| /* |
| * 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 <memory> |
| #include <utility> |
| |
| #include "absl/memory/memory.h" |
| #include "absl/strings/match.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/rtp_parameters.h" |
| #include "api/scoped_refptr.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "api/transport/field_trial_based_config.h" |
| #include "call/call.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_constants.h" |
| #include "media/engine/fake_webrtc_call.h" |
| #include "modules/audio_device/include/mock_audio_device.h" |
| #include "modules/audio_processing/include/mock_audio_processing.h" |
| #include "rtc_base/arraysize.h" |
| #include "rtc_base/byte_order.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "test/field_trial.h" |
| #include "test/gtest.h" |
| #include "test/mock_audio_decoder_factory.h" |
| #include "test/mock_audio_encoder_factory.h" |
| |
| using ::testing::_; |
| using ::testing::ContainerEq; |
| using ::testing::Contains; |
| using ::testing::Field; |
| using ::testing::Return; |
| using ::testing::ReturnPointee; |
| using ::testing::SaveArg; |
| using ::testing::StrictMock; |
| |
| namespace { |
| using webrtc::BitrateConstraints; |
| |
| constexpr uint32_t kMaxUnsignaledRecvStreams = 4; |
| |
| const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1); |
| const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1); |
| const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 32000, 2); |
| const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1); |
| const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1); |
| const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1); |
| const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1); |
| const cricket::AudioCodec kRed48000Codec(112, "RED", 48000, 32000, 2); |
| const cricket::AudioCodec kTelephoneEventCodec1(106, |
| "telephone-event", |
| 8000, |
| 0, |
| 1); |
| const cricket::AudioCodec kTelephoneEventCodec2(107, |
| "telephone-event", |
| 32000, |
| 0, |
| 1); |
| |
| const uint32_t kSsrc0 = 0; |
| const uint32_t kSsrc1 = 1; |
| const uint32_t kSsrcX = 0x99; |
| const uint32_t kSsrcY = 0x17; |
| const uint32_t kSsrcZ = 0x42; |
| const uint32_t kSsrcW = 0x02; |
| const uint32_t kSsrcs4[] = {11, 200, 30, 44}; |
| |
| constexpr int kRtpHistoryMs = 5000; |
| |
| 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)); |
| } |
| } // namespace |
| |
| // Tests that our stub library "works". |
| TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) { |
| for (bool use_null_apm : {false, true}) { |
| std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = |
| webrtc::CreateDefaultTaskQueueFactory(); |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = |
| webrtc::test::MockAudioDeviceModule::CreateStrict(); |
| AdmSetupExpectations(adm); |
| rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm = |
| use_null_apm ? nullptr |
| : new rtc::RefCountedObject< |
| 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()); |
| } |
| { |
| webrtc::FieldTrialBasedConfig trials; |
| cricket::WebRtcVoiceEngine engine( |
| task_queue_factory.get(), adm, |
| webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), |
| webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, |
| nullptr, 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() : WebRtcVoiceEngineTestFake("") {} |
| |
| explicit WebRtcVoiceEngineTestFake(const char* field_trials) |
| : use_null_apm_(GetParam()), |
| task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()), |
| adm_(webrtc::test::MockAudioDeviceModule::CreateStrict()), |
| apm_(use_null_apm_ |
| ? nullptr |
| : new rtc::RefCountedObject< |
| StrictMock<webrtc::test::MockAudioProcessing>>()), |
| call_(), |
| override_field_trials_(field_trials) { |
| // AudioDeviceModule. |
| AdmSetupExpectations(adm_); |
| |
| 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_.reset(new cricket::WebRtcVoiceEngine( |
| task_queue_factory_.get(), adm_, encoder_factory, decoder_factory, |
| nullptr, apm_, nullptr, trials_config_)); |
| 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(IsTypingDetectionEnabled()); |
| EXPECT_TRUE(apm_config_.noise_suppression.enabled); |
| EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); |
| VerifyGainControlEnabledCorrectly(); |
| VerifyGainControlDefaultSettings(); |
| } |
| } |
| |
| bool SetupChannel() { |
| channel_ = engine_->CreateMediaChannel(&call_, cricket::MediaConfig(), |
| cricket::AudioOptions(), |
| webrtc::CryptoOptions()); |
| return (channel_ != nullptr); |
| } |
| |
| 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 (!channel_->AddSendStream(sp)) { |
| return false; |
| } |
| if (!use_null_apm_) { |
| EXPECT_CALL(*apm_, set_output_will_be_muted(false)); |
| } |
| return channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_); |
| } |
| |
| bool AddRecvStream(uint32_t ssrc) { |
| EXPECT_TRUE(channel_); |
| return channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc)); |
| } |
| |
| void SetupForMultiSendStream() { |
| EXPECT_TRUE(SetupSendStream()); |
| // Remove stream added in Setup. |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrcX)); |
| // Verify the channel does not exist. |
| EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX)); |
| } |
| |
| void DeliverPacket(const void* data, int len) { |
| rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len); |
| channel_->OnPacketReceived(packet, /* packet_time_us */ -1); |
| rtc::Thread::Current()->ProcessMessages(0); |
| } |
| |
| void TearDown() override { delete channel_; } |
| |
| const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) { |
| const auto* send_stream = call_.GetAudioSendStream(ssrc); |
| EXPECT_TRUE(send_stream); |
| return *send_stream; |
| } |
| |
| const cricket::FakeAudioReceiveStream& GetRecvStream(uint32_t ssrc) { |
| const auto* recv_stream = call_.GetAudioReceiveStream(ssrc); |
| EXPECT_TRUE(recv_stream); |
| return *recv_stream; |
| } |
| |
| const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) { |
| return GetSendStream(ssrc).GetConfig(); |
| } |
| |
| const webrtc::AudioReceiveStream::Config& GetRecvStreamConfig(uint32_t ssrc) { |
| return GetRecvStream(ssrc).GetConfig(); |
| } |
| |
| void SetSend(bool enable) { |
| ASSERT_TRUE(channel_); |
| if (enable) { |
| EXPECT_CALL(*adm_, RecordingIsInitialized()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm_, Recording()).WillOnce(Return(false)); |
| EXPECT_CALL(*adm_, InitRecording()).WillOnce(Return(0)); |
| } |
| channel_->SetSend(enable); |
| } |
| |
| void SetSendParameters(const cricket::AudioSendParameters& params) { |
| ASSERT_TRUE(channel_); |
| EXPECT_TRUE(channel_->SetSendParameters(params)); |
| } |
| |
| void SetAudioSend(uint32_t ssrc, |
| bool enable, |
| cricket::AudioSource* source, |
| const cricket::AudioOptions* options = nullptr) { |
| ASSERT_TRUE(channel_); |
| if (!use_null_apm_) { |
| EXPECT_CALL(*apm_, set_output_will_be_muted(!enable)); |
| } |
| EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source)); |
| } |
| |
| void TestInsertDtmf(uint32_t ssrc, |
| bool caller, |
| const cricket::AudioCodec& codec) { |
| EXPECT_TRUE(SetupChannel()); |
| if (caller) { |
| // If this is a caller, local description will be applied and add the |
| // send stream. |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Test we can only InsertDtmf when the other side supports telephone-event. |
| SetSendParameters(send_parameters_); |
| SetSend(true); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111)); |
| send_parameters_.codecs.push_back(codec); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| |
| if (!caller) { |
| // If this is callee, there's no active send channel yet. |
| EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123)); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Check we fail if the ssrc is invalid. |
| EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111)); |
| |
| // Test send. |
| cricket::FakeAudioSendStream::TelephoneEvent telephone_event = |
| GetSendStream(kSsrcX).GetLatestTelephoneEvent(); |
| EXPECT_EQ(-1, telephone_event.payload_type); |
| EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123)); |
| telephone_event = GetSendStream(kSsrcX).GetLatestTelephoneEvent(); |
| EXPECT_EQ(codec.id, telephone_event.payload_type); |
| EXPECT_EQ(codec.clockrate, telephone_event.payload_frequency); |
| EXPECT_EQ(2, telephone_event.event_code); |
| EXPECT_EQ(123, telephone_event.duration_ms); |
| } |
| |
| void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) { |
| // For a caller, the answer will be applied in set remote description |
| // where SetSendParameters() is called. |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| send_parameters_.extmap_allow_mixed = extmap_allow_mixed; |
| SetSendParameters(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()); |
| channel_->SetExtmapAllowMixed(extmap_allow_mixed); |
| EXPECT_TRUE( |
| 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::AudioCodec& codec, |
| int max_bitrate, |
| bool expected_result, |
| int expected_bitrate) { |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| parameters.max_bandwidth_bps = max_bitrate; |
| if (expected_result) { |
| SetSendParameters(parameters); |
| } else { |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| EXPECT_EQ(expected_bitrate, GetCodecBitrate(kSsrcX)); |
| } |
| |
| // Sets the per-stream maximum bitrate limit for the specified SSRC. |
| bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) { |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrc); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| |
| parameters.encodings[0].max_bitrate_bps = bitrate; |
| return channel_->SetRtpSendParameters(ssrc, parameters).ok(); |
| } |
| |
| void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) { |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(codec); |
| send_parameters.max_bandwidth_bps = bitrate; |
| SetSendParameters(send_parameters); |
| } |
| |
| void CheckSendCodecBitrate(int32_t ssrc, |
| const char expected_name[], |
| int expected_bitrate) { |
| const auto& spec = GetSendStreamConfig(ssrc).send_codec_spec; |
| EXPECT_EQ(expected_name, spec->format.name); |
| EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps); |
| } |
| |
| absl::optional<int> GetCodecBitrate(int32_t ssrc) { |
| return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps; |
| } |
| |
| const absl::optional<std::string>& GetAudioNetworkAdaptorConfig( |
| int32_t ssrc) { |
| return GetSendStreamConfig(ssrc).audio_network_adaptor_config; |
| } |
| |
| void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec, |
| int global_max, |
| int stream_max, |
| bool expected_result, |
| int expected_codec_bitrate) { |
| // Clear the bitrate limit from the previous test case. |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcX, -1)); |
| |
| // Attempt to set the requested bitrate limits. |
| SetGlobalMaxBitrate(codec, global_max); |
| EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrcX, stream_max)); |
| |
| // Verify that reading back the parameters gives results |
| // consistent with the Set() result. |
| webrtc::RtpParameters resulting_parameters = |
| channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(1UL, resulting_parameters.encodings.size()); |
| EXPECT_EQ(expected_result ? stream_max : -1, |
| resulting_parameters.encodings[0].max_bitrate_bps); |
| |
| // Verify that the codec settings have the expected bitrate. |
| EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrcX)); |
| } |
| |
| void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps, |
| int expected_min_bitrate_bps, |
| const char* start_bitrate_kbps, |
| int expected_start_bitrate_bps, |
| const char* max_bitrate_kbps, |
| int expected_max_bitrate_bps) { |
| EXPECT_TRUE(SetupSendStream()); |
| auto& codecs = send_parameters_.codecs; |
| codecs.clear(); |
| codecs.push_back(kOpusCodec); |
| codecs[0].params[cricket::kCodecParamMinBitrate] = min_bitrate_kbps; |
| codecs[0].params[cricket::kCodecParamStartBitrate] = start_bitrate_kbps; |
| codecs[0].params[cricket::kCodecParamMaxBitrate] = max_bitrate_kbps; |
| 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)))); |
| |
| SetSendParameters(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)); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extensions stay off with an empty list of headers. |
| send_parameters_.extensions.clear(); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extension is set properly. |
| const int id = 1; |
| send_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(1u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetSendStreamConfig(kSsrcX).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id); |
| |
| // Ensure extension is set properly on new stream. |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY))); |
| EXPECT_NE(call_.GetAudioSendStream(kSsrcX), |
| call_.GetAudioSendStream(kSsrcY)); |
| EXPECT_EQ(1u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetSendStreamConfig(kSsrcY).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetSendStreamConfig(kSsrcY).rtp.extensions[0].id); |
| |
| // Ensure all extensions go back off with an empty list. |
| send_parameters_.codecs.push_back(kPcmuCodec); |
| send_parameters_.extensions.clear(); |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); |
| } |
| |
| void TestSetRecvRtpHeaderExtensions(const std::string& ext) { |
| EXPECT_TRUE(SetupRecvStream()); |
| |
| // Ensure extensions are off by default. |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure unknown extensions won't cause an error. |
| recv_parameters_.extensions.push_back( |
| webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extensions stay off with an empty list of headers. |
| recv_parameters_.extensions.clear(); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| |
| // Ensure extension is set properly. |
| const int id = 2; |
| recv_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].id); |
| |
| // Ensure extension is set properly on new stream. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| EXPECT_NE(call_.GetAudioReceiveStream(kSsrcX), |
| call_.GetAudioReceiveStream(kSsrcY)); |
| EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].id); |
| |
| // Ensure all extensions go back off with an empty list. |
| recv_parameters_.extensions.clear(); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size()); |
| } |
| |
| webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const { |
| webrtc::AudioSendStream::Stats stats; |
| stats.local_ssrc = 12; |
| stats.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 = 42; |
| 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; |
| stats.typing_noise_detected = true; |
| return stats; |
| } |
| void SetAudioSendStreamStats() { |
| for (auto* s : call_.GetAudioSendStreams()) { |
| s->SetStats(GetAudioSendStreamStats()); |
| } |
| } |
| void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info, |
| bool is_sending) { |
| const auto stats = GetAudioSendStreamStats(); |
| EXPECT_EQ(info.ssrc(), stats.local_ssrc); |
| EXPECT_EQ(info.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); |
| EXPECT_EQ(info.typing_noise_detected, |
| stats.typing_noise_detected && is_sending); |
| } |
| |
| webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const { |
| webrtc::AudioReceiveStream::Stats stats; |
| stats.remote_ssrc = 123; |
| stats.payload_bytes_rcvd = 456; |
| stats.header_and_padding_bytes_rcvd = 67; |
| stats.packets_rcvd = 768; |
| stats.packets_lost = 101; |
| stats.codec_name = "codec_name_recv"; |
| stats.codec_payload_type = 42; |
| 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.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_rcvd, stats.payload_bytes_rcvd); |
| EXPECT_EQ(info.header_and_padding_bytes_rcvd, |
| stats.header_and_padding_bytes_rcvd); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_rcvd), |
| stats.packets_rcvd); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(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.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::VoiceMediaInfo& info) const { |
| EXPECT_EQ(send_parameters_.codecs.size(), info.send_codecs.size()); |
| for (const cricket::AudioCodec& codec : send_parameters_.codecs) { |
| ASSERT_EQ(info.send_codecs.count(codec.id), 1U); |
| EXPECT_EQ(info.send_codecs.find(codec.id)->second, |
| codec.ToCodecParameters()); |
| } |
| EXPECT_EQ(recv_parameters_.codecs.size(), info.receive_codecs.size()); |
| for (const cricket::AudioCodec& codec : recv_parameters_.codecs) { |
| ASSERT_EQ(info.receive_codecs.count(codec.id), 1U); |
| EXPECT_EQ(info.receive_codecs.find(codec.id)->second, |
| codec.ToCodecParameters()); |
| } |
| } |
| |
| void VerifyGainControlEnabledCorrectly() { |
| EXPECT_TRUE(apm_config_.gain_controller1.enabled); |
| EXPECT_EQ(kDefaultAgcMode, apm_config_.gain_controller1.mode); |
| EXPECT_EQ(0, apm_config_.gain_controller1.analog_level_minimum); |
| EXPECT_EQ(255, apm_config_.gain_controller1.analog_level_maximum); |
| } |
| |
| 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; |
| } |
| |
| bool IsTypingDetectionEnabled() { |
| return apm_config_.voice_detection.enabled; |
| } |
| |
| protected: |
| const bool use_null_apm_; |
| std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_; |
| rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm_; |
| rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_; |
| cricket::FakeCall call_; |
| std::unique_ptr<cricket::WebRtcVoiceEngine> engine_; |
| cricket::VoiceMediaChannel* channel_ = nullptr; |
| cricket::AudioSendParameters send_parameters_; |
| cricket::AudioRecvParameters recv_parameters_; |
| FakeAudioSource fake_source_; |
| webrtc::AudioProcessing::Config apm_config_; |
| |
| private: |
| webrtc::test::ScopedFieldTrials override_field_trials_; |
| webrtc::FieldTrialBasedConfig trials_config_; |
| }; |
| |
| 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( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); |
| EXPECT_EQ(kSsrcX, config.rtp.ssrc); |
| EXPECT_EQ("", config.rtp.c_name); |
| EXPECT_EQ(0u, config.rtp.extensions.size()); |
| EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_), |
| config.send_transport); |
| } |
| |
| // Test that we can add a receive stream and that it has the correct defaults. |
| TEST_P(WebRtcVoiceEngineTestFake, CreateRecvStream) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| const webrtc::AudioReceiveStream::Config& config = |
| GetRecvStreamConfig(kSsrcX); |
| EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc); |
| EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc); |
| EXPECT_FALSE(config.rtp.transport_cc); |
| EXPECT_EQ(0u, config.rtp.extensions.size()); |
| EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_), |
| config.rtcp_send_transport); |
| EXPECT_EQ("", config.sync_group); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) { |
| const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs(); |
| bool opus_found = false; |
| for (const cricket::AudioCodec& codec : codecs) { |
| if (codec.name == "opus") { |
| EXPECT_TRUE(HasTransportCc(codec)); |
| opus_found = true; |
| } |
| } |
| EXPECT_TRUE(opus_found); |
| } |
| |
| // Test that we set our inbound codecs properly, including changing PT. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs[0].id = 106; // collide with existing CN 32k |
| parameters.codecs[2].id = 126; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {106, {"ISAC", 16000, 1}}, |
| {126, {"telephone-event", 8000, 1}}, |
| {107, {"telephone-event", 32000, 1}}}))); |
| } |
| |
| // Test that we fail to set an unknown inbound codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1)); |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| // Test that we fail if we have duplicate types in the inbound list. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = kIsacCodec.id; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| // Test that we can decode OPUS without stereo parameters. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {103, {"ISAC", 16000, 1}}, |
| {111, {"opus", 48000, 2}}}))); |
| } |
| |
| // Test that we can decode OPUS with stereo = 0. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[2].params["stereo"] = "0"; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {103, {"ISAC", 16000, 1}}, |
| {111, {"opus", 48000, 2, {{"stereo", "0"}}}}}))); |
| } |
| |
| // Test that we can decode OPUS with stereo = 1. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[2].params["stereo"] = "1"; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {103, {"ISAC", 16000, 1}}, |
| {111, {"opus", 48000, 2, {{"stereo", "1"}}}}}))); |
| } |
| |
| // Test that changes to recv codecs are applied to all streams. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs[0].id = 106; // collide with existing CN 32k |
| parameters.codecs[2].id = 126; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| for (const auto& ssrc : {kSsrcX, kSsrcY}) { |
| EXPECT_TRUE(AddRecvStream(ssrc)); |
| EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{0, {"PCMU", 8000, 1}}, |
| {106, {"ISAC", 16000, 1}}, |
| {126, {"telephone-event", 8000, 1}}, |
| {107, {"telephone-event", 32000, 1}}}))); |
| } |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].id = 106; // collide with existing CN 32k |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map; |
| ASSERT_EQ(1u, dm.count(106)); |
| EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106)); |
| } |
| |
| // Test that we can apply the same set of codecs again while playing. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| // Remapping a payload type to a different codec should fail. |
| parameters.codecs[0] = kOpusCodec; |
| parameters.codecs[0].id = kIsacCodec.id; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we can add a codec while playing. |
| TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| channel_->SetPlayout(true); |
| |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we accept adding the same codec with a different payload type. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847 |
| TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| ++parameters.codecs[0].id; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| // Test that we set Opus/Red under the field trial. |
| TEST_P(WebRtcVoiceEngineTestFake, RecvRed) { |
| webrtc::test::ScopedFieldTrials override_field_trials( |
| "WebRTC-Audio-Red-For-Opus/Enabled/"); |
| |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kRed48000Codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, |
| (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( |
| {{111, {"opus", 48000, 2}}, {112, {"red", 48000, 2}}}))); |
| } |
| |
| // Test that we do not allow setting Opus/Red by default. |
| TEST_P(WebRtcVoiceEngineTestFake, RecvRedDefault) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kRed48000Codec); |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| 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. |
| |
| // ISAC, default bitrate == 32000. |
| TestMaxSendBandwidth(kIsacCodec, 0, true, 32000); |
| |
| // PCMU, default bitrate == 64000. |
| TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000); |
| |
| // opus, default bitrate == 32000 in mono. |
| TestMaxSendBandwidth(kOpusCodec, -1, true, 32000); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // ISAC, default bitrate == 32000. |
| TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000); |
| // Rates above the max (56000) should be capped. |
| TestMaxSendBandwidth(kIsacCodec, 100000, true, 32000); |
| |
| // opus, default bitrate == 64000. |
| TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000); |
| TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000); |
| // Rates above the max (510000) should be capped. |
| TestMaxSendBandwidth(kOpusCodec, 600000, true, 510000); |
| } |
| |
| TEST_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::AudioSendParameters parameters; |
| parameters.codecs = engine_->send_codecs(); |
| parameters.max_bandwidth_bps = kDesiredBitrate; |
| SetSendParameters(parameters); |
| |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| |
| EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX)); |
| } |
| |
| // Test that bitrate cannot be set for CBR codecs. |
| // Bitrate is ignored if it is higher than the fixed bitrate. |
| // Bitrate less then the fixed bitrate is an error. |
| TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // PCMU, default bitrate == 64000. |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| |
| send_parameters_.max_bandwidth_bps = 128000; |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| |
| send_parameters_.max_bandwidth_bps = 128; |
| EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); |
| } |
| |
| // Test that the per-stream bitrate limit and the global |
| // bitrate limit both apply. |
| TEST_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 = |
| channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(0u, nonexistent_parameters.encodings.size()); |
| |
| nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE( |
| 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 = channel_->GetRtpSendParameters(kSsrcX); |
| // Two or more encodings should result in failure. |
| parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| // Zero encodings should also fail. |
| parameters.encodings.clear(); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| } |
| |
| // Changing the SSRC through RtpParameters is not allowed. |
| TEST_P(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); |
| parameters.encodings[0].ssrc = 0xdeadbeef; |
| EXPECT_FALSE(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 = channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_TRUE(parameters.encodings[0].active); |
| parameters.encodings[0].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).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 = absl::optional<int>(6000); |
| EXPECT_TRUE(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 = channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_FALSE(parameters.encodings[0].adaptive_ptime); |
| parameters.encodings[0].adaptive_ptime = true; |
| EXPECT_TRUE(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(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"}; |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); |
| parameters.encodings[0].adaptive_ptime = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, AdaptivePtimeFieldTrial) { |
| webrtc::test::ScopedFieldTrials override_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( |
| 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::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSendParameters(parameters); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); |
| EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); |
| } |
| |
| // Test that GetRtpSendParameters returns 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 = channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str()); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, |
| DetectRtpSendParameterHeaderExtensionsChange) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); |
| rtp_parameters.header_extensions.emplace_back(); |
| |
| EXPECT_NE(0u, rtp_parameters.header_extensions.size()); |
| |
| webrtc::RTCError result = |
| 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 = 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::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| SetSendParameters(parameters); |
| |
| webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrcX); |
| |
| // We should be able to set the params we just got. |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, initial_params).ok()); |
| |
| // ... And this shouldn't change the params returned by GetRtpSendParameters. |
| webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrcX); |
| EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrcX)); |
| } |
| |
| // Test that max_bitrate_bps in send stream config gets updated correctly when |
| // SetRtpSendParameters is called. |
| TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(send_parameters); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); |
| // Expect empty on parameters.encodings[0].max_bitrate_bps; |
| EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps); |
| |
| constexpr int kMaxBitrateBps = 6000; |
| rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps; |
| EXPECT_TRUE(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 = 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(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); |
| rtp_parameters.encodings[0].bitrate_priority = -1.0; |
| EXPECT_FALSE(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 = 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(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, |
| channel_->GetRtpSendParameters(kSsrcX).encodings[0].bitrate_priority); |
| EXPECT_EQ(new_bitrate_priority, GetSendStreamConfig(kSsrcX).bitrate_priority); |
| } |
| |
| // Test that GetRtpReceiveParameters returns the currently configured codecs. |
| TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpReceiveParameters(kSsrcX); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); |
| EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); |
| } |
| |
| // Test that GetRtpReceiveParameters returns an SSRC. |
| TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) { |
| EXPECT_TRUE(SetupRecvStream()); |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpReceiveParameters(kSsrcX); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc); |
| } |
| |
| // Test that if we set/get parameters multiple times, we get the same results. |
| TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| webrtc::RtpParameters initial_params = |
| channel_->GetRtpReceiveParameters(kSsrcX); |
| |
| // ... And this shouldn't change the params returned by |
| // GetRtpReceiveParameters. |
| webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrcX); |
| EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(kSsrcX)); |
| } |
| |
| // Test that GetRtpReceiveParameters returns parameters correctly when SSRCs |
| // aren't signaled. It should return an empty "RtpEncodingParameters" when |
| // configured to receive an unsignaled stream and no packets have been received |
| // yet, and start returning the SSRC once a packet has been received. |
| TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) { |
| ASSERT_TRUE(SetupChannel()); |
| // Call necessary methods to configure receiving a default stream as |
| // soon as it arrives. |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| // Call GetDefaultRtpReceiveParameters before configured to receive an |
| // unsignaled stream. Should return nothing. |
| EXPECT_EQ(webrtc::RtpParameters(), |
| channel_->GetDefaultRtpReceiveParameters()); |
| |
| // Set a sink for an unsignaled stream. |
| std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink()); |
| channel_->SetDefaultRawAudioSink(std::move(fake_sink)); |
| |
| // Call GetDefaultRtpReceiveParameters before the SSRC is known. |
| webrtc::RtpParameters rtp_parameters = |
| 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 = channel_->GetDefaultRtpReceiveParameters(); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); |
| } |
| |
| // Test that we apply codecs properly. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[0].bitrate = 22000; |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000); |
| EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we use Opus/Red under the field trial when it is |
| // listed as the first codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRed) { |
| webrtc::test::ScopedFieldTrials override_field_trials( |
| "WebRTC-Audio-Red-For-Opus/Enabled/"); |
| |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kRed48000Codec); |
| parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(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 under the field trial by default. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) { |
| webrtc::test::ScopedFieldTrials override_field_trials( |
| "WebRTC-Audio-Red-For-Opus/Enabled/"); |
| |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs.push_back(kRed48000Codec); |
| SetSendParameters(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(absl::nullopt, send_codec_spec.red_payload_type); |
| } |
| |
| // Test that WebRtcVoiceEngine reconfigures, rather than recreates its |
| // AudioSendStream. |
| TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[0].bitrate = 48000; |
| const int initial_num = call_.GetNumCreatedSendStreams(); |
| SetSendParameters(parameters); |
| EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); |
| // Calling SetSendCodec again with same codec which is already set. |
| // In this case media channel shouldn't send codec to VoE. |
| SetSendParameters(parameters); |
| EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); |
| } |
| |
| // TODO(ossu): Revisit if these tests need to be here, now that these kinds of |
| // tests should be available in AudioEncoderOpusTest. |
| |
| // Test that if clockrate is not 48000 for opus, we fail. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].clockrate = 50000; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channels=0 for opus, we fail. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 0; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channels=0 for opus, we fail. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channel is 1 for opus and there's no stereo, we fail. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channel is 1 for opus and stereo=0, we fail. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| parameters.codecs[0].params["stereo"] = "0"; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that if channel is 1 for opus and stereo=1, we fail. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].channels = 1; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that with bitrate=0 and no stereo, bitrate is 32000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 32000); |
| } |
| |
| // Test that with bitrate=0 and stereo=0, bitrate is 32000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "0"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 32000); |
| } |
| |
| // Test that with bitrate=invalid and stereo=0, bitrate is 32000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["stereo"] = "0"; |
| // bitrate that's out of the range between 6000 and 510000 will be clamped. |
| parameters.codecs[0].bitrate = 5999; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 6000); |
| |
| parameters.codecs[0].bitrate = 510001; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 510000); |
| } |
| |
| // Test that with bitrate=0 and stereo=1, bitrate is 64000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 64000); |
| } |
| |
| // Test that with bitrate=invalid and stereo=1, bitrate is 64000. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["stereo"] = "1"; |
| // bitrate that's out of the range between 6000 and 510000 will be clamped. |
| parameters.codecs[0].bitrate = 5999; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 6000); |
| |
| parameters.codecs[0].bitrate = 510001; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 510000); |
| } |
| |
| // Test that with bitrate=N and stereo unset, bitrate is N. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 96000; |
| SetSendParameters(parameters); |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_EQ(96000, spec.target_bitrate_bps); |
| EXPECT_EQ("opus", spec.format.name); |
| EXPECT_EQ(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::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "0"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| // Test that with bitrate=N and without any parameters, bitrate is N. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| // Test that with bitrate=N and stereo=1, bitrate is N. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "1"; |
| SetSendParameters(parameters); |
| CheckSendCodecBitrate(kSsrcX, "opus", 30000); |
| } |
| |
| TEST_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)))); |
| SetSendParameters(send_parameters_); |
| } |
| |
| // Test that we can enable NACK with opus as callee. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| SetSendParameters(parameters); |
| // NACK should be enabled even with no send stream. |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); |
| |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| } |
| |
| // Test that we can enable NACK on receive streams. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| SetSendParameters(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::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); |
| SetSendParameters(parameters); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); |
| |
| parameters.codecs.clear(); |
| parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(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::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); |
| SetSendParameters(parameters); |
| |
| EXPECT_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_P(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(send_parameters.codecs[0].feedback_params.params().empty()); |
| SetSendParameters(send_parameters); |
| |
| cricket::AudioRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(kIsacCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrcX)); |
| ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr); |
| EXPECT_FALSE( |
| call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc); |
| |
| send_parameters.codecs = engine_->send_codecs(); |
| SetSendParameters(send_parameters); |
| ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr); |
| EXPECT_TRUE( |
| call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc); |
| } |
| |
| // Test that we can switch back and forth between Opus and ISAC with CN. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| cricket::AudioSendParameters opus_parameters; |
| opus_parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(opus_parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); |
| } |
| |
| cricket::AudioSendParameters isac_parameters; |
| isac_parameters.codecs.push_back(kIsacCodec); |
| isac_parameters.codecs.push_back(kCn16000Codec); |
| isac_parameters.codecs.push_back(kOpusCodec); |
| SetSendParameters(isac_parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| } |
| |
| SetSendParameters(opus_parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); |
| } |
| } |
| |
| // Test that we handle various ways of specifying bitrate. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); // bitrate == 32000 |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| EXPECT_EQ(32000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| EXPECT_EQ(32000, spec.target_bitrate_bps); |
| } |
| parameters.codecs[0].bitrate = 28000; // bitrate == 28000 |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(103, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| EXPECT_EQ(28000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0] = kPcmuCodec; // bitrate == 64000 |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(0, spec.payload_type); |
| EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str()); |
| EXPECT_EQ(64000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(0, spec.payload_type); |
| EXPECT_STREQ("PCMU", spec.format.name.c_str()); |
| EXPECT_EQ(64000, spec.target_bitrate_bps); |
| } |
| |
| parameters.codecs[0] = kOpusCodec; |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| SetSendParameters(parameters); |
| { |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(111, spec.payload_type); |
| EXPECT_STREQ("opus", spec.format.name.c_str()); |
| EXPECT_EQ(32000, spec.target_bitrate_bps); |
| } |
| } |
| |
| // Test that we fail if no codecs are specified. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that we can set send codecs even with telephone-event codec as the first |
| // one on the list. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSendParameters(parameters); |
| const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); |
| SetSend(true); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that CanInsertDtmf() is governed by the send flag |
| TEST_P(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSendParameters(parameters); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| SetSend(true); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| SetSend(false); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that payload type range is limited for telephone-event codec. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].id = 0; // DTMF |
| parameters.codecs[1].id = 96; |
| SetSendParameters(parameters); |
| SetSend(true); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = 128; // DTMF |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = 127; |
| SetSendParameters(parameters); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| parameters.codecs[0].id = -1; // DTMF |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we can set send codecs even with CN codec as the first |
| // one on the list. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // wideband CN |
| parameters.codecs[1].id = 96; |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(98, send_codec_spec.cng_payload_type); |
| } |
| |
| // Test that we set VAD and DTMF types correctly as caller. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| // TODO(juberti): cn 32000 |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| SetSend(true); |
| EXPECT_TRUE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that we set VAD and DTMF types correctly as callee. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) { |
| EXPECT_TRUE(SetupChannel()); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| // TODO(juberti): cn 32000 |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kTelephoneEventCodec2); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| SetSendParameters(parameters); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); |
| |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| SetSend(true); |
| EXPECT_TRUE(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::AudioSendParameters parameters; |
| // Set ISAC(16K) and CN(16K). VAD should be activated. |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = 97; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| } |
| // Set PCMU(8K) and CN(16K). VAD should not be activated. |
| parameters.codecs[0] = kPcmuCodec; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type); |
| } |
| // Set PCMU(8K) and CN(8K). VAD should be activated. |
| parameters.codecs[1] = kCn8000Codec; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(13, send_codec_spec.cng_payload_type); |
| } |
| // Set ISAC(16K) and CN(8K). VAD should not be activated. |
| parameters.codecs[0] = kIsacCodec; |
| SetSendParameters(parameters); |
| { |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(absl::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::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs.push_back(kCn8000Codec); |
| parameters.codecs.push_back(kTelephoneEventCodec1); |
| parameters.codecs[0].name = "iSaC"; |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| SetSendParameters(parameters); |
| const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; |
| EXPECT_EQ(96, send_codec_spec.payload_type); |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(1u, send_codec_spec.format.num_channels); |
| EXPECT_EQ(97, send_codec_spec.cng_payload_type); |
| SetSend(true); |
| EXPECT_TRUE(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()); |
| SetSendParameters(send_parameters_); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that a channel will send if and only if it has a source and is enabled |
| // for sending. |
| TEST_P(WebRtcVoiceEngineTestFake, SendStateWithAndWithoutSource) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSendParameters(send_parameters_); |
| SetAudioSend(kSsrcX, true, nullptr); |
| SetSend(true); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| SetAudioSend(kSsrcX, true, &fake_source_); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| SetAudioSend(kSsrcX, true, nullptr); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that a channel is muted/unmuted. |
| TEST_P(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) { |
| EXPECT_TRUE(SetupSendStream()); |
| SetSendParameters(send_parameters_); |
| EXPECT_FALSE(GetSendStream(kSsrcX).muted()); |
| SetAudioSend(kSsrcX, true, nullptr); |
| EXPECT_FALSE(GetSendStream(kSsrcX).muted()); |
| SetAudioSend(kSsrcX, false, nullptr); |
| EXPECT_TRUE(GetSendStream(kSsrcX).muted()); |
| } |
| |
| // Test that SetSendParameters() does not alter a stream's send state. |
| TEST_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)); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Turn off sending. |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Changing RTP header extensions will recreate the AudioSendStream. |
| send_parameters_.extensions.clear(); |
| SetSendParameters(send_parameters_); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| } |
| |
| // Test that we can create a channel and start playing out on it. |
| TEST_P(WebRtcVoiceEngineTestFake, Playout) { |
| EXPECT_TRUE(SetupRecvStream()); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcX).started()); |
| channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrcX).started()); |
| } |
| |
| // Test that we can add and remove send streams. |
| TEST_P(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Set the global state for sending. |
| SetSend(true); |
| |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc))); |
| SetAudioSend(ssrc, true, &fake_source_); |
| // Verify that we are in a sending state for all the created streams. |
| EXPECT_TRUE(GetSendStream(ssrc).IsSending()); |
| } |
| EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size()); |
| |
| // Delete the send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(channel_->RemoveSendStream(ssrc)); |
| EXPECT_FALSE(call_.GetAudioSendStream(ssrc)); |
| EXPECT_FALSE(channel_->RemoveSendStream(ssrc)); |
| } |
| EXPECT_EQ(0u, call_.GetAudioSendStreams().size()); |
| } |
| |
| // Test SetSendCodecs correctly configure the codecs in all send streams. |
| TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Create send streams. |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| |
| cricket::AudioSendParameters parameters; |
| // Set ISAC(16K) and CN(16K). VAD should be activated. |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kCn16000Codec); |
| parameters.codecs[1].id = 97; |
| SetSendParameters(parameters); |
| |
| // Verify ISAC and VAD are corrected configured on all send channels. |
| for (uint32_t ssrc : kSsrcs4) { |
| ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); |
| const auto& send_codec_spec = |
| *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; |
| EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(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; |
| SetSendParameters(parameters); |
| for (uint32_t ssrc : kSsrcs4) { |
| ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); |
| const auto& send_codec_spec = |
| *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; |
| EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); |
| EXPECT_EQ(absl::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( |
| 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( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc))); |
| } |
| |
| // Create a receive stream to check that none of the send streams end up in |
| // the receive stream stats. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| |
| // We need send codec to be set to get all stats. |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| SetAudioSendStreamStats(); |
| |
| // Check stats for the added streams. |
| { |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| cricket::VoiceMediaInfo info; |
| EXPECT_EQ(true, |
| channel_->GetStats(&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)), info.senders.size()); |
| for (const auto& sender : info.senders) { |
| VerifyVoiceSenderInfo(sender, false); |
| } |
| VerifyVoiceSendRecvCodecs(info); |
| |
| // We have added one receive stream. We should see empty stats. |
| EXPECT_EQ(info.receivers.size(), 1u); |
| EXPECT_EQ(info.receivers[0].ssrc(), 0u); |
| } |
| |
| // Remove the kSsrcY stream. No receiver stats. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| EXPECT_EQ(true, |
| channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); |
| EXPECT_EQ(0u, info.receivers.size()); |
| } |
| |
| // Deliver a new packet - a default receive stream should be created and we |
| // should see stats again. |
| { |
| cricket::VoiceMediaInfo info; |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| SetAudioReceiveStreamStats(); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| EXPECT_EQ(true, |
| channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); |
| EXPECT_EQ(1u, info.receivers.size()); |
| VerifyVoiceReceiverInfo(info.receivers[0]); |
| VerifyVoiceSendRecvCodecs(info); |
| } |
| } |
| |
| // Test that we can add and remove receive streams, and do proper send/playout. |
| // We can receive on multiple streams while sending one stream. |
| TEST_P(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Start playout without a receive stream. |
| SetSendParameters(send_parameters_); |
| channel_->SetPlayout(true); |
| |
| // Adding another stream should enable playout on the new stream only. |
| EXPECT_TRUE(AddRecvStream(kSsrcY)); |
| SetSend(true); |
| EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Make sure only the new stream is played out. |
| EXPECT_TRUE(GetRecvStream(kSsrcY).started()); |
| |
| // Adding yet another stream should have stream 2 and 3 enabled for playout. |
| EXPECT_TRUE(AddRecvStream(kSsrcZ)); |
| EXPECT_TRUE(GetRecvStream(kSsrcY).started()); |
| EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); |
| |
| // Stop sending. |
| SetSend(false); |
| EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); |
| |
| // Stop playout. |
| channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrcY).started()); |
| EXPECT_FALSE(GetRecvStream(kSsrcZ).started()); |
| |
| // Restart playout and make sure recv streams are played out. |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrcY).started()); |
| EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); |
| |
| // Now remove the recv streams. |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcZ)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(*adm_, BuiltInAGCIsAvailable()) |
| .Times(::testing::AtLeast(1)) |
| .WillRepeatedly(Return(false)); |
| |
| if (!use_null_apm_) { |
| // Ensure default options. |
| VerifyGainControlEnabledCorrectly(); |
| VerifyGainControlDefaultSettings(); |
| } |
| |
| const auto& agc_config = apm_config_.gain_controller1; |
| |
| send_parameters_.options.auto_gain_control = false; |
| SetSendParameters(send_parameters_); |
| if (!use_null_apm_) { |
| EXPECT_FALSE(agc_config.enabled); |
| } |
| send_parameters_.options.auto_gain_control = absl::nullopt; |
| |
| send_parameters_.options.tx_agc_target_dbov = 5; |
| SetSendParameters(send_parameters_); |
| if (!use_null_apm_) { |
| EXPECT_EQ(5, agc_config.target_level_dbfs); |
| } |
| send_parameters_.options.tx_agc_target_dbov = absl::nullopt; |
| |
| send_parameters_.options.tx_agc_digital_compression_gain = 10; |
| SetSendParameters(send_parameters_); |
| if (!use_null_apm_) { |
| EXPECT_EQ(10, agc_config.compression_gain_db); |
| } |
| send_parameters_.options.tx_agc_digital_compression_gain = absl::nullopt; |
| |
| send_parameters_.options.tx_agc_limiter = false; |
| SetSendParameters(send_parameters_); |
| if (!use_null_apm_) { |
| EXPECT_FALSE(agc_config.enable_limiter); |
| } |
| send_parameters_.options.tx_agc_limiter = absl::nullopt; |
| |
| SetSendParameters(send_parameters_); |
| if (!use_null_apm_) { |
| // Expect all options to have been preserved. |
| EXPECT_FALSE(agc_config.enabled); |
| EXPECT_EQ(5, agc_config.target_level_dbfs); |
| EXPECT_EQ(10, agc_config.compression_gain_db); |
| EXPECT_FALSE(agc_config.enable_limiter); |
| } |
| } |
| |
| TEST_P(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| send_parameters_.options.audio_network_adaptor = true; |
| send_parameters_.options.audio_network_adaptor_config = {"1234"}; |
| SetSendParameters(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"}; |
| SetSendParameters(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(absl::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"}; |
| SetSendParameters(send_parameters_); |
| EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, |
| GetAudioNetworkAdaptorConfig(kSsrcX)); |
| const int initial_num = call_.GetNumCreatedSendStreams(); |
| cricket::AudioOptions options; |
| options.audio_network_adaptor = absl::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)); |
| } |
| |
| class WebRtcVoiceEngineWithSendSideBweWithOverheadTest |
| : public WebRtcVoiceEngineTestFake { |
| public: |
| WebRtcVoiceEngineWithSendSideBweWithOverheadTest() |
| : WebRtcVoiceEngineTestFake( |
| "WebRTC-Audio-Allocation/min:6000bps,max:32000bps/") {} |
| }; |
| |
| // 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)); |
| SetSendParameters(send_parameters_); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| SetAudioSendStreamStats(); |
| |
| // Check stats for the added streams. |
| { |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| cricket::VoiceMediaInfo info; |
| EXPECT_EQ(true, |
| channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true)); |
| |
| // We have added one send stream. We should see the stats we've set. |
| EXPECT_EQ(1u, info.senders.size()); |
| VerifyVoiceSenderInfo(info.senders[0], false); |
| // We have added one receive stream. We should see empty stats. |
| EXPECT_EQ(info.receivers.size(), 1u); |
| EXPECT_EQ(info.receivers[0].ssrc(), 0u); |
| } |
| |
| // Start sending - this affects some reported stats. |
| { |
| cricket::VoiceMediaInfo info; |
| SetSend(true); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| EXPECT_EQ(true, |
| channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true)); |
| VerifyVoiceSenderInfo(info.senders[0], true); |
| VerifyVoiceSendRecvCodecs(info); |
| } |
| |
| // Remove the kSsrcY stream. No receiver stats. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| EXPECT_EQ(true, |
| channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(1u, info.senders.size()); |
| EXPECT_EQ(0u, info.receivers.size()); |
| } |
| |
| // Deliver a new packet - a default receive stream should be created and we |
| // should see stats again. |
| { |
| cricket::VoiceMediaInfo info; |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| SetAudioReceiveStreamStats(); |
| EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); |
| EXPECT_EQ(true, |
| channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true)); |
| EXPECT_EQ(1u, info.senders.size()); |
| EXPECT_EQ(1u, info.receivers.size()); |
| VerifyVoiceReceiverInfo(info.receivers[0]); |
| VerifyVoiceSendRecvCodecs(info); |
| } |
| } |
| |
| // Test that we can set the outgoing SSRC properly with multiple streams. |
| // SSRC is set in SetupSendStream() by calling AddSendStream. |
| TEST_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( |
| 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(channel_->RemoveRecvStream(ssrc3)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(ssrc2)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(ssrc1)); |
| } |
| |
| // Test that receiving on an unsignaled stream works (a stream is created). |
| TEST_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(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. |
| channel_->ResetUnsignaledRecvStream(); |
| 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)); |
| |