000); |
| |
| parameters.codecs[0].bitrate = 510001; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 510000); |
| } |
| |
| // Test that with bitrate=0 and stereo=1, bitrate is 64000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 64000); |
| } |
| |
| // Test that with bitrate=invalid and stereo=1, bitrate is 64000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter 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; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 6000); |
| |
| parameters.codecs[0].bitrate = 510001; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 510000); |
| } |
| |
| // Test that with bitrate=N and stereo unset, bitrate is N. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 96000; |
| SetSenderParameters(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(2u, spec.format.num_channels); |
| EXPECT_EQ(48000, spec.format.clockrate_hz); |
| } |
| |
| // Test that with bitrate=N and stereo=0, bitrate is N. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "0"; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| // Test that with bitrate=N and without any parameters, bitrate is N. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| // Test that with bitrate=N and stereo=1, bitrate is N. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "1"; |
| SetSenderParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithBitrates) { |
| SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", |
| 200000); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithHighMaxBitrate) { |
| SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, |
| SetSendCodecsWithoutBitratesUsesCorrectDefaults) { |
| SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) { |
| SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) { |
| SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", |
| 200000); |
| send_parameters_.max_bandwidth_bps = 100000; |
| // Setting max bitrate should keep previous min bitrate |
| // Setting max bitrate should not reset start bitrate. |
| EXPECT_CALL(*call_.GetMockTransportControllerSend(), |
| SetSdpBitrateParameters( |
| AllOf(Field(&BitrateConstraints::min_bitrate_bps, 100000), |
| Field(&BitrateConstraints::start_bitrate_bps, -1), |
| Field(&BitrateConstraints::max_bitrate_bps, 200000)))); |
| SetSenderParameters(send_parameters_); |
| } |
| |
| // Test that we can enable NACK with opus as callee. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioSenderParameter 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); |
| SetSenderParameters(parameters); |
| // NACK should be enabled even with no send stream. |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| |
| EXPECT_TRUE(send_channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Test that we can enable NACK on receive streams. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| SetSenderParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that we can disable NACK on receive streams. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); |
| SetSenderParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| |
| parameters.codecs.clear(); |
| parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(parameters); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that NACK is enabled on a new receive stream. |
| TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); |
| SetSenderParameters(parameters); |
| |
| 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 that we can switch back and forth between Opus and PCMU with CN. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsOpusPcmuSwitching) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| cricket::AudioSenderParameter opus_parameters; |
| opus_parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(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::AudioSenderParameter pcmu_parameters; |
| pcmu_parameters.codecs.push_back(kPcmuCodec); |
| pcmu_parameters.codecs.push_back(kCn16000Codec); |
| pcmu_parameters.codecs.push_back(kOpusCodec); |
| SetSenderParameters(pcmu_parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(0, spec.payload_type); |
| EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str()); |
| } |
| |
| SetSenderParameters(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_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSenderParameters(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 |
| SetSenderParameters(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 |
| SetSenderParameters(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 do not fail if no codecs are specified. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt); |
| } |
| |
| // Test that we can set send codecs even with telephone-event codec as the first |
| // one on the list. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSenderParameters(parameters); |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, spec.payload_type); |
| EXPECT_STRCASEEQ("OPUS", spec.format.name.c_str()); |
| SetSend(true); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| } |
| |
| // Test that CanInsertDtmf() is governed by the send flag |
| TEST_P(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSenderParameters(parameters); |
| EXPECT_FALSE(send_channel_->CanInsertDtmf()); |
| SetSend(true); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| SetSend(false); |
| EXPECT_FALSE(send_channel_->CanInsertDtmf()); |
| } |
| |
| // Test that payload type range is limited for telephone-event codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].id = 0; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSenderParameters(parameters); |
| SetSend(true); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = 128; // DTMF |
| EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_FALSE(send_channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = 127; |
| SetSenderParameters(parameters); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = -1; // DTMF |
| EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); |
| EXPECT_FALSE(send_channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we can set send codecs even with CN codec as the first |
| // one on the list. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // narrowband CN |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(0, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("PCMU", 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_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| 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; // narrowband CN |
| parameters.codecs[3].id = 98; // DTMF |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| SetSend(true); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we set VAD and DTMF types correctly as callee. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| 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; // narrowband CN |
| parameters.codecs[3].id = 98; // DTMF |
| SetSenderParameters(parameters); |
| EXPECT_TRUE(send_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("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| SetSend(true); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we only apply VAD if we have a CN codec that matches the |
| // send codec clockrate. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| // Set PCMU(8K) and CN(16K). VAD should not be activated. |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = 97; |
| SetSenderParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(std::nullopt, send_codec_spec.cng_payload_type); |
| } |
| // Set PCMU(8K) and CN(8K). VAD should be activated. |
| parameters.codecs[1] = kCn8000Codec; |
| SetSenderParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(13, send_codec_spec.cng_payload_type); |
| } |
| // Set OPUS(48K) and CN(8K). VAD should not be activated. |
| parameters.codecs[0] = kOpusCodec; |
| SetSenderParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("OPUS", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(std::nullopt, send_codec_spec.cng_payload_type); |
| } |
| } |
| |
| // Test that we perform case-insensitive matching of codec names. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSenderParameter parameters; |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs[0].name = "PcMu"; |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // narrowband CN |
| parameters.codecs[3].id = 98; // DTMF |
| SetSenderParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| SetSend(true); |
| EXPECT_TRUE(send_channel_->CanInsertDtmf()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, |
| SupportsTransportSequenceNumberHeaderExtension) { |
| const std::vector<webrtc::RtpExtension> header_extensions = |
| GetDefaultEnabledRtpHeaderExtensions(*engine_); |
| EXPECT_THAT(header_extensions, |
| Contains(::testing::Field( |
| "uri", &RtpExtension::uri, |
| webrtc::RtpExtension::kTransportSequenceNumberUri))); |
| } |
| |
| // Test support for audio level header extension. |
| TEST_P(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); |
| } |
| TEST_P(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); |
| } |
| |
| // Test support for transport sequence number header extension. |
| TEST_P(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions( |
| webrtc::RtpExtension::kTransportSequenceNumberUri); |
| } |
| TEST_P(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions( |
| webrtc::RtpExtension::kTransportSequenceNumberUri); |
| } |
| |
| // Test that we can create a channel and start sending on it. |
| TEST_P(WebRtcVoiceEngineTestFake, Send) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSenderParameters(send_parameters_); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that a channel is muted/unmuted. |
| TEST_P(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSenderParameters(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 SetSenderParameters() does not alter a stream's send state. |
| TEST_P(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)); |
| SetSenderParameters(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(); |
| SetSenderParameters(send_parameters_); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that we can create a channel and start playing out on it. |
| TEST_P(WebRtcVoiceEngineTestFake, Playout) { |
| EXPECT_TRUE(SetupRecvStream()); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); |
| receive_channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| receive_channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we can add and remove send streams. |
| TEST_P(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Set the global state for sending. |
| SetSend(true); |
| |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(send_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(send_channel_->RemoveSendStream(ssrc)); |
| EXPECT_FALSE(call_.GetAudioSendStream(ssrc)); |
| EXPECT_FALSE(send_channel_->RemoveSendStream(ssrc)); |
| } |
| EXPECT_EQ(0u, call_.GetAudioSendStreams().size()); |
| } |
| |
| // Test SetSendCodecs correctly configure the codecs in all send streams. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Create send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(send_channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| |
| cricket::AudioSenderParameter parameters; |
| // Set PCMU and CN(8K). VAD should be activated. |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs[1].id = 97; |
| SetSenderParameters(parameters); |
| |
| // Verify PCMU 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("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, 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; |
| parameters.codecs[1] = kCn16000Codec; |
| SetSenderParameters(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(std::nullopt, send_codec_spec.cng_payload_type); |
| } |
| } |
| |
| // Test we can SetSend on all send streams correctly. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Create the send channels and they should be a "not sending" date. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(send_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_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Create send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(send_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. |
| SetSenderParameters(send_parameters_); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); |
| SetAudioSendStreamStats(); |
| SetAudioReceiveStreamStats(); |
| |
| // Check stats for the added streams. |
| { |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| cricket::VoiceMediaSendInfo send_info; |
| cricket::VoiceMediaReceiveInfo receive_info; |
| EXPECT_EQ(true, send_channel_->GetStats(&send_info)); |
| EXPECT_EQ(true, receive_channel_->GetStats( |
| &receive_info, /*get_and_clear_legacy_stats=*/true)); |
| |
| // We have added 4 send streams. We should see empty stats for all. |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), |
| send_info.senders.size()); |
| for (const auto& sender : send_info.senders) { |
| VerifyVoiceSenderInfo(sender, false); |
| } |
| VerifyVoiceSendRecvCodecs(send_info, receive_info); |
| |
| // We have added one receive stream. We should see empty stats. |
| EXPECT_EQ(receive_info.receivers.size(), 1u); |
| EXPECT_EQ(receive_info.receivers[0].ssrc(), 123u); |
| } |
| |
| // Remove the kSsrcY stream. No receiver stats. |
| { |
| cricket::VoiceMediaReceiveInfo receive_info; |
| cricket::VoiceMediaSendInfo send_info; |
| EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY)); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| EXPECT_EQ(true, send_channel_->GetStats(&send_info)); |
| EXPECT_EQ(true, receive_channel_->GetStats( |
| &receive_info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), |
| send_info.senders.size()); |
| EXPECT_EQ(0u, receive_info.receivers.size()); |
| } |
| |
| // Deliver a new packet - a default receive stream should be created and we |
| // should see stats again. |
| { |
| cricket::VoiceMediaSendInfo send_info; |
| cricket::VoiceMediaReceiveInfo receive_info; |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| SetAudioReceiveStreamStats(); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| EXPECT_EQ(true, send_channel_->GetStats(&send_info)); |
| EXPECT_EQ(true, receive_channel_->GetStats( |
| &receive_info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), |
| send_info.senders.size()); |
| EXPECT_EQ(1u, receive_info.receivers.size()); |
| VerifyVoiceReceiverInfo(receive_info.receivers[0]); |
| VerifyVoiceSendRecvCodecs(send_info, receive_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_P(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Start playout without a receive stream. |
| SetSenderParameters(send_parameters_); |
| receive_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. |
| receive_channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrcY).started()); |
| EXPECT_FALSE(GetRecvStream(kSsrcZ).started()); |
| |
| // Restart playout and make sure recv streams are played out. |
| receive_channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcY).started()); |
| EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); |
| |
| // Now remove the recv streams. |
| EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcZ)); |
| EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY)); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = true; |
| send_parameters_.options.audio_network_adaptor_config = {"1234"}; |
| SetSenderParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = true; |
| send_parameters_.options.audio_network_adaptor_config = {"1234"}; |
| SetSenderParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| cricket::AudioOptions options; |
| options.audio_network_adaptor = false; |
| SetAudioSend(kSsrcX, true, nullptr, &options); |
| EXPECT_EQ(std::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = true; |
| send_parameters_.options.audio_network_adaptor_config = {"1234"}; |
| SetSenderParameters(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 = std::nullopt; |
| // 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)); |
| } |
| |
| // Test that we can set the outgoing SSRC properly. |
| // SSRC is set in SetupSendStream() by calling AddSendStream. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrc) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); |
| } |
| |
| TEST_P(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)); |
| SetSenderParameters(send_parameters_); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); |
| SetAudioSendStreamStats(); |
| |
| // Check stats for the added streams. |
| { |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| cricket::VoiceMediaSendInfo send_info; |
| cricket::VoiceMediaReceiveInfo receive_info; |
| EXPECT_EQ(true, send_channel_->GetStats(&send_info)); |
| EXPECT_EQ(true, receive_channel_->GetStats( |
| &receive_info, /*get_and_clear_legacy_stats=*/true)); |
| |
| // We have added one send stream. We should see the stats we've set. |
| EXPECT_EQ(1u, send_info.senders.size()); |
| VerifyVoiceSenderInfo(send_info.senders[0], false); |
| // We have added one receive stream. We should see empty stats. |
| EXPECT_EQ(receive_info.receivers.size(), 1u); |
| EXPECT_EQ(receive_info.receivers[0].ssrc(), 0u); |
| } |
| |
| // Start sending - this affects some reported stats. |
| { |
| SetSend(true); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| cricket::VoiceMediaSendInfo send_info; |
| cricket::VoiceMediaReceiveInfo receive_info; |
| SetAudioReceiveStreamStats(); |
| EXPECT_EQ(true, send_channel_->GetStats(&send_info)); |
| EXPECT_EQ(true, receive_channel_->GetStats( |
| &receive_info, /*get_and_clear_legacy_stats=*/true)); |
| VerifyVoiceSenderInfo(send_info.senders[0], true); |
| VerifyVoiceSendRecvCodecs(send_info, receive_info); |
| } |
| |
| // Remove the kSsrcY stream. No receiver stats. |
| { |
| EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY)); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| cricket::VoiceMediaSendInfo send_info; |
| cricket::VoiceMediaReceiveInfo receive_info; |
| EXPECT_EQ(true, send_channel_->GetStats(&send_info)); |
| EXPECT_EQ(true, receive_channel_->GetStats( |
| &receive_info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(1u, send_info.senders.size()); |
| EXPECT_EQ(0u, receive_info.receivers.size()); |
| } |
| |
| // Deliver a new packet - a default receive stream should be created and we |
| // should see stats again. |
| { |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| SetAudioReceiveStreamStats(); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| cricket::VoiceMediaSendInfo send_info; |
| cricket::VoiceMediaReceiveInfo receive_info; |
| EXPECT_EQ(true, send_channel_->GetStats(&send_info)); |
| EXPECT_EQ(true, receive_channel_->GetStats( |
| &receive_info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(1u, send_info.senders.size()); |
| EXPECT_EQ(1u, receive_info.receivers.size()); |
| VerifyVoiceReceiverInfo(receive_info.receivers[0]); |
| VerifyVoiceSendRecvCodecs(send_info, receive_info); |
| } |
| } |
| |
| // Test that we can set the outgoing SSRC properly with multiple streams. |
| // SSRC is set in SetupSendStream() by calling AddSendStream. |
| TEST_P(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_P(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_TRUE(send_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_P(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_P(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(receive_channel_->RemoveRecvStream(ssrc3)); |
| EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc2)); |
| EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc1)); |
| } |
| |
| // Test that receiving on an unsignaled stream works (a stream is created). |
| TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaled) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size()); |
| |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE( |
| GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); |
| } |
| |
| // Tests that when we add a stream without SSRCs, but contains a stream_id |
| // that it is stored and its stream id is later used when the first packet |
| // arrives to properly create a receive stream with a sync label. |
| TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) { |
| const char kSyncLabel[] = "sync_label"; |
| EXPECT_TRUE(SetupChannel()); |
| cricket::StreamParams unsignaled_stream; |
| unsignaled_stream.set_stream_ids({kSyncLabel}); |
| ASSERT_TRUE(receive_channel_->AddRecvStream(unsignaled_stream)); |
| // The stream shouldn't have been created at this point because it doesn't |
| // have any SSRCs. |
| EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size()); |
| |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE( |
| GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); |
| EXPECT_EQ(kSyncLabel, GetRecvStream(kSsrc1).GetConfig().sync_group); |
| |
| // Remset the unsignaled stream to clear the cached parameters. If a new |
| // default unsignaled receive stream is created it will not have a sync group. |
| receive_channel_->ResetUnsignaledRecvStream(); |
| receive_channel_->RemoveRecvStream(kSsrc1); |
| |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE( |
| GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); |
| EXPECT_TRUE(GetRecvStream(kSsrc1).GetConfig().sync_group.empty()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, |
| ResetUnsignaledRecvStreamDeletesAllDefaultStreams) { |
| ASSERT_TRUE(SetupChannel()); |
| // No receive streams to start with. |
| ASSERT_TRUE(call_.GetAudioReceiveStreams().empty()); |
| |
| // Deliver a couple packets with unsignaled SSRCs. |
| unsigned char packet[sizeof(kPcmuFrame)]; |
| memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); |
| rtc::SetBE32(&packet[8], 0x1234); |
| DeliverPacket(packet, sizeof(packet)); |
| rtc::SetBE32(&packet[8], 0x5678); |
| DeliverPacket(packet, sizeof(packet)); |
| |
| // Verify that the receive streams were created. |
| const auto& receivers1 = call_.GetAudioReceiveStreams(); |
| ASSERT_EQ(receivers1.size(), 2u); |
| |
| // Should remove all default streams. |
| receive_channel_->ResetUnsignaledRecvStream(); |
| const auto& receivers2 = call_.GetAudioReceiveStreams(); |
| EXPECT_EQ(0u, receivers2.size()); |
| } |
| |
| // Test that receiving N unsignaled stream works (streams will be created), and |
| // that packets are forwarded to them all. |
| TEST_P(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_P(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(1u, 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(2u, 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(2u, 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_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) { |
| EXPECT_TRUE(SetupChannel()); |
| |
| // Spawn unsignaled stream with SSRC=1. |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_EQ(1u, 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(1u, streams.size()); |
| int audio_receive_stream_id = streams.front()->id(); |
| EXPECT_TRUE(AddRecvStream(1)); |
| EXPECT_EQ(1u, streams.size()); |
| EXPECT_EQ(audio_receive_stream_id, streams.front()->id()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Updates) { |
| EXPECT_TRUE(SetupChannel()); |
| |
| // Spawn unsignaled stream with SSRC=1. |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE( |
| GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); |
| |
| // Verify that the underlying stream object in Call gets updated when a |
| // stream with SSRC=1 is added, and which has changed stream parameters. |
| const auto& streams = call_.GetAudioReceiveStreams(); |
| EXPECT_EQ(1u, streams.size()); |
| // The sync_group id should be empty. |
| EXPECT_TRUE(streams.front()->GetConfig().sync_group.empty()); |
| |
| const std::string new_stream_id("stream_id"); |
| int audio_receive_stream_id = streams.front()->id(); |
| cricket::StreamParams stream_params; |
| stream_params.ssrcs.push_back(1); |
| stream_params.set_stream_ids({new_stream_id}); |
| |
| EXPECT_TRUE(receive_channel_->AddRecvStream(stream_params)); |
| EXPECT_EQ(1u, streams.size()); |
| // The audio receive stream should not have been recreated. |
| EXPECT_EQ(audio_receive_stream_id, streams.front()->id()); |
| |
| // The sync_group id should now match with the new stream params. |
| EXPECT_EQ(new_stream_id, streams.front()->GetConfig().sync_group); |
| } |
| |
| // Test that AddRecvStream creates new stream. |
| TEST_P(WebRtcVoiceEngineTestFake, AddRecvStream) { |
| EXPECT_TRUE(SetupRecvStream()); |
| EXPECT_TRUE(AddRecvStream(1)); |
| } |
| |
| // Test that after adding a recv stream, we do not decode more codecs than |
| // those previously passed into SetRecvCodecs. |
| TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, {111, {"OPUS", 48000, 2}}}))); |
| } |
| |
| // Test that we properly clean up any streams that were added, even if |
| // not explicitly removed. |
| TEST_P(WebRtcVoiceEngineTestFake, StreamCleanup) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSenderParameters(send_parameters_); |
| EXPECT_TRUE(AddRecvStream(1)); |
| EXPECT_TRUE(AddRecvStream(2)); |
| |
| EXPECT_EQ(1u, call_.GetAudioSendStreams().size()); |
| EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); |
| send_channel_.reset(); |
| receive_channel_.reset(); |
| EXPECT_EQ(0u, call_.GetAudioSendStreams().size()); |
| EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, TestAddRecvStreamSuccessWithZeroSsrc) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(0)); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithSameSsrc) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(1)); |
| EXPECT_FALSE(AddRecvStream(1)); |
| } |
| |
| // Test the InsertDtmf on default send stream as caller. |
| TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) { |
| TestInsertDtmf(0, true, kTelephoneEventCodec1); |
| } |
| |
| // Test the InsertDtmf on default send stream as callee |
| TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) { |
| TestInsertDtmf(0, false, kTelephoneEventCodec2); |
| } |
| |
| // Test the InsertDtmf on specified send stream as caller. |
| TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) { |
| TestInsertDtmf(kSsrcX, true, kTelephoneEventCodec2); |
| } |
| |
| // Test the InsertDtmf on specified send stream as callee. |
| TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) { |
| TestInsertDtmf(kSsrcX, false, kTelephoneEventCodec1); |
| } |
| |
| // Test propagation of extmap allow mixed setting. |
| TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCaller) { |
| TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true); |
| } |
| TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCaller) { |
| TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false); |
| } |
| TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCallee) { |
| TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true); |
| } |
| TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCallee) { |
| TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetAudioOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_CALL(*adm_, BuiltInAECIsAvailable()) |
| .Times(8) |
| .WillRepeatedly(Return(false)); |
| EXPECT_CALL(*adm_, BuiltInAGCIsAvailable()) |
| .Times(4) |
| .WillRepeatedly(Return(false)); |
| EXPECT_CALL(*adm_, BuiltInNSIsAvailable()) |
| .Times(2) |
| .WillRepeatedly(Return(false)); |
| |
| EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); |
| EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); |
| |
| // Nothing set in AudioOptions, so everything should be as default. |
| send_parameters_.options = cricket::AudioOptions(); |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_TRUE(IsHighPassFilterEnabled()); |
| } |
| EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); |
| EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); |
| |
| // Turn echo cancellation off |
| send_parameters_.options.echo_cancellation = false; |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/false); |
| } |
| |
| // Turn echo cancellation back on, with settings, and make sure |
| // nothing else changed. |
| send_parameters_.options.echo_cancellation = true; |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| } |
| |
| // Turn off echo cancellation and delay agnostic aec. |
| send_parameters_.options.echo_cancellation = false; |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/false); |
| } |
| |
| // Restore AEC to be on to work with the following tests. |
| send_parameters_.options.echo_cancellation = true; |
| SetSenderParameters(send_parameters_); |
| |
| // Turn off AGC |
| send_parameters_.options.auto_gain_control = false; |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_FALSE(apm_config_.gain_controller1.enabled); |
| } |
| |
| // Turn AGC back on |
| send_parameters_.options.auto_gain_control = true; |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_TRUE(apm_config_.gain_controller1.enabled); |
| } |
| |
| // Turn off other options. |
| send_parameters_.options.noise_suppression = false; |
| send_parameters_.options.highpass_filter = false; |
| send_parameters_.options.stereo_swapping = true; |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_FALSE(IsHighPassFilterEnabled()); |
| EXPECT_TRUE(apm_config_.gain_controller1.enabled); |
| EXPECT_FALSE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| } |
| |
| // Set options again to ensure it has no impact. |
| SetSenderParameters(send_parameters_); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_TRUE(apm_config_.gain_controller1.enabled); |
| EXPECT_FALSE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| } |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, InitRecordingOnSend) { |
| EXPECT_CALL(*adm_, RecordingIsInitialized()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm_, Recording()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm_, InitRecording()).Times(1); |
| |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel( |
| engine_->CreateSendChannel( |
| &call_, cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create())); |
| |
| send_channel->SetSend(true); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SkipInitRecordingOnSend) { |
| EXPECT_CALL(*adm_, RecordingIsInitialized()).Times(0); |
| EXPECT_CALL(*adm_, Recording()).Times(0); |
| EXPECT_CALL(*adm_, InitRecording()).Times(0); |
| |
| cricket::AudioOptions options; |
| options.init_recording_on_send = false; |
| |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel( |
| engine_->CreateSendChannel(&call_, cricket::MediaConfig(), options, |
| webrtc::CryptoOptions(), |
| webrtc::AudioCodecPairId::Create())); |
| |
| send_channel->SetSend(true); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(*adm_, BuiltInAECIsAvailable()) |
| .Times(use_null_apm_ ? 4 : 8) |
| .WillRepeatedly(Return(false)); |
| EXPECT_CALL(*adm_, BuiltInAGCIsAvailable()) |
| .Times(use_null_apm_ ? 7 : 8) |
| .WillRepeatedly(Return(false)); |
| EXPECT_CALL(*adm_, BuiltInNSIsAvailable()) |
| .Times(use_null_apm_ ? 5 : 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)); |
| |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel1( |
| engine_->CreateSendChannel( |
| &call_, cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create())); |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel2( |
| engine_->CreateSendChannel( |
| &call_, cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create())); |
| |
| // Have to add a stream to make SetSend work. |
| cricket::StreamParams stream1; |
| stream1.ssrcs.push_back(1); |
| send_channel1->AddSendStream(stream1); |
| cricket::StreamParams stream2; |
| stream2.ssrcs.push_back(2); |
| send_channel2->AddSendStream(stream2); |
| |
| // AEC and AGC and NS |
| cricket::AudioSenderParameter parameters_options_all = send_parameters_; |
| parameters_options_all.options.echo_cancellation = true; |
| parameters_options_all.options.auto_gain_control = true; |
| parameters_options_all.options.noise_suppression = true; |
| EXPECT_TRUE(send_channel1->SetSenderParameters(parameters_options_all)); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| VerifyGainControlEnabledCorrectly(); |
| EXPECT_TRUE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| EXPECT_EQ(parameters_options_all.options, |
| SendImplFromPointer(send_channel1.get())->options()); |
| EXPECT_TRUE(send_channel2->SetSenderParameters(parameters_options_all)); |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| VerifyGainControlEnabledCorrectly(); |
| EXPECT_EQ(parameters_options_all.options, |
| SendImplFromPointer(send_channel2.get())->options()); |
| } |
| |
| // unset NS |
| cricket::AudioSenderParameter parameters_options_no_ns = send_parameters_; |
| parameters_options_no_ns.options.noise_suppression = false; |
| EXPECT_TRUE(send_channel1->SetSenderParameters(parameters_options_no_ns)); |
| cricket::AudioOptions expected_options = parameters_options_all.options; |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_FALSE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| VerifyGainControlEnabledCorrectly(); |
| expected_options.echo_cancellation = true; |
| expected_options.auto_gain_control = true; |
| expected_options.noise_suppression = false; |
| EXPECT_EQ(expected_options, |
| SendImplFromPointer(send_channel1.get())->options()); |
| } |
| |
| // unset AGC |
| cricket::AudioSenderParameter parameters_options_no_agc = send_parameters_; |
| parameters_options_no_agc.options.auto_gain_control = false; |
| EXPECT_TRUE(send_channel2->SetSenderParameters(parameters_options_no_agc)); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_FALSE(apm_config_.gain_controller1.enabled); |
| EXPECT_TRUE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| expected_options.echo_cancellation = true; |
| expected_options.auto_gain_control = false; |
| expected_options.noise_suppression = true; |
| EXPECT_EQ(expected_options, |
| SendImplFromPointer(send_channel2.get())->options()); |
| } |
| |
| EXPECT_TRUE(send_channel_->SetSenderParameters(parameters_options_all)); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| VerifyGainControlEnabledCorrectly(); |
| EXPECT_TRUE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| } |
| |
| send_channel1->SetSend(true); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| VerifyGainControlEnabledCorrectly(); |
| EXPECT_FALSE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| } |
| |
| send_channel2->SetSend(true); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_FALSE(apm_config_.gain_controller1.enabled); |
| EXPECT_TRUE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| } |
| |
| // Make sure settings take effect while we are sending. |
| cricket::AudioSenderParameter parameters_options_no_agc_nor_ns = |
| send_parameters_; |
| parameters_options_no_agc_nor_ns.options.auto_gain_control = false; |
| parameters_options_no_agc_nor_ns.options.noise_suppression = false; |
| EXPECT_TRUE( |
| send_channel2->SetSenderParameters(parameters_options_no_agc_nor_ns)); |
| if (!use_null_apm_) { |
| VerifyEchoCancellationSettings(/*enabled=*/true); |
| EXPECT_FALSE(apm_config_.gain_controller1.enabled); |
| EXPECT_FALSE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| expected_options.echo_cancellation = true; |
| expected_options.auto_gain_control = false; |
| expected_options.noise_suppression = false; |
| EXPECT_EQ(expected_options, |
| SendImplFromPointer(send_channel2.get())->options()); |
| } |
| } |
| |
| // This test verifies DSCP settings are properly applied on voice media channel. |
| TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::FakeNetworkInterface network_interface; |
| cricket::MediaConfig config; |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> channel; |
| webrtc::RtpParameters parameters; |
| |
| channel = engine_->CreateSendChannel(&call_, config, cricket::AudioOptions(), |
| webrtc::CryptoOptions(), |
| webrtc::AudioCodecPairId::Create()); |
| 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); |
| |
| config.enable_dscp = true; |
| channel = engine_->CreateSendChannel(&call_, config, cricket::AudioOptions(), |
| webrtc::CryptoOptions(), |
| webrtc::AudioCodecPairId::Create()); |
| channel->SetInterface(&network_interface); |
| EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); |
| |
| // Create a send stream to configure |
| EXPECT_TRUE( |
| channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ))); |
| parameters = channel->GetRtpSendParameters(kSsrcZ); |
| ASSERT_FALSE(parameters.encodings.empty()); |
| |
| // Various priorities map to various dscp values. |
| parameters.encodings[0].network_priority = webrtc::Priority::kHigh; |
| ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters, nullptr).ok()); |
| EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp()); |
| parameters.encodings[0].network_priority = webrtc::Priority::kVeryLow; |
| ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters, nullptr).ok()); |
| EXPECT_EQ(rtc::DSCP_CS1, network_interface.dscp()); |
| |
| // Packets should also self-identify their dscp in PacketOptions. |
| const uint8_t kData[10] = {0}; |
| EXPECT_TRUE(SendImplFromPointer(channel.get())->transport()->SendRtcp(kData)); |
| EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp); |
| channel->SetInterface(nullptr); |
| |
| // Verify that setting the option to false resets the |
| // DiffServCodePoint. |
| config.enable_dscp = false; |
| channel = engine_->CreateSendChannel(&call_, config, cricket::AudioOptions(), |
| webrtc::CryptoOptions(), |
| webrtc::AudioCodecPairId::Create()); |
| 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_P(WebRtcVoiceEngineTestFake, SetOutputVolume) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_FALSE(receive_channel_->SetOutputVolume(kSsrcY, 0.5)); |
| cricket::StreamParams stream; |
| stream.ssrcs.push_back(kSsrcY); |
| EXPECT_TRUE(receive_channel_->AddRecvStream(stream)); |
| EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrcY).gain()); |
| EXPECT_TRUE(receive_channel_->SetOutputVolume(kSsrcY, 3)); |
| EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcY).gain()); |
| } |
| |
| TEST_P(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(receive_channel_->SetDefaultOutputVolume(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 for all unsignaled streams. |
| EXPECT_TRUE(receive_channel_->SetDefaultOutputVolume(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(receive_channel_->SetOutputVolume(kSsrcX, 4)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain()); |
| } |
| EXPECT_DOUBLE_EQ(4, GetRecvStream(kSsrcX).gain()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, BaseMinimumPlayoutDelayMs) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_FALSE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 200)); |
| EXPECT_FALSE( |
| receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); |
| |
| cricket::StreamParams stream; |
| stream.ssrcs.push_back(kSsrcY); |
| EXPECT_TRUE(receive_channel_->AddRecvStream(stream)); |
| EXPECT_EQ(0, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms()); |
| EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 300)); |
| EXPECT_EQ(300, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, |
| BaseMinimumPlayoutDelayMsUnsignaledRecvStream) { |
| // Here base minimum delay is abbreviated to delay in comments for shortness. |
| EXPECT_TRUE(SetupChannel()); |
| |
| // Spawn an unsignaled stream by sending a packet - delay should be 0. |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| EXPECT_EQ( |
| 0, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1)); |
| // Check that it doesn't provide default values for unknown ssrc. |
| EXPECT_FALSE( |
| receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); |
| |
| // Check that default value for unsignaled streams is 0. |
| EXPECT_EQ( |
| 0, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1)); |
| |
| // Should remember the delay 100 which will be set on new unsignaled streams, |
| // and also set the delay to 100 on existing unsignaled streams. |
| EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 100)); |
| EXPECT_EQ( |
| 100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1)); |
| // Check that it doesn't provide default values for unknown ssrc. |
| EXPECT_FALSE( |
| receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); |
| |
| // Spawn an unsignaled stream by sending a packet - delay should be 100. |
| unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; |
| memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); |
| rtc::SetBE32(&pcmuFrame2[8], kSsrcX); |
| DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); |
| EXPECT_EQ( |
| 100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1)); |
| |
| // Setting delay with SSRC=0 should affect all unsignaled streams. |
| EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 300)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_EQ( |
| 300, |
| receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1)); |
| } |
| EXPECT_EQ( |
| 300, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1)); |
| |
| // Setting delay on an individual stream affects only that. |
| EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcX, 400)); |
| if (kMaxUnsignaledRecvStreams > 1) { |
| EXPECT_EQ( |
| 300, |
| receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1)); |
| } |
| EXPECT_EQ( |
| 400, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1)); |
| EXPECT_EQ( |
| 300, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1)); |
| // Check that it doesn't provide default values for unknown ssrc. |
| EXPECT_FALSE( |
| receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetsSyncGroupFromStreamId) { |
| const uint32_t kAudioSsrc = 123; |
| const std::string kStreamId = "AvSyncLabel"; |
| |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kAudioSsrc); |
| sp.set_stream_ids({kStreamId}); |
| // Creating two channels to make sure that sync label is set properly for both |
| // the default voice channel and following ones. |
| EXPECT_TRUE(receive_channel_->AddRecvStream(sp)); |
| sp.ssrcs[0] += 1; |
| EXPECT_TRUE(receive_channel_->AddRecvStream(sp)); |
| |
| ASSERT_EQ(2u, call_.GetAudioReceiveStreams().size()); |
| EXPECT_EQ(kStreamId, |
| call_.GetAudioReceiveStream(kAudioSsrc)->GetConfig().sync_group) |
| << "SyncGroup should be set based on stream id"; |
| EXPECT_EQ(kStreamId, |
| call_.GetAudioReceiveStream(kAudioSsrc + 1)->GetConfig().sync_group) |
| << "SyncGroup should be set based on stream id"; |
| } |
| |
| // TODO(solenberg): Remove, once recv streams are configured through Call. |
| // (This is then covered by TestSetRecvRtpHeaderExtensions.) |
| TEST_P(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()); |
| SetSenderParameters(send_parameters_); |
| for (uint32_t ssrc : ssrcs) { |
| EXPECT_TRUE(receive_channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| |
| EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); |
| for (uint32_t ssrc : ssrcs) { |
| EXPECT_THAT( |
| receive_channel_->GetRtpReceiverParameters(ssrc).header_extensions, |
| IsEmpty()); |
| } |
| |
| // Set up receive extensions. |
| const std::vector<webrtc::RtpExtension> header_extensions = |
| GetDefaultEnabledRtpHeaderExtensions(*engine_); |
| cricket::AudioReceiverParameters recv_parameters; |
| recv_parameters.extensions = header_extensions; |
| receive_channel_->SetReceiverParameters(recv_parameters); |
| EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); |
| for (uint32_t ssrc : ssrcs) { |
| EXPECT_THAT( |
| receive_channel_->GetRtpReceiverParameters(ssrc).header_extensions, |
| testing::UnorderedElementsAreArray(header_extensions)); |
| } |
| |
| // Disable receive extensions. |
| receive_channel_->SetReceiverParameters(cricket::AudioReceiverParameters()); |
| for (uint32_t ssrc : ssrcs) { |
| EXPECT_THAT( |
| receive_channel_->GetRtpReceiverParameters(ssrc).header_extensions, |
| IsEmpty()); |
| } |
| } |
| |
| TEST_P(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::VoiceMediaReceiveChannelInterface* media_channel = ReceiveImpl(); |
| SetSenderParameters(send_parameters_); |
| EXPECT_TRUE(media_channel->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(kAudioSsrc))); |
| |
| EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); |
| const cricket::FakeAudioReceiveStream* s = |
| call_.GetAudioReceiveStream(kAudioSsrc); |
| EXPECT_EQ(0, s->received_packets()); |
| webrtc::RtpPacketReceived parsed_packet; |
| RTC_CHECK(parsed_packet.Parse(kPcmuPacket)); |
| receive_channel_->OnPacketReceived(parsed_packet); |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| EXPECT_EQ(1, s->received_packets()); |
| } |
| |
| // All receive channels should be associated with the first send channel, |
| // since they do not send RTCP SR. |
| TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); |
| EXPECT_TRUE(send_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_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) { |
| EXPECT_TRUE(SetupRecvStream()); |
| EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); |
| EXPECT_TRUE(send_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(send_channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcW))); |
| |
| EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); |
| EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc); |
| } |
| |
| TEST_P(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. |
| receive_channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_1)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); |
| |
| // Now try actually setting the sink. |
| receive_channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_2)); |
| EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); |
| |
| // Now try resetting it. |
| receive_channel_->SetRawAudioSink(kSsrcX, nullptr); |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); |
| } |
| |
| TEST_P(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. |
| receive_channel_->SetDefaultRawAudioSink(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. |
| receive_channel_->SetDefaultRawAudioSink(nullptr); |
| EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); |
| |
| // Try setting the default sink while the default stream exists. |
| receive_channel_->SetDefaultRawAudioSink(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(receive_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. |
| receive_channel_->SetDefaultRawAudioSink(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. |
| receive_channel_->SetDefaultRawAudioSink(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. |
| receive_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_P(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) { |
| EXPECT_TRUE(SetupChannel()); |
| |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::AUDIO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::VIDEO)); |
| |
| send_channel_->OnReadyToSend(false); |
| EXPECT_EQ(webrtc::kNetworkDown, |
| call_.GetNetworkState(webrtc::MediaType::AUDIO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| call_.GetNetworkState(webrtc::MediaType::VIDEO)); |
| |
| send_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_P(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) { |
| SetupRecvStream(); |
| receive_channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| |
| // Changing RTP header extensions will recreate the |
| // AudioReceiveStreamInterface. |
| cricket::AudioReceiverParameters parameters; |
| parameters.extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); |
| receive_channel_->SetReceiverParameters(parameters); |
| |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Tests when GetSources is called with non-existing ssrc, it will return an |
| // empty list of RtpSource without crashing. |
| TEST_P(WebRtcVoiceEngineTestFake, GetSourcesWithNonExistingSsrc) { |
| // Setup an recv stream with `kSsrcX`. |
| SetupRecvStream(); |
| cricket::WebRtcVoiceReceiveChannel* media_channel = ReceiveImpl(); |
| // Call GetSources with `kSsrcY` which doesn't exist. |
| std::vector<webrtc::RtpSource> sources = media_channel->GetSources(kSsrcY); |
| EXPECT_EQ(0u, sources.size()); |
| } |
| |
| // Tests that the library initializes and shuts down properly. |
| TEST(WebRtcVoiceEngineTest, StartupShutdown) { |
| rtc::AutoThread main_thread; |
| for (bool use_null_apm : {false, true}) { |
| // If the VoiceEngine wants to gather available codecs early, that's fine |
| // but we never want it to create a decoder at this stage. |
| Environment env = CreateEnvironment(); |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| scoped_refptr<AudioProcessing> apm = |
| use_null_apm ? nullptr : BuiltinAudioProcessingBuilder().Build(env); |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, |
| nullptr, env.field_trials()); |
| engine.Init(); |
| std::unique_ptr<Call> call = Call::Create(CallConfig(env)); |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel = |
| engine.CreateSendChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); |
| EXPECT_TRUE(send_channel); |
| std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> |
| receive_channel = engine.CreateReceiveChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); |
| EXPECT_TRUE(receive_channel); |
| } |
| } |
| |
| // Tests that reference counting on the external ADM is correct. |
| TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) { |
| rtc::AutoThread main_thread; |
| for (bool use_null_apm : {false, true}) { |
| Environment env = CreateEnvironment(); |
| auto adm = rtc::make_ref_counted< |
| ::testing::NiceMock<webrtc::test::MockAudioDeviceModule>>(); |
| { |
| scoped_refptr<AudioProcessing> apm = |
| use_null_apm ? nullptr : BuiltinAudioProcessingBuilder().Build(env); |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, |
| nullptr, env.field_trials()); |
| engine.Init(); |
| std::unique_ptr<Call> call = Call::Create(CallConfig(env)); |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel = |
| engine.CreateSendChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); |
| EXPECT_TRUE(send_channel); |
| std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> |
| receive_channel = engine.CreateReceiveChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); |
| EXPECT_TRUE(receive_channel); |
| } |
| // The engine/channel should have dropped their references. |
| EXPECT_EQ(adm.release()->Release(), |
| webrtc::RefCountReleaseStatus::kDroppedLastRef); |
| } |
| } |
| |
| // Verify the payload id of common audio codecs, including CN and G722. |
| TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) { |
| Environment env = CreateEnvironment(); |
| for (bool use_null_apm : {false, true}) { |
| // 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::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| scoped_refptr<AudioProcessing> apm = |
| use_null_apm ? nullptr : BuiltinAudioProcessingBuilder().Build(env); |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, |
| nullptr, env.field_trials()); |
| engine.Init(); |
| for (const cricket::Codec& codec : engine.send_codecs()) { |
| auto is_codec = [&codec](const char* name, int clockrate = 0) { |
| return absl::EqualsIgnoreCase(codec.name, name) && |
| (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("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::AutoThread main_thread; |
| for (bool use_null_apm : {false, true}) { |
| Environment env = CreateEnvironment(); |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| scoped_refptr<AudioProcessing> apm = |
| use_null_apm ? nullptr : BuiltinAudioProcessingBuilder().Build(env); |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, |
| nullptr, env.field_trials()); |
| engine.Init(); |
| std::unique_ptr<Call> call = Call::Create(CallConfig(env)); |
| |
| std::vector<std::unique_ptr<cricket::VoiceMediaSendChannelInterface>> |
| channels; |
| while (channels.size() < 32) { |
| std::unique_ptr<cricket::VoiceMediaSendChannelInterface> channel = |
| engine.CreateSendChannel( |
| call.get(), cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); |
| if (!channel) |
| break; |
| channels.emplace_back(std::move(channel)); |
| } |
| |
| EXPECT_EQ(channels.size(), 32u); |
| } |
| } |
| |
| // Test that we set our preferred codecs properly. |
| TEST(WebRtcVoiceEngineTest, SetRecvCodecs) { |
| rtc::AutoThread main_thread; |
| for (bool use_null_apm : {false, true}) { |
| Environment env = CreateEnvironment(); |
| // 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 |
| // SetReceiverParameters returns true. |
| // I think it will become clear once audio decoder injection is completed. |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| scoped_refptr<AudioProcessing> apm = |
| use_null_apm ? nullptr : BuiltinAudioProcessingBuilder().Build(env); |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm, nullptr, |
| env.field_trials()); |
| engine.Init(); |
| std::unique_ptr<Call> call = Call::Create(CallConfig(env)); |
| cricket::WebRtcVoiceReceiveChannel channel( |
| &engine, cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), call.get(), |
| webrtc::AudioCodecPairId::Create()); |
| cricket::AudioReceiverParameters parameters; |
| parameters.codecs = ReceiveCodecsWithId(engine); |
| EXPECT_TRUE(channel.SetReceiverParameters(parameters)); |
| } |
| } |
| |
| TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) { |
| rtc::AutoThread main_thread; |
| Environment env = CreateEnvironment(); |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| FakeAudioSource source; |
| cricket::WebRtcVoiceEngine engine(&env.task_queue_factory(), adm.get(), |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| nullptr, nullptr, nullptr, |
| env.field_trials()); |
| engine.Init(); |
| CallConfig call_config(env); |
| { |
| webrtc::AudioState::Config config; |
| config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
| config.audio_device_module = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| call_config.audio_state = webrtc::AudioState::Create(config); |
| } |
| std::unique_ptr<Call> call = Call::Create(std::move(call_config)); |
| cricket::WebRtcVoiceSendChannel channel( |
| &engine, cricket::MediaConfig(), cricket::AudioOptions(), |
| webrtc::CryptoOptions(), call.get(), webrtc::AudioCodecPairId::Create()); |
| { |
| cricket::AudioSenderParameter params; |
| params.codecs.push_back(cricket::CreateAudioCodec(1, "opus", 48000, 2)); |
| params.extensions.push_back(webrtc::RtpExtension( |
| webrtc::RtpExtension::kTransportSequenceNumberUri, 1)); |
| EXPECT_TRUE(channel.SetSenderParameters(params)); |
| } |
| constexpr int kSsrc = 1234; |
| { |
| cricket::StreamParams params; |
| params.add_ssrc(kSsrc); |
| channel.AddSendStream(params); |
| } |
| channel.SetAudioSend(kSsrc, true, nullptr, &source); |
| channel.SetSend(true); |
| webrtc::RtpParameters params = channel.GetRtpSendParameters(kSsrc); |
| for (int max_bitrate : {-10, -1, 0, 10000}) { |
| params.encodings[0].max_bitrate_bps = max_bitrate; |
| channel.SetRtpSendParameters( |
| kSsrc, params, [](webrtc::RTCError error) { EXPECT_TRUE(error.ok()); }); |
| } |
| } |
| |
| TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) { |
| Environment env = CreateEnvironment(); |
| for (bool use_null_apm : {false, true}) { |
| 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 = |
| rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>(); |
| EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders()) |
| .WillOnce(Return(specs)); |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| |
| scoped_refptr<AudioProcessing> apm = |
| use_null_apm ? nullptr : BuiltinAudioProcessingBuilder().Build(env); |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), unused_encoder_factory, |
| mock_decoder_factory, nullptr, apm, nullptr, env.field_trials()); |
| engine.Init(); |
| auto codecs = engine.recv_codecs(); |
| EXPECT_EQ(11u, 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::Codec& { |
| static const cricket::Codec missing_codec = |
| cricket::CreateAudioCodec(0, "<missing>", 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::Codec& codec = codecs[i]; |
| if (absl::EqualsIgnoreCase(codec.name, format.name) && |
| 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); |
| } |
| } |
| |
| TEST(WebRtcVoiceEngineTest, CollectRecvCodecsWithLatePtAssignment) { |
| webrtc::test::ScopedKeyValueConfig field_trials( |
| "WebRTC-PayloadTypesInTransport/Enabled/"); |
| Environment env = CreateEnvironment(&field_trials); |
| |
| for (bool use_null_apm : {false, true}) { |
| 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 = |
| rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>(); |
| EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders()) |
| .WillOnce(Return(specs)); |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateNice(); |
| |
| scoped_refptr<AudioProcessing> apm = |
| use_null_apm ? nullptr : BuiltinAudioProcessingBuilder().Build(env); |
| cricket::WebRtcVoiceEngine engine( |
| &env.task_queue_factory(), adm.get(), unused_encoder_factory, |
| mock_decoder_factory, nullptr, apm, nullptr, env.field_trials()); |
| engine.Init(); |
| auto codecs = engine.recv_codecs(); |
| EXPECT_EQ(11u, 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::Codec& { |
| static const cricket::Codec missing_codec = |
| cricket::CreateAudioCodec(0, "<missing>", 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::Codec& codec = codecs[i]; |
| if (absl::EqualsIgnoreCase(codec.name, format.name) && |
| 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); |
| } |
| } |
