diff options
Diffstat (limited to 'third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine_unittest.cc')
| -rw-r--r-- | third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine_unittest.cc | 3575 |
1 files changed, 3575 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine_unittest.cc new file mode 100644 index 0000000000..d604d5b919 --- /dev/null +++ b/third_party/libwebrtc/webrtc/media/engine/webrtcvoiceengine_unittest.cc @@ -0,0 +1,3575 @@ +/* + * Copyright (c) 2008 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <memory> +#include <utility> + +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "call/call.h" +#include "logging/rtc_event_log/rtc_event_log.h" +#include "media/base/fakemediaengine.h" +#include "media/base/fakenetworkinterface.h" +#include "media/base/fakertp.h" +#include "media/base/mediaconstants.h" +#include "media/engine/fakewebrtccall.h" +#include "media/engine/fakewebrtcvoiceengine.h" +#include "media/engine/webrtcvoiceengine.h" +#include "modules/audio_device/include/mock_audio_device.h" +#include "modules/audio_processing/include/mock_audio_processing.h" +#include "pc/channel.h" +#include "rtc_base/arraysize.h" +#include "rtc_base/byteorder.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/scoped_ref_ptr.h" +#include "test/field_trial.h" +#include "test/gtest.h" +#include "test/mock_audio_decoder_factory.h" +#include "test/mock_audio_encoder_factory.h" +#include "voice_engine/transmit_mixer.h" + +using testing::_; +using testing::ContainerEq; +using testing::Return; +using testing::ReturnPointee; +using testing::SaveArg; +using testing::StrictMock; + +namespace { + +constexpr uint32_t kMaxUnsignaledRecvStreams = 4; + +const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1); +const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1); +const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 32000, 2); +const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1); +const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1); +const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1); +const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1); +const cricket::AudioCodec + kTelephoneEventCodec1(106, "telephone-event", 8000, 0, 1); +const cricket::AudioCodec + kTelephoneEventCodec2(107, "telephone-event", 32000, 0, 1); + +const uint32_t kSsrc0 = 0; +const uint32_t kSsrc1 = 1; +const uint32_t kSsrcX = 0x99; +const uint32_t kSsrcY = 0x17; +const uint32_t kSsrcZ = 0x42; +const uint32_t kSsrcW = 0x02; +const uint32_t kSsrcs4[] = { 11, 200, 30, 44 }; + +constexpr int kRtpHistoryMs = 5000; + +constexpr webrtc::GainControl::Mode kDefaultAgcMode = +#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) + webrtc::GainControl::kFixedDigital; +#else + webrtc::GainControl::kAdaptiveAnalog; +#endif + +constexpr webrtc::NoiseSuppression::Level kDefaultNsLevel = + webrtc::NoiseSuppression::kHigh; + +class FakeVoEWrapper : public cricket::VoEWrapper { + public: + explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine) + : cricket::VoEWrapper(engine) { + } +}; + +class MockTransmitMixer : public webrtc::voe::TransmitMixer { + public: + MockTransmitMixer() = default; + virtual ~MockTransmitMixer() = default; + + MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable)); +}; + +void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) { + RTC_DCHECK(adm); + + // Setup. + EXPECT_CALL(*adm, AddRef()).Times(1); + EXPECT_CALL(*adm, Init()).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)); + EXPECT_CALL(*adm, SetAGC(true)).WillOnce(Return(0)); + + // 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)); + EXPECT_CALL(*adm, Release()) + .WillOnce(Return(rtc::RefCountReleaseStatus::kDroppedLastRef)); +} +} // namespace + +// Tests that our stub library "works". +TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) { + StrictMock<webrtc::test::MockAudioDeviceModule> adm; + AdmSetupExpectations(&adm); + rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm = + new rtc::RefCountedObject< + StrictMock<webrtc::test::MockAudioProcessing>>(); + webrtc::AudioProcessing::Config apm_config; + EXPECT_CALL(*apm, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config)); + EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config)); + EXPECT_CALL(*apm, SetExtraOptions(testing::_)); + EXPECT_CALL(*apm, DetachAecDump()); + StrictMock<MockTransmitMixer> transmit_mixer; + EXPECT_CALL(transmit_mixer, EnableStereoChannelSwapping(false)); + cricket::FakeWebRtcVoiceEngine voe(&transmit_mixer); + EXPECT_FALSE(voe.IsInited()); + { + cricket::WebRtcVoiceEngine engine( + &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm, + new FakeVoEWrapper(&voe)); + engine.Init(); + EXPECT_TRUE(voe.IsInited()); + } + EXPECT_FALSE(voe.IsInited()); +} + +class FakeAudioSink : public webrtc::AudioSinkInterface { + public: + void OnData(const Data& audio) override {} +}; + +class FakeAudioSource : public cricket::AudioSource { + void SetSink(Sink* sink) override {} +}; + +class WebRtcVoiceEngineTestFake : public testing::Test { + public: + WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {} + + explicit WebRtcVoiceEngineTestFake(const char* field_trials) + : apm_(new rtc::RefCountedObject< + StrictMock<webrtc::test::MockAudioProcessing>>()), + apm_gc_(*apm_->gain_control()), + apm_ec_(*apm_->echo_cancellation()), + apm_ns_(*apm_->noise_suppression()), + apm_vd_(*apm_->voice_detection()), + call_(webrtc::Call::Config(&event_log_)), + voe_(&transmit_mixer_), + override_field_trials_(field_trials) { + // AudioDeviceModule. + AdmSetupExpectations(&adm_); + // AudioProcessing. + EXPECT_CALL(*apm_, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config_)); + EXPECT_CALL(*apm_, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config_)); + EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); + EXPECT_CALL(*apm_, DetachAecDump()); + // Default Options. + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_drift_compensation(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_analog_level_limits(0, 255)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_vd_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(false)); + // Init does not overwrite default AGC config. + EXPECT_CALL(apm_gc_, target_level_dbfs()).WillOnce(Return(1)); + EXPECT_CALL(apm_gc_, compression_gain_db()).WillRepeatedly(Return(5)); + EXPECT_CALL(apm_gc_, is_limiter_enabled()).WillRepeatedly(Return(true)); + EXPECT_CALL(apm_gc_, set_target_level_dbfs(1)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_compression_gain_db(5)).WillRepeatedly(Return(0)); + EXPECT_CALL(apm_gc_, enable_limiter(true)).WillRepeatedly(Return(0)); + // TODO(kwiberg): We should use mock factories here, but a bunch of + // the tests here probe the specific set of codecs provided by the builtin + // factories. Those tests should probably be moved elsewhere. + auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); + auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); + engine_.reset(new cricket::WebRtcVoiceEngine(&adm_, encoder_factory, + decoder_factory, nullptr, apm_, + new FakeVoEWrapper(&voe_))); + engine_->Init(); + send_parameters_.codecs.push_back(kPcmuCodec); + recv_parameters_.codecs.push_back(kPcmuCodec); + + // Default Options. + EXPECT_TRUE(IsHighPassFilterEnabled()); + } + + bool SetupChannel() { + EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); + channel_ = engine_->CreateChannel(&call_, cricket::MediaConfig(), + cricket::AudioOptions()); + return (channel_ != nullptr); + } + + bool SetupRecvStream() { + if (!SetupChannel()) { + return false; + } + return AddRecvStream(kSsrcX); + } + + bool SetupSendStream() { + if (!SetupChannel()) { + return false; + } + if (!channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))) { + return false; + } + EXPECT_CALL(*apm_, set_output_will_be_muted(false)); + return channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_); + } + + bool AddRecvStream(uint32_t ssrc) { + EXPECT_TRUE(channel_); + return channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc)); + } + + void SetupForMultiSendStream() { + EXPECT_TRUE(SetupSendStream()); + // Remove stream added in Setup. + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); + EXPECT_TRUE(channel_->RemoveSendStream(kSsrcX)); + // Verify the channel does not exist. + EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX)); + } + + void DeliverPacket(const void* data, int len) { + rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len); + channel_->OnPacketReceived(&packet, rtc::PacketTime()); + } + + void TearDown() override { + delete channel_; + } + + const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) { + const auto* send_stream = call_.GetAudioSendStream(ssrc); + EXPECT_TRUE(send_stream); + return *send_stream; + } + + const cricket::FakeAudioReceiveStream& GetRecvStream(uint32_t ssrc) { + const auto* recv_stream = call_.GetAudioReceiveStream(ssrc); + EXPECT_TRUE(recv_stream); + return *recv_stream; + } + + const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) { + return GetSendStream(ssrc).GetConfig(); + } + + const webrtc::AudioReceiveStream::Config& GetRecvStreamConfig(uint32_t ssrc) { + return GetRecvStream(ssrc).GetConfig(); + } + + void SetSend(bool enable) { + ASSERT_TRUE(channel_); + if (enable) { + EXPECT_CALL(adm_, RecordingIsInitialized()).WillOnce(Return(false)); + EXPECT_CALL(adm_, Recording()).WillOnce(Return(false)); + EXPECT_CALL(adm_, InitRecording()).WillOnce(Return(0)); + EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); + } + channel_->SetSend(enable); + } + + void SetSendParameters(const cricket::AudioSendParameters& params) { + EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); + ASSERT_TRUE(channel_); + EXPECT_TRUE(channel_->SetSendParameters(params)); + } + + void SetAudioSend(uint32_t ssrc, bool enable, cricket::AudioSource* source, + const cricket::AudioOptions* options = nullptr) { + EXPECT_CALL(*apm_, set_output_will_be_muted(!enable)); + ASSERT_TRUE(channel_); + if (enable && options) { + EXPECT_CALL(*apm_, SetExtraOptions(testing::_)); + } + EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source)); + } + + void TestInsertDtmf(uint32_t ssrc, bool caller, + const cricket::AudioCodec& codec) { + EXPECT_TRUE(SetupChannel()); + if (caller) { + // If this is a caller, local description will be applied and add the + // send stream. + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + } + + // Test we can only InsertDtmf when the other side supports telephone-event. + SetSendParameters(send_parameters_); + SetSend(true); + EXPECT_FALSE(channel_->CanInsertDtmf()); + EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111)); + send_parameters_.codecs.push_back(codec); + SetSendParameters(send_parameters_); + EXPECT_TRUE(channel_->CanInsertDtmf()); + + if (!caller) { + // If this is callee, there's no active send channel yet. + EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123)); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + } + + // Check we fail if the ssrc is invalid. + EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111)); + + // Test send. + cricket::FakeAudioSendStream::TelephoneEvent telephone_event = + GetSendStream(kSsrcX).GetLatestTelephoneEvent(); + EXPECT_EQ(-1, telephone_event.payload_type); + EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123)); + telephone_event = GetSendStream(kSsrcX).GetLatestTelephoneEvent(); + EXPECT_EQ(codec.id, telephone_event.payload_type); + EXPECT_EQ(codec.clockrate, telephone_event.payload_frequency); + EXPECT_EQ(2, telephone_event.event_code); + EXPECT_EQ(123, telephone_event.duration_ms); + } + + // Test that send bandwidth is set correctly. + // |codec| is the codec under test. + // |max_bitrate| is a parameter to set to SetMaxSendBandwidth(). + // |expected_result| is the expected result from SetMaxSendBandwidth(). + // |expected_bitrate| is the expected audio bitrate afterward. + void TestMaxSendBandwidth(const cricket::AudioCodec& codec, + int max_bitrate, + bool expected_result, + int expected_bitrate) { + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(codec); + parameters.max_bandwidth_bps = max_bitrate; + if (expected_result) { + SetSendParameters(parameters); + } else { + EXPECT_FALSE(channel_->SetSendParameters(parameters)); + } + EXPECT_EQ(expected_bitrate, GetCodecBitrate(kSsrcX)); + } + + // Sets the per-stream maximum bitrate limit for the specified SSRC. + bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) { + webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrc); + EXPECT_EQ(1UL, parameters.encodings.size()); + + parameters.encodings[0].max_bitrate_bps = bitrate; + return channel_->SetRtpSendParameters(ssrc, parameters); + } + + void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) { + cricket::AudioSendParameters send_parameters; + send_parameters.codecs.push_back(codec); + send_parameters.max_bandwidth_bps = bitrate; + SetSendParameters(send_parameters); + } + + void CheckSendCodecBitrate(int32_t ssrc, + const char expected_name[], + int expected_bitrate) { + const auto& spec = GetSendStreamConfig(ssrc).send_codec_spec; + EXPECT_EQ(expected_name, spec->format.name); + EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps); + } + + rtc::Optional<int> GetCodecBitrate(int32_t ssrc) { + return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps; + } + + const rtc::Optional<std::string>& GetAudioNetworkAdaptorConfig(int32_t ssrc) { + return GetSendStreamConfig(ssrc).audio_network_adaptor_config; + } + + void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec, + int global_max, + int stream_max, + bool expected_result, + int expected_codec_bitrate) { + // Clear the bitrate limit from the previous test case. + EXPECT_TRUE(SetMaxBitrateForStream(kSsrcX, -1)); + + // Attempt to set the requested bitrate limits. + SetGlobalMaxBitrate(codec, global_max); + EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrcX, stream_max)); + + // Verify that reading back the parameters gives results + // consistent with the Set() result. + webrtc::RtpParameters resulting_parameters = + channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(1UL, resulting_parameters.encodings.size()); + EXPECT_EQ(expected_result ? stream_max : -1, + resulting_parameters.encodings[0].max_bitrate_bps); + + // Verify that the codec settings have the expected bitrate. + EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrcX)); + } + + void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps, + int expected_min_bitrate_bps, + const char* start_bitrate_kbps, + int expected_start_bitrate_bps, + const char* max_bitrate_kbps, + int expected_max_bitrate_bps) { + EXPECT_TRUE(SetupSendStream()); + auto& codecs = send_parameters_.codecs; + codecs.clear(); + codecs.push_back(kOpusCodec); + codecs[0].params[cricket::kCodecParamMinBitrate] = min_bitrate_kbps; + codecs[0].params[cricket::kCodecParamStartBitrate] = start_bitrate_kbps; + codecs[0].params[cricket::kCodecParamMaxBitrate] = max_bitrate_kbps; + SetSendParameters(send_parameters_); + + EXPECT_EQ(expected_min_bitrate_bps, + call_.GetConfig().bitrate_config.min_bitrate_bps); + EXPECT_EQ(expected_start_bitrate_bps, + call_.GetConfig().bitrate_config.start_bitrate_bps); + EXPECT_EQ(expected_max_bitrate_bps, + call_.GetConfig().bitrate_config.max_bitrate_bps); + } + + void TestSetSendRtpHeaderExtensions(const std::string& ext) { + EXPECT_TRUE(SetupSendStream()); + + // Ensure extensions are off by default. + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure unknown extensions won't cause an error. + send_parameters_.extensions.push_back( + webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); + SetSendParameters(send_parameters_); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure extensions stay off with an empty list of headers. + send_parameters_.extensions.clear(); + SetSendParameters(send_parameters_); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure extension is set properly. + const int id = 1; + send_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); + SetSendParameters(send_parameters_); + EXPECT_EQ(1u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + EXPECT_EQ(ext, GetSendStreamConfig(kSsrcX).rtp.extensions[0].uri); + EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id); + + // Ensure extension is set properly on new stream. + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcY))); + EXPECT_NE(call_.GetAudioSendStream(kSsrcX), + call_.GetAudioSendStream(kSsrcY)); + EXPECT_EQ(1u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); + EXPECT_EQ(ext, GetSendStreamConfig(kSsrcY).rtp.extensions[0].uri); + EXPECT_EQ(id, GetSendStreamConfig(kSsrcY).rtp.extensions[0].id); + + // Ensure all extensions go back off with an empty list. + send_parameters_.codecs.push_back(kPcmuCodec); + send_parameters_.extensions.clear(); + SetSendParameters(send_parameters_); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size()); + EXPECT_EQ(0u, GetSendStreamConfig(kSsrcY).rtp.extensions.size()); + } + + void TestSetRecvRtpHeaderExtensions(const std::string& ext) { + EXPECT_TRUE(SetupRecvStream()); + + // Ensure extensions are off by default. + EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure unknown extensions won't cause an error. + recv_parameters_.extensions.push_back( + webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); + EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure extensions stay off with an empty list of headers. + recv_parameters_.extensions.clear(); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); + EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); + + // Ensure extension is set properly. + const int id = 2; + recv_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); + EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); + EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].uri); + EXPECT_EQ(id, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].id); + + // Ensure extension is set properly on new stream. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_NE(call_.GetAudioReceiveStream(kSsrcX), + call_.GetAudioReceiveStream(kSsrcY)); + EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size()); + EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].uri); + EXPECT_EQ(id, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].id); + + // Ensure all extensions go back off with an empty list. + recv_parameters_.extensions.clear(); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); + EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size()); + EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size()); + } + + webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const { + webrtc::AudioSendStream::Stats stats; + stats.local_ssrc = 12; + stats.bytes_sent = 345; + stats.packets_sent = 678; + stats.packets_lost = 9012; + stats.fraction_lost = 34.56f; + stats.codec_name = "codec_name_send"; + stats.codec_payload_type = 42; + stats.ext_seqnum = 789; + stats.jitter_ms = 12; + stats.rtt_ms = 345; + stats.audio_level = 678; + stats.apm_statistics.delay_median_ms = 234; + stats.apm_statistics.delay_standard_deviation_ms = 567; + stats.apm_statistics.echo_return_loss = 890; + stats.apm_statistics.echo_return_loss_enhancement = 1234; + stats.apm_statistics.residual_echo_likelihood = 0.432f; + stats.apm_statistics.residual_echo_likelihood_recent_max = 0.6f; + stats.ana_statistics.bitrate_action_counter = 321; + stats.ana_statistics.channel_action_counter = 432; + stats.ana_statistics.dtx_action_counter = 543; + stats.ana_statistics.fec_action_counter = 654; + stats.ana_statistics.frame_length_increase_counter = 765; + stats.ana_statistics.frame_length_decrease_counter = 876; + stats.ana_statistics.uplink_packet_loss_fraction = 987.0; + stats.typing_noise_detected = true; + return stats; + } + void SetAudioSendStreamStats() { + for (auto* s : call_.GetAudioSendStreams()) { + s->SetStats(GetAudioSendStreamStats()); + } + } + void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info, + bool is_sending) { + const auto stats = GetAudioSendStreamStats(); + EXPECT_EQ(info.ssrc(), stats.local_ssrc); + EXPECT_EQ(info.bytes_sent, stats.bytes_sent); + EXPECT_EQ(info.packets_sent, stats.packets_sent); + EXPECT_EQ(info.packets_lost, stats.packets_lost); + EXPECT_EQ(info.fraction_lost, stats.fraction_lost); + EXPECT_EQ(info.codec_name, stats.codec_name); + EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); + EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum); + EXPECT_EQ(info.jitter_ms, stats.jitter_ms); + EXPECT_EQ(info.rtt_ms, stats.rtt_ms); + EXPECT_EQ(info.audio_level, stats.audio_level); + EXPECT_EQ(info.apm_statistics.delay_median_ms, + stats.apm_statistics.delay_median_ms); + EXPECT_EQ(info.apm_statistics.delay_standard_deviation_ms, + stats.apm_statistics.delay_standard_deviation_ms); + EXPECT_EQ(info.apm_statistics.echo_return_loss, + stats.apm_statistics.echo_return_loss); + EXPECT_EQ(info.apm_statistics.echo_return_loss_enhancement, + stats.apm_statistics.echo_return_loss_enhancement); + EXPECT_EQ(info.apm_statistics.residual_echo_likelihood, + stats.apm_statistics.residual_echo_likelihood); + EXPECT_EQ(info.apm_statistics.residual_echo_likelihood_recent_max, + stats.apm_statistics.residual_echo_likelihood_recent_max); + EXPECT_EQ(info.ana_statistics.bitrate_action_counter, + stats.ana_statistics.bitrate_action_counter); + EXPECT_EQ(info.ana_statistics.channel_action_counter, + stats.ana_statistics.channel_action_counter); + EXPECT_EQ(info.ana_statistics.dtx_action_counter, + stats.ana_statistics.dtx_action_counter); + EXPECT_EQ(info.ana_statistics.fec_action_counter, + stats.ana_statistics.fec_action_counter); + EXPECT_EQ(info.ana_statistics.frame_length_increase_counter, + stats.ana_statistics.frame_length_increase_counter); + EXPECT_EQ(info.ana_statistics.frame_length_decrease_counter, + stats.ana_statistics.frame_length_decrease_counter); + EXPECT_EQ(info.ana_statistics.uplink_packet_loss_fraction, + stats.ana_statistics.uplink_packet_loss_fraction); + EXPECT_EQ(info.typing_noise_detected, + stats.typing_noise_detected && is_sending); + } + + webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const { + webrtc::AudioReceiveStream::Stats stats; + stats.remote_ssrc = 123; + stats.bytes_rcvd = 456; + stats.packets_rcvd = 768; + stats.packets_lost = 101; + stats.fraction_lost = 23.45f; + stats.codec_name = "codec_name_recv"; + stats.codec_payload_type = 42; + stats.ext_seqnum = 678; + stats.jitter_ms = 901; + stats.jitter_buffer_ms = 234; + stats.jitter_buffer_preferred_ms = 567; + stats.delay_estimate_ms = 890; + stats.audio_level = 1234; + stats.total_samples_received = 5678901; + stats.concealed_samples = 234; + stats.concealment_events = 12; + stats.jitter_buffer_delay_seconds = 34; + stats.expand_rate = 5.67f; + stats.speech_expand_rate = 8.90f; + stats.secondary_decoded_rate = 1.23f; + stats.secondary_discarded_rate = 0.12f; + stats.accelerate_rate = 4.56f; + stats.preemptive_expand_rate = 7.89f; + stats.decoding_calls_to_silence_generator = 12; + stats.decoding_calls_to_neteq = 345; + stats.decoding_normal = 67890; + stats.decoding_plc = 1234; + stats.decoding_cng = 5678; + stats.decoding_plc_cng = 9012; + stats.decoding_muted_output = 3456; + stats.capture_start_ntp_time_ms = 7890; + return stats; + } + void SetAudioReceiveStreamStats() { + for (auto* s : call_.GetAudioReceiveStreams()) { + s->SetStats(GetAudioReceiveStreamStats()); + } + } + void VerifyVoiceReceiverInfo(const cricket::VoiceReceiverInfo& info) { + const auto stats = GetAudioReceiveStreamStats(); + EXPECT_EQ(info.ssrc(), stats.remote_ssrc); + EXPECT_EQ(info.bytes_rcvd, stats.bytes_rcvd); + EXPECT_EQ(info.packets_rcvd, stats.packets_rcvd); + EXPECT_EQ(info.packets_lost, stats.packets_lost); + EXPECT_EQ(info.fraction_lost, stats.fraction_lost); + EXPECT_EQ(info.codec_name, stats.codec_name); + EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); + EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum); + EXPECT_EQ(info.jitter_ms, stats.jitter_ms); + EXPECT_EQ(info.jitter_buffer_ms, stats.jitter_buffer_ms); + EXPECT_EQ(info.jitter_buffer_preferred_ms, + stats.jitter_buffer_preferred_ms); + EXPECT_EQ(info.delay_estimate_ms, stats.delay_estimate_ms); + EXPECT_EQ(info.audio_level, stats.audio_level); + EXPECT_EQ(info.total_samples_received, stats.total_samples_received); + EXPECT_EQ(info.concealed_samples, stats.concealed_samples); + EXPECT_EQ(info.concealment_events, stats.concealment_events); + EXPECT_EQ(info.jitter_buffer_delay_seconds, + stats.jitter_buffer_delay_seconds); + EXPECT_EQ(info.expand_rate, stats.expand_rate); + EXPECT_EQ(info.speech_expand_rate, stats.speech_expand_rate); + EXPECT_EQ(info.secondary_decoded_rate, stats.secondary_decoded_rate); + EXPECT_EQ(info.secondary_discarded_rate, stats.secondary_discarded_rate); + EXPECT_EQ(info.accelerate_rate, stats.accelerate_rate); + EXPECT_EQ(info.preemptive_expand_rate, stats.preemptive_expand_rate); + EXPECT_EQ(info.decoding_calls_to_silence_generator, + stats.decoding_calls_to_silence_generator); + EXPECT_EQ(info.decoding_calls_to_neteq, stats.decoding_calls_to_neteq); + EXPECT_EQ(info.decoding_normal, stats.decoding_normal); + EXPECT_EQ(info.decoding_plc, stats.decoding_plc); + EXPECT_EQ(info.decoding_cng, stats.decoding_cng); + EXPECT_EQ(info.decoding_plc_cng, stats.decoding_plc_cng); + EXPECT_EQ(info.decoding_muted_output, stats.decoding_muted_output); + EXPECT_EQ(info.capture_start_ntp_time_ms, stats.capture_start_ntp_time_ms); + } + void VerifyVoiceSendRecvCodecs(const cricket::VoiceMediaInfo& info) const { + EXPECT_EQ(send_parameters_.codecs.size(), info.send_codecs.size()); + for (const cricket::AudioCodec& codec : send_parameters_.codecs) { + ASSERT_EQ(info.send_codecs.count(codec.id), 1U); + EXPECT_EQ(info.send_codecs.find(codec.id)->second, + codec.ToCodecParameters()); + } + EXPECT_EQ(recv_parameters_.codecs.size(), info.receive_codecs.size()); + for (const cricket::AudioCodec& codec : recv_parameters_.codecs) { + ASSERT_EQ(info.receive_codecs.count(codec.id), 1U); + EXPECT_EQ(info.receive_codecs.find(codec.id)->second, + codec.ToCodecParameters()); + } + } + + bool IsHighPassFilterEnabled() { + return engine_->GetApmConfigForTest().high_pass_filter.enabled; + } + + protected: + StrictMock<webrtc::test::MockAudioDeviceModule> adm_; + rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_; + webrtc::test::MockGainControl& apm_gc_; + webrtc::test::MockEchoCancellation& apm_ec_; + webrtc::test::MockNoiseSuppression& apm_ns_; + webrtc::test::MockVoiceDetection& apm_vd_; + StrictMock<MockTransmitMixer> transmit_mixer_; + webrtc::RtcEventLogNullImpl event_log_; + cricket::FakeCall call_; + cricket::FakeWebRtcVoiceEngine voe_; + std::unique_ptr<cricket::WebRtcVoiceEngine> engine_; + cricket::VoiceMediaChannel* channel_ = nullptr; + cricket::AudioSendParameters send_parameters_; + cricket::AudioRecvParameters recv_parameters_; + FakeAudioSource fake_source_; + webrtc::AudioProcessing::Config apm_config_; + + private: + webrtc::test::ScopedFieldTrials override_field_trials_; +}; + +// Tests that we can create and destroy a channel. +TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) { + EXPECT_TRUE(SetupChannel()); +} + +// Test that we can add a send stream and that it has the correct defaults. +TEST_F(WebRtcVoiceEngineTestFake, CreateSendStream) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE( + channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); + const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX); + EXPECT_EQ(kSsrcX, config.rtp.ssrc); + EXPECT_EQ("", config.rtp.c_name); + EXPECT_EQ(0u, config.rtp.extensions.size()); + EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_), + config.send_transport); +} + +// Test that we can add a receive stream and that it has the correct defaults. +TEST_F(WebRtcVoiceEngineTestFake, CreateRecvStream) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + const webrtc::AudioReceiveStream::Config& config = + GetRecvStreamConfig(kSsrcX); + EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc); + EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc); + EXPECT_FALSE(config.rtp.transport_cc); + EXPECT_EQ(0u, config.rtp.extensions.size()); + EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_), + config.rtcp_send_transport); + EXPECT_EQ("", config.sync_group); +} + +TEST_F(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) { + const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs(); + bool opus_found = false; + for (cricket::AudioCodec codec : codecs) { + if (codec.name == "opus") { + EXPECT_TRUE(HasTransportCc(codec)); + opus_found = true; + } + } + EXPECT_TRUE(opus_found); +} + +// Test that we set our inbound codecs properly, including changing PT. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs[0].id = 106; // collide with existing CN 32k + parameters.codecs[2].id = 126; + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {106, {"ISAC", 16000, 1}}, + {126, {"telephone-event", 8000, 1}}, + {107, {"telephone-event", 32000, 1}}}))); +} + +// Test that we fail to set an unknown inbound codec. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1)); + EXPECT_FALSE(channel_->SetRecvParameters(parameters)); +} + +// Test that we fail if we have duplicate types in the inbound list. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs[1].id = kIsacCodec.id; + EXPECT_FALSE(channel_->SetRecvParameters(parameters)); +} + +// Test that we can decode OPUS without stereo parameters. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kOpusCodec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {103, {"ISAC", 16000, 1}}, + {111, {"opus", 48000, 2}}}))); +} + +// Test that we can decode OPUS with stereo = 0. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[2].params["stereo"] = "0"; + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {103, {"ISAC", 16000, 1}}, + {111, {"opus", 48000, 2, {{"stereo", "0"}}}}}))); +} + +// Test that we can decode OPUS with stereo = 1. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[2].params["stereo"] = "1"; + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {103, {"ISAC", 16000, 1}}, + {111, {"opus", 48000, 2, {{"stereo", "1"}}}}}))); +} + +// Test that changes to recv codecs are applied to all streams. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs[0].id = 106; // collide with existing CN 32k + parameters.codecs[2].id = 126; + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + for (const auto& ssrc : {kSsrcX, kSsrcY}) { + EXPECT_TRUE(AddRecvStream(ssrc)); + EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, + {106, {"ISAC", 16000, 1}}, + {126, {"telephone-event", 8000, 1}}, + {107, {"telephone-event", 32000, 1}}}))); + } +} + +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs[0].id = 106; // collide with existing CN 32k + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + + const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map; + ASSERT_EQ(1, dm.count(106)); + EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106)); +} + +// Test that we can apply the same set of codecs again while playing. +TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kCn16000Codec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + channel_->SetPlayout(true); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + + // Remapping a payload type to a different codec should fail. + parameters.codecs[0] = kOpusCodec; + parameters.codecs[0].id = kIsacCodec.id; + EXPECT_FALSE(channel_->SetRecvParameters(parameters)); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); +} + +// Test that we can add a codec while playing. +TEST_F(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kCn16000Codec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + channel_->SetPlayout(true); + + parameters.codecs.push_back(kOpusCodec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); +} + +// Test that we accept adding the same codec with a different payload type. +// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847 +TEST_F(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + + ++parameters.codecs[0].id; + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) { + EXPECT_TRUE(SetupSendStream()); + + // Test that when autobw is enabled, bitrate is kept as the default + // value. autobw is enabled for the following tests because the target + // bitrate is <= 0. + + // ISAC, default bitrate == 32000. + TestMaxSendBandwidth(kIsacCodec, 0, true, 32000); + + // PCMU, default bitrate == 64000. + TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000); + + // opus, default bitrate == 32000 in mono. + TestMaxSendBandwidth(kOpusCodec, -1, true, 32000); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) { + EXPECT_TRUE(SetupSendStream()); + + // ISAC, default bitrate == 32000. + TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000); + // Rates above the max (56000) should be capped. + TestMaxSendBandwidth(kIsacCodec, 100000, true, 32000); + + // opus, default bitrate == 64000. + TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000); + TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000); + // Rates above the max (510000) should be capped. + TestMaxSendBandwidth(kOpusCodec, 600000, true, 510000); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) { + EXPECT_TRUE(SetupSendStream()); + + // Test that we can only set a maximum bitrate for a fixed-rate codec + // if it's bigger than the fixed rate. + + // PCMU, fixed bitrate == 64000. + TestMaxSendBandwidth(kPcmuCodec, 0, true, 64000); + TestMaxSendBandwidth(kPcmuCodec, 1, false, 64000); + TestMaxSendBandwidth(kPcmuCodec, 128000, true, 64000); + TestMaxSendBandwidth(kPcmuCodec, 32000, false, 64000); + TestMaxSendBandwidth(kPcmuCodec, 64000, true, 64000); + TestMaxSendBandwidth(kPcmuCodec, 63999, false, 64000); + TestMaxSendBandwidth(kPcmuCodec, 64001, true, 64000); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) { + EXPECT_TRUE(SetupChannel()); + const int kDesiredBitrate = 128000; + cricket::AudioSendParameters parameters; + parameters.codecs = engine_->send_codecs(); + parameters.max_bandwidth_bps = kDesiredBitrate; + SetSendParameters(parameters); + + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + + EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX)); +} + +// Test that bitrate cannot be set for CBR codecs. +// Bitrate is ignored if it is higher than the fixed bitrate. +// Bitrate less then the fixed bitrate is an error. +TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) { + EXPECT_TRUE(SetupSendStream()); + + // PCMU, default bitrate == 64000. + SetSendParameters(send_parameters_); + EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); + + send_parameters_.max_bandwidth_bps = 128000; + SetSendParameters(send_parameters_); + EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); + + send_parameters_.max_bandwidth_bps = 128; + EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)); + EXPECT_EQ(64000, GetCodecBitrate(kSsrcX)); +} + +// Test that the per-stream bitrate limit and the global +// bitrate limit both apply. +TEST_F(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) { + EXPECT_TRUE(SetupSendStream()); + + // opus, default bitrate == 32000. + SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 32000); + SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000); + SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000); + SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000); + + // CBR codecs allow both maximums to exceed the bitrate. + SetAndExpectMaxBitrate(kPcmuCodec, 0, 0, true, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 64001, 0, true, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 0, 64001, true, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 64001, 64001, true, 64000); + + // CBR codecs don't allow per stream maximums to be too low. + SetAndExpectMaxBitrate(kPcmuCodec, 0, 63999, false, 64000); + SetAndExpectMaxBitrate(kPcmuCodec, 64001, 63999, false, 64000); +} + +// Test that an attempt to set RtpParameters for a stream that does not exist +// fails. +TEST_F(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) { + EXPECT_TRUE(SetupChannel()); + webrtc::RtpParameters nonexistent_parameters = + channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(0, nonexistent_parameters.encodings.size()); + + nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters()); + EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters)); +} + +TEST_F(WebRtcVoiceEngineTestFake, + CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) { + // This test verifies that setting RtpParameters succeeds only if + // the structure contains exactly one encoding. + // TODO(skvlad): Update this test when we start supporting setting parameters + // for each encoding individually. + + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); + // Two or more encodings should result in failure. + parameters.encodings.push_back(webrtc::RtpEncodingParameters()); + EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters)); + // Zero encodings should also fail. + parameters.encodings.clear(); + EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters)); +} + +// Changing the SSRC through RtpParameters is not allowed. +TEST_F(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) { + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); + parameters.encodings[0].ssrc = 0xdeadbeef; + EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters)); +} + +// Test that a stream will not be sending if its encoding is made +// inactive through SetRtpSendParameters. +TEST_F(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) { + EXPECT_TRUE(SetupSendStream()); + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + // Get current parameters and change "active" to false. + webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX); + ASSERT_EQ(1u, parameters.encodings.size()); + ASSERT_TRUE(parameters.encodings[0].active); + parameters.encodings[0].active = false; + EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters)); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Now change it back to active and verify we resume sending. + parameters.encodings[0].active = true; + EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters)); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); +} + +// Test that SetRtpSendParameters configures the correct encoding channel for +// each SSRC. +TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) { + SetupForMultiSendStream(); + // Create send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE( + channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc))); + } + // Configure one stream to be limited by the stream config, another to be + // limited by the global max, and the third one with no per-stream limit + // (still subject to the global limit). + SetGlobalMaxBitrate(kOpusCodec, 32000); + EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 24000)); + EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 48000)); + EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1)); + + EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0])); + EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[1])); + EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2])); + + // Remove the global cap; the streams should switch to their respective + // maximums (or remain unchanged if there was no other limit on them.) + SetGlobalMaxBitrate(kOpusCodec, -1); + EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0])); + EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[1])); + EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2])); +} + +// Test that GetRtpSendParameters returns the currently configured codecs. +TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + SetSendParameters(parameters); + + webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); + ASSERT_EQ(2u, rtp_parameters.codecs.size()); + EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); + EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); +} + +// Test that GetRtpSendParameters returns an SSRC. +TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) { + EXPECT_TRUE(SetupSendStream()); + webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc); +} + +// Test that if we set/get parameters multiple times, we get the same results. +TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + SetSendParameters(parameters); + + webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrcX); + + // We should be able to set the params we just got. + EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, initial_params)); + + // ... And this shouldn't change the params returned by GetRtpSendParameters. + webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrcX); + EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrcX)); +} + +// Test that max_bitrate_bps in send stream config gets updated correctly when +// SetRtpSendParameters is called. +TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) { + webrtc::test::ScopedFieldTrials override_field_trials( + "WebRTC-Audio-SendSideBwe/Enabled/"); + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters send_parameters; + send_parameters.codecs.push_back(kOpusCodec); + SetSendParameters(send_parameters); + + webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); + // Expect empty on parameters.encodings[0].max_bitrate_bps; + EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps); + + constexpr int kMaxBitrateBps = 6000; + rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps; + EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters)); + + const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps; + EXPECT_EQ(max_bitrate, kMaxBitrateBps); +} + +// Test that GetRtpReceiveParameters returns the currently configured codecs. +TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + + webrtc::RtpParameters rtp_parameters = + channel_->GetRtpReceiveParameters(kSsrcX); + ASSERT_EQ(2u, rtp_parameters.codecs.size()); + EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); + EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); +} + +// Test that GetRtpReceiveParameters returns an SSRC. +TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) { + EXPECT_TRUE(SetupRecvStream()); + webrtc::RtpParameters rtp_parameters = + channel_->GetRtpReceiveParameters(kSsrcX); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc); +} + +// Test that if we set/get parameters multiple times, we get the same results. +TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + + webrtc::RtpParameters initial_params = + channel_->GetRtpReceiveParameters(kSsrcX); + + // We should be able to set the params we just got. + EXPECT_TRUE(channel_->SetRtpReceiveParameters(kSsrcX, initial_params)); + + // ... And this shouldn't change the params returned by + // GetRtpReceiveParameters. + webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrcX); + EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(kSsrcX)); +} + +// Test that GetRtpReceiveParameters returns parameters correctly when SSRCs +// aren't signaled. It should return an empty "RtpEncodingParameters" when +// configured to receive an unsignaled stream and no packets have been received +// yet, and start returning the SSRC once a packet has been received. +TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) { + ASSERT_TRUE(SetupChannel()); + // Call necessary methods to configure receiving a default stream as + // soon as it arrives. + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + + // Call GetRtpReceiveParameters before configured to receive an unsignaled + // stream. Should return nothing. + EXPECT_EQ(webrtc::RtpParameters(), channel_->GetRtpReceiveParameters(0)); + + // Set a sink for an unsignaled stream. + std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink()); + // Value of "0" means "unsignaled stream". + channel_->SetRawAudioSink(0, std::move(fake_sink)); + + // Call GetRtpReceiveParameters before the SSRC is known. Value of "0" + // in this method means "unsignaled stream". + webrtc::RtpParameters rtp_parameters = channel_->GetRtpReceiveParameters(0); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); + + // Receive PCMU packet (SSRC=1). + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + // The |ssrc| member should still be unset. + rtp_parameters = channel_->GetRtpReceiveParameters(0); + ASSERT_EQ(1u, rtp_parameters.encodings.size()); + EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); +} + +// Test that we apply codecs properly. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs[0].id = 96; + parameters.codecs[0].bitrate = 22000; + SetSendParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps); + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000); + EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type); + EXPECT_FALSE(channel_->CanInsertDtmf()); +} + +// Test that WebRtcVoiceEngine reconfigures, rather than recreates its +// AudioSendStream. +TEST_F(WebRtcVoiceEngineTestFake, DontRecreateSendStream) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs[0].id = 96; + parameters.codecs[0].bitrate = 48000; + const int initial_num = call_.GetNumCreatedSendStreams(); + SetSendParameters(parameters); + EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); + // Calling SetSendCodec again with same codec which is already set. + // In this case media channel shouldn't send codec to VoE. + SetSendParameters(parameters); + EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); +} + +// TODO(ossu): Revisit if these tests need to be here, now that these kinds of +// tests should be available in AudioEncoderOpusTest. + +// Test that if clockrate is not 48000 for opus, we fail. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].clockrate = 50000; + EXPECT_FALSE(channel_->SetSendParameters(parameters)); +} + +// Test that if channels=0 for opus, we fail. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 0; + EXPECT_FALSE(channel_->SetSendParameters(parameters)); +} + +// Test that if channels=0 for opus, we fail. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 0; + parameters.codecs[0].params["stereo"] = "1"; + EXPECT_FALSE(channel_->SetSendParameters(parameters)); +} + +// Test that if channel is 1 for opus and there's no stereo, we fail. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 1; + EXPECT_FALSE(channel_->SetSendParameters(parameters)); +} + +// Test that if channel is 1 for opus and stereo=0, we fail. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 1; + parameters.codecs[0].params["stereo"] = "0"; + EXPECT_FALSE(channel_->SetSendParameters(parameters)); +} + +// Test that if channel is 1 for opus and stereo=1, we fail. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].channels = 1; + parameters.codecs[0].params["stereo"] = "1"; + EXPECT_FALSE(channel_->SetSendParameters(parameters)); +} + +// Test that with bitrate=0 and no stereo, bitrate is 32000. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 32000); +} + +// Test that with bitrate=0 and stereo=0, bitrate is 32000. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].params["stereo"] = "0"; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 32000); +} + +// Test that with bitrate=invalid and stereo=0, bitrate is 32000. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].params["stereo"] = "0"; + // bitrate that's out of the range between 6000 and 510000 will be clamped. + parameters.codecs[0].bitrate = 5999; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 6000); + + parameters.codecs[0].bitrate = 510001; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 510000); +} + +// Test that with bitrate=0 and stereo=1, bitrate is 64000. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 0; + parameters.codecs[0].params["stereo"] = "1"; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 64000); +} + +// Test that with bitrate=invalid and stereo=1, bitrate is 64000. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].params["stereo"] = "1"; + // bitrate that's out of the range between 6000 and 510000 will be clamped. + parameters.codecs[0].bitrate = 5999; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 6000); + + parameters.codecs[0].bitrate = 510001; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 510000); +} + +// Test that with bitrate=N and stereo unset, bitrate is N. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 96000; + SetSendParameters(parameters); + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_EQ(96000, spec.target_bitrate_bps); + EXPECT_EQ("opus", spec.format.name); + EXPECT_EQ(2, spec.format.num_channels); + EXPECT_EQ(48000, spec.format.clockrate_hz); +} + +// Test that with bitrate=N and stereo=0, bitrate is N. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 30000; + parameters.codecs[0].params["stereo"] = "0"; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 30000); +} + +// Test that with bitrate=N and without any parameters, bitrate is N. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 30000; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 30000); +} + +// Test that with bitrate=N and stereo=1, bitrate is N. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].bitrate = 30000; + parameters.codecs[0].params["stereo"] = "1"; + SetSendParameters(parameters); + CheckSendCodecBitrate(kSsrcX, "opus", 30000); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithBitrates) { + SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", + 200000); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithHighMaxBitrate) { + SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000); +} + +TEST_F(WebRtcVoiceEngineTestFake, + SetSendCodecsWithoutBitratesUsesCorrectDefaults) { + SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) { + SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1); +} + +TEST_F(WebRtcVoiceEngineTestFake, + SetMaxSendBandwidthForAudioDoesntAffectBwe) { + SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", + 200000); + send_parameters_.max_bandwidth_bps = 100000; + SetSendParameters(send_parameters_); + EXPECT_EQ(100000, call_.GetConfig().bitrate_config.min_bitrate_bps) + << "Setting max bitrate should keep previous min bitrate."; + EXPECT_EQ(-1, call_.GetConfig().bitrate_config.start_bitrate_bps) + << "Setting max bitrate should not reset start bitrate."; + EXPECT_EQ(200000, call_.GetConfig().bitrate_config.max_bitrate_bps); +} + +// Test that we can enable NACK with opus as caller. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCaller) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam( + cricket::FeedbackParam(cricket::kRtcpFbParamNack, + cricket::kParamValueEmpty)); + EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + SetSendParameters(parameters); + EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); +} + +// Test that we can enable NACK with opus as callee. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) { + EXPECT_TRUE(SetupRecvStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam( + cricket::FeedbackParam(cricket::kRtcpFbParamNack, + cricket::kParamValueEmpty)); + EXPECT_EQ(0, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + SetSendParameters(parameters); + // NACK should be enabled even with no send stream. + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); +} + +// Test that we can enable NACK on receive streams. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam( + cricket::FeedbackParam(cricket::kRtcpFbParamNack, + cricket::kParamValueEmpty)); + EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); + SetSendParameters(parameters); + EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); +} + +// Test that we can disable NACK. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNack) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam( + cricket::FeedbackParam(cricket::kRtcpFbParamNack, + cricket::kParamValueEmpty)); + SetSendParameters(parameters); + EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + + parameters.codecs.clear(); + parameters.codecs.push_back(kOpusCodec); + SetSendParameters(parameters); + EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); +} + +// Test that we can disable NACK on receive streams. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + parameters.codecs[0].AddFeedbackParam( + cricket::FeedbackParam(cricket::kRtcpFbParamNack, + cricket::kParamValueEmpty)); + SetSendParameters(parameters); + EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); + + parameters.codecs.clear(); + parameters.codecs.push_back(kOpusCodec); + SetSendParameters(parameters); + EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); +} + +// Test that NACK is enabled on a new receive stream. +TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs[0].AddFeedbackParam( + cricket::FeedbackParam(cricket::kRtcpFbParamNack, + cricket::kParamValueEmpty)); + SetSendParameters(parameters); + EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms); + + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms); + EXPECT_TRUE(AddRecvStream(kSsrcZ)); + EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcZ).rtp.nack.rtp_history_ms); +} + +TEST_F(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioSendParameters send_parameters; + send_parameters.codecs.push_back(kOpusCodec); + EXPECT_TRUE(send_parameters.codecs[0].feedback_params.params().empty()); + SetSendParameters(send_parameters); + + cricket::AudioRecvParameters recv_parameters; + recv_parameters.codecs.push_back(kIsacCodec); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr); + EXPECT_FALSE( + call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc); + + send_parameters.codecs = engine_->send_codecs(); + SetSendParameters(send_parameters); + ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr); + EXPECT_TRUE( + call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc); +} + +// Test that we can switch back and forth between Opus and ISAC with CN. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) { + EXPECT_TRUE(SetupSendStream()); + + cricket::AudioSendParameters opus_parameters; + opus_parameters.codecs.push_back(kOpusCodec); + SetSendParameters(opus_parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); + } + + cricket::AudioSendParameters isac_parameters; + isac_parameters.codecs.push_back(kIsacCodec); + isac_parameters.codecs.push_back(kCn16000Codec); + isac_parameters.codecs.push_back(kOpusCodec); + SetSendParameters(isac_parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(103, spec.payload_type); + EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); + } + + SetSendParameters(opus_parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); + } +} + +// Test that we handle various ways of specifying bitrate. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); // bitrate == 32000 + SetSendParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(103, spec.payload_type); + EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); + EXPECT_EQ(32000, spec.target_bitrate_bps); + } + + parameters.codecs[0].bitrate = 0; // bitrate == default + SetSendParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(103, spec.payload_type); + EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); + EXPECT_EQ(32000, spec.target_bitrate_bps); + } + parameters.codecs[0].bitrate = 28000; // bitrate == 28000 + SetSendParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(103, spec.payload_type); + EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); + EXPECT_EQ(28000, spec.target_bitrate_bps); + } + + parameters.codecs[0] = kPcmuCodec; // bitrate == 64000 + SetSendParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(0, spec.payload_type); + EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str()); + EXPECT_EQ(64000, spec.target_bitrate_bps); + } + + parameters.codecs[0].bitrate = 0; // bitrate == default + SetSendParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(0, spec.payload_type); + EXPECT_STREQ("PCMU", spec.format.name.c_str()); + EXPECT_EQ(64000, spec.target_bitrate_bps); + } + + parameters.codecs[0] = kOpusCodec; + parameters.codecs[0].bitrate = 0; // bitrate == default + SetSendParameters(parameters); + { + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(111, spec.payload_type); + EXPECT_STREQ("opus", spec.format.name.c_str()); + EXPECT_EQ(32000, spec.target_bitrate_bps); + } +} + +// Test that we fail if no codecs are specified. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + EXPECT_FALSE(channel_->SetSendParameters(parameters)); +} + +// Test that we can set send codecs even with telephone-event codec as the first +// one on the list. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs[0].id = 98; // DTMF + parameters.codecs[1].id = 96; + SetSendParameters(parameters); + const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, spec.payload_type); + EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); + EXPECT_TRUE(channel_->CanInsertDtmf()); +} + +// Test that payload type range is limited for telephone-event codec. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs.push_back(kIsacCodec); + parameters.codecs[0].id = 0; // DTMF + parameters.codecs[1].id = 96; + SetSendParameters(parameters); + EXPECT_TRUE(channel_->CanInsertDtmf()); + parameters.codecs[0].id = 128; // DTMF + EXPECT_FALSE(channel_->SetSendParameters(parameters)); + EXPECT_FALSE(channel_->CanInsertDtmf()); + parameters.codecs[0].id = 127; + SetSendParameters(parameters); + EXPECT_TRUE(channel_->CanInsertDtmf()); + parameters.codecs[0].id = -1; // DTMF + EXPECT_FALSE(channel_->SetSendParameters(parameters)); + EXPECT_FALSE(channel_->CanInsertDtmf()); +} + +// Test that we can set send codecs even with CN codec as the first +// one on the list. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs[0].id = 98; // wideband CN + parameters.codecs[1].id = 96; + SetSendParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(98, send_codec_spec.cng_payload_type); +} + +// Test that we set VAD and DTMF types correctly as caller. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + // TODO(juberti): cn 32000 + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs[0].id = 96; + parameters.codecs[2].id = 97; // wideband CN + parameters.codecs[4].id = 98; // DTMF + SetSendParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + EXPECT_TRUE(channel_->CanInsertDtmf()); +} + +// Test that we set VAD and DTMF types correctly as callee. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) { + EXPECT_TRUE(SetupChannel()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + // TODO(juberti): cn 32000 + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs.push_back(kTelephoneEventCodec2); + parameters.codecs[0].id = 96; + parameters.codecs[2].id = 97; // wideband CN + parameters.codecs[4].id = 98; // DTMF + SetSendParameters(parameters); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + EXPECT_TRUE(channel_->CanInsertDtmf()); +} + +// Test that we only apply VAD if we have a CN codec that matches the +// send codec clockrate. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + // Set ISAC(16K) and CN(16K). VAD should be activated. + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs[1].id = 97; + SetSendParameters(parameters); + { + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + } + // Set PCMU(8K) and CN(16K). VAD should not be activated. + parameters.codecs[0] = kPcmuCodec; + SetSendParameters(parameters); + { + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type); + } + // Set PCMU(8K) and CN(8K). VAD should be activated. + parameters.codecs[1] = kCn8000Codec; + SetSendParameters(parameters); + { + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1, send_codec_spec.format.num_channels); + EXPECT_EQ(13, send_codec_spec.cng_payload_type); + } + // Set ISAC(16K) and CN(8K). VAD should not be activated. + parameters.codecs[0] = kIsacCodec; + SetSendParameters(parameters); + { + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type); + } +} + +// Test that we perform case-insensitive matching of codec names. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs.push_back(kCn8000Codec); + parameters.codecs.push_back(kTelephoneEventCodec1); + parameters.codecs[0].name = "iSaC"; + parameters.codecs[0].id = 96; + parameters.codecs[2].id = 97; // wideband CN + parameters.codecs[4].id = 98; // DTMF + SetSendParameters(parameters); + const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; + EXPECT_EQ(96, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + EXPECT_TRUE(channel_->CanInsertDtmf()); +} + +class WebRtcVoiceEngineWithSendSideBweTest : public WebRtcVoiceEngineTestFake { + public: + WebRtcVoiceEngineWithSendSideBweTest() + : WebRtcVoiceEngineTestFake("WebRTC-Audio-SendSideBwe/Enabled/") {} +}; + +TEST_F(WebRtcVoiceEngineWithSendSideBweTest, + SupportsTransportSequenceNumberHeaderExtension) { + cricket::RtpCapabilities capabilities = engine_->GetCapabilities(); + ASSERT_FALSE(capabilities.header_extensions.empty()); + for (const webrtc::RtpExtension& extension : capabilities.header_extensions) { + if (extension.uri == webrtc::RtpExtension::kTransportSequenceNumberUri) { + EXPECT_EQ(webrtc::RtpExtension::kTransportSequenceNumberDefaultId, + extension.id); + return; + } + } + FAIL() << "Transport sequence number extension not in header-extension list."; +} + +// Test support for audio level header extension. +TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) { + TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); +} +TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) { + TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); +} + +// Test support for transport sequence number header extension. +TEST_F(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) { + TestSetSendRtpHeaderExtensions( + webrtc::RtpExtension::kTransportSequenceNumberUri); +} +TEST_F(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) { + TestSetRecvRtpHeaderExtensions( + webrtc::RtpExtension::kTransportSequenceNumberUri); +} + +// Test that we can create a channel and start sending on it. +TEST_F(WebRtcVoiceEngineTestFake, Send) { + EXPECT_TRUE(SetupSendStream()); + SetSendParameters(send_parameters_); + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + SetSend(false); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); +} + +// Test that a channel will send if and only if it has a source and is enabled +// for sending. +TEST_F(WebRtcVoiceEngineTestFake, SendStateWithAndWithoutSource) { + EXPECT_TRUE(SetupSendStream()); + SetSendParameters(send_parameters_); + SetAudioSend(kSsrcX, true, nullptr); + SetSend(true); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + SetAudioSend(kSsrcX, true, &fake_source_); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + SetAudioSend(kSsrcX, true, nullptr); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); +} + +// Test that a channel is muted/unmuted. +TEST_F(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) { + EXPECT_TRUE(SetupSendStream()); + SetSendParameters(send_parameters_); + EXPECT_FALSE(GetSendStream(kSsrcX).muted()); + SetAudioSend(kSsrcX, true, nullptr); + EXPECT_FALSE(GetSendStream(kSsrcX).muted()); + SetAudioSend(kSsrcX, false, nullptr); + EXPECT_TRUE(GetSendStream(kSsrcX).muted()); +} + +// Test that SetSendParameters() does not alter a stream's send state. +TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Turn on sending. + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + + // Changing RTP header extensions will recreate the AudioSendStream. + send_parameters_.extensions.push_back( + webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); + SetSendParameters(send_parameters_); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + + // Turn off sending. + SetSend(false); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Changing RTP header extensions will recreate the AudioSendStream. + send_parameters_.extensions.clear(); + SetSendParameters(send_parameters_); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); +} + +// Test that we can create a channel and start playing out on it. +TEST_F(WebRtcVoiceEngineTestFake, Playout) { + EXPECT_TRUE(SetupRecvStream()); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); + channel_->SetPlayout(true); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); + channel_->SetPlayout(false); + EXPECT_FALSE(GetRecvStream(kSsrcX).started()); +} + +// Test that we can add and remove send streams. +TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) { + SetupForMultiSendStream(); + + // Set the global state for sending. + SetSend(true); + + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + SetAudioSend(ssrc, true, &fake_source_); + // Verify that we are in a sending state for all the created streams. + EXPECT_TRUE(GetSendStream(ssrc).IsSending()); + } + EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size()); + + // Delete the send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(channel_->RemoveSendStream(ssrc)); + EXPECT_FALSE(call_.GetAudioSendStream(ssrc)); + EXPECT_FALSE(channel_->RemoveSendStream(ssrc)); + } + EXPECT_EQ(0u, call_.GetAudioSendStreams().size()); +} + +// Test SetSendCodecs correctly configure the codecs in all send streams. +TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) { + SetupForMultiSendStream(); + + // Create send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + } + + cricket::AudioSendParameters parameters; + // Set ISAC(16K) and CN(16K). VAD should be activated. + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kCn16000Codec); + parameters.codecs[1].id = 97; + SetSendParameters(parameters); + + // Verify ISAC and VAD are corrected configured on all send channels. + for (uint32_t ssrc : kSsrcs4) { + ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); + const auto& send_codec_spec = + *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; + EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(1, send_codec_spec.format.num_channels); + EXPECT_EQ(97, send_codec_spec.cng_payload_type); + } + + // Change to PCMU(8K) and CN(16K). + parameters.codecs[0] = kPcmuCodec; + SetSendParameters(parameters); + for (uint32_t ssrc : kSsrcs4) { + ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); + const auto& send_codec_spec = + *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); + EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type); + } +} + +// Test we can SetSend on all send streams correctly. +TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) { + SetupForMultiSendStream(); + + // Create the send channels and they should be a "not sending" date. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + SetAudioSend(ssrc, true, &fake_source_); + EXPECT_FALSE(GetSendStream(ssrc).IsSending()); + } + + // Set the global state for starting sending. + SetSend(true); + for (uint32_t ssrc : kSsrcs4) { + // Verify that we are in a sending state for all the send streams. + EXPECT_TRUE(GetSendStream(ssrc).IsSending()); + } + + // Set the global state for stopping sending. + SetSend(false); + for (uint32_t ssrc : kSsrcs4) { + // Verify that we are in a stop state for all the send streams. + EXPECT_FALSE(GetSendStream(ssrc).IsSending()); + } +} + +// Test we can set the correct statistics on all send streams. +TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) { + SetupForMultiSendStream(); + + // Create send streams. + for (uint32_t ssrc : kSsrcs4) { + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(ssrc))); + } + + // Create a receive stream to check that none of the send streams end up in + // the receive stream stats. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + + // We need send codec to be set to get all stats. + SetSendParameters(send_parameters_); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); + SetAudioSendStreamStats(); + + // Check stats for the added streams. + { + cricket::VoiceMediaInfo info; + EXPECT_EQ(true, channel_->GetStats(&info)); + + // We have added 4 send streams. We should see empty stats for all. + EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); + for (const auto& sender : info.senders) { + VerifyVoiceSenderInfo(sender, false); + } + VerifyVoiceSendRecvCodecs(info); + + // We have added one receive stream. We should see empty stats. + EXPECT_EQ(info.receivers.size(), 1u); + EXPECT_EQ(info.receivers[0].ssrc(), 0); + } + + // Remove the kSsrcY stream. No receiver stats. + { + cricket::VoiceMediaInfo info; + EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); + EXPECT_EQ(true, channel_->GetStats(&info)); + EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); + EXPECT_EQ(0u, info.receivers.size()); + } + + // Deliver a new packet - a default receive stream should be created and we + // should see stats again. + { + cricket::VoiceMediaInfo info; + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + SetAudioReceiveStreamStats(); + EXPECT_EQ(true, channel_->GetStats(&info)); + EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); + EXPECT_EQ(1u, info.receivers.size()); + VerifyVoiceReceiverInfo(info.receivers[0]); + VerifyVoiceSendRecvCodecs(info); + } +} + +// Test that we can add and remove receive streams, and do proper send/playout. +// We can receive on multiple streams while sending one stream. +TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) { + EXPECT_TRUE(SetupSendStream()); + + // Start playout without a receive stream. + SetSendParameters(send_parameters_); + channel_->SetPlayout(true); + + // Adding another stream should enable playout on the new stream only. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + + // Make sure only the new stream is played out. + EXPECT_TRUE(GetRecvStream(kSsrcY).started()); + + // Adding yet another stream should have stream 2 and 3 enabled for playout. + EXPECT_TRUE(AddRecvStream(kSsrcZ)); + EXPECT_TRUE(GetRecvStream(kSsrcY).started()); + EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); + + // Stop sending. + SetSend(false); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); + + // Stop playout. + channel_->SetPlayout(false); + EXPECT_FALSE(GetRecvStream(kSsrcY).started()); + EXPECT_FALSE(GetRecvStream(kSsrcZ).started()); + + // Restart playout and make sure recv streams are played out. + channel_->SetPlayout(true); + EXPECT_TRUE(GetRecvStream(kSsrcY).started()); + EXPECT_TRUE(GetRecvStream(kSsrcZ).started()); + + // Now remove the recv streams. + EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcZ)); + EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); +} + +// Test that we can create a channel configured for Codian bridges, +// and start sending on it. +TEST_F(WebRtcVoiceEngineTestFake, CodianSend) { + EXPECT_TRUE(SetupSendStream()); + send_parameters_.options.adjust_agc_delta = -10; + EXPECT_CALL(apm_gc_, + set_target_level_dbfs(11)).Times(2).WillRepeatedly(Return(0)); + SetSendParameters(send_parameters_); + SetSend(true); + EXPECT_TRUE(GetSendStream(kSsrcX).IsSending()); + SetSend(false); + EXPECT_FALSE(GetSendStream(kSsrcX).IsSending()); +} + +TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_CALL(adm_, + BuiltInAGCIsAvailable()).Times(2).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, SetAGC(true)).Times(2).WillRepeatedly(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).Times(2).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).Times(2).WillOnce(Return(0)); + send_parameters_.options.tx_agc_target_dbov = 3; + send_parameters_.options.tx_agc_digital_compression_gain = 9; + send_parameters_.options.tx_agc_limiter = true; + send_parameters_.options.auto_gain_control = true; + EXPECT_CALL(apm_gc_, set_target_level_dbfs(3)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_compression_gain_db(9)).WillRepeatedly(Return(0)); + EXPECT_CALL(apm_gc_, enable_limiter(true)).WillRepeatedly(Return(0)); + SetSendParameters(send_parameters_); + + // Check interaction with adjust_agc_delta. Both should be respected, for + // backwards compatibility. + send_parameters_.options.adjust_agc_delta = -10; + EXPECT_CALL(apm_gc_, set_target_level_dbfs(13)).WillOnce(Return(0)); + SetSendParameters(send_parameters_); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) { + EXPECT_TRUE(SetupSendStream()); + send_parameters_.options.audio_network_adaptor = true; + send_parameters_.options.audio_network_adaptor_config = {"1234"}; + SetSendParameters(send_parameters_); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); +} + +TEST_F(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) { + EXPECT_TRUE(SetupSendStream()); + send_parameters_.options.audio_network_adaptor = true; + send_parameters_.options.audio_network_adaptor_config = {"1234"}; + SetSendParameters(send_parameters_); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); + cricket::AudioOptions options; + options.audio_network_adaptor = false; + SetAudioSend(kSsrcX, true, nullptr, &options); + EXPECT_EQ(rtc::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX)); +} + +TEST_F(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) { + EXPECT_TRUE(SetupSendStream()); + send_parameters_.options.audio_network_adaptor = true; + send_parameters_.options.audio_network_adaptor_config = {"1234"}; + SetSendParameters(send_parameters_); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); + const int initial_num = call_.GetNumCreatedSendStreams(); + cricket::AudioOptions options; + options.audio_network_adaptor = rtc::nullopt; + // Unvalued |options.audio_network_adaptor|.should not reset audio network + // adaptor. + SetAudioSend(kSsrcX, true, nullptr, &options); + // AudioSendStream not expected to be recreated. + EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); + EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config, + GetAudioNetworkAdaptorConfig(kSsrcX)); +} + +class WebRtcVoiceEngineWithSendSideBweWithOverheadTest + : public WebRtcVoiceEngineTestFake { + public: + WebRtcVoiceEngineWithSendSideBweWithOverheadTest() + : WebRtcVoiceEngineTestFake( + "WebRTC-Audio-SendSideBwe/Enabled/WebRTC-SendSideBwe-WithOverhead/" + "Enabled/") {} +}; + +TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest, MinAndMaxBitrate) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters parameters; + parameters.codecs.push_back(kOpusCodec); + SetSendParameters(parameters); + const int initial_num = call_.GetNumCreatedSendStreams(); + EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams()); + + // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12) + constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12; + constexpr int kOpusMaxPtimeMs = WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60; + constexpr int kMinOverheadBps = + kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs; + + constexpr int kOpusMinBitrateBps = 6000; + EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadBps, + GetSendStreamConfig(kSsrcX).min_bitrate_bps); + constexpr int kOpusBitrateFbBps = 32000; + EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadBps, + GetSendStreamConfig(kSsrcX).max_bitrate_bps); + + parameters.options.audio_network_adaptor = true; + parameters.options.audio_network_adaptor_config = {"1234"}; + SetSendParameters(parameters); + + constexpr int kMinOverheadWithAnaBps = + kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs; + + EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadWithAnaBps, + GetSendStreamConfig(kSsrcX).min_bitrate_bps); + + EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadWithAnaBps, + GetSendStreamConfig(kSsrcX).max_bitrate_bps); +} + +// This test is similar to +// WebRtcVoiceEngineTestFake.SetRtpSendParameterUpdatesMaxBitrate but with an +// additional field trial. +TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest, + SetRtpSendParameterUpdatesMaxBitrate) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioSendParameters send_parameters; + send_parameters.codecs.push_back(kOpusCodec); + SetSendParameters(send_parameters); + + webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); + // Expect empty on parameters.encodings[0].max_bitrate_bps; + EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps); + + constexpr int kMaxBitrateBps = 6000; + rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps; + EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters)); + + const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps; +#if WEBRTC_OPUS_SUPPORT_120MS_PTIME + constexpr int kMinOverhead = 3333; +#else + constexpr int kMinOverhead = 6666; +#endif + EXPECT_EQ(max_bitrate, kMaxBitrateBps + kMinOverhead); +} + +// Test that we can set the outgoing SSRC properly. +// SSRC is set in SetupSendStream() by calling AddSendStream. +TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); +} + +TEST_F(WebRtcVoiceEngineTestFake, GetStats) { + // Setup. We need send codec to be set to get all stats. + EXPECT_TRUE(SetupSendStream()); + // SetupSendStream adds a send stream with kSsrcX, so the receive + // stream has to use a different SSRC. + EXPECT_TRUE(AddRecvStream(kSsrcY)); + SetSendParameters(send_parameters_); + EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); + SetAudioSendStreamStats(); + + // Check stats for the added streams. + { + cricket::VoiceMediaInfo info; + EXPECT_EQ(true, channel_->GetStats(&info)); + + // We have added one send stream. We should see the stats we've set. + EXPECT_EQ(1u, info.senders.size()); + VerifyVoiceSenderInfo(info.senders[0], false); + // We have added one receive stream. We should see empty stats. + EXPECT_EQ(info.receivers.size(), 1u); + EXPECT_EQ(info.receivers[0].ssrc(), 0); + } + + // Start sending - this affects some reported stats. + { + cricket::VoiceMediaInfo info; + SetSend(true); + EXPECT_EQ(true, channel_->GetStats(&info)); + VerifyVoiceSenderInfo(info.senders[0], true); + VerifyVoiceSendRecvCodecs(info); + } + + // Remove the kSsrcY stream. No receiver stats. + { + cricket::VoiceMediaInfo info; + EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY)); + EXPECT_EQ(true, channel_->GetStats(&info)); + EXPECT_EQ(1u, info.senders.size()); + EXPECT_EQ(0u, info.receivers.size()); + } + + // Deliver a new packet - a default receive stream should be created and we + // should see stats again. + { + cricket::VoiceMediaInfo info; + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + SetAudioReceiveStreamStats(); + EXPECT_EQ(true, channel_->GetStats(&info)); + EXPECT_EQ(1u, info.senders.size()); + EXPECT_EQ(1u, info.receivers.size()); + VerifyVoiceReceiverInfo(info.receivers[0]); + VerifyVoiceSendRecvCodecs(info); + } +} + +// Test that we can set the outgoing SSRC properly with multiple streams. +// SSRC is set in SetupSendStream() by calling AddSendStream. +TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); +} + +// Test that the local SSRC is the same on sending and receiving channels if the +// receive channel is created before the send channel. +TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); +} + +// Test that we can properly receive packets. +TEST_F(WebRtcVoiceEngineTestFake, Recv) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(1)); + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame, + sizeof(kPcmuFrame))); +} + +// Test that we can properly receive packets on multiple streams. +TEST_F(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) { + EXPECT_TRUE(SetupChannel()); + const uint32_t ssrc1 = 1; + const uint32_t ssrc2 = 2; + const uint32_t ssrc3 = 3; + EXPECT_TRUE(AddRecvStream(ssrc1)); + EXPECT_TRUE(AddRecvStream(ssrc2)); + EXPECT_TRUE(AddRecvStream(ssrc3)); + // Create packets with the right SSRCs. + unsigned char packets[4][sizeof(kPcmuFrame)]; + for (size_t i = 0; i < arraysize(packets); ++i) { + memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i)); + } + + const cricket::FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1); + const cricket::FakeAudioReceiveStream& s2 = GetRecvStream(ssrc2); + const cricket::FakeAudioReceiveStream& s3 = GetRecvStream(ssrc3); + + EXPECT_EQ(s1.received_packets(), 0); + EXPECT_EQ(s2.received_packets(), 0); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[0], sizeof(packets[0])); + EXPECT_EQ(s1.received_packets(), 0); + EXPECT_EQ(s2.received_packets(), 0); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[1], sizeof(packets[1])); + EXPECT_EQ(s1.received_packets(), 1); + EXPECT_TRUE(s1.VerifyLastPacket(packets[1], sizeof(packets[1]))); + EXPECT_EQ(s2.received_packets(), 0); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[2], sizeof(packets[2])); + EXPECT_EQ(s1.received_packets(), 1); + EXPECT_EQ(s2.received_packets(), 1); + EXPECT_TRUE(s2.VerifyLastPacket(packets[2], sizeof(packets[2]))); + EXPECT_EQ(s3.received_packets(), 0); + + DeliverPacket(packets[3], sizeof(packets[3])); + EXPECT_EQ(s1.received_packets(), 1); + EXPECT_EQ(s2.received_packets(), 1); + EXPECT_EQ(s3.received_packets(), 1); + EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3]))); + + EXPECT_TRUE(channel_->RemoveRecvStream(ssrc3)); + EXPECT_TRUE(channel_->RemoveRecvStream(ssrc2)); + EXPECT_TRUE(channel_->RemoveRecvStream(ssrc1)); +} + +// Test that receiving on an unsignaled stream works (a stream is created). +TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaled) { + EXPECT_TRUE(SetupChannel()); + EXPECT_EQ(0, call_.GetAudioReceiveStreams().size()); + + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + + EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE(GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, + sizeof(kPcmuFrame))); +} + +// Test that receiving N unsignaled stream works (streams will be created), and +// that packets are forwarded to them all. +TEST_F(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) { + EXPECT_TRUE(SetupChannel()); + unsigned char packet[sizeof(kPcmuFrame)]; + memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); + + // Note that SSRC = 0 is not supported. + for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) { + rtc::SetBE32(&packet[8], ssrc); + DeliverPacket(packet, sizeof(packet)); + + // Verify we have one new stream for each loop iteration. + EXPECT_EQ(ssrc, call_.GetAudioReceiveStreams().size()); + EXPECT_EQ(1, GetRecvStream(ssrc).received_packets()); + EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); + } + + // Sending on the same SSRCs again should not create new streams. + for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) { + rtc::SetBE32(&packet[8], ssrc); + DeliverPacket(packet, sizeof(packet)); + + EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size()); + EXPECT_EQ(2, GetRecvStream(ssrc).received_packets()); + EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); + } + + // Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced. + constexpr uint32_t kAnotherSsrc = 667; + rtc::SetBE32(&packet[8], kAnotherSsrc); + DeliverPacket(packet, sizeof(packet)); + + const auto& streams = call_.GetAudioReceiveStreams(); + EXPECT_EQ(kMaxUnsignaledRecvStreams, streams.size()); + size_t i = 0; + for (uint32_t ssrc = 2; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc, ++i) { + EXPECT_EQ(ssrc, streams[i]->GetConfig().rtp.remote_ssrc); + EXPECT_EQ(2, streams[i]->received_packets()); + } + EXPECT_EQ(kAnotherSsrc, streams[i]->GetConfig().rtp.remote_ssrc); + EXPECT_EQ(1, streams[i]->received_packets()); + // Sanity check that we've checked all streams. + EXPECT_EQ(kMaxUnsignaledRecvStreams, (i + 1)); +} + +// Test that a default channel is created even after a signaled stream has been +// added, and that this stream will get any packets for unknown SSRCs. +TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) { + EXPECT_TRUE(SetupChannel()); + unsigned char packet[sizeof(kPcmuFrame)]; + memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); + + // Add a known stream, send packet and verify we got it. + const uint32_t signaled_ssrc = 1; + rtc::SetBE32(&packet[8], signaled_ssrc); + EXPECT_TRUE(AddRecvStream(signaled_ssrc)); + DeliverPacket(packet, sizeof(packet)); + EXPECT_TRUE(GetRecvStream(signaled_ssrc).VerifyLastPacket( + packet, sizeof(packet))); + EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); + + // Note that the first unknown SSRC cannot be 0, because we only support + // creating receive streams for SSRC!=0. + const uint32_t unsignaled_ssrc = 7011; + rtc::SetBE32(&packet[8], unsignaled_ssrc); + DeliverPacket(packet, sizeof(packet)); + EXPECT_TRUE(GetRecvStream(unsignaled_ssrc).VerifyLastPacket( + packet, sizeof(packet))); + EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); + + DeliverPacket(packet, sizeof(packet)); + EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets()); + + rtc::SetBE32(&packet[8], signaled_ssrc); + DeliverPacket(packet, sizeof(packet)); + EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets()); + EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); +} + +// Two tests to verify that adding a receive stream with the same SSRC as a +// previously added unsignaled stream will only recreate underlying stream +// objects if the stream parameters have changed. +TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) { + EXPECT_TRUE(SetupChannel()); + + // Spawn unsignaled stream with SSRC=1. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame, + sizeof(kPcmuFrame))); + + // Verify that the underlying stream object in Call is not recreated when a + // stream with SSRC=1 is added. + const auto& streams = call_.GetAudioReceiveStreams(); + EXPECT_EQ(1, streams.size()); + int audio_receive_stream_id = streams.front()->id(); + EXPECT_TRUE(AddRecvStream(1)); + EXPECT_EQ(1, streams.size()); + EXPECT_EQ(audio_receive_stream_id, streams.front()->id()); +} + +TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Recreate) { + EXPECT_TRUE(SetupChannel()); + + // Spawn unsignaled stream with SSRC=1. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); + EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame, + sizeof(kPcmuFrame))); + + // Verify that the underlying stream object in Call *is* recreated when a + // stream with SSRC=1 is added, and which has changed stream parameters. + const auto& streams = call_.GetAudioReceiveStreams(); + EXPECT_EQ(1, streams.size()); + int audio_receive_stream_id = streams.front()->id(); + cricket::StreamParams stream_params; + stream_params.ssrcs.push_back(1); + stream_params.sync_label = "sync_label"; + EXPECT_TRUE(channel_->AddRecvStream(stream_params)); + EXPECT_EQ(1, streams.size()); + EXPECT_NE(audio_receive_stream_id, streams.front()->id()); +} + +// Test that we properly handle failures to add a receive stream. +TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamFail) { + EXPECT_TRUE(SetupChannel()); + voe_.set_fail_create_channel(true); + EXPECT_FALSE(AddRecvStream(2)); +} + +// Test that we properly handle failures to add a send stream. +TEST_F(WebRtcVoiceEngineTestFake, AddSendStreamFail) { + EXPECT_TRUE(SetupChannel()); + voe_.set_fail_create_channel(true); + EXPECT_FALSE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2))); +} + +// Test that AddRecvStream creates new stream. +TEST_F(WebRtcVoiceEngineTestFake, AddRecvStream) { + EXPECT_TRUE(SetupRecvStream()); + int channel_num = voe_.GetLastChannel(); + EXPECT_TRUE(AddRecvStream(1)); + EXPECT_NE(channel_num, voe_.GetLastChannel()); +} + +// Test that after adding a recv stream, we do not decode more codecs than +// those previously passed into SetRecvCodecs. +TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) { + EXPECT_TRUE(SetupSendStream()); + cricket::AudioRecvParameters parameters; + parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); + EXPECT_TRUE(channel_->SetRecvParameters(parameters)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, + (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>( + {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}}))); +} + +// Test that we properly clean up any streams that were added, even if +// not explicitly removed. +TEST_F(WebRtcVoiceEngineTestFake, StreamCleanup) { + EXPECT_TRUE(SetupSendStream()); + SetSendParameters(send_parameters_); + EXPECT_TRUE(AddRecvStream(1)); + EXPECT_TRUE(AddRecvStream(2)); + EXPECT_EQ(3, voe_.GetNumChannels()); // default channel + 2 added + delete channel_; + channel_ = NULL; + EXPECT_EQ(0, voe_.GetNumChannels()); +} + +TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithZeroSsrc) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_FALSE(AddRecvStream(0)); +} + +TEST_F(WebRtcVoiceEngineTestFake, TestNoLeakingWhenAddRecvStreamFail) { + EXPECT_TRUE(SetupChannel()); + EXPECT_TRUE(AddRecvStream(1)); + // Manually delete channel to simulate a failure. + int channel = voe_.GetLastChannel(); + EXPECT_EQ(0, voe_.DeleteChannel(channel)); + // Add recv stream 2 should work. + EXPECT_TRUE(AddRecvStream(2)); + int new_channel = voe_.GetLastChannel(); + EXPECT_NE(channel, new_channel); + // The last created channel is deleted too. + EXPECT_EQ(0, voe_.DeleteChannel(new_channel)); +} + +// Test the InsertDtmf on default send stream as caller. +TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) { + TestInsertDtmf(0, true, kTelephoneEventCodec1); +} + +// Test the InsertDtmf on default send stream as callee +TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) { + TestInsertDtmf(0, false, kTelephoneEventCodec2); +} + +// Test the InsertDtmf on specified send stream as caller. +TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) { + TestInsertDtmf(kSsrcX, true, kTelephoneEventCodec2); +} + +// Test the InsertDtmf on specified send stream as callee. +TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) { + TestInsertDtmf(kSsrcX, false, kTelephoneEventCodec1); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_CALL(adm_, + BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, + BuiltInAGCIsAvailable()).Times(4).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, + BuiltInNSIsAvailable()).Times(2).WillRepeatedly(Return(false)); + + EXPECT_EQ(50, voe_.GetNetEqCapacity()); + EXPECT_FALSE(voe_.GetNetEqFastAccelerate()); + + // Nothing set in AudioOptions, so everything should be as default. + send_parameters_.options = cricket::AudioOptions(); + SetSendParameters(send_parameters_); + EXPECT_TRUE(IsHighPassFilterEnabled()); + EXPECT_EQ(50, voe_.GetNetEqCapacity()); + EXPECT_FALSE(voe_.GetNetEqFastAccelerate()); + + // Turn echo cancellation off + EXPECT_CALL(apm_ec_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(false)).WillOnce(Return(0)); + send_parameters_.options.echo_cancellation = false; + SetSendParameters(send_parameters_); + + // Turn echo cancellation back on, with settings, and make sure + // nothing else changed. + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + send_parameters_.options.echo_cancellation = true; + SetSendParameters(send_parameters_); + + // Turn on delay agnostic aec and make sure nothing change w.r.t. echo + // control. + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + send_parameters_.options.delay_agnostic_aec = true; + SetSendParameters(send_parameters_); + + // Turn off echo cancellation and delay agnostic aec. + EXPECT_CALL(apm_ec_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(false)).WillOnce(Return(0)); + send_parameters_.options.delay_agnostic_aec = false; + send_parameters_.options.extended_filter_aec = false; + send_parameters_.options.echo_cancellation = false; + SetSendParameters(send_parameters_); + + // Turning delay agnostic aec back on should also turn on echo cancellation. + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + send_parameters_.options.delay_agnostic_aec = true; + SetSendParameters(send_parameters_); + + // Turn off AGC + EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); + send_parameters_.options.auto_gain_control = false; + SetSendParameters(send_parameters_); + + // Turn AGC back on + EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); + send_parameters_.options.auto_gain_control = true; + send_parameters_.options.adjust_agc_delta = rtc::nullopt; + SetSendParameters(send_parameters_); + + // Turn off other options (and stereo swapping on). + EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_vd_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(true)); + send_parameters_.options.noise_suppression = false; + send_parameters_.options.highpass_filter = false; + send_parameters_.options.typing_detection = false; + send_parameters_.options.stereo_swapping = true; + SetSendParameters(send_parameters_); + EXPECT_FALSE(IsHighPassFilterEnabled()); + + // Set options again to ensure it has no impact. + EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_vd_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(transmit_mixer_, EnableStereoChannelSwapping(true)); + SetSendParameters(send_parameters_); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_CALL(adm_, + BuiltInAECIsAvailable()).Times(8).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, + BuiltInAGCIsAvailable()).Times(8).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, + BuiltInNSIsAvailable()).Times(8).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, + RecordingIsInitialized()).Times(2).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, Recording()).Times(2).WillRepeatedly(Return(false)); + EXPECT_CALL(adm_, InitRecording()).Times(2).WillRepeatedly(Return(0)); + webrtc::AudioProcessing::Config apm_config; + EXPECT_CALL(*apm_, GetConfig()) + .Times(10) + .WillRepeatedly(ReturnPointee(&apm_config)); + EXPECT_CALL(*apm_, ApplyConfig(_)) + .Times(10) + .WillRepeatedly(SaveArg<0>(&apm_config)); + EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(10); + + std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1( + static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel( + &call_, cricket::MediaConfig(), cricket::AudioOptions()))); + std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel2( + static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel( + &call_, cricket::MediaConfig(), cricket::AudioOptions()))); + + // Have to add a stream to make SetSend work. + cricket::StreamParams stream1; + stream1.ssrcs.push_back(1); + channel1->AddSendStream(stream1); + cricket::StreamParams stream2; + stream2.ssrcs.push_back(2); + channel2->AddSendStream(stream2); + + // AEC and AGC and NS + cricket::AudioSendParameters parameters_options_all = send_parameters_; + parameters_options_all.options.echo_cancellation = true; + parameters_options_all.options.auto_gain_control = true; + parameters_options_all.options.noise_suppression = true; + EXPECT_CALL(adm_, SetAGC(true)).Times(2).WillRepeatedly(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).Times(2).WillRepeatedly(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).Times(2).WillRepeatedly(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).Times(2).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).Times(2).WillRepeatedly(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).Times(2).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(true)).Times(2).WillRepeatedly(Return(0)); + EXPECT_TRUE(channel1->SetSendParameters(parameters_options_all)); + EXPECT_EQ(parameters_options_all.options, channel1->options()); + EXPECT_TRUE(channel2->SetSendParameters(parameters_options_all)); + EXPECT_EQ(parameters_options_all.options, channel2->options()); + + // unset NS + cricket::AudioSendParameters parameters_options_no_ns = send_parameters_; + parameters_options_no_ns.options.noise_suppression = false; + EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); + EXPECT_TRUE(channel1->SetSendParameters(parameters_options_no_ns)); + cricket::AudioOptions expected_options = parameters_options_all.options; + expected_options.echo_cancellation = true; + expected_options.auto_gain_control = true; + expected_options.noise_suppression = false; + EXPECT_EQ(expected_options, channel1->options()); + + // unset AGC + cricket::AudioSendParameters parameters_options_no_agc = send_parameters_; + parameters_options_no_agc.options.auto_gain_control = false; + EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); + EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc)); + expected_options.echo_cancellation = true; + expected_options.auto_gain_control = false; + expected_options.noise_suppression = true; + EXPECT_EQ(expected_options, channel2->options()); + + EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); + EXPECT_TRUE(channel_->SetSendParameters(parameters_options_all)); + + EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); + channel1->SetSend(true); + + EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0)); + channel2->SetSend(true); + + // Make sure settings take effect while we are sending. + cricket::AudioSendParameters parameters_options_no_agc_nor_ns = + send_parameters_; + parameters_options_no_agc_nor_ns.options.auto_gain_control = false; + parameters_options_no_agc_nor_ns.options.noise_suppression = false; + EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0)); + EXPECT_CALL(apm_gc_, Enable(false)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0)); + EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0)); + EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc_nor_ns)); + expected_options.echo_cancellation = true; + expected_options.auto_gain_control = false; + expected_options.noise_suppression = false; + EXPECT_EQ(expected_options, channel2->options()); +} + +// This test verifies DSCP settings are properly applied on voice media channel. +TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) { + EXPECT_TRUE(SetupSendStream()); + cricket::FakeNetworkInterface network_interface; + cricket::MediaConfig config; + std::unique_ptr<cricket::VoiceMediaChannel> channel; + + webrtc::AudioProcessing::Config apm_config; + EXPECT_CALL(*apm_, GetConfig()) + .Times(3) + .WillRepeatedly(ReturnPointee(&apm_config)); + EXPECT_CALL(*apm_, ApplyConfig(_)) + .Times(3) + .WillRepeatedly(SaveArg<0>(&apm_config)); + EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(3); + + channel.reset( + engine_->CreateChannel(&call_, config, cricket::AudioOptions())); + channel->SetInterface(&network_interface); + // Default value when DSCP is disabled should be DSCP_DEFAULT. + EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); + + config.enable_dscp = true; + channel.reset( + engine_->CreateChannel(&call_, config, cricket::AudioOptions())); + channel->SetInterface(&network_interface); + EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp()); + + // Verify that setting the option to false resets the + // DiffServCodePoint. + config.enable_dscp = false; + channel.reset( + engine_->CreateChannel(&call_, config, cricket::AudioOptions())); + channel->SetInterface(&network_interface); + // Default value when DSCP is disabled should be DSCP_DEFAULT. + EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp()); + + channel->SetInterface(nullptr); +} + +TEST_F(WebRtcVoiceEngineTestFake, TestGetReceiveChannelId) { + EXPECT_TRUE(SetupChannel()); + cricket::WebRtcVoiceMediaChannel* media_channel = + static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); + EXPECT_EQ(-1, media_channel->GetReceiveChannelId(0)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + int channel_id = voe_.GetLastChannel(); + EXPECT_EQ(channel_id, media_channel->GetReceiveChannelId(kSsrcX)); + EXPECT_EQ(-1, media_channel->GetReceiveChannelId(kSsrcY)); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + int channel_id2 = voe_.GetLastChannel(); + EXPECT_EQ(channel_id2, media_channel->GetReceiveChannelId(kSsrcY)); +} + +TEST_F(WebRtcVoiceEngineTestFake, TestGetSendChannelId) { + EXPECT_TRUE(SetupChannel()); + cricket::WebRtcVoiceMediaChannel* media_channel = + static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); + EXPECT_EQ(-1, media_channel->GetSendChannelId(0)); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcX))); + int channel_id = voe_.GetLastChannel(); + EXPECT_EQ(channel_id, media_channel->GetSendChannelId(kSsrcX)); + EXPECT_EQ(-1, media_channel->GetSendChannelId(kSsrcY)); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcY))); + int channel_id2 = voe_.GetLastChannel(); + EXPECT_EQ(channel_id2, media_channel->GetSendChannelId(kSsrcY)); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolume) { + EXPECT_TRUE(SetupChannel()); + EXPECT_FALSE(channel_->SetOutputVolume(kSsrcY, 0.5)); + cricket::StreamParams stream; + stream.ssrcs.push_back(kSsrcY); + EXPECT_TRUE(channel_->AddRecvStream(stream)); + EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrcY).gain()); + EXPECT_TRUE(channel_->SetOutputVolume(kSsrcY, 3)); + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcY).gain()); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) { + EXPECT_TRUE(SetupChannel()); + + // Spawn an unsignaled stream by sending a packet - gain should be 1. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrc1).gain()); + + // Should remember the volume "2" which will be set on new unsignaled streams, + // and also set the gain to 2 on existing unsignaled streams. + EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 2)); + EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrc1).gain()); + + // Spawn an unsignaled stream by sending a packet - gain should be 2. + unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; + memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(&pcmuFrame2[8], kSsrcX); + DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); + EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain()); + + // Setting gain with SSRC=0 should affect all unsignaled streams. + EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 3)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain()); + } + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcX).gain()); + + // Setting gain on an individual stream affects only that. + EXPECT_TRUE(channel_->SetOutputVolume(kSsrcX, 4)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain()); + } + EXPECT_DOUBLE_EQ(4, GetRecvStream(kSsrcX).gain()); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromSyncLabel) { + const uint32_t kAudioSsrc = 123; + const std::string kSyncLabel = "AvSyncLabel"; + + EXPECT_TRUE(SetupSendStream()); + cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kAudioSsrc); + sp.sync_label = kSyncLabel; + // Creating two channels to make sure that sync label is set properly for both + // the default voice channel and following ones. + EXPECT_TRUE(channel_->AddRecvStream(sp)); + sp.ssrcs[0] += 1; + EXPECT_TRUE(channel_->AddRecvStream(sp)); + + ASSERT_EQ(2, call_.GetAudioReceiveStreams().size()); + EXPECT_EQ(kSyncLabel, + call_.GetAudioReceiveStream(kAudioSsrc)->GetConfig().sync_group) + << "SyncGroup should be set based on sync_label"; + EXPECT_EQ(kSyncLabel, + call_.GetAudioReceiveStream(kAudioSsrc + 1)->GetConfig().sync_group) + << "SyncGroup should be set based on sync_label"; +} + +// TODO(solenberg): Remove, once recv streams are configured through Call. +// (This is then covered by TestSetRecvRtpHeaderExtensions.) +TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) { + // Test that setting the header extensions results in the expected state + // changes on an associated Call. + std::vector<uint32_t> ssrcs; + ssrcs.push_back(223); + ssrcs.push_back(224); + + EXPECT_TRUE(SetupSendStream()); + SetSendParameters(send_parameters_); + for (uint32_t ssrc : ssrcs) { + EXPECT_TRUE(channel_->AddRecvStream( + cricket::StreamParams::CreateLegacy(ssrc))); + } + + EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); + for (uint32_t ssrc : ssrcs) { + const auto* s = call_.GetAudioReceiveStream(ssrc); + EXPECT_NE(nullptr, s); + EXPECT_EQ(0, s->GetConfig().rtp.extensions.size()); + } + + // Set up receive extensions. + cricket::RtpCapabilities capabilities = engine_->GetCapabilities(); + cricket::AudioRecvParameters recv_parameters; + recv_parameters.extensions = capabilities.header_extensions; + channel_->SetRecvParameters(recv_parameters); + EXPECT_EQ(2, call_.GetAudioReceiveStreams().size()); + for (uint32_t ssrc : ssrcs) { + const auto* s = call_.GetAudioReceiveStream(ssrc); + EXPECT_NE(nullptr, s); + const auto& s_exts = s->GetConfig().rtp.extensions; + EXPECT_EQ(capabilities.header_extensions.size(), s_exts.size()); + for (const auto& e_ext : capabilities.header_extensions) { + for (const auto& s_ext : s_exts) { + if (e_ext.id == s_ext.id) { + EXPECT_EQ(e_ext.uri, s_ext.uri); + } + } + } + } + + // Disable receive extensions. + channel_->SetRecvParameters(cricket::AudioRecvParameters()); + for (uint32_t ssrc : ssrcs) { + const auto* s = call_.GetAudioReceiveStream(ssrc); + EXPECT_NE(nullptr, s); + EXPECT_EQ(0, s->GetConfig().rtp.extensions.size()); + } +} + +TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) { + // Test that packets are forwarded to the Call when configured accordingly. + const uint32_t kAudioSsrc = 1; + rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame)); + static const unsigned char kRtcp[] = { + 0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, + 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 + }; + rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp)); + + EXPECT_TRUE(SetupSendStream()); + cricket::WebRtcVoiceMediaChannel* media_channel = + static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); + SetSendParameters(send_parameters_); + EXPECT_TRUE(media_channel->AddRecvStream( + cricket::StreamParams::CreateLegacy(kAudioSsrc))); + + EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); + const cricket::FakeAudioReceiveStream* s = + call_.GetAudioReceiveStream(kAudioSsrc); + EXPECT_EQ(0, s->received_packets()); + channel_->OnPacketReceived(&kPcmuPacket, rtc::PacketTime()); + EXPECT_EQ(1, s->received_packets()); + channel_->OnRtcpReceived(&kRtcpPacket, rtc::PacketTime()); + EXPECT_EQ(2, s->received_packets()); +} + +// All receive channels should be associated with the first send channel, +// since they do not send RTCP SR. +TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) { + EXPECT_TRUE(SetupSendStream()); + EXPECT_TRUE(AddRecvStream(kSsrcY)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcZ))); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc); + EXPECT_TRUE(AddRecvStream(kSsrcW)); + EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcW).rtp.local_ssrc); +} + +TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) { + EXPECT_TRUE(SetupRecvStream()); + EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcY))); + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); + EXPECT_TRUE(AddRecvStream(kSsrcZ)); + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc); + EXPECT_TRUE(channel_->AddSendStream( + cricket::StreamParams::CreateLegacy(kSsrcW))); + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc); + EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetRawAudioSink) { + EXPECT_TRUE(SetupChannel()); + std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink()); + + // Setting the sink before a recv stream exists should do nothing. + channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_1)); + EXPECT_TRUE(AddRecvStream(kSsrcX)); + EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); + + // Now try actually setting the sink. + channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_2)); + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + + // Now try resetting it. + channel_->SetRawAudioSink(kSsrcX, nullptr); + EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); +} + +TEST_F(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) { + EXPECT_TRUE(SetupChannel()); + std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_3(new FakeAudioSink()); + std::unique_ptr<FakeAudioSink> fake_sink_4(new FakeAudioSink()); + + // Should be able to set a default sink even when no stream exists. + channel_->SetRawAudioSink(0, std::move(fake_sink_1)); + + // Spawn an unsignaled stream by sending a packet - it should be assigned the + // default sink. + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + + // Try resetting the default sink. + channel_->SetRawAudioSink(kSsrc0, nullptr); + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + + // Try setting the default sink while the default stream exists. + channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_2)); + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + + // If we remove and add a default stream, it should get the same sink. + EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc1)); + DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + + // Spawn another unsignaled stream - it should be assigned the default sink + // and the previous unsignaled stream should lose it. + unsigned char pcmuFrame2[sizeof(kPcmuFrame)]; + memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame)); + rtc::SetBE32(&pcmuFrame2[8], kSsrcX); + DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + + // Reset the default sink - the second unsignaled stream should lose it. + channel_->SetRawAudioSink(kSsrc0, nullptr); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink()); + + // Try setting the default sink while two streams exists. + channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_3)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + + // Try setting the sink for the first unsignaled stream using its known SSRC. + channel_->SetRawAudioSink(kSsrc1, std::move(fake_sink_4)); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink()); + } + EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink()); + if (kMaxUnsignaledRecvStreams > 1) { + EXPECT_NE(GetRecvStream(kSsrc1).sink(), GetRecvStream(kSsrcX).sink()); + } +} + +// Test that, just like the video channel, the voice channel communicates the +// network state to the call. +TEST_F(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) { + EXPECT_TRUE(SetupChannel()); + + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::AUDIO)); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::VIDEO)); + + channel_->OnReadyToSend(false); + EXPECT_EQ(webrtc::kNetworkDown, + call_.GetNetworkState(webrtc::MediaType::AUDIO)); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::VIDEO)); + + channel_->OnReadyToSend(true); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::AUDIO)); + EXPECT_EQ(webrtc::kNetworkUp, + call_.GetNetworkState(webrtc::MediaType::VIDEO)); +} + +// Test that playout is still started after changing parameters +TEST_F(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) { + SetupRecvStream(); + channel_->SetPlayout(true); + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); + + // Changing RTP header extensions will recreate the AudioReceiveStream. + cricket::AudioRecvParameters parameters; + parameters.extensions.push_back( + webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); + channel_->SetRecvParameters(parameters); + + EXPECT_TRUE(GetRecvStream(kSsrcX).started()); +} + +// Tests when GetSources is called with non-existing ssrc, it will return an +// empty list of RtpSource without crashing. +TEST_F(WebRtcVoiceEngineTestFake, GetSourcesWithNonExistingSsrc) { + // Setup an recv stream with |kSsrcX|. + SetupRecvStream(); + cricket::WebRtcVoiceMediaChannel* media_channel = + static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); + // 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) { + // If the VoiceEngine wants to gather available codecs early, that's fine but + // we never want it to create a decoder at this stage. + testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + webrtc::AudioProcessing::Create(); + cricket::WebRtcVoiceEngine engine( + &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); + engine.Init(); + webrtc::RtcEventLogNullImpl event_log; + std::unique_ptr<webrtc::Call> call( + webrtc::Call::Create(webrtc::Call::Config(&event_log))); + cricket::VoiceMediaChannel* channel = engine.CreateChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions()); + EXPECT_TRUE(channel != nullptr); + delete channel; +} + +// Tests that reference counting on the external ADM is correct. +TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) { + testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; + EXPECT_CALL(adm, AddRef()).Times(3); + EXPECT_CALL(adm, Release()) + .Times(3) + .WillRepeatedly(Return(rtc::RefCountReleaseStatus::kDroppedLastRef)); + { + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + webrtc::AudioProcessing::Create(); + cricket::WebRtcVoiceEngine engine( + &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); + engine.Init(); + webrtc::RtcEventLogNullImpl event_log; + std::unique_ptr<webrtc::Call> call( + webrtc::Call::Create(webrtc::Call::Config(&event_log))); + cricket::VoiceMediaChannel* channel = engine.CreateChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions()); + EXPECT_TRUE(channel != nullptr); + delete channel; + } +} + +// Verify the payload id of common audio codecs, including CN, ISAC, and G722. +TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) { + // TODO(ossu): Why are the payload types of codecs with non-static payload + // type assignments checked here? It shouldn't really matter. + testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + webrtc::AudioProcessing::Create(); + cricket::WebRtcVoiceEngine engine( + &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); + engine.Init(); + for (const cricket::AudioCodec& codec : engine.send_codecs()) { + auto is_codec = [&codec](const char* name, int clockrate = 0) { + return STR_CASE_CMP(codec.name.c_str(), name) == 0 && + (clockrate == 0 || codec.clockrate == clockrate); + }; + if (is_codec("CN", 16000)) { + EXPECT_EQ(105, codec.id); + } else if (is_codec("CN", 32000)) { + EXPECT_EQ(106, codec.id); + } else if (is_codec("ISAC", 16000)) { + EXPECT_EQ(103, codec.id); + } else if (is_codec("ISAC", 32000)) { + EXPECT_EQ(104, codec.id); + } else if (is_codec("G722", 8000)) { + EXPECT_EQ(9, codec.id); + } else if (is_codec("telephone-event", 8000)) { + EXPECT_EQ(126, codec.id); + // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned. + // Remove these checks once both send and receive side assigns payload types + // dynamically. + } else if (is_codec("telephone-event", 16000)) { + EXPECT_EQ(113, codec.id); + } else if (is_codec("telephone-event", 32000)) { + EXPECT_EQ(112, codec.id); + } else if (is_codec("telephone-event", 48000)) { + EXPECT_EQ(110, codec.id); + } else if (is_codec("opus")) { + EXPECT_EQ(111, codec.id); + ASSERT_TRUE(codec.params.find("minptime") != codec.params.end()); + EXPECT_EQ("10", codec.params.find("minptime")->second); + ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end()); + EXPECT_EQ("1", codec.params.find("useinbandfec")->second); + } + } +} + +// Tests that VoE supports at least 32 channels +TEST(WebRtcVoiceEngineTest, Has32Channels) { + testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + webrtc::AudioProcessing::Create(); + cricket::WebRtcVoiceEngine engine( + &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm); + engine.Init(); + webrtc::RtcEventLogNullImpl event_log; + std::unique_ptr<webrtc::Call> call( + webrtc::Call::Create(webrtc::Call::Config(&event_log))); + + cricket::VoiceMediaChannel* channels[32]; + int num_channels = 0; + while (num_channels < arraysize(channels)) { + cricket::VoiceMediaChannel* channel = engine.CreateChannel( + call.get(), cricket::MediaConfig(), cricket::AudioOptions()); + if (!channel) + break; + channels[num_channels++] = channel; + } + + int expected = arraysize(channels); + EXPECT_EQ(expected, num_channels); + + while (num_channels > 0) { + delete channels[--num_channels]; + } +} + +// Test that we set our preferred codecs properly. +TEST(WebRtcVoiceEngineTest, SetRecvCodecs) { + // TODO(ossu): I'm not sure of the intent of this test. It's either: + // - Check that our builtin codecs are usable by Channel. + // - The codecs provided by the engine is usable by Channel. + // It does not check that the codecs in the RecvParameters are actually + // what we sent in - though it's probably reasonable to expect so, if + // SetRecvParameters returns true. + // I think it will become clear once audio decoder injection is completed. + testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + webrtc::AudioProcessing::Create(); + cricket::WebRtcVoiceEngine engine( + &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(), + webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm); + engine.Init(); + webrtc::RtcEventLogNullImpl event_log; + std::unique_ptr<webrtc::Call> call( + webrtc::Call::Create(webrtc::Call::Config(&event_log))); + cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(), + cricket::AudioOptions(), call.get()); + cricket::AudioRecvParameters parameters; + parameters.codecs = engine.recv_codecs(); + EXPECT_TRUE(channel.SetRecvParameters(parameters)); +} + +TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) { + std::vector<webrtc::AudioCodecSpec> specs; + webrtc::AudioCodecSpec spec1{{"codec1", 48000, 2, {{"param1", "value1"}}}, + {48000, 2, 16000, 10000, 20000}}; + spec1.info.allow_comfort_noise = false; + spec1.info.supports_network_adaption = true; + specs.push_back(spec1); + webrtc::AudioCodecSpec spec2{{"codec2", 32000, 1}, {32000, 1, 32000}}; + spec2.info.allow_comfort_noise = false; + specs.push_back(spec2); + specs.push_back(webrtc::AudioCodecSpec{ + {"codec3", 16000, 1, {{"param1", "value1b"}, {"param2", "value2"}}}, + {16000, 1, 13300}}); + specs.push_back( + webrtc::AudioCodecSpec{{"codec4", 8000, 1}, {8000, 1, 64000}}); + specs.push_back( + webrtc::AudioCodecSpec{{"codec5", 8000, 2}, {8000, 1, 64000}}); + + rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> unused_encoder_factory = + webrtc::MockAudioEncoderFactory::CreateUnusedFactory(); + rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory = + new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>; + EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders()) + .WillOnce(Return(specs)); + testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm; + + rtc::scoped_refptr<webrtc::AudioProcessing> apm = + webrtc::AudioProcessing::Create(); + cricket::WebRtcVoiceEngine engine(&adm, unused_encoder_factory, + mock_decoder_factory, nullptr, apm); + engine.Init(); + auto codecs = engine.recv_codecs(); + EXPECT_EQ(11, codecs.size()); + + // Rather than just ASSERTing that there are enough codecs, ensure that we can + // check the actual values safely, to provide better test results. + auto get_codec = + [&codecs](size_t index) -> const cricket::AudioCodec& { + static const cricket::AudioCodec missing_codec(0, "<missing>", 0, 0, 0); + if (codecs.size() > index) + return codecs[index]; + return missing_codec; + }; + + // Ensure the general codecs are generated first and in order. + for (size_t i = 0; i != specs.size(); ++i) { + EXPECT_EQ(specs[i].format.name, get_codec(i).name); + EXPECT_EQ(specs[i].format.clockrate_hz, get_codec(i).clockrate); + EXPECT_EQ(specs[i].format.num_channels, get_codec(i).channels); + EXPECT_EQ(specs[i].format.parameters, get_codec(i).params); + } + + // Find the index of a codec, or -1 if not found, so that we can easily check + // supplementary codecs are ordered after the general codecs. + auto find_codec = + [&codecs](const webrtc::SdpAudioFormat& format) -> int { + for (size_t i = 0; i != codecs.size(); ++i) { + const cricket::AudioCodec& codec = codecs[i]; + if (STR_CASE_CMP(codec.name.c_str(), format.name.c_str()) == 0 && + codec.clockrate == format.clockrate_hz && + codec.channels == format.num_channels) { + return rtc::checked_cast<int>(i); + } + } + return -1; + }; + + // Ensure all supplementary codecs are generated last. Their internal ordering + // is not important. + // Without this cast, the comparison turned unsigned and, thus, failed for -1. + const int num_specs = static_cast<int>(specs.size()); + EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs); + EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs); + EXPECT_EQ(find_codec({"cn", 32000, 1}), -1); + EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs); + EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs); + EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs); + EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs); +} |