diff options
Diffstat (limited to 'third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc')
| -rw-r--r-- | third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc | 4017 |
1 files changed, 4017 insertions, 0 deletions
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc new file mode 100644 index 0000000000..b1393eec74 --- /dev/null +++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc @@ -0,0 +1,4017 @@ +/* + * 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 "absl/types/optional.h" +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/media_types.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/codec.h" +#include "media/base/fake_media_engine.h" +#include "media/base/fake_network_interface.h" +#include "media/base/fake_rtp.h" +#include "media/base/media_channel.h" +#include "media/base/media_constants.h" +#include "media/engine/fake_webrtc_call.h" +#include "modules/audio_device/include/mock_audio_device.h" +#include "modules/audio_mixer/audio_mixer_impl.h" +#include "modules/audio_processing/include/mock_audio_processing.h" +#include "modules/rtp_rtcp/source/rtp_header_extensions.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "rtc_base/arraysize.h" +#include "rtc_base/byte_order.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "test/gtest.h" +#include "test/mock_audio_decoder_factory.h" +#include "test/mock_audio_encoder_factory.h" +#include "test/scoped_key_value_config.h" + +using ::testing::_; +using ::testing::ContainerEq; +using ::testing::Contains; +using ::testing::Field; +using ::testing::IsEmpty; +using ::testing::Return; +using ::testing::ReturnPointee; +using ::testing::SaveArg; +using ::testing::StrictMock; +using ::testing::UnorderedElementsAreArray; +using ::webrtc::Call; +using ::webrtc::CallConfig; + +namespace { +using webrtc::BitrateConstraints; + +constexpr uint32_t kMaxUnsignaledRecvStreams = 4; + +const cricket::AudioCodec kPcmuCodec = + cricket::CreateAudioCodec(0, "PCMU", 8000, 1); +const cricket::AudioCodec kOpusCodec = + cricket::CreateAudioCodec(111, "opus", 48000, 2); +const cricket::AudioCodec kG722CodecVoE = + cricket::CreateAudioCodec(9, "G722", 16000, 1); +const cricket::AudioCodec kG722CodecSdp = + cricket::CreateAudioCodec(9, "G722", 8000, 1); +const cricket::AudioCodec kCn8000Codec = + cricket::CreateAudioCodec(13, "CN", 8000, 1); +const cricket::AudioCodec kCn16000Codec = + cricket::CreateAudioCodec(105, "CN", 16000, 1); +const cricket::AudioCodec kRed48000Codec = + cricket::CreateAudioCodec(112, "RED", 48000, 2); +const cricket::AudioCodec kTelephoneEventCodec1 = + cricket::CreateAudioCodec(106, "telephone-event", 8000, 1); +const cricket::AudioCodec kTelephoneEventCodec2 = + cricket::CreateAudioCodec(107, "telephone-event", 32000, 1); + +const uint32_t kSsrc0 = 0; +const uint32_t kSsrc1 = 1; +const uint32_t kSsrcX = 0x99; +const uint32_t kSsrcY = 0x17; +const uint32_t kSsrcZ = 0x42; +const uint32_t kSsrcW = 0x02; +const uint32_t kSsrcs4[] = {11, 200, 30, 44}; + +constexpr int kRtpHistoryMs = 5000; + +constexpr webrtc::AudioProcessing::Config::GainController1::Mode + kDefaultAgcMode = +#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) + webrtc::AudioProcessing::Config::GainController1::kFixedDigital; +#else + webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog; +#endif + +constexpr webrtc::AudioProcessing::Config::NoiseSuppression::Level + kDefaultNsLevel = + webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh; + +void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) { + RTC_DCHECK(adm); + + // Setup. + EXPECT_CALL(*adm, Init()).WillOnce(Return(0)); + EXPECT_CALL(*adm, RegisterAudioCallback(_)).WillOnce(Return(0)); +#if defined(WEBRTC_WIN) + EXPECT_CALL( + *adm, + SetPlayoutDevice( + ::testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>( + webrtc::AudioDeviceModule::kDefaultCommunicationDevice))) + .WillOnce(Return(0)); +#else + EXPECT_CALL(*adm, SetPlayoutDevice(0)).WillOnce(Return(0)); +#endif // #if defined(WEBRTC_WIN) + EXPECT_CALL(*adm, InitSpeaker()).WillOnce(Return(0)); + EXPECT_CALL(*adm, StereoPlayoutIsAvailable(::testing::_)).WillOnce(Return(0)); + EXPECT_CALL(*adm, SetStereoPlayout(false)).WillOnce(Return(0)); +#if defined(WEBRTC_WIN) + EXPECT_CALL( + *adm, + SetRecordingDevice( + ::testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>( + webrtc::AudioDeviceModule::kDefaultCommunicationDevice))) + .WillOnce(Return(0)); +#else + EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0)); +#endif // #if defined(WEBRTC_WIN) + EXPECT_CALL(*adm, InitMicrophone()).WillOnce(Return(0)); + EXPECT_CALL(*adm, StereoRecordingIsAvailable(::testing::_)) + .WillOnce(Return(0)); + EXPECT_CALL(*adm, SetStereoRecording(false)).WillOnce(Return(0)); + EXPECT_CALL(*adm, BuiltInAECIsAvailable()).WillOnce(Return(false)); + EXPECT_CALL(*adm, BuiltInAGCIsAvailable()).WillOnce(Return(false)); + EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false)); + + // Teardown. + EXPECT_CALL(*adm, StopPlayout()).WillOnce(Return(0)); + EXPECT_CALL(*adm, StopRecording()).WillOnce(Return(0)); + EXPECT_CALL(*adm, RegisterAudioCallback(nullptr)).WillOnce(Return(0)); + EXPECT_CALL(*adm, Terminate()).WillOnce(Return(0)); +} +} // 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.get()); + rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm = + use_null_apm ? nullptr + : rtc::make_ref_counted< + StrictMock<webrtc::test::MockAudioProcessing>>(); + + webrtc::AudioProcessing::Config apm_config; + if (!use_null_apm) { + EXPECT_CALL(*apm, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config)); + EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config)); + EXPECT_CALL(*apm, DetachAecDump()); + } + { + webrtc::FieldTrialBasedConfig trials; + cricket::WebRtcVoiceEngine engine( + task_queue_factory.get(), adm.get(), + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, + nullptr, 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() + : use_null_apm_(GetParam()), + task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()), + adm_(webrtc::test::MockAudioDeviceModule::CreateStrict()), + apm_(use_null_apm_ + ? nullptr + : rtc::make_ref_counted< + StrictMock<webrtc::test::MockAudioProcessing>>()), + call_(&field_trials_) { + // AudioDeviceModule. + AdmSetupExpectations(adm_.get()); + + if (!use_null_apm_) { + // AudioProcessing. + EXPECT_CALL(*apm_, GetConfig()) + .WillRepeatedly(ReturnPointee(&apm_config_)); + EXPECT_CALL(*apm_, ApplyConfig(_)) + .WillRepeatedly(SaveArg<0>(&apm_config_)); + EXPECT_CALL(*apm_, DetachAecDump()); + } + + // Default Options. + // TODO(kwiberg): We should use mock factories here, but a bunch of + // the tests here probe the specific set of codecs provided by the builtin + // factories. Those tests should probably be moved elsewhere. + auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); + auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); + engine_.reset(new cricket::WebRtcVoiceEngine( + task_queue_factory_.get(), adm_.get(), encoder_factory, decoder_factory, + nullptr, apm_, nullptr, nullptr, field_trials_)); + engine_->Init(); + send_parameters_.codecs.push_back(kPcmuCodec); + recv_parameters_.codecs.push_back(kPcmuCodec); + + if (!use_null_apm_) { + // Default Options. + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_TRUE(IsHighPassFilterEnabled()); + EXPECT_TRUE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + VerifyGainControlEnabledCorrectly(); + VerifyGainControlDefaultSettings(); + } + } + + bool SetupChannel() { + send_channel_ = engine_->CreateSendChannel( + &call_, cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + receive_channel_ = engine_->CreateReceiveChannel( + &call_, cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + send_channel_->SetSsrcListChangedCallback( + [receive_channel = + receive_channel_.get()](const std::set<uint32_t>& choices) { + receive_channel->ChooseReceiverReportSsrc(choices); + }); + send_channel_->SetSendCodecChangedCallback( + [receive_channel = receive_channel_.get(), + send_channel = send_channel_.get()]() { + receive_channel->SetReceiveNackEnabled( + send_channel->SendCodecHasNack()); + receive_channel->SetReceiveNonSenderRttEnabled( + send_channel->SenderNonSenderRttEnabled()); + }); + return true; + } + + bool SetupRecvStream() { + if (!SetupChannel()) { + return false; + } + return AddRecvStream(kSsrcX); + } + + bool SetupSendStream() { + return SetupSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)); + } + + bool SetupSendStream(const cricket::StreamParams& sp) { + if (!SetupChannel()) { + return false; + } + if (!send_channel_->AddSendStream(sp)) { + return false; + } + if (!use_null_apm_) { + EXPECT_CALL(*apm_, set_output_will_be_muted(false)); + } + return send_channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_); + } + + bool AddRecvStream(uint32_t ssrc) { + EXPECT_TRUE(receive_channel_); + return receive_channel_->AddRecvStream( + cricket::StreamParams::CreateLegacy(ssrc)); + } + + void SetupForMultiSendStream() { + EXPECT_TRUE(SetupSendStream()); + // Remove stream added in Setup. + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); + EXPECT_TRUE(send_channel_->RemoveSendStream(kSsrcX)); + // Verify the channel does not exist. + EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX)); + } + + void DeliverPacket(const void* data, int len) { + webrtc::RtpPacketReceived packet; + packet.Parse(reinterpret_cast<const uint8_t*>(data), len); + receive_channel_->OnPacketReceived(packet); + rtc::Thread::Current()->ProcessMessages(0); + } + + const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) { + const auto* send_stream = call_.GetAudioSendStream(ssrc); + EXPECT_TRUE(send_stream); + return *send_stream; + } + + const cricket::FakeAudioReceiveStream& GetRecvStream(uint32_t ssrc) { + const auto* recv_stream = call_.GetAudioReceiveStream(ssrc); + EXPECT_TRUE(recv_stream); + return *recv_stream; + } + + const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) { + return GetSendStream(ssrc).GetConfig(); + } + + const webrtc::AudioReceiveStreamInterface::Config& GetRecvStreamConfig( + uint32_t ssrc) { + return GetRecvStream(ssrc).GetConfig(); + } + + void SetSend(bool enable) { + ASSERT_TRUE(send_channel_); + if (enable) { + EXPECT_CALL(*adm_, RecordingIsInitialized()) + .Times(::testing::AtMost(1)) + .WillOnce(Return(false)); + EXPECT_CALL(*adm_, Recording()) + .Times(::testing::AtMost(1)) + .WillOnce(Return(false)); + EXPECT_CALL(*adm_, InitRecording()) + .Times(::testing::AtMost(1)) + .WillOnce(Return(0)); + } + send_channel_->SetSend(enable); + } + + void SetSenderParameters(const cricket::AudioSenderParameter& params) { + ASSERT_TRUE(send_channel_); + EXPECT_TRUE(send_channel_->SetSenderParameters(params)); + } + + void SetAudioSend(uint32_t ssrc, + bool enable, + cricket::AudioSource* source, + const cricket::AudioOptions* options = nullptr) { + ASSERT_TRUE(send_channel_); + if (!use_null_apm_) { + EXPECT_CALL(*apm_, set_output_will_be_muted(!enable)); + } + EXPECT_TRUE(send_channel_->SetAudioSend(ssrc, enable, options, source)); + } + + void TestInsertDtmf(uint32_t ssrc, + bool caller, + const cricket::AudioCodec& codec) { + EXPECT_TRUE(SetupChannel()); + if (caller) { + // If this is a caller, local description will be applied and add the + // send stream. + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + } + + // Test we can only InsertDtmf when the other side supports telephone-event. + SetSenderParameters(send_parameters_); + SetSend(true); + EXPECT_FALSE(send_channel_->CanInsertDtmf()); + EXPECT_FALSE(send_channel_->InsertDtmf(ssrc, 1, 111)); + send_parameters_.codecs.push_back(codec); + SetSenderParameters(send_parameters_); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); + + if (!caller) { + // If this is callee, there's no active send channel yet. + EXPECT_FALSE(send_channel_->InsertDtmf(ssrc, 2, 123)); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + } + + // Check we fail if the ssrc is invalid. + EXPECT_FALSE(send_channel_->InsertDtmf(-1, 1, 111)); + + // Test send. + cricket::FakeAudioSendStream::TelephoneEvent telephone_event = + GetSendStream(kSsrcX).GetLatestTelephoneEvent(); + EXPECT_EQ(-1, telephone_event.payload_type); + EXPECT_TRUE(send_channel_->InsertDtmf(ssrc, 2, 123)); + telephone_event = GetSendStream(kSsrcX).GetLatestTelephoneEvent(); + EXPECT_EQ(codec.id, telephone_event.payload_type); + EXPECT_EQ(codec.clockrate, telephone_event.payload_frequency); + EXPECT_EQ(2, telephone_event.event_code); + EXPECT_EQ(123, telephone_event.duration_ms); + } + + void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) { + // For a caller, the answer will be applied in set remote description + // where SetSenderParameters() is called. + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + send_parameters_.extmap_allow_mixed = extmap_allow_mixed; + SetSenderParameters(send_parameters_); + const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); + EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed); + } + + void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) { + // For a callee, the answer will be applied in set local description + // where SetExtmapAllowMixed() and AddSendStream() are called. + EXPECT_TRUE(SetupChannel()); + send_channel_->SetExtmapAllowMixed(extmap_allow_mixed); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + + const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); + EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed); + } + + // Test that send bandwidth is set correctly. + // `codec` is the codec under test. + // `max_bitrate` is a parameter to set to SetMaxSendBandwidth(). + // `expected_result` is the expected result from SetMaxSendBandwidth(). + // `expected_bitrate` is the expected audio bitrate afterward. + void TestMaxSendBandwidth(const cricket::AudioCodec& codec, + int max_bitrate, + bool expected_result, + int expected_bitrate) { + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(codec); + parameters.max_bandwidth_bps = max_bitrate; + if (expected_result) { + SetSenderParameters(parameters); + } else { + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + } + EXPECT_EQ(expected_bitrate, GetCodecBitrate(kSsrcX)); + } + + // Sets the per-stream maximum bitrate limit for the specified SSRC. + bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) { + webrtc::RtpParameters parameters = + send_channel_->GetRtpSendParameters(ssrc); + EXPECT_EQ(1UL, parameters.encodings.size()); + + parameters.encodings[0].max_bitrate_bps = bitrate; + return send_channel_->SetRtpSendParameters(ssrc, parameters).ok(); + } + + void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) { + cricket::AudioSenderParameter send_parameters; + send_parameters.codecs.push_back(codec); + send_parameters.max_bandwidth_bps = bitrate; + SetSenderParameters(send_parameters); + } + + void CheckSendCodecBitrate(int32_t ssrc, + const char expected_name[], + int expected_bitrate) { + const auto& spec = GetSendStreamConfig(ssrc).send_codec_spec; + EXPECT_EQ(expected_name, spec->format.name); + EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps); + } + + absl::optional<int> GetCodecBitrate(int32_t ssrc) { + return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps; + } + + int GetMaxBitrate(int32_t ssrc) { + return GetSendStreamConfig(ssrc).max_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 = + send_channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(1UL, resulting_parameters.encodings.size()); + EXPECT_EQ(expected_result ? stream_max : -1, + resulting_parameters.encodings[0].max_bitrate_bps); + + // Verify that the codec settings have the expected bitrate. + EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrcX)); + EXPECT_EQ(expected_codec_bitrate, GetMaxBitrate(kSsrcX)); + } + + void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps, + int expected_min_bitrate_bps, + const char* start_bitrate_kbps, + int expected_start_bitrate_bps, + const char* max_bitrate_kbps, + int expected_max_bitrate_bps) { + EXPECT_TRUE(SetupSendStream()); + auto& codecs = send_parameters_.codecs; + codecs.clear(); + codecs.push_back(kOpusCodec); + codecs[0].params[cricket::kCodecParamMinBitrate] = min_bitrate_kbps; + codecs[0].params[cricket::kCodecParamStartBitrate] = start_bitrate_kbps; + codecs[0].params[cricket::kCodecParamMaxBitrate] = max_bitrate_kbps; + EXPECT_CALL(*call_.GetMockTransportControllerSend(), + SetSdpBitrateParameters( + AllOf(Field(&BitrateConstraints::min_bitrate_bps, + expected_min_bitrate_bps), + Field(&BitrateConstraints::start_bitrate_bps, + expected_start_bitrate_bps), + Field(&BitrateConstraints::max_bitrate_bps, + expected_max_bitrate_bps)))); + + SetSenderParameters(send_parameters_); + } + + void TestSetSendRtpHeaderExtensions(const std::string& ext) { + EXPECT_TRUE(SetupSendStream()); + + // Ensure extensions are off by default. + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure unknown extensions won't cause an error. + send_parameters_.extensions.push_back( + webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); + SetSenderParameters(send_parameters_); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure extensions stay off with an empty list of headers. + send_parameters_.extensions.clear(); + SetSenderParameters(send_parameters_); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure extension is set properly. + const int id = 1; + send_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); + SetSenderParameters(send_parameters_); + EXPECT_EQ(1u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + EXPECT_EQ(ext, GetSendStreamConfig(kSsrcX).rtp.extensions[0].uri); + EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id); + + // Ensure extension is set properly on new stream. + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcY))); + EXPECT_NE(call_.GetAudioSendStream(kSsrcX), + call_.GetAudioSendStream(kSsrcY)); + EXPECT_EQ(1u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); + EXPECT_EQ(ext, GetSendStreamConfig(kSsrcY).rtp.extensions[0].uri); + EXPECT_EQ(id, GetSendStreamConfig(kSsrcY).rtp.extensions[0].id); + + // Ensure all extensions go back off with an empty list. + send_parameters_.codecs.push_back(kPcmuCodec); + send_parameters_.extensions.clear(); + SetSenderParameters(send_parameters_); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); + } + + void TestSetRecvRtpHeaderExtensions(const std::string& ext) { + EXPECT_TRUE(SetupRecvStream()); + + // Ensure extensions are off by default. + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, + IsEmpty()); + + // Ensure unknown extensions won't cause an error. + recv_parameters_.extensions.push_back( + webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, + IsEmpty()); + + // Ensure extensions stay off with an empty list of headers. + recv_parameters_.extensions.clear(); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, + IsEmpty()); + + // Ensure extension is set properly. + const int id = 2; + recv_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); + EXPECT_EQ( + receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, + recv_parameters_.extensions); + + // Ensure extension is set properly on new stream. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_EQ( + receive_channel_->GetRtpReceiverParameters(kSsrcY).header_extensions, + recv_parameters_.extensions); + + // Ensure all extensions go back off with an empty list. + recv_parameters_.extensions.clear(); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(kSsrcX).header_extensions, + IsEmpty()); + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(kSsrcY).header_extensions, + IsEmpty()); + } + + webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const { + webrtc::AudioSendStream::Stats stats; + stats.local_ssrc = 12; + stats.payload_bytes_sent = 345; + stats.header_and_padding_bytes_sent = 56; + stats.packets_sent = 678; + stats.packets_lost = 9012; + stats.fraction_lost = 34.56f; + stats.codec_name = "codec_name_send"; + stats.codec_payload_type = 0; + stats.jitter_ms = 12; + stats.rtt_ms = 345; + stats.audio_level = 678; + stats.apm_statistics.delay_median_ms = 234; + stats.apm_statistics.delay_standard_deviation_ms = 567; + stats.apm_statistics.echo_return_loss = 890; + stats.apm_statistics.echo_return_loss_enhancement = 1234; + stats.apm_statistics.residual_echo_likelihood = 0.432f; + stats.apm_statistics.residual_echo_likelihood_recent_max = 0.6f; + stats.ana_statistics.bitrate_action_counter = 321; + stats.ana_statistics.channel_action_counter = 432; + stats.ana_statistics.dtx_action_counter = 543; + stats.ana_statistics.fec_action_counter = 654; + stats.ana_statistics.frame_length_increase_counter = 765; + stats.ana_statistics.frame_length_decrease_counter = 876; + stats.ana_statistics.uplink_packet_loss_fraction = 987.0; + return stats; + } + void SetAudioSendStreamStats() { + for (auto* s : call_.GetAudioSendStreams()) { + s->SetStats(GetAudioSendStreamStats()); + } + } + void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info, + bool is_sending) { + const auto stats = GetAudioSendStreamStats(); + EXPECT_EQ(info.ssrc(), stats.local_ssrc); + EXPECT_EQ(info.payload_bytes_sent, stats.payload_bytes_sent); + EXPECT_EQ(info.header_and_padding_bytes_sent, + stats.header_and_padding_bytes_sent); + EXPECT_EQ(info.packets_sent, stats.packets_sent); + EXPECT_EQ(info.packets_lost, stats.packets_lost); + EXPECT_EQ(info.fraction_lost, stats.fraction_lost); + EXPECT_EQ(info.codec_name, stats.codec_name); + EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); + EXPECT_EQ(info.jitter_ms, stats.jitter_ms); + EXPECT_EQ(info.rtt_ms, stats.rtt_ms); + EXPECT_EQ(info.audio_level, stats.audio_level); + EXPECT_EQ(info.apm_statistics.delay_median_ms, + stats.apm_statistics.delay_median_ms); + EXPECT_EQ(info.apm_statistics.delay_standard_deviation_ms, + stats.apm_statistics.delay_standard_deviation_ms); + EXPECT_EQ(info.apm_statistics.echo_return_loss, + stats.apm_statistics.echo_return_loss); + EXPECT_EQ(info.apm_statistics.echo_return_loss_enhancement, + stats.apm_statistics.echo_return_loss_enhancement); + EXPECT_EQ(info.apm_statistics.residual_echo_likelihood, + stats.apm_statistics.residual_echo_likelihood); + EXPECT_EQ(info.apm_statistics.residual_echo_likelihood_recent_max, + stats.apm_statistics.residual_echo_likelihood_recent_max); + EXPECT_EQ(info.ana_statistics.bitrate_action_counter, + stats.ana_statistics.bitrate_action_counter); + EXPECT_EQ(info.ana_statistics.channel_action_counter, + stats.ana_statistics.channel_action_counter); + EXPECT_EQ(info.ana_statistics.dtx_action_counter, + stats.ana_statistics.dtx_action_counter); + EXPECT_EQ(info.ana_statistics.fec_action_counter, + stats.ana_statistics.fec_action_counter); + EXPECT_EQ(info.ana_statistics.frame_length_increase_counter, + stats.ana_statistics.frame_length_increase_counter); + EXPECT_EQ(info.ana_statistics.frame_length_decrease_counter, + stats.ana_statistics.frame_length_decrease_counter); + EXPECT_EQ(info.ana_statistics.uplink_packet_loss_fraction, + stats.ana_statistics.uplink_packet_loss_fraction); + } + + webrtc::AudioReceiveStreamInterface::Stats GetAudioReceiveStreamStats() + const { + webrtc::AudioReceiveStreamInterface::Stats stats; + stats.remote_ssrc = 123; + stats.payload_bytes_received = 456; + stats.header_and_padding_bytes_received = 67; + stats.packets_received = 768; + stats.packets_lost = 101; + stats.codec_name = "codec_name_recv"; + stats.codec_payload_type = 0; + stats.jitter_ms = 901; + stats.jitter_buffer_ms = 234; + stats.jitter_buffer_preferred_ms = 567; + stats.delay_estimate_ms = 890; + stats.audio_level = 1234; + stats.total_samples_received = 5678901; + stats.concealed_samples = 234; + stats.concealment_events = 12; + stats.jitter_buffer_delay_seconds = 34; + stats.jitter_buffer_emitted_count = 77; + stats.expand_rate = 5.67f; + stats.speech_expand_rate = 8.90f; + stats.secondary_decoded_rate = 1.23f; + stats.secondary_discarded_rate = 0.12f; + stats.accelerate_rate = 4.56f; + stats.preemptive_expand_rate = 7.89f; + stats.decoding_calls_to_silence_generator = 12; + stats.decoding_calls_to_neteq = 345; + stats.decoding_normal = 67890; + stats.decoding_plc = 1234; + stats.decoding_codec_plc = 1236; + stats.decoding_cng = 5678; + stats.decoding_plc_cng = 9012; + stats.decoding_muted_output = 3456; + stats.capture_start_ntp_time_ms = 7890; + return stats; + } + void SetAudioReceiveStreamStats() { + for (auto* s : call_.GetAudioReceiveStreams()) { + s->SetStats(GetAudioReceiveStreamStats()); + } + } + void VerifyVoiceReceiverInfo(const cricket::VoiceReceiverInfo& info) { + const auto stats = GetAudioReceiveStreamStats(); + EXPECT_EQ(info.ssrc(), stats.remote_ssrc); + EXPECT_EQ(info.payload_bytes_received, stats.payload_bytes_received); + EXPECT_EQ(info.header_and_padding_bytes_received, + stats.header_and_padding_bytes_received); + EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_received), + stats.packets_received); + EXPECT_EQ(info.packets_lost, stats.packets_lost); + EXPECT_EQ(info.codec_name, stats.codec_name); + EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); + EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_ms), stats.jitter_ms); + EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_ms), + stats.jitter_buffer_ms); + EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_preferred_ms), + stats.jitter_buffer_preferred_ms); + EXPECT_EQ(rtc::checked_cast<unsigned int>(info.delay_estimate_ms), + stats.delay_estimate_ms); + EXPECT_EQ(info.audio_level, stats.audio_level); + EXPECT_EQ(info.total_samples_received, stats.total_samples_received); + EXPECT_EQ(info.concealed_samples, stats.concealed_samples); + EXPECT_EQ(info.concealment_events, stats.concealment_events); + EXPECT_EQ(info.jitter_buffer_delay_seconds, + stats.jitter_buffer_delay_seconds); + EXPECT_EQ(info.jitter_buffer_emitted_count, + stats.jitter_buffer_emitted_count); + EXPECT_EQ(info.expand_rate, stats.expand_rate); + EXPECT_EQ(info.speech_expand_rate, stats.speech_expand_rate); + EXPECT_EQ(info.secondary_decoded_rate, stats.secondary_decoded_rate); + EXPECT_EQ(info.secondary_discarded_rate, stats.secondary_discarded_rate); + EXPECT_EQ(info.accelerate_rate, stats.accelerate_rate); + EXPECT_EQ(info.preemptive_expand_rate, stats.preemptive_expand_rate); + EXPECT_EQ(info.decoding_calls_to_silence_generator, + stats.decoding_calls_to_silence_generator); + EXPECT_EQ(info.decoding_calls_to_neteq, stats.decoding_calls_to_neteq); + EXPECT_EQ(info.decoding_normal, stats.decoding_normal); + EXPECT_EQ(info.decoding_plc, stats.decoding_plc); + EXPECT_EQ(info.decoding_codec_plc, stats.decoding_codec_plc); + EXPECT_EQ(info.decoding_cng, stats.decoding_cng); + EXPECT_EQ(info.decoding_plc_cng, stats.decoding_plc_cng); + EXPECT_EQ(info.decoding_muted_output, stats.decoding_muted_output); + EXPECT_EQ(info.capture_start_ntp_time_ms, stats.capture_start_ntp_time_ms); + } + void VerifyVoiceSendRecvCodecs( + const cricket::VoiceMediaSendInfo& send_info, + const cricket::VoiceMediaReceiveInfo& receive_info) const { + EXPECT_EQ(send_parameters_.codecs.size(), send_info.send_codecs.size()); + for (const cricket::AudioCodec& codec : send_parameters_.codecs) { + ASSERT_EQ(send_info.send_codecs.count(codec.id), 1U); + EXPECT_EQ(send_info.send_codecs.find(codec.id)->second, + codec.ToCodecParameters()); + } + EXPECT_EQ(recv_parameters_.codecs.size(), + receive_info.receive_codecs.size()); + for (const cricket::AudioCodec& codec : recv_parameters_.codecs) { + ASSERT_EQ(receive_info.receive_codecs.count(codec.id), 1U); + EXPECT_EQ(receive_info.receive_codecs.find(codec.id)->second, + codec.ToCodecParameters()); + } + } + + void VerifyGainControlEnabledCorrectly() { + EXPECT_TRUE(apm_config_.gain_controller1.enabled); + EXPECT_EQ(kDefaultAgcMode, apm_config_.gain_controller1.mode); + } + + void VerifyGainControlDefaultSettings() { + EXPECT_EQ(3, apm_config_.gain_controller1.target_level_dbfs); + EXPECT_EQ(9, apm_config_.gain_controller1.compression_gain_db); + EXPECT_TRUE(apm_config_.gain_controller1.enable_limiter); + } + + void VerifyEchoCancellationSettings(bool enabled) { + constexpr bool kDefaultUseAecm = +#if defined(WEBRTC_ANDROID) + true; +#else + false; +#endif + EXPECT_EQ(apm_config_.echo_canceller.enabled, enabled); + EXPECT_EQ(apm_config_.echo_canceller.mobile_mode, kDefaultUseAecm); + } + + bool IsHighPassFilterEnabled() { + return apm_config_.high_pass_filter.enabled; + } + + cricket::WebRtcVoiceSendChannel* SendImplFromPointer( + cricket::VoiceMediaSendChannelInterface* channel) { + return static_cast<cricket::WebRtcVoiceSendChannel*>(channel); + } + + cricket::WebRtcVoiceSendChannel* SendImpl() { + return SendImplFromPointer(send_channel_.get()); + } + cricket::WebRtcVoiceReceiveChannel* ReceiveImpl() { + return static_cast<cricket::WebRtcVoiceReceiveChannel*>( + receive_channel_.get()); + } + + protected: + rtc::AutoThread main_thread_; + const bool use_null_apm_; + webrtc::test::ScopedKeyValueConfig field_trials_; + 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_; + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel_; + std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> receive_channel_; + cricket::AudioSenderParameter send_parameters_; + cricket::AudioReceiverParameters recv_parameters_; + FakeAudioSource fake_source_; + webrtc::AudioProcessing::Config apm_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(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); + EXPECT_EQ(kSsrcX, config.rtp.ssrc); + EXPECT_EQ("", config.rtp.c_name); + EXPECT_EQ(0u, config.rtp.extensions.size()); + EXPECT_EQ(SendImpl()->transport(), config.send_transport); +} + +// Test that we can add a receive stream and that it has the correct defaults. +TEST_P(WebRtcVoiceEngineTestFake, CreateRecvStream) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + const webrtc::AudioReceiveStreamInterface::Config& config = + GetRecvStreamConfig(kSsrcX); + EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc); + EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc); + EXPECT_EQ(ReceiveImpl()->transport(), config.rtcp_send_transport); + EXPECT_EQ("", config.sync_group); +} + +TEST_P(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) { + const std::vector<cricket::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::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs[0].id = 106; // collide with existing CN 32k + parameters.codecs[2].id = 126; + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {106, {"OPUS", 48000, 2}}, + {126, {"telephone-event", 8000, 1}}, + {107, {"telephone-event", 32000, 1}}}))); +} + +// Test that we fail to set an unknown inbound codec. +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(cricket::CreateAudioCodec(127, "XYZ", 32000, 1)); + EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)); +} + +// Test that we fail if we have duplicate types in the inbound list. +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs[1].id = kOpusCodec.id; + EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)); +} + +// Test that we can decode OPUS without stereo parameters. +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kOpusCodec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, {111, {"opus", 48000, 2}}}))); +} + +// Test that we can decode OPUS with stereo = 0. +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[1].params["stereo"] = "0"; + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {111, {"opus", 48000, 2, {{"stereo", "0"}}}}}))); +} + +// Test that we can decode OPUS with stereo = 1. +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[1].params["stereo"] = "1"; + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {111, {"opus", 48000, 2, {{"stereo", "1"}}}}}))); +} + +// Test that changes to recv codecs are applied to all streams. +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs[0].id = 106; // collide with existing CN 32k + parameters.codecs[2].id = 126; + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + for (const auto& ssrc : {kSsrcX, kSsrcY}) { + EXPECT_TRUE(AddRecvStream(ssrc)); + EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {106, {"OPUS", 48000, 2}}, + {126, {"telephone-event", 8000, 1}}, + {107, {"telephone-event", 32000, 1}}}))); + } +} + +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].id = 106; // collide with existing CN 32k + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + + const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map; + ASSERT_EQ(1u, dm.count(106)); + EXPECT_EQ(webrtc::SdpAudioFormat("opus", 48000, 2), dm.at(106)); +} + +// Test that we can apply the same set of codecs again while playing. +TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn16000Codec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + receive_channel_->SetPlayout(true); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + + // Remapping a payload type to a different codec should fail. + parameters.codecs[0] = kOpusCodec; + parameters.codecs[0].id = kPcmuCodec.id; + EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); +} + +// Test that we can add a codec while playing. +TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn16000Codec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + receive_channel_->SetPlayout(true); + + parameters.codecs.push_back(kOpusCodec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); +} + +// Test that we accept adding the same codec with a different payload type. +// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847 +TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + + ++parameters.codecs[0].id; + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); +} + +// Test that we do allow setting Opus/Red by default. +TEST_P(WebRtcVoiceEngineTestFake, RecvRedDefault) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kRed48000Codec); + parameters.codecs[1].params[""] = "111/111"; + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{111, {"opus", 48000, 2}}, + {112, {"red", 48000, 2, {{"", "111/111"}}}}}))); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) { + EXPECT_TRUE(SetupSendStream()); + + // Test that when autobw is enabled, bitrate is kept as the default + // value. autobw is enabled for the following tests because the target + // bitrate is <= 0. + + // PCMU, default bitrate == 64000. + TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000); + + // opus, default bitrate == 32000 in mono. + TestMaxSendBandwidth(kOpusCodec, -1, true, 32000); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) { + EXPECT_TRUE(SetupSendStream()); + + // opus, default bitrate == 64000. + TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000); + TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000); + // Rates above the max (510000) should be capped. + TestMaxSendBandwidth(kOpusCodec, 600000, true, 510000); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) { + EXPECT_TRUE(SetupSendStream()); + + // Test that we can only set a maximum bitrate for a fixed-rate codec + // if it's bigger than the fixed rate. + + // PCMU, fixed bitrate == 64000. + TestMaxSendBandwidth(kPcmuCodec, 0, true, 64000); + TestMaxSendBandwidth(kPcmuCodec, 1, false, 64000); + TestMaxSendBandwidth(kPcmuCodec, 128000, true, 64000); + TestMaxSendBandwidth(kPcmuCodec, 32000, false, 64000); + TestMaxSendBandwidth(kPcmuCodec, 64000, true, 64000); + TestMaxSendBandwidth(kPcmuCodec, 63999, false, 64000); + TestMaxSendBandwidth(kPcmuCodec, 64001, true, 64000); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) { + EXPECT_TRUE(SetupChannel()); + const int kDesiredBitrate = 128000; + cricket::AudioSenderParameter parameters; + parameters.codecs = engine_->send_codecs(); + parameters.max_bandwidth_bps = kDesiredBitrate; + SetSenderParameters(parameters); + + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + + EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX)); +} + +// Test that bitrate cannot be set for CBR codecs. +// Bitrate is ignored if it is higher than the fixed bitrate. +// Bitrate less then the fixed bitrate is an error. +TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) { + EXPECT_TRUE(SetupSendStream()); + + // PCMU, default bitrate == 64000. + SetSenderParameters(send_parameters_); + EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); + + send_parameters_.max_bandwidth_bps = 128000; + SetSenderParameters(send_parameters_); + EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); + + send_parameters_.max_bandwidth_bps = 128; + EXPECT_FALSE(send_channel_->SetSenderParameters(send_parameters_)); + EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); +} + +// Test that the per-stream bitrate limit and the global +// bitrate limit both apply. +TEST_P(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) { + EXPECT_TRUE(SetupSendStream()); + + // opus, default bitrate == 32000. + SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 32000); + SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000); + SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000); + SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000); + + // CBR codecs allow both maximums to exceed the bitrate. + SetAndExpectMaxBitrate(kPcmuCodec, 0, 0, true, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 64001, 0, true, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 0, 64001, true, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 64001, 64001, true, 64000); + + // CBR codecs don't allow per stream maximums to be too low. + SetAndExpectMaxBitrate(kPcmuCodec, 0, 63999, false, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 64001, 63999, false, 64000); +} + +// Test that an attempt to set RtpParameters for a stream that does not exist +// fails. +TEST_P(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) { + EXPECT_TRUE(SetupChannel()); + webrtc::RtpParameters nonexistent_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(0u, nonexistent_parameters.encodings.size()); + + nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters()); + EXPECT_FALSE( + send_channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters).ok()); +} + +TEST_P(WebRtcVoiceEngineTestFake, + CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) { + // This test verifies that setting RtpParameters succeeds only if + // the structure contains exactly one encoding. + // TODO(skvlad): Update this test when we start supporting setting parameters + // for each encoding individually. + + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + // Two or more encodings should result in failure. + parameters.encodings.push_back(webrtc::RtpEncodingParameters()); + EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); + // Zero encodings should also fail. + parameters.encodings.clear(); + EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); +} + +// Changing the SSRC through RtpParameters is not allowed. +TEST_P(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) { + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + parameters.encodings[0].ssrc = 0xdeadbeef; + EXPECT_FALSE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); +} + +// Test that a stream will not be sending if its encoding is made +// inactive through SetRtpSendParameters. +TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) { + EXPECT_TRUE(SetupSendStream()); + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + // Get current parameters and change "active" to false. + webrtc::RtpParameters parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + ASSERT_EQ(1u, parameters.encodings.size()); + ASSERT_TRUE(parameters.encodings[0].active); + parameters.encodings[0].active = false; + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Now change it back to active and verify we resume sending. + // This should occur even when other parameters are updated. + parameters.encodings[0].active = true; + parameters.encodings[0].max_bitrate_bps = absl::optional<int>(6000); + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersAdaptivePtime) { + EXPECT_TRUE(SetupSendStream()); + // Get current parameters and change "adaptive_ptime" to true. + webrtc::RtpParameters parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + ASSERT_EQ(1u, parameters.encodings.size()); + ASSERT_FALSE(parameters.encodings[0].adaptive_ptime); + parameters.encodings[0].adaptive_ptime = true; + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); + EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX)); + EXPECT_EQ(16000, GetSendStreamConfig(kSsrcX).min_bitrate_bps); + + parameters.encodings[0].adaptive_ptime = false; + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); + EXPECT_FALSE(GetAudioNetworkAdaptorConfig(kSsrcX)); + EXPECT_EQ(32000, GetSendStreamConfig(kSsrcX).min_bitrate_bps); +} + +TEST_P(WebRtcVoiceEngineTestFake, + DisablingAdaptivePtimeDoesNotRemoveAudioNetworkAdaptorFromOptions) { + EXPECT_TRUE(SetupSendStream()); + send_parameters_.options.audio_network_adaptor = true; + send_parameters_.options.audio_network_adaptor_config = {"1234"}; + SetSenderParameters(send_parameters_); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); + + webrtc::RtpParameters parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + parameters.encodings[0].adaptive_ptime = false; + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok()); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); +} + +TEST_P(WebRtcVoiceEngineTestFake, AdaptivePtimeFieldTrial) { + webrtc::test::ScopedKeyValueConfig override_field_trials( + field_trials_, "WebRTC-Audio-AdaptivePtime/enabled:true/"); + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX)); +} + +// Test that SetRtpSendParameters configures the correct encoding channel for +// each SSRC. +TEST_P(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) { + SetupForMultiSendStream(); + // Create send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + } + // Configure one stream to be limited by the stream config, another to be + // limited by the global max, and the third one with no per-stream limit + // (still subject to the global limit). + SetGlobalMaxBitrate(kOpusCodec, 32000); + EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 24000)); + EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 48000)); + EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1)); + + EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0])); + EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[1])); + EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2])); + + // Remove the global cap; the streams should switch to their respective + // maximums (or remain unchanged if there was no other limit on them.) + SetGlobalMaxBitrate(kOpusCodec, -1); + EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0])); + EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[1])); + EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2])); +} + +// Test that GetRtpSendParameters returns the currently configured codecs. +TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + SetSenderParameters(parameters); + + webrtc::RtpParameters rtp_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + ASSERT_EQ(2u, rtp_parameters.codecs.size()); + EXPECT_EQ(kOpusCodec.ToCodecParameters(), rtp_parameters.codecs[0]); + EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); +} + +// Test that GetRtpSendParameters returns the currently configured RTCP CNAME. +TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) { + cricket::StreamParams params = cricket::StreamParams::CreateLegacy(kSsrcX); + params.cname = "rtcpcname"; + EXPECT_TRUE(SetupSendStream(params)); + + webrtc::RtpParameters rtp_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str()); +} + +TEST_P(WebRtcVoiceEngineTestFake, + DetectRtpSendParameterHeaderExtensionsChange) { + EXPECT_TRUE(SetupSendStream()); + + webrtc::RtpParameters rtp_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + rtp_parameters.header_extensions.emplace_back(); + + EXPECT_NE(0u, rtp_parameters.header_extensions.size()); + + webrtc::RTCError result = + send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters); + EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type()); +} + +// Test that GetRtpSendParameters returns an SSRC. +TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) { + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters rtp_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc); +} + +// Test that if we set/get parameters multiple times, we get the same results. +TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + SetSenderParameters(parameters); + + webrtc::RtpParameters initial_params = + send_channel_->GetRtpSendParameters(kSsrcX); + + // We should be able to set the params we just got. + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, initial_params).ok()); + + // ... And this shouldn't change the params returned by GetRtpSendParameters. + webrtc::RtpParameters new_params = + send_channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(initial_params, send_channel_->GetRtpSendParameters(kSsrcX)); +} + +// Test that we remove the codec from RTP parameters if it's not negotiated +// anymore. +TEST_P(WebRtcVoiceEngineTestFake, + SetSendParametersRemovesSelectedCodecFromRtpParameters) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + SetSenderParameters(parameters); + + webrtc::RtpParameters initial_params = + send_channel_->GetRtpSendParameters(kSsrcX); + + webrtc::RtpCodec opus_rtp_codec; + opus_rtp_codec.name = "opus"; + opus_rtp_codec.kind = cricket::MEDIA_TYPE_AUDIO; + opus_rtp_codec.num_channels = 2; + opus_rtp_codec.clock_rate = 48000; + initial_params.encodings[0].codec = opus_rtp_codec; + + // We should be able to set the params with the opus codec that has been + // negotiated. + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, initial_params).ok()); + + parameters.codecs.clear(); + parameters.codecs.push_back(kPcmuCodec); + SetSenderParameters(parameters); + + // Since Opus is no longer negotiated, the RTP parameters should not have a + // forced codec anymore. + webrtc::RtpParameters new_params = + send_channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(new_params.encodings[0].codec, absl::nullopt); +} + +// Test that max_bitrate_bps in send stream config gets updated correctly when +// SetRtpSendParameters is called. +TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter send_parameters; + send_parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(send_parameters); + + webrtc::RtpParameters rtp_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + // Expect empty on parameters.encodings[0].max_bitrate_bps; + EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps); + + constexpr int kMaxBitrateBps = 6000; + rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps; + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); + + const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps; + EXPECT_EQ(max_bitrate, kMaxBitrateBps); +} + +// Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to +// a value <= 0, setting the parameters returns false. +TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterInvalidBitratePriority) { + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters rtp_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(1UL, rtp_parameters.encodings.size()); + EXPECT_EQ(webrtc::kDefaultBitratePriority, + rtp_parameters.encodings[0].bitrate_priority); + + rtp_parameters.encodings[0].bitrate_priority = 0; + EXPECT_FALSE( + send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); + rtp_parameters.encodings[0].bitrate_priority = -1.0; + EXPECT_FALSE( + send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); +} + +// Test that the bitrate_priority in the send stream config gets updated when +// SetRtpSendParameters is set for the VoiceMediaChannel. +TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) { + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters rtp_parameters = + send_channel_->GetRtpSendParameters(kSsrcX); + + EXPECT_EQ(1UL, rtp_parameters.encodings.size()); + EXPECT_EQ(webrtc::kDefaultBitratePriority, + rtp_parameters.encodings[0].bitrate_priority); + double new_bitrate_priority = 2.0; + rtp_parameters.encodings[0].bitrate_priority = new_bitrate_priority; + EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok()); + + // The priority should get set for both the audio channel's rtp parameters + // and the audio send stream's audio config. + EXPECT_EQ(new_bitrate_priority, send_channel_->GetRtpSendParameters(kSsrcX) + .encodings[0] + .bitrate_priority); + EXPECT_EQ(new_bitrate_priority, GetSendStreamConfig(kSsrcX).bitrate_priority); +} + +// Test that GetRtpReceiverParameters returns the currently configured codecs. +TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + + webrtc::RtpParameters rtp_parameters = + receive_channel_->GetRtpReceiverParameters(kSsrcX); + ASSERT_EQ(2u, rtp_parameters.codecs.size()); + EXPECT_EQ(kOpusCodec.ToCodecParameters(), rtp_parameters.codecs[0]); + EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); +} + +// Test that GetRtpReceiverParameters returns an SSRC. +TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) { + EXPECT_TRUE(SetupRecvStream()); + webrtc::RtpParameters rtp_parameters = + receive_channel_->GetRtpReceiverParameters(kSsrcX); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc); +} + +// Test that if we set/get parameters multiple times, we get the same results. +TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + + webrtc::RtpParameters initial_params = + receive_channel_->GetRtpReceiverParameters(kSsrcX); + + // ... And this shouldn't change the params returned by + // GetRtpReceiverParameters. + webrtc::RtpParameters new_params = + receive_channel_->GetRtpReceiverParameters(kSsrcX); + EXPECT_EQ(initial_params, receive_channel_->GetRtpReceiverParameters(kSsrcX)); +} + +// Test that GetRtpReceiverParameters returns parameters correctly when SSRCs +// aren't signaled. It should return an empty "RtpEncodingParameters" when +// configured to receive an unsignaled stream and no packets have been received +// yet, and start returning the SSRC once a packet has been received. +TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) { + ASSERT_TRUE(SetupChannel()); + // Call necessary methods to configure receiving a default stream as + // soon as it arrives. + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + + // Call GetDefaultRtpReceiveParameters before configured to receive an + // unsignaled stream. Should return nothing. + EXPECT_EQ(webrtc::RtpParameters(), + receive_channel_->GetDefaultRtpReceiveParameters()); + + // Set a sink for an unsignaled stream. + std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink()); + receive_channel_->SetDefaultRawAudioSink(std::move(fake_sink)); + + // Call GetDefaultRtpReceiveParameters before the SSRC is known. + webrtc::RtpParameters rtp_parameters = + receive_channel_->GetDefaultRtpReceiveParameters(); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); + + // Receive PCMU packet (SSRC=1). + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + // The `ssrc` member should still be unset. + rtp_parameters = receive_channel_->GetDefaultRtpReceiveParameters(); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); +} + +TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) { + ASSERT_TRUE(SetupChannel()); + cricket::AudioReceiverParameters parameters = recv_parameters_; + parameters.extensions.push_back( + RtpExtension(RtpExtension::kAudioLevelUri, /*id=*/1)); + ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + webrtc::RtpHeaderExtensionMap extension_map(parameters.extensions); + webrtc::RtpPacketReceived reference_packet(&extension_map); + constexpr uint8_t kAudioLevel = 123; + reference_packet.SetExtension<webrtc::AudioLevel>(/*voice_activity=*/true, + kAudioLevel); + // Create a packet without the extension map but with the same content. + webrtc::RtpPacketReceived received_packet; + ASSERT_TRUE(received_packet.Parse(reference_packet.Buffer())); + + receive_channel_->OnPacketReceived(received_packet); + rtc::Thread::Current()->ProcessMessages(0); + + bool voice_activity; + uint8_t audio_level; + EXPECT_TRUE(call_.last_received_rtp_packet().GetExtension<webrtc::AudioLevel>( + &voice_activity, &audio_level)); + EXPECT_EQ(audio_level, kAudioLevel); +} + +// Test that we apply codecs properly. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs[0].id = 96; + parameters.codecs[0].bitrate = 22000; + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps); + EXPECT_STRCASEEQ("OPUS", send_codec_spec.format.name.c_str()); + EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000); + EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type); + EXPECT_FALSE(send_channel_->CanInsertDtmf()); +} + +// Test that we use Opus/Red by default when it is +// listed as the first codec and there is an fmtp line. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRed) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kRed48000Codec); + parameters.codecs[0].params[""] = "111/111"; + parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); + EXPECT_EQ(112, send_codec_spec.red_payload_type); +} + +// Test that we do not use Opus/Red by default when it is +// listed as the first codec but there is no fmtp line. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedNoFmtp) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kRed48000Codec); + parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); + EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type); +} + +// Test that we do not use Opus/Red by default. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kRed48000Codec); + parameters.codecs[1].params[""] = "111/111"; + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); + EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type); +} + +// Test that the RED fmtp line must match the payload type. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpMismatch) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kRed48000Codec); + parameters.codecs[0].params[""] = "8/8"; + parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); + EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type); +} + +// Test that the RED fmtp line must show 2..32 payloads. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kRed48000Codec); + parameters.codecs[0].params[""] = "111"; + parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str()); + EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type); + for (int i = 1; i < 32; i++) { + parameters.codecs[0].params[""] += "/111"; + SetSenderParameters(parameters); + const auto& send_codec_spec2 = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, send_codec_spec2.payload_type); + EXPECT_STRCASEEQ("opus", send_codec_spec2.format.name.c_str()); + EXPECT_EQ(112, send_codec_spec2.red_payload_type); + } + parameters.codecs[0].params[""] += "/111"; + SetSenderParameters(parameters); + const auto& send_codec_spec3 = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, send_codec_spec3.payload_type); + EXPECT_STRCASEEQ("opus", send_codec_spec3.format.name.c_str()); + EXPECT_EQ(absl::nullopt, send_codec_spec3.red_payload_type); +} + +// Test that WebRtcVoiceEngine reconfigures, rather than recreates its +// AudioSendStream. +TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs[0].id = 96; + parameters.codecs[0].bitrate = 48000; + const int initial_num = call_.GetNumCreatedSendStreams(); + SetSenderParameters(parameters); + EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); + // Calling SetSendCodec again with same codec which is already set. + // In this case media channel shouldn't send codec to VoE. + SetSenderParameters(parameters); + EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); +} + +// TODO(ossu): Revisit if these tests need to be here, now that these kinds of +// tests should be available in AudioEncoderOpusTest. + +// Test that if clockrate is not 48000 for opus, we fail. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].clockrate = 50000; + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); +} + +// Test that if channels=0 for opus, we fail. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 0; + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); +} + +// Test that if channels=0 for opus, we fail. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 0; + parameters.codecs[0].params["stereo"] = "1"; + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); +} + +// Test that if channel is 1 for opus and there's no stereo, we fail. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 1; + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); +} + +// Test that if channel is 1 for opus and stereo=0, we fail. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 1; + parameters.codecs[0].params["stereo"] = "0"; + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); +} + +// Test that if channel is 1 for opus and stereo=1, we fail. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 1; + parameters.codecs[0].params["stereo"] = "1"; + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); +} + +// Test that with bitrate=0 and no stereo, bitrate is 32000. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 32000); +} + +// Test that with bitrate=0 and stereo=0, bitrate is 32000. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].params["stereo"] = "0"; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 32000); +} + +// Test that with bitrate=invalid and stereo=0, bitrate is 32000. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].params["stereo"] = "0"; + // bitrate that's out of the range between 6000 and 510000 will be clamped. + parameters.codecs[0].bitrate = 5999; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 6000); + + parameters.codecs[0].bitrate = 510001; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 510000); +} + +// Test that with bitrate=0 and stereo=1, bitrate is 64000. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].params["stereo"] = "1"; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 64000); +} + +// Test that with bitrate=invalid and stereo=1, bitrate is 64000. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].params["stereo"] = "1"; + // bitrate that's out of the range between 6000 and 510000 will be clamped. + parameters.codecs[0].bitrate = 5999; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 6000); + + parameters.codecs[0].bitrate = 510001; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 510000); +} + +// Test that with bitrate=N and stereo unset, bitrate is N. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 96000; + SetSenderParameters(parameters); + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_EQ(96000, spec.target_bitrate_bps); + EXPECT_EQ("opus", spec.format.name); + EXPECT_EQ(2u, spec.format.num_channels); + EXPECT_EQ(48000, spec.format.clockrate_hz); +} + +// Test that with bitrate=N and stereo=0, bitrate is N. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 30000; + parameters.codecs[0].params["stereo"] = "0"; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 30000); +} + +// Test that with bitrate=N and without any parameters, bitrate is N. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 30000; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 30000); +} + +// Test that with bitrate=N and stereo=1, bitrate is N. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 30000; + parameters.codecs[0].params["stereo"] = "1"; + SetSenderParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 30000); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithBitrates) { + SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", + 200000); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithHighMaxBitrate) { + SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000); +} + +TEST_P(WebRtcVoiceEngineTestFake, + SetSendCodecsWithoutBitratesUsesCorrectDefaults) { + SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) { + SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) { + SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", + 200000); + send_parameters_.max_bandwidth_bps = 100000; + // Setting max bitrate should keep previous min bitrate + // Setting max bitrate should not reset start bitrate. + EXPECT_CALL(*call_.GetMockTransportControllerSend(), + SetSdpBitrateParameters( + AllOf(Field(&BitrateConstraints::min_bitrate_bps, 100000), + Field(&BitrateConstraints::start_bitrate_bps, -1), + Field(&BitrateConstraints::max_bitrate_bps, 200000)))); + SetSenderParameters(send_parameters_); +} + +// Test that we can enable NACK with opus as callee. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( + cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); + EXPECT_EQ(0, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + SetSenderParameters(parameters); + // NACK should be enabled even with no send stream. + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); +} + +// Test that we can enable NACK on receive streams. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( + cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); + EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); + SetSenderParameters(parameters); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); +} + +// Test that we can disable NACK on receive streams. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( + cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); + SetSenderParameters(parameters); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); + + parameters.codecs.clear(); + parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(parameters); + EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); +} + +// Test that NACK is enabled on a new receive stream. +TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( + cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); + SetSenderParameters(parameters); + + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); + EXPECT_TRUE(AddRecvStream(kSsrcZ)); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcZ).rtp.nack.rtp_history_ms); +} + +// Test that we can switch back and forth between Opus and PCMU with CN. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsOpusPcmuSwitching) { + EXPECT_TRUE(SetupSendStream()); + + cricket::AudioSenderParameter opus_parameters; + opus_parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(opus_parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); + } + + cricket::AudioSenderParameter pcmu_parameters; + pcmu_parameters.codecs.push_back(kPcmuCodec); + pcmu_parameters.codecs.push_back(kCn16000Codec); + pcmu_parameters.codecs.push_back(kOpusCodec); + SetSenderParameters(pcmu_parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(0, spec.payload_type); + EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str()); + } + + SetSenderParameters(opus_parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); + } +} + +// Test that we handle various ways of specifying bitrate. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kPcmuCodec); + SetSenderParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(0, spec.payload_type); + EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str()); + EXPECT_EQ(64000, spec.target_bitrate_bps); + } + + parameters.codecs[0].bitrate = 0; // bitrate == default + SetSenderParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(0, spec.payload_type); + EXPECT_STREQ("PCMU", spec.format.name.c_str()); + EXPECT_EQ(64000, spec.target_bitrate_bps); + } + + parameters.codecs[0] = kOpusCodec; + parameters.codecs[0].bitrate = 0; // bitrate == default + SetSenderParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_STREQ("opus", spec.format.name.c_str()); + EXPECT_EQ(32000, spec.target_bitrate_bps); + } +} + +// Test that we fail if no codecs are specified. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + EXPECT_FALSE(send_channel_->SetSenderParameters(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::AudioSenderParameter parameters; + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs[0].id = 98; // DTMF + parameters.codecs[1].id = 96; + SetSenderParameters(parameters); + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, spec.payload_type); + EXPECT_STRCASEEQ("OPUS", spec.format.name.c_str()); + SetSend(true); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); +} + +// Test that CanInsertDtmf() is governed by the send flag +TEST_P(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs[0].id = 98; // DTMF + parameters.codecs[1].id = 96; + SetSenderParameters(parameters); + EXPECT_FALSE(send_channel_->CanInsertDtmf()); + SetSend(true); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); + SetSend(false); + EXPECT_FALSE(send_channel_->CanInsertDtmf()); +} + +// Test that payload type range is limited for telephone-event codec. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].id = 0; // DTMF + parameters.codecs[1].id = 96; + SetSenderParameters(parameters); + SetSend(true); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); + parameters.codecs[0].id = 128; // DTMF + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_FALSE(send_channel_->CanInsertDtmf()); + parameters.codecs[0].id = 127; + SetSenderParameters(parameters); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); + parameters.codecs[0].id = -1; // DTMF + EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_FALSE(send_channel_->CanInsertDtmf()); +} + +// Test that we can set send codecs even with CN codec as the first +// one on the list. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs[0].id = 98; // narrowband CN + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(0, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(98, send_codec_spec.cng_payload_type); +} + +// Test that we set VAD and DTMF types correctly as caller. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs[0].id = 96; + parameters.codecs[2].id = 97; // narrowband CN + parameters.codecs[3].id = 98; // DTMF + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1u, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + SetSend(true); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); +} + +// Test that we set VAD and DTMF types correctly as callee. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioSenderParameter parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs[0].id = 96; + parameters.codecs[2].id = 97; // narrowband CN + parameters.codecs[3].id = 98; // DTMF + SetSenderParameters(parameters); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1u, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + SetSend(true); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); +} + +// Test that we only apply VAD if we have a CN codec that matches the +// send codec clockrate. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSenderParameter parameters; + // Set PCMU(8K) and CN(16K). VAD should not be activated. + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs[1].id = 97; + SetSenderParameters(parameters); + { + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type); + } + // Set PCMU(8K) and CN(8K). VAD should be activated. + parameters.codecs[1] = kCn8000Codec; + SetSenderParameters(parameters); + { + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1u, send_codec_spec.format.num_channels); + EXPECT_EQ(13, send_codec_spec.cng_payload_type); + } + // Set OPUS(48K) and CN(8K). VAD should not be activated. + parameters.codecs[0] = kOpusCodec; + SetSenderParameters(parameters); + { + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_STRCASEEQ("OPUS", send_codec_spec.format.name.c_str()); + EXPECT_EQ(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::AudioSenderParameter parameters; + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs[0].name = "PcMu"; + parameters.codecs[0].id = 96; + parameters.codecs[2].id = 97; // narrowband CN + parameters.codecs[3].id = 98; // DTMF + SetSenderParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1u, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + SetSend(true); + EXPECT_TRUE(send_channel_->CanInsertDtmf()); +} + +TEST_P(WebRtcVoiceEngineTestFake, + SupportsTransportSequenceNumberHeaderExtension) { + const std::vector<webrtc::RtpExtension> header_extensions = + GetDefaultEnabledRtpHeaderExtensions(*engine_); + EXPECT_THAT(header_extensions, + Contains(::testing::Field( + "uri", &RtpExtension::uri, + webrtc::RtpExtension::kTransportSequenceNumberUri))); +} + +// Test support for audio level header extension. +TEST_P(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) { + TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); +} +TEST_P(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) { + TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); +} + +// Test support for transport sequence number header extension. +TEST_P(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) { + TestSetSendRtpHeaderExtensions( + webrtc::RtpExtension::kTransportSequenceNumberUri); +} +TEST_P(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) { + TestSetRecvRtpHeaderExtensions( + webrtc::RtpExtension::kTransportSequenceNumberUri); +} + +// Test that we can create a channel and start sending on it. +TEST_P(WebRtcVoiceEngineTestFake, Send) { + EXPECT_TRUE(SetupSendStream()); + SetSenderParameters(send_parameters_); + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + SetSend(false); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); +} + +// Test that a channel is muted/unmuted. +TEST_P(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) { + EXPECT_TRUE(SetupSendStream()); + SetSenderParameters(send_parameters_); + EXPECT_FALSE(GetSendStream(kSsrcX).muted()); + SetAudioSend(kSsrcX, true, nullptr); + EXPECT_FALSE(GetSendStream(kSsrcX).muted()); + SetAudioSend(kSsrcX, false, nullptr); + EXPECT_TRUE(GetSendStream(kSsrcX).muted()); +} + +// Test that SetSenderParameters() does not alter a stream's send state. +TEST_P(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Turn on sending. + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + + // Changing RTP header extensions will recreate the AudioSendStream. + send_parameters_.extensions.push_back( + webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); + SetSenderParameters(send_parameters_); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + + // Turn off sending. + SetSend(false); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Changing RTP header extensions will recreate the AudioSendStream. + send_parameters_.extensions.clear(); + SetSenderParameters(send_parameters_); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); +} + +// Test that we can create a channel and start playing out on it. +TEST_P(WebRtcVoiceEngineTestFake, Playout) { + EXPECT_TRUE(SetupRecvStream()); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); + receive_channel_->SetPlayout(true); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); + receive_channel_->SetPlayout(false); + EXPECT_FALSE(GetRecvStream(kSsrcX).started()); +} + +// Test that we can add and remove send streams. +TEST_P(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) { + SetupForMultiSendStream(); + + // Set the global state for sending. + SetSend(true); + + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + SetAudioSend(ssrc, true, &fake_source_); + // Verify that we are in a sending state for all the created streams. + EXPECT_TRUE(GetSendStream(ssrc).IsSending()); + } + EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size()); + + // Delete the send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(send_channel_->RemoveSendStream(ssrc)); + EXPECT_FALSE(call_.GetAudioSendStream(ssrc)); + EXPECT_FALSE(send_channel_->RemoveSendStream(ssrc)); + } + EXPECT_EQ(0u, call_.GetAudioSendStreams().size()); +} + +// Test SetSendCodecs correctly configure the codecs in all send streams. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) { + SetupForMultiSendStream(); + + // Create send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + } + + cricket::AudioSenderParameter parameters; + // Set PCMU and CN(8K). VAD should be activated. + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs[1].id = 97; + SetSenderParameters(parameters); + + // Verify PCMU and VAD are corrected configured on all send channels. + for (uint32_t ssrc : kSsrcs4) { + ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); + const auto& send_codec_spec = + *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1u, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + } + + // Change to PCMU(8K) and CN(16K). + parameters.codecs[0] = kPcmuCodec; + parameters.codecs[1] = kCn16000Codec; + SetSenderParameters(parameters); + for (uint32_t ssrc : kSsrcs4) { + ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); + const auto& send_codec_spec = + *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(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(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + SetAudioSend(ssrc, true, &fake_source_); + EXPECT_FALSE(GetSendStream(ssrc).IsSending()); + } + + // Set the global state for starting sending. + SetSend(true); + for (uint32_t ssrc : kSsrcs4) { + // Verify that we are in a sending state for all the send streams. + EXPECT_TRUE(GetSendStream(ssrc).IsSending()); + } + + // Set the global state for stopping sending. + SetSend(false); + for (uint32_t ssrc : kSsrcs4) { + // Verify that we are in a stop state for all the send streams. + EXPECT_FALSE(GetSendStream(ssrc).IsSending()); + } +} + +// Test we can set the correct statistics on all send streams. +TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) { + SetupForMultiSendStream(); + + // Create send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + } + + // Create a receive stream to check that none of the send streams end up in + // the receive stream stats. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + + // We need send codec to be set to get all stats. + SetSenderParameters(send_parameters_); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); + SetAudioSendStreamStats(); + SetAudioReceiveStreamStats(); + + // Check stats for the added streams. + { + EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); + cricket::VoiceMediaSendInfo send_info; + cricket::VoiceMediaReceiveInfo receive_info; + EXPECT_EQ(true, send_channel_->GetStats(&send_info)); + EXPECT_EQ(true, receive_channel_->GetStats( + &receive_info, /*get_and_clear_legacy_stats=*/true)); + + // We have added 4 send streams. We should see empty stats for all. + EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), + send_info.senders.size()); + for (const auto& sender : send_info.senders) { + VerifyVoiceSenderInfo(sender, false); + } + VerifyVoiceSendRecvCodecs(send_info, receive_info); + + // We have added one receive stream. We should see empty stats. + EXPECT_EQ(receive_info.receivers.size(), 1u); + EXPECT_EQ(receive_info.receivers[0].ssrc(), 123u); + } + + // Remove the kSsrcY stream. No receiver stats. + { + cricket::VoiceMediaReceiveInfo receive_info; + cricket::VoiceMediaSendInfo send_info; + EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY)); + EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); + EXPECT_EQ(true, send_channel_->GetStats(&send_info)); + EXPECT_EQ(true, receive_channel_->GetStats( + &receive_info, /*get_and_clear_legacy_stats=*/true)); + EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), + send_info.senders.size()); + EXPECT_EQ(0u, receive_info.receivers.size()); + } + + // Deliver a new packet - a default receive stream should be created and we + // should see stats again. + { + cricket::VoiceMediaSendInfo send_info; + cricket::VoiceMediaReceiveInfo receive_info; + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + SetAudioReceiveStreamStats(); + EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); + EXPECT_EQ(true, send_channel_->GetStats(&send_info)); + EXPECT_EQ(true, receive_channel_->GetStats( + &receive_info, /*get_and_clear_legacy_stats=*/true)); + EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), + send_info.senders.size()); + EXPECT_EQ(1u, receive_info.receivers.size()); + VerifyVoiceReceiverInfo(receive_info.receivers[0]); + VerifyVoiceSendRecvCodecs(send_info, receive_info); + } +} + +// Test that we can add and remove receive streams, and do proper send/playout. +// We can receive on multiple streams while sending one stream. +TEST_P(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) { + EXPECT_TRUE(SetupSendStream()); + + // Start playout without a receive stream. + SetSenderParameters(send_parameters_); + receive_channel_->SetPlayout(true); + + // Adding another stream should enable playout on the new stream only. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + + // Make sure only the new stream is played out. + EXPECT_TRUE(GetRecvStream(kSsrcY).started()); + + // Adding yet another stream should have stream 2 and 3 enabled for playout. + EXPECT_TRUE(AddRecvStream(kSsrcZ)); + EXPECT_TRUE(GetRecvStream(kSsrcY).started()); + EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); + + // Stop sending. + SetSend(false); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Stop playout. + receive_channel_->SetPlayout(false); + EXPECT_FALSE(GetRecvStream(kSsrcY).started()); + EXPECT_FALSE(GetRecvStream(kSsrcZ).started()); + + // Restart playout and make sure recv streams are played out. + receive_channel_->SetPlayout(true); + EXPECT_TRUE(GetRecvStream(kSsrcY).started()); + EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); + + // Now remove the recv streams. + EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcZ)); + EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY)); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) { + EXPECT_TRUE(SetupSendStream()); + send_parameters_.options.audio_network_adaptor = true; + send_parameters_.options.audio_network_adaptor_config = {"1234"}; + SetSenderParameters(send_parameters_); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); +} + +TEST_P(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) { + EXPECT_TRUE(SetupSendStream()); + send_parameters_.options.audio_network_adaptor = true; + send_parameters_.options.audio_network_adaptor_config = {"1234"}; + SetSenderParameters(send_parameters_); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); + cricket::AudioOptions options; + options.audio_network_adaptor = false; + SetAudioSend(kSsrcX, true, nullptr, &options); + EXPECT_EQ(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"}; + SetSenderParameters(send_parameters_); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); + const int initial_num = call_.GetNumCreatedSendStreams(); + cricket::AudioOptions options; + options.audio_network_adaptor = 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)); +} + +// Test that we can set the outgoing SSRC properly. +// SSRC is set in SetupSendStream() by calling AddSendStream. +TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrc) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); +} + +TEST_P(WebRtcVoiceEngineTestFake, GetStats) { + // Setup. We need send codec to be set to get all stats. + EXPECT_TRUE(SetupSendStream()); + // SetupSendStream adds a send stream with kSsrcX, so the receive + // stream has to use a different SSRC. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + SetSenderParameters(send_parameters_); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); + SetAudioSendStreamStats(); + + // Check stats for the added streams. + { + EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); + cricket::VoiceMediaSendInfo send_info; + cricket::VoiceMediaReceiveInfo receive_info; + EXPECT_EQ(true, send_channel_->GetStats(&send_info)); + EXPECT_EQ(true, receive_channel_->GetStats( + &receive_info, /*get_and_clear_legacy_stats=*/true)); + + // We have added one send stream. We should see the stats we've set. + EXPECT_EQ(1u, send_info.senders.size()); + VerifyVoiceSenderInfo(send_info.senders[0], false); + // We have added one receive stream. We should see empty stats. + EXPECT_EQ(receive_info.receivers.size(), 1u); + EXPECT_EQ(receive_info.receivers[0].ssrc(), 0u); + } + + // Start sending - this affects some reported stats. + { + SetSend(true); + EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); + cricket::VoiceMediaSendInfo send_info; + cricket::VoiceMediaReceiveInfo receive_info; + SetAudioReceiveStreamStats(); + EXPECT_EQ(true, send_channel_->GetStats(&send_info)); + EXPECT_EQ(true, receive_channel_->GetStats( + &receive_info, /*get_and_clear_legacy_stats=*/true)); + VerifyVoiceSenderInfo(send_info.senders[0], true); + VerifyVoiceSendRecvCodecs(send_info, receive_info); + } + + // Remove the kSsrcY stream. No receiver stats. + { + EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrcY)); + EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); + cricket::VoiceMediaSendInfo send_info; + cricket::VoiceMediaReceiveInfo receive_info; + EXPECT_EQ(true, send_channel_->GetStats(&send_info)); + EXPECT_EQ(true, receive_channel_->GetStats( + &receive_info, /*get_and_clear_legacy_stats=*/true)); + EXPECT_EQ(1u, send_info.senders.size()); + EXPECT_EQ(0u, receive_info.receivers.size()); + } + + // Deliver a new packet - a default receive stream should be created and we + // should see stats again. + { + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + SetAudioReceiveStreamStats(); + EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0)); + cricket::VoiceMediaSendInfo send_info; + cricket::VoiceMediaReceiveInfo receive_info; + EXPECT_EQ(true, send_channel_->GetStats(&send_info)); + EXPECT_EQ(true, receive_channel_->GetStats( + &receive_info, /*get_and_clear_legacy_stats=*/true)); + EXPECT_EQ(1u, send_info.senders.size()); + EXPECT_EQ(1u, receive_info.receivers.size()); + VerifyVoiceReceiverInfo(receive_info.receivers[0]); + VerifyVoiceSendRecvCodecs(send_info, receive_info); + } +} + +// Test that we can set the outgoing SSRC properly with multiple streams. +// SSRC is set in SetupSendStream() by calling AddSendStream. +TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); +} + +// Test that the local SSRC is the same on sending and receiving channels if the +// receive channel is created before the send channel. +TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); +} + +// Test that we can properly receive packets. +TEST_P(WebRtcVoiceEngineTestFake, Recv) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(1)); + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + EXPECT_TRUE( + GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); +} + +// Test that we can properly receive packets on multiple streams. +TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) { + EXPECT_TRUE(SetupChannel()); + const uint32_t ssrc1 = 1; + const uint32_t ssrc2 = 2; + const uint32_t ssrc3 = 3; + EXPECT_TRUE(AddRecvStream(ssrc1)); + EXPECT_TRUE(AddRecvStream(ssrc2)); + EXPECT_TRUE(AddRecvStream(ssrc3)); + // Create packets with the right SSRCs. + unsigned char packets[4][sizeof(kPcmuFrame)]; + for (size_t i = 0; i < arraysize(packets); ++i) { + memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i)); + } + + const cricket::FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1); + const cricket::FakeAudioReceiveStream& s2 = GetRecvStream(ssrc2); + const cricket::FakeAudioReceiveStream& s3 = GetRecvStream(ssrc3); + + EXPECT_EQ(s1.received_packets(), 0); + EXPECT_EQ(s2.received_packets(), 0); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[0], sizeof(packets[0])); + EXPECT_EQ(s1.received_packets(), 0); + EXPECT_EQ(s2.received_packets(), 0); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[1], sizeof(packets[1])); + EXPECT_EQ(s1.received_packets(), 1); + EXPECT_TRUE(s1.VerifyLastPacket(packets[1], sizeof(packets[1]))); + EXPECT_EQ(s2.received_packets(), 0); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[2], sizeof(packets[2])); + EXPECT_EQ(s1.received_packets(), 1); + EXPECT_EQ(s2.received_packets(), 1); + EXPECT_TRUE(s2.VerifyLastPacket(packets[2], sizeof(packets[2]))); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[3], sizeof(packets[3])); + EXPECT_EQ(s1.received_packets(), 1); + EXPECT_EQ(s2.received_packets(), 1); + EXPECT_EQ(s3.received_packets(), 1); + EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3]))); + + EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc3)); + EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc2)); + EXPECT_TRUE(receive_channel_->RemoveRecvStream(ssrc1)); +} + +// Test that receiving on an unsignaled stream works (a stream is created). +TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaled) { + EXPECT_TRUE(SetupChannel()); + EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size()); + + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE( + GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); +} + +// Tests that when we add a stream without SSRCs, but contains a stream_id +// that it is stored and its stream id is later used when the first packet +// arrives to properly create a receive stream with a sync label. +TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) { + const char kSyncLabel[] = "sync_label"; + EXPECT_TRUE(SetupChannel()); + cricket::StreamParams unsignaled_stream; + unsignaled_stream.set_stream_ids({kSyncLabel}); + ASSERT_TRUE(receive_channel_->AddRecvStream(unsignaled_stream)); + // The stream shouldn't have been created at this point because it doesn't + // have any SSRCs. + EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size()); + + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE( + GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); + EXPECT_EQ(kSyncLabel, GetRecvStream(kSsrc1).GetConfig().sync_group); + + // Remset the unsignaled stream to clear the cached parameters. If a new + // default unsignaled receive stream is created it will not have a sync group. + receive_channel_->ResetUnsignaledRecvStream(); + receive_channel_->RemoveRecvStream(kSsrc1); + + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE( + GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); + EXPECT_TRUE(GetRecvStream(kSsrc1).GetConfig().sync_group.empty()); +} + +TEST_P(WebRtcVoiceEngineTestFake, + ResetUnsignaledRecvStreamDeletesAllDefaultStreams) { + ASSERT_TRUE(SetupChannel()); + // No receive streams to start with. + ASSERT_TRUE(call_.GetAudioReceiveStreams().empty()); + + // Deliver a couple packets with unsignaled SSRCs. + unsigned char packet[sizeof(kPcmuFrame)]; + memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(&packet[8], 0x1234); + DeliverPacket(packet, sizeof(packet)); + rtc::SetBE32(&packet[8], 0x5678); + DeliverPacket(packet, sizeof(packet)); + + // Verify that the receive streams were created. + const auto& receivers1 = call_.GetAudioReceiveStreams(); + ASSERT_EQ(receivers1.size(), 2u); + + // Should remove all default streams. + receive_channel_->ResetUnsignaledRecvStream(); + const auto& receivers2 = call_.GetAudioReceiveStreams(); + EXPECT_EQ(0u, receivers2.size()); +} + +// Test that receiving N unsignaled stream works (streams will be created), and +// that packets are forwarded to them all. +TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) { + EXPECT_TRUE(SetupChannel()); + unsigned char packet[sizeof(kPcmuFrame)]; + memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); + + // Note that SSRC = 0 is not supported. + for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) { + rtc::SetBE32(&packet[8], ssrc); + DeliverPacket(packet, sizeof(packet)); + + // Verify we have one new stream for each loop iteration. + EXPECT_EQ(ssrc, call_.GetAudioReceiveStreams().size()); + EXPECT_EQ(1, GetRecvStream(ssrc).received_packets()); + EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); + } + + // Sending on the same SSRCs again should not create new streams. + for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) { + rtc::SetBE32(&packet[8], ssrc); + DeliverPacket(packet, sizeof(packet)); + + EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size()); + EXPECT_EQ(2, GetRecvStream(ssrc).received_packets()); + EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); + } + + // Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced. + constexpr uint32_t kAnotherSsrc = 667; + rtc::SetBE32(&packet[8], kAnotherSsrc); + DeliverPacket(packet, sizeof(packet)); + + const auto& streams = call_.GetAudioReceiveStreams(); + EXPECT_EQ(kMaxUnsignaledRecvStreams, streams.size()); + size_t i = 0; + for (uint32_t ssrc = 2; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc, ++i) { + EXPECT_EQ(ssrc, streams[i]->GetConfig().rtp.remote_ssrc); + EXPECT_EQ(2, streams[i]->received_packets()); + } + EXPECT_EQ(kAnotherSsrc, streams[i]->GetConfig().rtp.remote_ssrc); + EXPECT_EQ(1, streams[i]->received_packets()); + // Sanity check that we've checked all streams. + EXPECT_EQ(kMaxUnsignaledRecvStreams, (i + 1)); +} + +// Test that a default channel is created even after a signaled stream has been +// added, and that this stream will get any packets for unknown SSRCs. +TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) { + EXPECT_TRUE(SetupChannel()); + unsigned char packet[sizeof(kPcmuFrame)]; + memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); + + // Add a known stream, send packet and verify we got it. + const uint32_t signaled_ssrc = 1; + rtc::SetBE32(&packet[8], signaled_ssrc); + EXPECT_TRUE(AddRecvStream(signaled_ssrc)); + DeliverPacket(packet, sizeof(packet)); + EXPECT_TRUE( + GetRecvStream(signaled_ssrc).VerifyLastPacket(packet, sizeof(packet))); + EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); + + // Note that the first unknown SSRC cannot be 0, because we only support + // creating receive streams for SSRC!=0. + const uint32_t unsignaled_ssrc = 7011; + rtc::SetBE32(&packet[8], unsignaled_ssrc); + DeliverPacket(packet, sizeof(packet)); + EXPECT_TRUE( + GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet, sizeof(packet))); + EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); + + DeliverPacket(packet, sizeof(packet)); + EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets()); + + rtc::SetBE32(&packet[8], signaled_ssrc); + DeliverPacket(packet, sizeof(packet)); + EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets()); + EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); +} + +// Two tests to verify that adding a receive stream with the same SSRC as a +// previously added unsignaled stream will only recreate underlying stream +// objects if the stream parameters have changed. +TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) { + EXPECT_TRUE(SetupChannel()); + + // Spawn unsignaled stream with SSRC=1. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE( + GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); + + // Verify that the underlying stream object in Call is not recreated when a + // stream with SSRC=1 is added. + const auto& streams = call_.GetAudioReceiveStreams(); + EXPECT_EQ(1u, streams.size()); + int audio_receive_stream_id = streams.front()->id(); + EXPECT_TRUE(AddRecvStream(1)); + EXPECT_EQ(1u, streams.size()); + EXPECT_EQ(audio_receive_stream_id, streams.front()->id()); +} + +TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Updates) { + EXPECT_TRUE(SetupChannel()); + + // Spawn unsignaled stream with SSRC=1. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE( + GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame))); + + // Verify that the underlying stream object in Call gets updated when a + // stream with SSRC=1 is added, and which has changed stream parameters. + const auto& streams = call_.GetAudioReceiveStreams(); + EXPECT_EQ(1u, streams.size()); + // The sync_group id should be empty. + EXPECT_TRUE(streams.front()->GetConfig().sync_group.empty()); + + const std::string new_stream_id("stream_id"); + int audio_receive_stream_id = streams.front()->id(); + cricket::StreamParams stream_params; + stream_params.ssrcs.push_back(1); + stream_params.set_stream_ids({new_stream_id}); + + EXPECT_TRUE(receive_channel_->AddRecvStream(stream_params)); + EXPECT_EQ(1u, streams.size()); + // The audio receive stream should not have been recreated. + EXPECT_EQ(audio_receive_stream_id, streams.front()->id()); + + // The sync_group id should now match with the new stream params. + EXPECT_EQ(new_stream_id, streams.front()->GetConfig().sync_group); +} + +// Test that AddRecvStream creates new stream. +TEST_P(WebRtcVoiceEngineTestFake, AddRecvStream) { + EXPECT_TRUE(SetupRecvStream()); + EXPECT_TRUE(AddRecvStream(1)); +} + +// Test that after adding a recv stream, we do not decode more codecs than +// those previously passed into SetRecvCodecs. +TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioReceiverParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, {111, {"OPUS", 48000, 2}}}))); +} + +// Test that we properly clean up any streams that were added, even if +// not explicitly removed. +TEST_P(WebRtcVoiceEngineTestFake, StreamCleanup) { + EXPECT_TRUE(SetupSendStream()); + SetSenderParameters(send_parameters_); + EXPECT_TRUE(AddRecvStream(1)); + EXPECT_TRUE(AddRecvStream(2)); + + EXPECT_EQ(1u, call_.GetAudioSendStreams().size()); + EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); + send_channel_.reset(); + receive_channel_.reset(); + EXPECT_EQ(0u, call_.GetAudioSendStreams().size()); + EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size()); +} + +TEST_P(WebRtcVoiceEngineTestFake, TestAddRecvStreamSuccessWithZeroSsrc) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(0)); +} + +TEST_P(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithSameSsrc) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(1)); + EXPECT_FALSE(AddRecvStream(1)); +} + +// Test the InsertDtmf on default send stream as caller. +TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) { + TestInsertDtmf(0, true, kTelephoneEventCodec1); +} + +// Test the InsertDtmf on default send stream as callee +TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) { + TestInsertDtmf(0, false, kTelephoneEventCodec2); +} + +// Test the InsertDtmf on specified send stream as caller. +TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) { + TestInsertDtmf(kSsrcX, true, kTelephoneEventCodec2); +} + +// Test the InsertDtmf on specified send stream as callee. +TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) { + TestInsertDtmf(kSsrcX, false, kTelephoneEventCodec1); +} + +// Test propagation of extmap allow mixed setting. +TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCaller) { + TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true); +} +TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCaller) { + TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false); +} +TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCallee) { + TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true); +} +TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCallee) { + TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetAudioOptions) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_CALL(*adm_, BuiltInAECIsAvailable()) + .Times(8) + .WillRepeatedly(Return(false)); + EXPECT_CALL(*adm_, BuiltInAGCIsAvailable()) + .Times(4) + .WillRepeatedly(Return(false)); + EXPECT_CALL(*adm_, BuiltInNSIsAvailable()) + .Times(2) + .WillRepeatedly(Return(false)); + + EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); + EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); + + // Nothing set in AudioOptions, so everything should be as default. + send_parameters_.options = cricket::AudioOptions(); + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_TRUE(IsHighPassFilterEnabled()); + } + EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets); + EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate); + + // Turn echo cancellation off + send_parameters_.options.echo_cancellation = false; + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/false); + } + + // Turn echo cancellation back on, with settings, and make sure + // nothing else changed. + send_parameters_.options.echo_cancellation = true; + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + } + + // Turn off echo cancellation and delay agnostic aec. + send_parameters_.options.echo_cancellation = false; + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/false); + } + + // Restore AEC to be on to work with the following tests. + send_parameters_.options.echo_cancellation = true; + SetSenderParameters(send_parameters_); + + // Turn off AGC + send_parameters_.options.auto_gain_control = false; + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_FALSE(apm_config_.gain_controller1.enabled); + } + + // Turn AGC back on + send_parameters_.options.auto_gain_control = true; + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_TRUE(apm_config_.gain_controller1.enabled); + } + + // Turn off other options. + send_parameters_.options.noise_suppression = false; + send_parameters_.options.highpass_filter = false; + send_parameters_.options.stereo_swapping = true; + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_FALSE(IsHighPassFilterEnabled()); + EXPECT_TRUE(apm_config_.gain_controller1.enabled); + EXPECT_FALSE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + } + + // Set options again to ensure it has no impact. + SetSenderParameters(send_parameters_); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_TRUE(apm_config_.gain_controller1.enabled); + EXPECT_FALSE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + } +} + +TEST_P(WebRtcVoiceEngineTestFake, InitRecordingOnSend) { + EXPECT_CALL(*adm_, RecordingIsInitialized()).WillOnce(Return(false)); + EXPECT_CALL(*adm_, Recording()).WillOnce(Return(false)); + EXPECT_CALL(*adm_, InitRecording()).Times(1); + + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel( + engine_->CreateSendChannel( + &call_, cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create())); + + send_channel->SetSend(true); +} + +TEST_P(WebRtcVoiceEngineTestFake, SkipInitRecordingOnSend) { + EXPECT_CALL(*adm_, RecordingIsInitialized()).Times(0); + EXPECT_CALL(*adm_, Recording()).Times(0); + EXPECT_CALL(*adm_, InitRecording()).Times(0); + + cricket::AudioOptions options; + options.init_recording_on_send = false; + + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel( + engine_->CreateSendChannel(&call_, cricket::MediaConfig(), options, + webrtc::CryptoOptions(), + webrtc::AudioCodecPairId::Create())); + + send_channel->SetSend(true); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_CALL(*adm_, BuiltInAECIsAvailable()) + .Times(use_null_apm_ ? 4 : 8) + .WillRepeatedly(Return(false)); + EXPECT_CALL(*adm_, BuiltInAGCIsAvailable()) + .Times(use_null_apm_ ? 7 : 8) + .WillRepeatedly(Return(false)); + EXPECT_CALL(*adm_, BuiltInNSIsAvailable()) + .Times(use_null_apm_ ? 5 : 8) + .WillRepeatedly(Return(false)); + EXPECT_CALL(*adm_, RecordingIsInitialized()) + .Times(2) + .WillRepeatedly(Return(false)); + + EXPECT_CALL(*adm_, Recording()).Times(2).WillRepeatedly(Return(false)); + EXPECT_CALL(*adm_, InitRecording()).Times(2).WillRepeatedly(Return(0)); + + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel1( + engine_->CreateSendChannel( + &call_, cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create())); + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel2( + engine_->CreateSendChannel( + &call_, cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create())); + + // Have to add a stream to make SetSend work. + cricket::StreamParams stream1; + stream1.ssrcs.push_back(1); + send_channel1->AddSendStream(stream1); + cricket::StreamParams stream2; + stream2.ssrcs.push_back(2); + send_channel2->AddSendStream(stream2); + + // AEC and AGC and NS + cricket::AudioSenderParameter parameters_options_all = send_parameters_; + parameters_options_all.options.echo_cancellation = true; + parameters_options_all.options.auto_gain_control = true; + parameters_options_all.options.noise_suppression = true; + EXPECT_TRUE(send_channel1->SetSenderParameters(parameters_options_all)); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + VerifyGainControlEnabledCorrectly(); + EXPECT_TRUE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + EXPECT_EQ(parameters_options_all.options, + SendImplFromPointer(send_channel1.get())->options()); + EXPECT_TRUE(send_channel2->SetSenderParameters(parameters_options_all)); + VerifyEchoCancellationSettings(/*enabled=*/true); + VerifyGainControlEnabledCorrectly(); + EXPECT_EQ(parameters_options_all.options, + SendImplFromPointer(send_channel2.get())->options()); + } + + // unset NS + cricket::AudioSenderParameter parameters_options_no_ns = send_parameters_; + parameters_options_no_ns.options.noise_suppression = false; + EXPECT_TRUE(send_channel1->SetSenderParameters(parameters_options_no_ns)); + cricket::AudioOptions expected_options = parameters_options_all.options; + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_FALSE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + VerifyGainControlEnabledCorrectly(); + expected_options.echo_cancellation = true; + expected_options.auto_gain_control = true; + expected_options.noise_suppression = false; + EXPECT_EQ(expected_options, + SendImplFromPointer(send_channel1.get())->options()); + } + + // unset AGC + cricket::AudioSenderParameter parameters_options_no_agc = send_parameters_; + parameters_options_no_agc.options.auto_gain_control = false; + EXPECT_TRUE(send_channel2->SetSenderParameters(parameters_options_no_agc)); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_FALSE(apm_config_.gain_controller1.enabled); + EXPECT_TRUE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + expected_options.echo_cancellation = true; + expected_options.auto_gain_control = false; + expected_options.noise_suppression = true; + EXPECT_EQ(expected_options, + SendImplFromPointer(send_channel2.get())->options()); + } + + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters_options_all)); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + VerifyGainControlEnabledCorrectly(); + EXPECT_TRUE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + } + + send_channel1->SetSend(true); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + VerifyGainControlEnabledCorrectly(); + EXPECT_FALSE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + } + + send_channel2->SetSend(true); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_FALSE(apm_config_.gain_controller1.enabled); + EXPECT_TRUE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + } + + // Make sure settings take effect while we are sending. + cricket::AudioSenderParameter parameters_options_no_agc_nor_ns = + send_parameters_; + parameters_options_no_agc_nor_ns.options.auto_gain_control = false; + parameters_options_no_agc_nor_ns.options.noise_suppression = false; + EXPECT_TRUE( + send_channel2->SetSenderParameters(parameters_options_no_agc_nor_ns)); + if (!use_null_apm_) { + VerifyEchoCancellationSettings(/*enabled=*/true); + EXPECT_FALSE(apm_config_.gain_controller1.enabled); + EXPECT_FALSE(apm_config_.noise_suppression.enabled); + EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel); + expected_options.echo_cancellation = true; + expected_options.auto_gain_control = false; + expected_options.noise_suppression = false; + EXPECT_EQ(expected_options, + SendImplFromPointer(send_channel2.get())->options()); + } +} + +// This test verifies DSCP settings are properly applied on voice media channel. +TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) { + EXPECT_TRUE(SetupSendStream()); + cricket::FakeNetworkInterface network_interface; + cricket::MediaConfig config; + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> channel; + webrtc::RtpParameters parameters; + + channel = engine_->CreateSendChannel(&call_, config, cricket::AudioOptions(), + webrtc::CryptoOptions(), + webrtc::AudioCodecPairId::Create()); + channel->SetInterface(&network_interface); + // Default value when DSCP is disabled should be DSCP_DEFAULT. + EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); + channel->SetInterface(nullptr); + + config.enable_dscp = true; + channel = engine_->CreateSendChannel(&call_, config, cricket::AudioOptions(), + webrtc::CryptoOptions(), + webrtc::AudioCodecPairId::Create()); + channel->SetInterface(&network_interface); + EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); + + // Create a send stream to configure + EXPECT_TRUE( + channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ))); + parameters = channel->GetRtpSendParameters(kSsrcZ); + ASSERT_FALSE(parameters.encodings.empty()); + + // Various priorities map to various dscp values. + parameters.encodings[0].network_priority = webrtc::Priority::kHigh; + ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters, nullptr).ok()); + EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp()); + parameters.encodings[0].network_priority = webrtc::Priority::kVeryLow; + ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters, nullptr).ok()); + EXPECT_EQ(rtc::DSCP_CS1, network_interface.dscp()); + + // Packets should also self-identify their dscp in PacketOptions. + const uint8_t kData[10] = {0}; + EXPECT_TRUE(SendImplFromPointer(channel.get())->transport()->SendRtcp(kData)); + EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp); + channel->SetInterface(nullptr); + + // Verify that setting the option to false resets the + // DiffServCodePoint. + config.enable_dscp = false; + channel = engine_->CreateSendChannel(&call_, config, cricket::AudioOptions(), + webrtc::CryptoOptions(), + webrtc::AudioCodecPairId::Create()); + channel->SetInterface(&network_interface); + // Default value when DSCP is disabled should be DSCP_DEFAULT. + EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); + + channel->SetInterface(nullptr); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolume) { + EXPECT_TRUE(SetupChannel()); + EXPECT_FALSE(receive_channel_->SetOutputVolume(kSsrcY, 0.5)); + cricket::StreamParams stream; + stream.ssrcs.push_back(kSsrcY); + EXPECT_TRUE(receive_channel_->AddRecvStream(stream)); + EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrcY).gain()); + EXPECT_TRUE(receive_channel_->SetOutputVolume(kSsrcY, 3)); + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcY).gain()); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) { + EXPECT_TRUE(SetupChannel()); + + // Spawn an unsignaled stream by sending a packet - gain should be 1. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrc1).gain()); + + // Should remember the volume "2" which will be set on new unsignaled streams, + // and also set the gain to 2 on existing unsignaled streams. + EXPECT_TRUE(receive_channel_->SetDefaultOutputVolume(2)); + EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrc1).gain()); + + // Spawn an unsignaled stream by sending a packet - gain should be 2. + unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; + memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(&pcmuFrame2[8], kSsrcX); + DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); + EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain()); + + // Setting gain for all unsignaled streams. + EXPECT_TRUE(receive_channel_->SetDefaultOutputVolume(3)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain()); + } + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcX).gain()); + + // Setting gain on an individual stream affects only that. + EXPECT_TRUE(receive_channel_->SetOutputVolume(kSsrcX, 4)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain()); + } + EXPECT_DOUBLE_EQ(4, GetRecvStream(kSsrcX).gain()); +} + +TEST_P(WebRtcVoiceEngineTestFake, BaseMinimumPlayoutDelayMs) { + EXPECT_TRUE(SetupChannel()); + EXPECT_FALSE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 200)); + EXPECT_FALSE( + receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); + + cricket::StreamParams stream; + stream.ssrcs.push_back(kSsrcY); + EXPECT_TRUE(receive_channel_->AddRecvStream(stream)); + EXPECT_EQ(0, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms()); + EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 300)); + EXPECT_EQ(300, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms()); +} + +TEST_P(WebRtcVoiceEngineTestFake, + BaseMinimumPlayoutDelayMsUnsignaledRecvStream) { + // Here base minimum delay is abbreviated to delay in comments for shortness. + EXPECT_TRUE(SetupChannel()); + + // Spawn an unsignaled stream by sending a packet - delay should be 0. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_EQ( + 0, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1)); + // Check that it doesn't provide default values for unknown ssrc. + EXPECT_FALSE( + receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); + + // Check that default value for unsignaled streams is 0. + EXPECT_EQ( + 0, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1)); + + // Should remember the delay 100 which will be set on new unsignaled streams, + // and also set the delay to 100 on existing unsignaled streams. + EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 100)); + EXPECT_EQ( + 100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1)); + // Check that it doesn't provide default values for unknown ssrc. + EXPECT_FALSE( + receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); + + // Spawn an unsignaled stream by sending a packet - delay should be 100. + unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; + memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(&pcmuFrame2[8], kSsrcX); + DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); + EXPECT_EQ( + 100, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1)); + + // Setting delay with SSRC=0 should affect all unsignaled streams. + EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 300)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ( + 300, + receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1)); + } + EXPECT_EQ( + 300, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1)); + + // Setting delay on an individual stream affects only that. + EXPECT_TRUE(receive_channel_->SetBaseMinimumPlayoutDelayMs(kSsrcX, 400)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ( + 300, + receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1)); + } + EXPECT_EQ( + 400, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1)); + EXPECT_EQ( + 300, receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1)); + // Check that it doesn't provide default values for unknown ssrc. + EXPECT_FALSE( + receive_channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value()); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetsSyncGroupFromStreamId) { + const uint32_t kAudioSsrc = 123; + const std::string kStreamId = "AvSyncLabel"; + + EXPECT_TRUE(SetupSendStream()); + cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kAudioSsrc); + sp.set_stream_ids({kStreamId}); + // Creating two channels to make sure that sync label is set properly for both + // the default voice channel and following ones. + EXPECT_TRUE(receive_channel_->AddRecvStream(sp)); + sp.ssrcs[0] += 1; + EXPECT_TRUE(receive_channel_->AddRecvStream(sp)); + + ASSERT_EQ(2u, call_.GetAudioReceiveStreams().size()); + EXPECT_EQ(kStreamId, + call_.GetAudioReceiveStream(kAudioSsrc)->GetConfig().sync_group) + << "SyncGroup should be set based on stream id"; + EXPECT_EQ(kStreamId, + call_.GetAudioReceiveStream(kAudioSsrc + 1)->GetConfig().sync_group) + << "SyncGroup should be set based on stream id"; +} + +// TODO(solenberg): Remove, once recv streams are configured through Call. +// (This is then covered by TestSetRecvRtpHeaderExtensions.) +TEST_P(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) { + // Test that setting the header extensions results in the expected state + // changes on an associated Call. + std::vector<uint32_t> ssrcs; + ssrcs.push_back(223); + ssrcs.push_back(224); + + EXPECT_TRUE(SetupSendStream()); + SetSenderParameters(send_parameters_); + for (uint32_t ssrc : ssrcs) { + EXPECT_TRUE(receive_channel_->AddRecvStream( + cricket::StreamParams::CreateLegacy(ssrc))); + } + + EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); + for (uint32_t ssrc : ssrcs) { + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(ssrc).header_extensions, + IsEmpty()); + } + + // Set up receive extensions. + const std::vector<webrtc::RtpExtension> header_extensions = + GetDefaultEnabledRtpHeaderExtensions(*engine_); + cricket::AudioReceiverParameters recv_parameters; + recv_parameters.extensions = header_extensions; + receive_channel_->SetReceiverParameters(recv_parameters); + EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size()); + for (uint32_t ssrc : ssrcs) { + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(ssrc).header_extensions, + testing::UnorderedElementsAreArray(header_extensions)); + } + + // Disable receive extensions. + receive_channel_->SetReceiverParameters(cricket::AudioReceiverParameters()); + for (uint32_t ssrc : ssrcs) { + EXPECT_THAT( + receive_channel_->GetRtpReceiverParameters(ssrc).header_extensions, + IsEmpty()); + } +} + +TEST_P(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) { + // Test that packets are forwarded to the Call when configured accordingly. + const uint32_t kAudioSsrc = 1; + rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame)); + static const unsigned char kRtcp[] = { + 0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00, + 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}; + rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp)); + + EXPECT_TRUE(SetupSendStream()); + cricket::VoiceMediaReceiveChannelInterface* media_channel = ReceiveImpl(); + SetSenderParameters(send_parameters_); + EXPECT_TRUE(media_channel->AddRecvStream( + cricket::StreamParams::CreateLegacy(kAudioSsrc))); + + EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size()); + const cricket::FakeAudioReceiveStream* s = + call_.GetAudioReceiveStream(kAudioSsrc); + EXPECT_EQ(0, s->received_packets()); + webrtc::RtpPacketReceived parsed_packet; + RTC_CHECK(parsed_packet.Parse(kPcmuPacket)); + receive_channel_->OnPacketReceived(parsed_packet); + rtc::Thread::Current()->ProcessMessages(0); + + EXPECT_EQ(1, s->received_packets()); +} + +// All receive channels should be associated with the first send channel, +// since they do not send RTCP SR. +TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcZ))); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); + EXPECT_TRUE(AddRecvStream(kSsrcW)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcW).rtp.local_ssrc); +} + +TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) { + EXPECT_TRUE(SetupRecvStream()); + EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcY))); + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); + EXPECT_TRUE(AddRecvStream(kSsrcZ)); + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc); + EXPECT_TRUE(send_channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcW))); + + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSink) { + EXPECT_TRUE(SetupChannel()); + std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink()); + + // Setting the sink before a recv stream exists should do nothing. + receive_channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_1)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); + + // Now try actually setting the sink. + receive_channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_2)); + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + + // Now try resetting it. + receive_channel_->SetRawAudioSink(kSsrcX, nullptr); + EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); +} + +TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) { + EXPECT_TRUE(SetupChannel()); + std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_3(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_4(new FakeAudioSink()); + + // Should be able to set a default sink even when no stream exists. + receive_channel_->SetDefaultRawAudioSink(std::move(fake_sink_1)); + + // Spawn an unsignaled stream by sending a packet - it should be assigned the + // default sink. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + + // Try resetting the default sink. + receive_channel_->SetDefaultRawAudioSink(nullptr); + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + + // Try setting the default sink while the default stream exists. + receive_channel_->SetDefaultRawAudioSink(std::move(fake_sink_2)); + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + + // If we remove and add a default stream, it should get the same sink. + EXPECT_TRUE(receive_channel_->RemoveRecvStream(kSsrc1)); + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + + // Spawn another unsignaled stream - it should be assigned the default sink + // and the previous unsignaled stream should lose it. + unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; + memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(&pcmuFrame2[8], kSsrcX); + DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + + // Reset the default sink - the second unsignaled stream should lose it. + receive_channel_->SetDefaultRawAudioSink(nullptr); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); + + // Try setting the default sink while two streams exists. + receive_channel_->SetDefaultRawAudioSink(std::move(fake_sink_3)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + + // Try setting the sink for the first unsignaled stream using its known SSRC. + receive_channel_->SetRawAudioSink(kSsrc1, std::move(fake_sink_4)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_NE(GetRecvStream(kSsrc1).sink(), GetRecvStream(kSsrcX).sink()); + } +} + +// Test that, just like the video channel, the voice channel communicates the +// network state to the call. +TEST_P(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) { + EXPECT_TRUE(SetupChannel()); + + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::AUDIO)); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::VIDEO)); + + send_channel_->OnReadyToSend(false); + EXPECT_EQ(webrtc::kNetworkDown, + call_.GetNetworkState(webrtc::MediaType::AUDIO)); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::VIDEO)); + + send_channel_->OnReadyToSend(true); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::AUDIO)); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::VIDEO)); +} + +// Test that playout is still started after changing parameters +TEST_P(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) { + SetupRecvStream(); + receive_channel_->SetPlayout(true); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); + + // Changing RTP header extensions will recreate the + // AudioReceiveStreamInterface. + cricket::AudioReceiverParameters parameters; + parameters.extensions.push_back( + webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); + receive_channel_->SetReceiverParameters(parameters); + + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); +} + +// Tests when GetSources is called with non-existing ssrc, it will return an +// empty list of RtpSource without crashing. +TEST_P(WebRtcVoiceEngineTestFake, GetSourcesWithNonExistingSsrc) { + // Setup an recv stream with `kSsrcX`. + SetupRecvStream(); + cricket::WebRtcVoiceReceiveChannel* media_channel = ReceiveImpl(); + // Call GetSources with `kSsrcY` which doesn't exist. + std::vector<webrtc::RtpSource> sources = media_channel->GetSources(kSsrcY); + EXPECT_EQ(0u, sources.size()); +} + +// Tests that the library initializes and shuts down properly. +TEST(WebRtcVoiceEngineTest, StartupShutdown) { + rtc::AutoThread main_thread; + for (bool use_null_apm : {false, true}) { + // If the VoiceEngine wants to gather available codecs early, that's fine + // but we never want it to create a decoder at this stage. + std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = + webrtc::CreateDefaultTaskQueueFactory(); + rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = + webrtc::test::MockAudioDeviceModule::CreateNice(); + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create(); + webrtc::FieldTrialBasedConfig field_trials; + cricket::WebRtcVoiceEngine engine( + task_queue_factory.get(), adm.get(), + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, + nullptr, nullptr, field_trials); + engine.Init(); + webrtc::RtcEventLogNull event_log; + CallConfig call_config(&event_log); + call_config.trials = &field_trials; + call_config.task_queue_factory = task_queue_factory.get(); + std::unique_ptr<Call> call = Call::Create(call_config); + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel = + engine.CreateSendChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + EXPECT_TRUE(send_channel); + std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> + receive_channel = engine.CreateReceiveChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + EXPECT_TRUE(receive_channel); + } +} + +// Tests that reference counting on the external ADM is correct. +TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) { + rtc::AutoThread main_thread; + for (bool use_null_apm : {false, true}) { + std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = + webrtc::CreateDefaultTaskQueueFactory(); + auto adm = rtc::make_ref_counted< + ::testing::NiceMock<webrtc::test::MockAudioDeviceModule>>(); + { + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create(); + webrtc::FieldTrialBasedConfig field_trials; + cricket::WebRtcVoiceEngine engine( + task_queue_factory.get(), adm.get(), + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, + nullptr, nullptr, field_trials); + engine.Init(); + webrtc::RtcEventLogNull event_log; + CallConfig call_config(&event_log); + call_config.trials = &field_trials; + call_config.task_queue_factory = task_queue_factory.get(); + std::unique_ptr<Call> call = Call::Create(call_config); + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel = + engine.CreateSendChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + EXPECT_TRUE(send_channel); + std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> + receive_channel = engine.CreateReceiveChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + EXPECT_TRUE(receive_channel); + } + // The engine/channel should have dropped their references. + EXPECT_EQ(adm.release()->Release(), + rtc::RefCountReleaseStatus::kDroppedLastRef); + } +} + +// Verify the payload id of common audio codecs, including CN and G722. +TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) { + for (bool use_null_apm : {false, true}) { + std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = + webrtc::CreateDefaultTaskQueueFactory(); + // TODO(ossu): Why are the payload types of codecs with non-static payload + // type assignments checked here? It shouldn't really matter. + rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = + webrtc::test::MockAudioDeviceModule::CreateNice(); + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create(); + webrtc::FieldTrialBasedConfig field_trials; + cricket::WebRtcVoiceEngine engine( + task_queue_factory.get(), adm.get(), + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, + nullptr, nullptr, field_trials); + engine.Init(); + for (const cricket::AudioCodec& codec : engine.send_codecs()) { + auto is_codec = [&codec](const char* name, int clockrate = 0) { + return absl::EqualsIgnoreCase(codec.name, name) && + (clockrate == 0 || codec.clockrate == clockrate); + }; + if (is_codec("CN", 16000)) { + EXPECT_EQ(105, codec.id); + } else if (is_codec("CN", 32000)) { + EXPECT_EQ(106, codec.id); + } else if (is_codec("G722", 8000)) { + EXPECT_EQ(9, codec.id); + } else if (is_codec("telephone-event", 8000)) { + EXPECT_EQ(126, codec.id); + // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned. + // Remove these checks once both send and receive side assigns payload + // types dynamically. + } else if (is_codec("telephone-event", 16000)) { + EXPECT_EQ(113, codec.id); + } else if (is_codec("telephone-event", 32000)) { + EXPECT_EQ(112, codec.id); + } else if (is_codec("telephone-event", 48000)) { + EXPECT_EQ(110, codec.id); + } else if (is_codec("opus")) { + EXPECT_EQ(111, codec.id); + ASSERT_TRUE(codec.params.find("minptime") != codec.params.end()); + EXPECT_EQ("10", codec.params.find("minptime")->second); + ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end()); + EXPECT_EQ("1", codec.params.find("useinbandfec")->second); + } + } + } +} + +// Tests that VoE supports at least 32 channels +TEST(WebRtcVoiceEngineTest, Has32Channels) { + rtc::AutoThread main_thread; + for (bool use_null_apm : {false, true}) { + std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = + webrtc::CreateDefaultTaskQueueFactory(); + rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = + webrtc::test::MockAudioDeviceModule::CreateNice(); + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create(); + webrtc::FieldTrialBasedConfig field_trials; + cricket::WebRtcVoiceEngine engine( + task_queue_factory.get(), adm.get(), + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, + nullptr, nullptr, field_trials); + engine.Init(); + webrtc::RtcEventLogNull event_log; + CallConfig call_config(&event_log); + call_config.trials = &field_trials; + call_config.task_queue_factory = task_queue_factory.get(); + std::unique_ptr<Call> call = Call::Create(call_config); + + std::vector<std::unique_ptr<cricket::VoiceMediaSendChannelInterface>> + channels; + while (channels.size() < 32) { + std::unique_ptr<cricket::VoiceMediaSendChannelInterface> channel = + engine.CreateSendChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + if (!channel) + break; + channels.emplace_back(std::move(channel)); + } + + EXPECT_EQ(channels.size(), 32u); + } +} + +// Test that we set our preferred codecs properly. +TEST(WebRtcVoiceEngineTest, SetRecvCodecs) { + rtc::AutoThread main_thread; + for (bool use_null_apm : {false, true}) { + std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = + webrtc::CreateDefaultTaskQueueFactory(); + // TODO(ossu): I'm not sure of the intent of this test. It's either: + // - Check that our builtin codecs are usable by Channel. + // - The codecs provided by the engine is usable by Channel. + // It does not check that the codecs in the RecvParameters are actually + // what we sent in - though it's probably reasonable to expect so, if + // SetReceiverParameters returns true. + // I think it will become clear once audio decoder injection is completed. + rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = + webrtc::test::MockAudioDeviceModule::CreateNice(); + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create(); + webrtc::FieldTrialBasedConfig field_trials; + cricket::WebRtcVoiceEngine engine( + task_queue_factory.get(), adm.get(), + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm, nullptr, + nullptr, field_trials); + engine.Init(); + webrtc::RtcEventLogNull event_log; + CallConfig call_config(&event_log); + call_config.trials = &field_trials; + call_config.task_queue_factory = task_queue_factory.get(); + std::unique_ptr<Call> call = Call::Create(call_config); + cricket::WebRtcVoiceReceiveChannel channel( + &engine, cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), call.get(), + webrtc::AudioCodecPairId::Create()); + cricket::AudioReceiverParameters parameters; + parameters.codecs = engine.recv_codecs(); + EXPECT_TRUE(channel.SetReceiverParameters(parameters)); + } +} + +TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) { + rtc::AutoThread main_thread; + std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = + webrtc::CreateDefaultTaskQueueFactory(); + rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = + webrtc::test::MockAudioDeviceModule::CreateNice(); + webrtc::FieldTrialBasedConfig field_trials; + FakeAudioSource source; + cricket::WebRtcVoiceEngine engine(task_queue_factory.get(), adm.get(), + webrtc::CreateBuiltinAudioEncoderFactory(), + webrtc::CreateBuiltinAudioDecoderFactory(), + nullptr, nullptr, nullptr, nullptr, + field_trials); + engine.Init(); + webrtc::RtcEventLogNull event_log; + CallConfig call_config(&event_log); + call_config.trials = &field_trials; + call_config.task_queue_factory = task_queue_factory.get(); + { + webrtc::AudioState::Config config; + config.audio_mixer = webrtc::AudioMixerImpl::Create(); + config.audio_device_module = + webrtc::test::MockAudioDeviceModule::CreateNice(); + call_config.audio_state = webrtc::AudioState::Create(config); + } + std::unique_ptr<Call> call = Call::Create(call_config); + cricket::WebRtcVoiceSendChannel channel( + &engine, cricket::MediaConfig(), cricket::AudioOptions(), + webrtc::CryptoOptions(), call.get(), webrtc::AudioCodecPairId::Create()); + { + cricket::AudioSenderParameter params; + params.codecs.push_back(cricket::CreateAudioCodec(1, "opus", 48000, 2)); + params.extensions.push_back(webrtc::RtpExtension( + webrtc::RtpExtension::kTransportSequenceNumberUri, 1)); + EXPECT_TRUE(channel.SetSenderParameters(params)); + } + constexpr int kSsrc = 1234; + { + cricket::StreamParams params; + params.add_ssrc(kSsrc); + channel.AddSendStream(params); + } + channel.SetAudioSend(kSsrc, true, nullptr, &source); + channel.SetSend(true); + webrtc::RtpParameters params = channel.GetRtpSendParameters(kSsrc); + for (int max_bitrate : {-10, -1, 0, 10000}) { + params.encodings[0].max_bitrate_bps = max_bitrate; + channel.SetRtpSendParameters( + kSsrc, params, [](webrtc::RTCError error) { EXPECT_TRUE(error.ok()); }); + } +} + +TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) { + for (bool use_null_apm : {false, true}) { + std::vector<webrtc::AudioCodecSpec> specs; + webrtc::AudioCodecSpec spec1{{"codec1", 48000, 2, {{"param1", "value1"}}}, + {48000, 2, 16000, 10000, 20000}}; + spec1.info.allow_comfort_noise = false; + spec1.info.supports_network_adaption = true; + specs.push_back(spec1); + webrtc::AudioCodecSpec spec2{{"codec2", 32000, 1}, {32000, 1, 32000}}; + spec2.info.allow_comfort_noise = false; + specs.push_back(spec2); + specs.push_back(webrtc::AudioCodecSpec{ + {"codec3", 16000, 1, {{"param1", "value1b"}, {"param2", "value2"}}}, + {16000, 1, 13300}}); + specs.push_back( + webrtc::AudioCodecSpec{{"codec4", 8000, 1}, {8000, 1, 64000}}); + specs.push_back( + webrtc::AudioCodecSpec{{"codec5", 8000, 2}, {8000, 1, 64000}}); + + std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory = + webrtc::CreateDefaultTaskQueueFactory(); + rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> unused_encoder_factory = + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(); + rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory = + rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>(); + EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders()) + .WillOnce(Return(specs)); + rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm = + webrtc::test::MockAudioDeviceModule::CreateNice(); + + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create(); + webrtc::FieldTrialBasedConfig field_trials; + cricket::WebRtcVoiceEngine engine( + task_queue_factory.get(), adm.get(), unused_encoder_factory, + mock_decoder_factory, nullptr, apm, nullptr, nullptr, field_trials); + engine.Init(); + auto codecs = engine.recv_codecs(); + EXPECT_EQ(11u, codecs.size()); + + // Rather than just ASSERTing that there are enough codecs, ensure that we + // can check the actual values safely, to provide better test results. + auto get_codec = [&codecs](size_t index) -> const cricket::AudioCodec& { + static const cricket::AudioCodec missing_codec = + cricket::CreateAudioCodec(0, "<missing>", 0, 0); + if (codecs.size() > index) + return codecs[index]; + return missing_codec; + }; + + // Ensure the general codecs are generated first and in order. + for (size_t i = 0; i != specs.size(); ++i) { + EXPECT_EQ(specs[i].format.name, get_codec(i).name); + EXPECT_EQ(specs[i].format.clockrate_hz, get_codec(i).clockrate); + EXPECT_EQ(specs[i].format.num_channels, get_codec(i).channels); + EXPECT_EQ(specs[i].format.parameters, get_codec(i).params); + } + + // Find the index of a codec, or -1 if not found, so that we can easily + // check supplementary codecs are ordered after the general codecs. + auto find_codec = [&codecs](const webrtc::SdpAudioFormat& format) -> int { + for (size_t i = 0; i != codecs.size(); ++i) { + const cricket::AudioCodec& codec = codecs[i]; + if (absl::EqualsIgnoreCase(codec.name, format.name) && + codec.clockrate == format.clockrate_hz && + codec.channels == format.num_channels) { + return rtc::checked_cast<int>(i); + } + } + return -1; + }; + + // Ensure all supplementary codecs are generated last. Their internal + // ordering is not important. Without this cast, the comparison turned + // unsigned and, thus, failed for -1. + const int num_specs = static_cast<int>(specs.size()); + EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs); + EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs); + EXPECT_EQ(find_codec({"cn", 32000, 1}), -1); + EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs); + EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs); + EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs); + EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs); + } +} |