| /* |
| * Copyright (c) 2008 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <memory> |
| |
| #include "webrtc/pc/channel.h" |
| #include "webrtc/base/arraysize.h" |
| #include "webrtc/base/byteorder.h" |
| #include "webrtc/base/gunit.h" |
| #include "webrtc/call.h" |
| #include "webrtc/p2p/base/faketransportcontroller.h" |
| #include "webrtc/test/field_trial.h" |
| #include "webrtc/media/base/fakemediaengine.h" |
| #include "webrtc/media/base/fakenetworkinterface.h" |
| #include "webrtc/media/base/fakertp.h" |
| #include "webrtc/media/base/mediaconstants.h" |
| #include "webrtc/media/engine/fakewebrtccall.h" |
| #include "webrtc/media/engine/fakewebrtcvoiceengine.h" |
| #include "webrtc/media/engine/webrtcvoiceengine.h" |
| #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" |
| #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" |
| #include "webrtc/modules/audio_device/include/mock_audio_device.h" |
| |
| using testing::Return; |
| using testing::StrictMock; |
| |
| namespace { |
| |
| 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, 64000, 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 kTelephoneEventCodec(106, |
| "telephone-event", |
| 8000, |
| 0, |
| 1); |
| const uint32_t kSsrc1 = 0x99; |
| const uint32_t kSsrc2 = 2; |
| const uint32_t kSsrc3 = 3; |
| const uint32_t kSsrcs4[] = { 1, 2, 3, 4 }; |
| |
| constexpr int kRtpHistoryMs = 5000; |
| |
| class FakeVoEWrapper : public cricket::VoEWrapper { |
| public: |
| explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine) |
| : cricket::VoEWrapper(engine, // processing |
| engine, // base |
| engine, // codec |
| engine, // hw |
| engine) { // volume |
| } |
| }; |
| } // namespace |
| |
| // Tests that our stub library "works". |
| TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) { |
| StrictMock<webrtc::test::MockAudioDeviceModule> adm; |
| EXPECT_CALL(adm, AddRef()).WillOnce(Return(0)); |
| EXPECT_CALL(adm, Release()).WillOnce(Return(0)); |
| EXPECT_CALL(adm, BuiltInAECIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(adm, BuiltInAGCIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(adm, BuiltInNSIsAvailable()).WillOnce(Return(false)); |
| cricket::FakeWebRtcVoiceEngine voe; |
| EXPECT_FALSE(voe.IsInited()); |
| { |
| cricket::WebRtcVoiceEngine engine( |
| &adm, webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), |
| new FakeVoEWrapper(&voe)); |
| 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) |
| : call_(webrtc::Call::Config()), override_field_trials_(field_trials) { |
| auto factory = webrtc::MockAudioDecoderFactory::CreateUnusedFactory(); |
| EXPECT_CALL(adm_, AddRef()).WillOnce(Return(0)); |
| EXPECT_CALL(adm_, Release()).WillOnce(Return(0)); |
| EXPECT_CALL(adm_, BuiltInAECIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(adm_, BuiltInAGCIsAvailable()).WillOnce(Return(false)); |
| EXPECT_CALL(adm_, BuiltInNSIsAvailable()).WillOnce(Return(false)); |
| engine_.reset(new cricket::WebRtcVoiceEngine(&adm_, factory, |
| new FakeVoEWrapper(&voe_))); |
| send_parameters_.codecs.push_back(kPcmuCodec); |
| recv_parameters_.codecs.push_back(kPcmuCodec); |
| } |
| |
| bool SetupChannel() { |
| channel_ = engine_->CreateChannel(&call_, cricket::MediaConfig(), |
| cricket::AudioOptions()); |
| return (channel_ != nullptr); |
| } |
| |
| bool SetupRecvStream() { |
| if (!SetupChannel()) { |
| return false; |
| } |
| return AddRecvStream(kSsrc1); |
| } |
| |
| bool SetupSendStream() { |
| if (!SetupChannel()) { |
| return false; |
| } |
| if (!channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc1))) { |
| return false; |
| } |
| return channel_->SetAudioSend(kSsrc1, 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(kSsrc1)); |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrc1)); |
| // Verify the channel does not exist. |
| EXPECT_FALSE(call_.GetAudioSendStream(kSsrc1)); |
| } |
| |
| 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(cricket::VoiceMediaChannel* channel, bool enable) { |
| ASSERT_TRUE(channel); |
| if (enable) { |
| EXPECT_CALL(adm_, RecordingIsInitialized()).WillOnce(Return(false)); |
| EXPECT_CALL(adm_, Recording()).WillOnce(Return(false)); |
| EXPECT_CALL(adm_, InitRecording()).WillOnce(Return(0)); |
| } |
| channel->SetSend(enable); |
| } |
| |
| void TestInsertDtmf(uint32_t ssrc, bool caller) { |
| 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(kSsrc1))); |
| } |
| |
| // Test we can only InsertDtmf when the other side supports telephone-event. |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| SetSend(channel_, true); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111)); |
| send_parameters_.codecs.push_back(kTelephoneEventCodec); |
| EXPECT_TRUE(channel_->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(kSsrc1))); |
| } |
| |
| // Check we fail if the ssrc is invalid. |
| EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111)); |
| |
| // Test send. |
| cricket::FakeAudioSendStream::TelephoneEvent telephone_event = |
| GetSendStream(kSsrc1).GetLatestTelephoneEvent(); |
| EXPECT_EQ(-1, telephone_event.payload_type); |
| EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123)); |
| telephone_event = GetSendStream(kSsrc1).GetLatestTelephoneEvent(); |
| EXPECT_EQ(kTelephoneEventCodec.id, telephone_event.payload_type); |
| 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; |
| EXPECT_EQ(expected_result, channel_->SetSendParameters(parameters)); |
| |
| int channel_num = voe_.GetLastChannel(); |
| webrtc::CodecInst temp_codec; |
| EXPECT_FALSE(voe_.GetSendCodec(channel_num, temp_codec)); |
| EXPECT_EQ(expected_bitrate, temp_codec.rate); |
| } |
| |
| // 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); |
| } |
| |
| bool SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) { |
| cricket::AudioSendParameters send_parameters; |
| send_parameters.codecs.push_back(codec); |
| send_parameters.max_bandwidth_bps = bitrate; |
| return channel_->SetSendParameters(send_parameters); |
| } |
| |
| int GetCodecBitrate(int32_t ssrc) { |
| cricket::WebRtcVoiceMediaChannel* media_channel = |
| static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_); |
| int channel = media_channel->GetSendChannelId(ssrc); |
| EXPECT_NE(-1, channel); |
| webrtc::CodecInst codec; |
| EXPECT_FALSE(voe_.GetSendCodec(channel, codec)); |
| return codec.rate; |
| } |
| |
| 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(kSsrc1, -1)); |
| |
| // Attempt to set the requested bitrate limits. |
| EXPECT_TRUE(SetGlobalMaxBitrate(codec, global_max)); |
| EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrc1, stream_max)); |
| |
| // Verify that reading back the parameters gives results |
| // consistent with the Set() result. |
| webrtc::RtpParameters resulting_parameters = |
| channel_->GetRtpSendParameters(kSsrc1); |
| 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(kSsrc1)); |
| } |
| |
| void TestSetSendRtpHeaderExtensions(const std::string& ext) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Ensure extensions are off by default. |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size()); |
| |
| // Ensure unknown extensions won't cause an error. |
| send_parameters_.extensions.push_back( |
| webrtc::RtpExtension("urn:ietf:params:unknownextention", 1)); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size()); |
| |
| // Ensure extensions stay off with an empty list of headers. |
| send_parameters_.extensions.clear(); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size()); |
| |
| // Ensure extension is set properly. |
| const int id = 1; |
| send_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id)); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(1u, GetSendStreamConfig(kSsrc1).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetSendStreamConfig(kSsrc1).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetSendStreamConfig(kSsrc1).rtp.extensions[0].id); |
| |
| // Ensure extension is set properly on new stream. |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrc2))); |
| EXPECT_NE(call_.GetAudioSendStream(kSsrc1), |
| call_.GetAudioSendStream(kSsrc2)); |
| EXPECT_EQ(1u, GetSendStreamConfig(kSsrc2).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetSendStreamConfig(kSsrc2).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetSendStreamConfig(kSsrc2).rtp.extensions[0].id); |
| |
| // Ensure all extensions go back off with an empty list. |
| send_parameters_.codecs.push_back(kPcmuCodec); |
| send_parameters_.extensions.clear(); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size()); |
| EXPECT_EQ(0u, GetSendStreamConfig(kSsrc2).rtp.extensions.size()); |
| } |
| |
| void TestSetRecvRtpHeaderExtensions(const std::string& ext) { |
| EXPECT_TRUE(SetupRecvStream()); |
| |
| // Ensure extensions are off by default. |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc1).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(kSsrc1).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(kSsrc1).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(kSsrc1).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetRecvStreamConfig(kSsrc1).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetRecvStreamConfig(kSsrc1).rtp.extensions[0].id); |
| |
| // Ensure extension is set properly on new stream. |
| EXPECT_TRUE(AddRecvStream(kSsrc2)); |
| EXPECT_NE(call_.GetAudioReceiveStream(kSsrc1), |
| call_.GetAudioReceiveStream(kSsrc2)); |
| EXPECT_EQ(1u, GetRecvStreamConfig(kSsrc2).rtp.extensions.size()); |
| EXPECT_EQ(ext, GetRecvStreamConfig(kSsrc2).rtp.extensions[0].uri); |
| EXPECT_EQ(id, GetRecvStreamConfig(kSsrc2).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(kSsrc1).rtp.extensions.size()); |
| EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc2).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.ext_seqnum = 789; |
| stats.jitter_ms = 12; |
| stats.rtt_ms = 345; |
| stats.audio_level = 678; |
| stats.aec_quality_min = 9.01f; |
| stats.echo_delay_median_ms = 234; |
| stats.echo_delay_std_ms = 567; |
| stats.echo_return_loss = 890; |
| stats.echo_return_loss_enhancement = 1234; |
| stats.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.ext_seqnum, stats.ext_seqnum); |
| EXPECT_EQ(info.jitter_ms, stats.jitter_ms); |
| EXPECT_EQ(info.rtt_ms, stats.rtt_ms); |
| EXPECT_EQ(info.audio_level, stats.audio_level); |
| EXPECT_EQ(info.aec_quality_min, stats.aec_quality_min); |
| EXPECT_EQ(info.echo_delay_median_ms, stats.echo_delay_median_ms); |
| EXPECT_EQ(info.echo_delay_std_ms, stats.echo_delay_std_ms); |
| EXPECT_EQ(info.echo_return_loss, stats.echo_return_loss); |
| EXPECT_EQ(info.echo_return_loss_enhancement, |
| stats.echo_return_loss_enhancement); |
| EXPECT_EQ(info.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.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.expand_rate = 5.67f; |
| stats.speech_expand_rate = 8.90f; |
| stats.secondary_decoded_rate = 1.23f; |
| 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.capture_start_ntp_time_ms = 3456; |
| 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.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.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.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.capture_start_ntp_time_ms, stats.capture_start_ntp_time_ms); |
| } |
| |
| protected: |
| StrictMock<webrtc::test::MockAudioDeviceModule> adm_; |
| 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_; |
| 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(kSsrc1))); |
| const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrc1); |
| EXPECT_EQ(kSsrc1, 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(kSsrc1)); |
| const webrtc::AudioReceiveStream::Config& config = |
| GetRecvStreamConfig(kSsrc1); |
| EXPECT_EQ(kSsrc1, 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); |
| } |
| |
| // Tests that the list of supported codecs is created properly and ordered |
| // correctly (such that opus appears first). |
| // TODO(ossu): This test should move into a separate builtin audio codecs |
| // module. |
| TEST_F(WebRtcVoiceEngineTestFake, CodecOrder) { |
| const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs(); |
| ASSERT_FALSE(codecs.empty()); |
| EXPECT_STRCASEEQ("opus", codecs[0].name.c_str()); |
| EXPECT_EQ(48000, codecs[0].clockrate); |
| EXPECT_EQ(2, codecs[0].channels); |
| EXPECT_EQ(64000, codecs[0].bitrate); |
| } |
| |
| 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); |
| } |
| |
| // Tests that we can find codecs by name or id, and that we interpret the |
| // clockrate and bitrate fields properly. |
| TEST_F(WebRtcVoiceEngineTestFake, FindCodec) { |
| cricket::AudioCodec codec; |
| webrtc::CodecInst codec_inst; |
| // Find PCMU with explicit clockrate and bitrate. |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kPcmuCodec, &codec_inst)); |
| // Find ISAC with explicit clockrate and 0 bitrate. |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kIsacCodec, &codec_inst)); |
| // Find telephone-event with explicit clockrate and 0 bitrate. |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec, |
| &codec_inst)); |
| // Find ISAC with a different payload id. |
| codec = kIsacCodec; |
| codec.id = 127; |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst)); |
| EXPECT_EQ(codec.id, codec_inst.pltype); |
| // Find PCMU with a 0 clockrate. |
| codec = kPcmuCodec; |
| codec.clockrate = 0; |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst)); |
| EXPECT_EQ(codec.id, codec_inst.pltype); |
| EXPECT_EQ(8000, codec_inst.plfreq); |
| // Find PCMU with a 0 bitrate. |
| codec = kPcmuCodec; |
| codec.bitrate = 0; |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst)); |
| EXPECT_EQ(codec.id, codec_inst.pltype); |
| EXPECT_EQ(64000, codec_inst.rate); |
| // Find ISAC with an explicit bitrate. |
| codec = kIsacCodec; |
| codec.bitrate = 32000; |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst)); |
| EXPECT_EQ(codec.id, codec_inst.pltype); |
| EXPECT_EQ(32000, codec_inst.rate); |
| } |
| |
| // 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(kTelephoneEventCodec); |
| parameters.codecs[0].id = 106; // collide with existing telephone-event |
| parameters.codecs[2].id = 126; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrc1)); |
| int channel_num = voe_.GetLastChannel(); |
| webrtc::CodecInst gcodec; |
| rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC"); |
| gcodec.plfreq = 16000; |
| gcodec.channels = 1; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec)); |
| EXPECT_EQ(106, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event"); |
| gcodec.plfreq = 8000; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec)); |
| EXPECT_EQ(126, gcodec.pltype); |
| EXPECT_STREQ("telephone-event", gcodec.plname); |
| } |
| |
| // 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(kSsrc1)); |
| int channel_num = voe_.GetLastChannel(); |
| webrtc::CodecInst opus; |
| cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus); |
| // Even without stereo parameters, recv codecs still specify channels = 2. |
| EXPECT_EQ(2, opus.channels); |
| EXPECT_EQ(111, opus.pltype); |
| EXPECT_STREQ("opus", opus.plname); |
| opus.pltype = 0; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, opus)); |
| EXPECT_EQ(111, opus.pltype); |
| } |
| |
| // 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(kSsrc1)); |
| int channel_num2 = voe_.GetLastChannel(); |
| webrtc::CodecInst opus; |
| cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus); |
| // Even when stereo is off, recv codecs still specify channels = 2. |
| EXPECT_EQ(2, opus.channels); |
| EXPECT_EQ(111, opus.pltype); |
| EXPECT_STREQ("opus", opus.plname); |
| opus.pltype = 0; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus)); |
| EXPECT_EQ(111, opus.pltype); |
| } |
| |
| // 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(kSsrc1)); |
| int channel_num2 = voe_.GetLastChannel(); |
| webrtc::CodecInst opus; |
| cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus); |
| EXPECT_EQ(2, opus.channels); |
| EXPECT_EQ(111, opus.pltype); |
| EXPECT_STREQ("opus", opus.plname); |
| opus.pltype = 0; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus)); |
| EXPECT_EQ(111, opus.pltype); |
| } |
| |
| // 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(kTelephoneEventCodec); |
| parameters.codecs[0].id = 106; // collide with existing telephone-event |
| parameters.codecs[2].id = 126; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrc1)); |
| int channel_num2 = voe_.GetLastChannel(); |
| webrtc::CodecInst gcodec; |
| rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC"); |
| gcodec.plfreq = 16000; |
| gcodec.channels = 1; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec)); |
| EXPECT_EQ(106, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event"); |
| gcodec.plfreq = 8000; |
| gcodec.channels = 1; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec)); |
| EXPECT_EQ(126, gcodec.pltype); |
| EXPECT_STREQ("telephone-event", gcodec.plname); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { |
| EXPECT_TRUE(SetupRecvStream()); |
| cricket::AudioRecvParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].id = 106; // collide with existing telephone-event |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| int channel_num2 = voe_.GetLastChannel(); |
| webrtc::CodecInst gcodec; |
| rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC"); |
| gcodec.plfreq = 16000; |
| gcodec.channels = 1; |
| EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec)); |
| EXPECT_EQ(106, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| } |
| |
| // 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)); |
| |
| // Changing the payload type of a codec should fail. |
| parameters.codecs[0].id = 127; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| EXPECT_TRUE(GetRecvStream(kSsrc1).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(kSsrc1).started()); |
| webrtc::CodecInst gcodec; |
| EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &gcodec)); |
| EXPECT_EQ(kOpusCodec.id, gcodec.pltype); |
| } |
| |
| 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 == 64000. |
| TestMaxSendBandwidth(kOpusCodec, -1, true, 64000); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // Test that the bitrate of a multi-rate codec is always the maximum. |
| |
| // ISAC, default bitrate == 32000. |
| TestMaxSendBandwidth(kIsacCodec, 40000, true, 40000); |
| TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000); |
| // Rates above the max (56000) should be capped. |
| TestMaxSendBandwidth(kIsacCodec, 100000, true, 56000); |
| |
| // 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; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrc1))); |
| |
| int channel_num = voe_.GetLastChannel(); |
| webrtc::CodecInst codec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec)); |
| EXPECT_EQ(kDesiredBitrate, codec.rate); |
| } |
| |
| // 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. |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| int channel_num = voe_.GetLastChannel(); |
| webrtc::CodecInst codec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec)); |
| EXPECT_EQ(64000, codec.rate); |
| |
| send_parameters_.max_bandwidth_bps = 128000; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec)); |
| EXPECT_EQ(64000, codec.rate); |
| |
| send_parameters_.max_bandwidth_bps = 128; |
| EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec)); |
| EXPECT_EQ(64000, codec.rate); |
| } |
| |
| // Test that the per-stream bitrate limit and the global |
| // bitrate limit both apply. |
| TEST_F(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) { |
| EXPECT_TRUE(SetupSendStream()); |
| |
| // opus, default bitrate == 64000. |
| SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 64000); |
| 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(kSsrc1); |
| EXPECT_EQ(0, nonexistent_parameters.encodings.size()); |
| |
| nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, 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()); |
| // Setting RtpParameters with no encoding is expected to fail. |
| webrtc::RtpParameters parameters; |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters)); |
| // Setting RtpParameters with exactly one encoding should succeed. |
| parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters)); |
| // Two or more encodings should result in failure. |
| parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, 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(channel_, true); |
| EXPECT_TRUE(GetSendStream(kSsrc1).IsSending()); |
| // Get current parameters and change "active" to false. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrc1); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_TRUE(parameters.encodings[0].active); |
| parameters.encodings[0].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters)); |
| EXPECT_FALSE(GetSendStream(kSsrc1).IsSending()); |
| |
| // Now change it back to active and verify we resume sending. |
| parameters.encodings[0].active = true; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters)); |
| EXPECT_TRUE(GetSendStream(kSsrc1).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). |
| EXPECT_TRUE(SetGlobalMaxBitrate(kOpusCodec, 64000)); |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 48000)); |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 96000)); |
| EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1)); |
| |
| EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0])); |
| EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[1])); |
| EXPECT_EQ(64000, 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.) |
| EXPECT_TRUE(SetGlobalMaxBitrate(kOpusCodec, -1)); |
| EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0])); |
| EXPECT_EQ(96000, GetCodecBitrate(kSsrcs4[1])); |
| EXPECT_EQ(64000, 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); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc1); |
| 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 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); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrc1); |
| |
| // We should be able to set the params we just got. |
| EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, initial_params)); |
| |
| // ... And this shouldn't change the params returned by GetRtpSendParameters. |
| webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrc1); |
| EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrc1)); |
| } |
| |
| // 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(kSsrc1); |
| 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 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(kSsrc1); |
| |
| // We should be able to set the params we just got. |
| EXPECT_TRUE(channel_->SetRtpReceiveParameters(kSsrc1, initial_params)); |
| |
| // ... And this shouldn't change the params returned by |
| // GetRtpReceiveParameters. |
| webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrc1); |
| EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(kSsrc1)); |
| } |
| |
| // 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 = 48000; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(1, voe_.GetNumSetSendCodecs()); |
| int channel_num = voe_.GetLastChannel(); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(96, gcodec.pltype); |
| EXPECT_EQ(48000, gcodec.rate); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_FALSE(voe_.GetVAD(channel_num)); |
| EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false)); |
| EXPECT_EQ(105, voe_.GetSendCNPayloadType(channel_num, true)); |
| EXPECT_FALSE(channel_->CanInsertDtmf()); |
| } |
| |
| // Test that VoE Channel doesn't call SetSendCodec again if same codec is tried |
| // to apply. |
| TEST_F(WebRtcVoiceEngineTestFake, DontResetSetSendCodec) { |
| 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; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(1, voe_.GetNumSetSendCodecs()); |
| // Calling SetSendCodec again with same codec which is already set. |
| // In this case media channel shouldn't send codec to VoE. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(1, voe_.GetNumSetSendCodecs()); |
| } |
| |
| // Verify that G722 is set with 16000 samples per second to WebRTC. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecG722) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kG722CodecSdp); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("G722", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(16000, gcodec.plfreq); |
| } |
| |
| // 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, |
| // channels and bitrate are 1 and 32000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(32000, gcodec.rate); |
| } |
| |
| // Test that with bitrate=0 and stereo=0, |
| // channels and bitrate are 1 and 32000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "0"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(32000, gcodec.rate); |
| } |
| |
| // Test that with bitrate=invalid and stereo=0, |
| // channels and bitrate are 1 and 32000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["stereo"] = "0"; |
| webrtc::CodecInst gcodec; |
| |
| // bitrate that's out of the range between 6000 and 510000 will be clamped. |
| parameters.codecs[0].bitrate = 5999; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(6000, gcodec.rate); |
| |
| parameters.codecs[0].bitrate = 510001; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(510000, gcodec.rate); |
| } |
| |
| // Test that with bitrate=0 and stereo=1, |
| // channels and bitrate are 2 and 64000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(2, gcodec.channels); |
| EXPECT_EQ(64000, gcodec.rate); |
| } |
| |
| // Test that with bitrate=invalid and stereo=1, |
| // channels and bitrate are 2 and 64000. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["stereo"] = "1"; |
| webrtc::CodecInst gcodec; |
| |
| // bitrate that's out of the range between 6000 and 510000 will be clamped. |
| parameters.codecs[0].bitrate = 5999; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(2, gcodec.channels); |
| EXPECT_EQ(6000, gcodec.rate); |
| |
| parameters.codecs[0].bitrate = 510001; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(2, gcodec.channels); |
| EXPECT_EQ(510000, gcodec.rate); |
| } |
| |
| // Test that with bitrate=N and stereo unset, |
| // channels and bitrate are 1 and N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 96000; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(111, gcodec.pltype); |
| EXPECT_EQ(96000, gcodec.rate); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(48000, gcodec.plfreq); |
| } |
| |
| // Test that with bitrate=N and stereo=0, |
| // channels and bitrate are 1 and N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "0"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(30000, gcodec.rate); |
| EXPECT_STREQ("opus", gcodec.plname); |
| } |
| |
| // Test that with bitrate=N and without any parameters, |
| // channels and bitrate are 1 and N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(30000, gcodec.rate); |
| EXPECT_STREQ("opus", gcodec.plname); |
| } |
| |
| // Test that with bitrate=N and stereo=1, |
| // channels and bitrate are 2 and N. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(2, gcodec.channels); |
| EXPECT_EQ(30000, gcodec.rate); |
| EXPECT_STREQ("opus", gcodec.plname); |
| } |
| |
| // Test that bitrate will be overridden by the "maxaveragebitrate" parameter. |
| // Also test that the "maxaveragebitrate" can't be set to values outside the |
| // range of 6000 and 510000 |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusMaxAverageBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 30000; |
| webrtc::CodecInst gcodec; |
| |
| // Ignore if less than 6000. |
| parameters.codecs[0].params["maxaveragebitrate"] = "5999"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(6000, gcodec.rate); |
| |
| // Ignore if larger than 510000. |
| parameters.codecs[0].params["maxaveragebitrate"] = "510001"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(510000, gcodec.rate); |
| |
| parameters.codecs[0].params["maxaveragebitrate"] = "200000"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(200000, gcodec.rate); |
| } |
| |
| // 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(kSsrc1).rtp.nack.rtp_history_ms); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrc1).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(kSsrc1).rtp.nack.rtp_history_ms); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| // NACK should be enabled even with no send stream. |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrc1))); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that we can enable NACK on receive streams. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrc2)); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| EXPECT_EQ(0, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrc2).rtp.nack.rtp_history_ms); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrc2).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)); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| |
| parameters.codecs.clear(); |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that we can disable NACK on receive streams. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(AddRecvStream(kSsrc2)); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].AddFeedbackParam( |
| cricket::FeedbackParam(cricket::kRtcpFbParamNack, |
| cricket::kParamValueEmpty)); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrc2).rtp.nack.rtp_history_ms); |
| |
| parameters.codecs.clear(); |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| EXPECT_EQ(0, GetRecvStreamConfig(kSsrc2).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)); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrc1).rtp.nack.rtp_history_ms); |
| |
| EXPECT_TRUE(AddRecvStream(kSsrc2)); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrc2).rtp.nack.rtp_history_ms); |
| EXPECT_TRUE(AddRecvStream(kSsrc3)); |
| EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrc3).rtp.nack.rtp_history_ms); |
| } |
| |
| // Test that without useinbandfec, Opus FEC is off. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFec) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(voe_.GetCodecFEC(channel_num)); |
| } |
| |
| // Test that with useinbandfec=0, Opus FEC is off. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusDisableFec) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["useinbandfec"] = "0"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(voe_.GetCodecFEC(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(32000, gcodec.rate); |
| } |
| |
| // Test that with useinbandfec=1, Opus FEC is on. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFec) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["useinbandfec"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(voe_.GetCodecFEC(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(1, gcodec.channels); |
| EXPECT_EQ(32000, gcodec.rate); |
| } |
| |
| // Test that with useinbandfec=1, stereo=1, Opus FEC is on. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFecStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].params["stereo"] = "1"; |
| parameters.codecs[0].params["useinbandfec"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(voe_.GetCodecFEC(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(2, gcodec.channels); |
| EXPECT_EQ(64000, gcodec.rate); |
| } |
| |
| // Test that with non-Opus, codec FEC is off. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacNoFec) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(voe_.GetCodecFEC(channel_num)); |
| } |
| |
| // Test the with non-Opus, even if useinbandfec=1, FEC is off. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacWithParamNoFec) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].params["useinbandfec"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(voe_.GetCodecFEC(channel_num)); |
| } |
| |
| // Test that Opus FEC status can be changed. |
| TEST_F(WebRtcVoiceEngineTestFake, ChangeOpusFecStatus) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(voe_.GetCodecFEC(channel_num)); |
| parameters.codecs[0].params["useinbandfec"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(voe_.GetCodecFEC(channel_num)); |
| } |
| |
| 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()); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters)); |
| |
| cricket::AudioRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(kIsacCodec); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| EXPECT_TRUE(AddRecvStream(kSsrc1)); |
| ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrc1) != nullptr); |
| EXPECT_FALSE( |
| call_.GetAudioReceiveStream(kSsrc1)->GetConfig().rtp.transport_cc); |
| |
| send_parameters.codecs = engine_->send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters)); |
| ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrc1) != nullptr); |
| EXPECT_TRUE( |
| call_.GetAudioReceiveStream(kSsrc1)->GetConfig().rtp.transport_cc); |
| } |
| |
| // Test maxplaybackrate <= 8000 triggers Opus narrow band mode. |
| TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateNb) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(cricket::kOpusBandwidthNb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| |
| EXPECT_EQ(12000, gcodec.rate); |
| parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1"); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(24000, gcodec.rate); |
| } |
| |
| // Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode. |
| TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateMb) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8001); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(cricket::kOpusBandwidthMb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| |
| EXPECT_EQ(20000, gcodec.rate); |
| parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1"); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(40000, gcodec.rate); |
| } |
| |
| // Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode. |
| TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateWb) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 12001); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(cricket::kOpusBandwidthWb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| |
| EXPECT_EQ(20000, gcodec.rate); |
| parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1"); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(40000, gcodec.rate); |
| } |
| |
| // Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode. |
| TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateSwb) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 16001); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(cricket::kOpusBandwidthSwb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| |
| EXPECT_EQ(32000, gcodec.rate); |
| parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1"); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(64000, gcodec.rate); |
| } |
| |
| // Test 24000 < maxplaybackrate triggers Opus full band mode. |
| TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateFb) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].bitrate = 0; |
| parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 24001); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(cricket::kOpusBandwidthFb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("opus", gcodec.plname); |
| |
| EXPECT_EQ(32000, gcodec.rate); |
| parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1"); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(64000, gcodec.rate); |
| } |
| |
| // Test Opus that without maxplaybackrate, default playback rate is used. |
| TEST_F(WebRtcVoiceEngineTestFake, DefaultOpusMaxPlaybackRate) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(cricket::kOpusBandwidthFb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| } |
| |
| // Test the with non-Opus, maxplaybackrate has no effect. |
| TEST_F(WebRtcVoiceEngineTestFake, SetNonOpusMaxPlaybackRate) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 32000); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetMaxEncodingBandwidth(channel_num)); |
| } |
| |
| // Test maxplaybackrate can be set on two streams. |
| TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| // Default bandwidth is 24000. |
| EXPECT_EQ(cricket::kOpusBandwidthFb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| |
| parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(cricket::kOpusBandwidthNb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc2)); |
| channel_num = voe_.GetLastChannel(); |
| EXPECT_EQ(cricket::kOpusBandwidthNb, |
| voe_.GetMaxEncodingBandwidth(channel_num)); |
| } |
| |
| // Test that with usedtx=0, Opus DTX is off. |
| TEST_F(WebRtcVoiceEngineTestFake, DisableOpusDtxOnOpus) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["usedtx"] = "0"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(voe_.GetOpusDtx(channel_num)); |
| } |
| |
| // Test that with usedtx=1, Opus DTX is on. |
| TEST_F(WebRtcVoiceEngineTestFake, EnableOpusDtxOnOpus) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["usedtx"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(voe_.GetOpusDtx(channel_num)); |
| EXPECT_FALSE(voe_.GetVAD(channel_num)); // Opus DTX should not affect VAD. |
| } |
| |
| // Test that usedtx=1 works with stereo Opus. |
| TEST_F(WebRtcVoiceEngineTestFake, EnableOpusDtxOnOpusStereo) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].params["usedtx"] = "1"; |
| parameters.codecs[0].params["stereo"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(voe_.GetOpusDtx(channel_num)); |
| EXPECT_FALSE(voe_.GetVAD(channel_num)); // Opus DTX should not affect VAD. |
| } |
| |
| // Test that usedtx=1 does not work with non Opus. |
| TEST_F(WebRtcVoiceEngineTestFake, CannotEnableOpusDtxOnNonOpus) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].params["usedtx"] = "1"; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_FALSE(voe_.GetOpusDtx(channel_num)); |
| } |
| |
| // Test that we can switch back and forth between Opus and ISAC with CN. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters opus_parameters; |
| opus_parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(opus_parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(111, gcodec.pltype); |
| EXPECT_STREQ("opus", gcodec.plname); |
| |
| cricket::AudioSendParameters isac_parameters; |
| isac_parameters.codecs.push_back(kIsacCodec); |
| isac_parameters.codecs.push_back(kCn16000Codec); |
| isac_parameters.codecs.push_back(kOpusCodec); |
| EXPECT_TRUE(channel_->SetSendParameters(isac_parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(103, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(opus_parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(111, gcodec.pltype); |
| EXPECT_STREQ("opus", gcodec.plname); |
| } |
| |
| // Test that we handle various ways of specifying bitrate. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kIsacCodec); // bitrate == 32000 |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(103, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_EQ(32000, gcodec.rate); |
| |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(103, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_EQ(-1, gcodec.rate); |
| |
| parameters.codecs[0].bitrate = 28000; // bitrate == 28000 |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(103, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_EQ(28000, gcodec.rate); |
| |
| parameters.codecs[0] = kPcmuCodec; // bitrate == 64000 |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(0, gcodec.pltype); |
| EXPECT_STREQ("PCMU", gcodec.plname); |
| EXPECT_EQ(64000, gcodec.rate); |
| |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(0, gcodec.pltype); |
| EXPECT_STREQ("PCMU", gcodec.plname); |
| EXPECT_EQ(64000, gcodec.rate); |
| |
| parameters.codecs[0] = kOpusCodec; |
| parameters.codecs[0].bitrate = 0; // bitrate == default |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(111, gcodec.pltype); |
| EXPECT_STREQ("opus", gcodec.plname); |
| EXPECT_EQ(32000, gcodec.rate); |
| } |
| |
| // Test that we could set packet size specified in kCodecParamPTime. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsPTimeAsPacketSize) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kOpusCodec); |
| parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 40); // Within range. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(1920, gcodec.pacsize); // Opus gets 40ms. |
| |
| parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 5); // Below range. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(480, gcodec.pacsize); // Opus gets 10ms. |
| |
| parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 80); // Beyond range. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(2880, gcodec.pacsize); // Opus gets 60ms. |
| |
| parameters.codecs[0] = kIsacCodec; // Also try Isac, with unsupported size. |
| parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 40); // Within range. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(480, gcodec.pacsize); // Isac gets 30ms as the next smallest value. |
| |
| parameters.codecs[0] = kG722CodecSdp; // Try G722 @8kHz as negotiated in SDP. |
| parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 40); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(640, gcodec.pacsize); // G722 gets 40ms @16kHz as defined in VoE. |
| } |
| |
| // 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()); |
| int channel_num = voe_.GetLastChannel(); |
| cricket::AudioSendParameters parameters; |
| parameters.codecs.push_back(kTelephoneEventCodec); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs.push_back(kPcmuCodec); |
| parameters.codecs[0].id = 98; // DTMF |
| parameters.codecs[1].id = 96; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(96, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| 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(kTelephoneEventCodec); |
| parameters.codecs.push_back(kIsacCodec); |
| parameters.codecs[0].id = 0; // DTMF |
| parameters.codecs[1].id = 96; |
| EXPECT_TRUE(channel_->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; |
| EXPECT_TRUE(channel_->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()); |
| int channel_num = voe_.GetLastChannel(); |
| 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; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(96, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_EQ(98, voe_.GetSendCNPayloadType(channel_num, true)); |
| } |
| |
| // Test that we set VAD and DTMF types correctly as caller. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| 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(kTelephoneEventCodec); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(96, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_TRUE(voe_.GetVAD(channel_num)); |
| EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false)); |
| EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true)); |
| 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(kTelephoneEventCodec); |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrc1))); |
| int channel_num = voe_.GetLastChannel(); |
| |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(96, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_TRUE(voe_.GetVAD(channel_num)); |
| EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false)); |
| EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true)); |
| 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()); |
| int channel_num = voe_.GetLastChannel(); |
| 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; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_TRUE(voe_.GetVAD(channel_num)); |
| EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true)); |
| // Set PCMU(8K) and CN(16K). VAD should not be activated. |
| parameters.codecs[0] = kPcmuCodec; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("PCMU", gcodec.plname); |
| EXPECT_FALSE(voe_.GetVAD(channel_num)); |
| // Set PCMU(8K) and CN(8K). VAD should be activated. |
| parameters.codecs[1] = kCn8000Codec; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("PCMU", gcodec.plname); |
| EXPECT_TRUE(voe_.GetVAD(channel_num)); |
| EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false)); |
| // Set ISAC(16K) and CN(8K). VAD should not be activated. |
| parameters.codecs[0] = kIsacCodec; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_FALSE(voe_.GetVAD(channel_num)); |
| } |
| |
| // Test that we perform case-insensitive matching of codec names. |
| TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) { |
| EXPECT_TRUE(SetupSendStream()); |
| int channel_num = voe_.GetLastChannel(); |
| 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(kTelephoneEventCodec); |
| parameters.codecs[0].name = "iSaC"; |
| parameters.codecs[0].id = 96; |
| parameters.codecs[2].id = 97; // wideband CN |
| parameters.codecs[4].id = 98; // DTMF |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| webrtc::CodecInst gcodec; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_EQ(96, gcodec.pltype); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_TRUE(voe_.GetVAD(channel_num)); |
| EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false)); |
| EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true)); |
| 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 absolute send time header extension. |
| TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri); |
| } |
| TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri); |
| } |
| |
| // Test that we can create a channel and start sending on it. |
| TEST_F(WebRtcVoiceEngineTestFake, Send) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| SetSend(channel_, true); |
| EXPECT_TRUE(GetSendStream(kSsrc1).IsSending()); |
| SetSend(channel_, false); |
| EXPECT_FALSE(GetSendStream(kSsrc1).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()); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, nullptr)); |
| SetSend(channel_, true); |
| EXPECT_FALSE(GetSendStream(kSsrc1).IsSending()); |
| EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, &fake_source_)); |
| EXPECT_TRUE(GetSendStream(kSsrc1).IsSending()); |
| EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, nullptr)); |
| EXPECT_FALSE(GetSendStream(kSsrc1).IsSending()); |
| } |
| |
| // Test that a channel is muted/unmuted. |
| TEST_F(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_FALSE(GetSendStream(kSsrc1).muted()); |
| EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, nullptr)); |
| EXPECT_FALSE(GetSendStream(kSsrc1).muted()); |
| EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, false, nullptr, nullptr)); |
| EXPECT_TRUE(GetSendStream(kSsrc1).muted()); |
| } |
| |
| // Test that SetSendParameters() does not alter a stream's send state. |
| TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_FALSE(GetSendStream(kSsrc1).IsSending()); |
| |
| // Turn on sending. |
| SetSend(channel_, true); |
| EXPECT_TRUE(GetSendStream(kSsrc1).IsSending()); |
| |
| // Changing RTP header extensions will recreate the AudioSendStream. |
| send_parameters_.extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_TRUE(GetSendStream(kSsrc1).IsSending()); |
| |
| // Turn off sending. |
| SetSend(channel_, false); |
| EXPECT_FALSE(GetSendStream(kSsrc1).IsSending()); |
| |
| // Changing RTP header extensions will recreate the AudioSendStream. |
| send_parameters_.extensions.clear(); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_FALSE(GetSendStream(kSsrc1).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(kSsrc1).started()); |
| channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrc1).started()); |
| } |
| |
| // Test that we can add and remove send streams. |
| TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) { |
| SetupForMultiSendStream(); |
| |
| // Set the global state for sending. |
| SetSend(channel_, true); |
| |
| for (uint32_t ssrc : kSsrcs4) { |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(ssrc))); |
| EXPECT_TRUE(channel_->SetAudioSend(ssrc, true, nullptr, &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; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| // Verify ISAC and VAD are corrected configured on all send channels. |
| webrtc::CodecInst gcodec; |
| for (uint32_t ssrc : kSsrcs4) { |
| int channel_num = GetSendStreamConfig(ssrc).voe_channel_id; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("ISAC", gcodec.plname); |
| EXPECT_TRUE(voe_.GetVAD(channel_num)); |
| EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true)); |
| } |
| |
| // Change to PCMU(8K) and CN(16K). VAD should not be activated. |
| parameters.codecs[0] = kPcmuCodec; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| for (uint32_t ssrc : kSsrcs4) { |
| int channel_num = GetSendStreamConfig(ssrc).voe_channel_id; |
| EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); |
| EXPECT_STREQ("PCMU", gcodec.plname); |
| EXPECT_FALSE(voe_.GetVAD(channel_num)); |
| } |
| } |
| |
| // 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))); |
| EXPECT_TRUE(channel_->SetAudioSend(ssrc, true, nullptr, &fake_source_)); |
| EXPECT_FALSE(GetSendStream(ssrc).IsSending()); |
| } |
| |
| // Set the global state for starting sending. |
| SetSend(channel_, 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(channel_, 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(kSsrc2)); |
| |
| // We need send codec to be set to get all stats. |
| EXPECT_TRUE(channel_->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); |
| } |
| |
| // 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 kSsrc2 stream. No receiver stats. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc2)); |
| 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]); |
| } |
| } |
| |
| // 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. |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| channel_->SetPlayout(true); |
| |
| // Adding another stream should enable playout on the new stream only. |
| EXPECT_TRUE(AddRecvStream(kSsrc2)); |
| SetSend(channel_, true); |
| EXPECT_TRUE(GetSendStream(kSsrc1).IsSending()); |
| |
| // Make sure only the new stream is played out. |
| EXPECT_TRUE(GetRecvStream(kSsrc2).started()); |
| |
| // Adding yet another stream should have stream 2 and 3 enabled for playout. |
| EXPECT_TRUE(AddRecvStream(kSsrc3)); |
| EXPECT_TRUE(GetRecvStream(kSsrc2).started()); |
| EXPECT_TRUE(GetRecvStream(kSsrc3).started()); |
| |
| // Stop sending. |
| SetSend(channel_, false); |
| EXPECT_FALSE(GetSendStream(kSsrc1).IsSending()); |
| |
| // Stop playout. |
| channel_->SetPlayout(false); |
| EXPECT_FALSE(GetRecvStream(kSsrc2).started()); |
| EXPECT_FALSE(GetRecvStream(kSsrc3).started()); |
| |
| // Restart playout and make sure recv streams are played out. |
| channel_->SetPlayout(true); |
| EXPECT_TRUE(GetRecvStream(kSsrc2).started()); |
| EXPECT_TRUE(GetRecvStream(kSsrc3).started()); |
| |
| // Now remove the recv streams. |
| EXPECT_TRUE(channel_->RemoveRecvStream(3)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(2)); |
| } |
| |
| // Test that we can create a channel configured for Codian bridges, |
| // and start sending on it. |
| TEST_F(WebRtcVoiceEngineTestFake, CodianSend) { |
| EXPECT_TRUE(SetupSendStream()); |
| cricket::AudioOptions options_adjust_agc; |
| options_adjust_agc.adjust_agc_delta = rtc::Optional<int>(-10); |
| webrtc::AgcConfig agc_config; |
| EXPECT_EQ(0, voe_.GetAgcConfig(agc_config)); |
| EXPECT_EQ(0, agc_config.targetLeveldBOv); |
| send_parameters_.options = options_adjust_agc; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| SetSend(channel_, true); |
| EXPECT_TRUE(GetSendStream(kSsrc1).IsSending()); |
| EXPECT_EQ(0, voe_.GetAgcConfig(agc_config)); |
| EXPECT_EQ(agc_config.targetLeveldBOv, 10); // level was attenuated |
| SetSend(channel_, false); |
| EXPECT_FALSE(GetSendStream(kSsrc1).IsSending()); |
| EXPECT_EQ(0, voe_.GetAgcConfig(agc_config)); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(adm_, |
| BuiltInAGCIsAvailable()).Times(2).WillRepeatedly(Return(false)); |
| webrtc::AgcConfig agc_config; |
| EXPECT_EQ(0, voe_.GetAgcConfig(agc_config)); |
| EXPECT_EQ(0, agc_config.targetLeveldBOv); |
| send_parameters_.options.tx_agc_target_dbov = rtc::Optional<uint16_t>(3); |
| send_parameters_.options.tx_agc_digital_compression_gain = |
| rtc::Optional<uint16_t>(9); |
| send_parameters_.options.tx_agc_limiter = rtc::Optional<bool>(true); |
| send_parameters_.options.auto_gain_control = rtc::Optional<bool>(true); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(0, voe_.GetAgcConfig(agc_config)); |
| EXPECT_EQ(3, agc_config.targetLeveldBOv); |
| EXPECT_EQ(9, agc_config.digitalCompressionGaindB); |
| EXPECT_TRUE(agc_config.limiterEnable); |
| |
| // Check interaction with adjust_agc_delta. Both should be respected, for |
| // backwards compatibility. |
| send_parameters_.options.adjust_agc_delta = rtc::Optional<int>(-10); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(0, voe_.GetAgcConfig(agc_config)); |
| EXPECT_EQ(13, agc_config.targetLeveldBOv); |
| } |
| |
| TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_CALL(adm_, SetRecordingSampleRate(48000)).WillOnce(Return(0)); |
| EXPECT_CALL(adm_, SetPlayoutSampleRate(44100)).WillOnce(Return(0)); |
| send_parameters_.options.recording_sample_rate = |
| rtc::Optional<uint32_t>(48000); |
| send_parameters_.options.playout_sample_rate = rtc::Optional<uint32_t>(44100); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| // 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(kSsrc1)); |
| } |
| |
| 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 kSsrc1, so the receive |
| // stream has to use a different SSRC. |
| EXPECT_TRUE(AddRecvStream(kSsrc2)); |
| EXPECT_TRUE(channel_->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(channel_, true); |
| EXPECT_EQ(true, channel_->GetStats(&info)); |
| VerifyVoiceSenderInfo(info.senders[0], true); |
| } |
| |
| // Remove the kSsrc2 stream. No receiver stats. |
| { |
| cricket::VoiceMediaInfo info; |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc2)); |
| 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]); |
| } |
| } |
| |
| // 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(kSsrc1)); |
| EXPECT_TRUE(AddRecvStream(kSsrc2)); |
| EXPECT_EQ(kSsrc1, GetRecvStreamConfig(kSsrc2).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(kSsrc2)); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrc1))); |
| EXPECT_TRUE(call_.GetAudioSendStream(kSsrc1)); |
| EXPECT_EQ(kSsrc1, GetRecvStreamConfig(kSsrc2).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 unsignalled stream works (default channel will be |
| // created). |
| TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalled) { |
| EXPECT_TRUE(SetupChannel()); |
| EXPECT_EQ(0, call_.GetAudioReceiveStreams().size()); |
| |
| DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame, |
| sizeof(kPcmuFrame))); |
| } |
| |
| // Test that receiving on an unsignalled stream works (default channel will be |
| // created), and that packets will be forwarded to the default channel |
| // regardless of their SSRCs. |
| TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalledWithSsrcSwitch) { |
| EXPECT_TRUE(SetupChannel()); |
| unsigned char packet[sizeof(kPcmuFrame)]; |
| memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame)); |
| |
| // Note that ssrc = 0 is not supported. |
| uint32_t ssrc = 1; |
| for (; ssrc < 10; ++ssrc) { |
| rtc::SetBE32(&packet[8], ssrc); |
| DeliverPacket(packet, sizeof(packet)); |
| |
| // Verify we only have one default stream. |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| EXPECT_EQ(1, GetRecvStream(ssrc).received_packets()); |
| EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); |
| } |
| |
| // Sending the same ssrc again should not create a new stream. |
| --ssrc; |
| DeliverPacket(packet, sizeof(packet)); |
| EXPECT_EQ(1, call_.GetAudioReceiveStreams().size()); |
| EXPECT_EQ(2, GetRecvStream(ssrc).received_packets()); |
| EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet))); |
| } |
| |
| // Test that a default channel is created even after a signalled stream has been |
| // added, and that this stream will get any packets for unknown SSRCs. |
| TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalledAfterSignalled) { |
| 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))); |
| |
| // 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()); |
| } |
| |
| // 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(kSsrc1)); |
| int channel_num2 = voe_.GetLastChannel(); |
| webrtc::CodecInst gcodec; |
| rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "opus"); |
| gcodec.plfreq = 48000; |
| gcodec.channels = 2; |
| EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec)); |
| } |
| |
| // Test that we properly clean up any streams that were added, even if |
| // not explicitly removed. |
| TEST_F(WebRtcVoiceEngineTestFake, StreamCleanup) { |
| EXPECT_TRUE(SetupSendStream()); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_TRUE(AddRecvStream(1)); |
| EXPECT_TRUE(AddRecvStream( |
