| /* |
| * Copyright (c) 2015 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 <string> |
| #include <vector> |
| |
| #include "webrtc/audio/audio_receive_stream.h" |
| #include "webrtc/audio/conversion.h" |
| #include "webrtc/call/mock/mock_rtc_event_log.h" |
| #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" |
| #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h" |
| #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h" |
| #include "webrtc/modules/pacing/packet_router.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/test/gtest.h" |
| #include "webrtc/test/mock_voe_channel_proxy.h" |
| #include "webrtc/test/mock_voice_engine.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| using testing::_; |
| using testing::FloatEq; |
| using testing::Return; |
| using testing::ReturnRef; |
| |
| AudioDecodingCallStats MakeAudioDecodeStatsForTest() { |
| AudioDecodingCallStats audio_decode_stats; |
| audio_decode_stats.calls_to_silence_generator = 234; |
| audio_decode_stats.calls_to_neteq = 567; |
| audio_decode_stats.decoded_normal = 890; |
| audio_decode_stats.decoded_plc = 123; |
| audio_decode_stats.decoded_cng = 456; |
| audio_decode_stats.decoded_plc_cng = 789; |
| audio_decode_stats.decoded_muted_output = 987; |
| return audio_decode_stats; |
| } |
| |
| const int kChannelId = 2; |
| const uint32_t kRemoteSsrc = 1234; |
| const uint32_t kLocalSsrc = 5678; |
| const size_t kOneByteExtensionHeaderLength = 4; |
| const size_t kOneByteExtensionLength = 4; |
| const int kAbsSendTimeId = 2; |
| const int kAudioLevelId = 3; |
| const int kTransportSequenceNumberId = 4; |
| const int kJitterBufferDelay = -7; |
| const int kPlayoutBufferDelay = 302; |
| const unsigned int kSpeechOutputLevel = 99; |
| const CallStatistics kCallStats = { |
| 345, 678, 901, 234, -12, 3456, 7890, 567, 890, 123}; |
| const CodecInst kCodecInst = { |
| 123, "codec_name_recv", 96000, -187, 0, -103}; |
| const NetworkStatistics kNetworkStats = { |
| 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0}; |
| const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest(); |
| |
| struct ConfigHelper { |
| ConfigHelper() |
| : simulated_clock_(123456), |
| decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>), |
| congestion_controller_(&simulated_clock_, |
| &bitrate_observer_, |
| &remote_bitrate_observer_, |
| &event_log_) { |
| using testing::Invoke; |
| |
| EXPECT_CALL(voice_engine_, |
| RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); |
| EXPECT_CALL(voice_engine_, |
| DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); |
| AudioState::Config config; |
| config.voice_engine = &voice_engine_; |
| audio_state_ = AudioState::Create(config); |
| |
| EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) |
| .WillOnce(Invoke([this](int channel_id) { |
| EXPECT_FALSE(channel_proxy_); |
| channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); |
| EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1); |
| EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1); |
| EXPECT_CALL(*channel_proxy_, |
| SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId)) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, |
| SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId)) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, |
| EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId)) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, |
| RegisterReceiverCongestionControlObjects(&packet_router_)) |
| .Times(1); |
| EXPECT_CALL(congestion_controller_, packet_router()) |
| .WillOnce(Return(&packet_router_)); |
| EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory()) |
| .WillOnce(ReturnRef(decoder_factory_)); |
| testing::Expectation expect_set = |
| EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_)) |
| .Times(1); |
| EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) |
| .Times(1) |
| .After(expect_set); |
| return channel_proxy_; |
| })); |
| EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0)); |
| stream_config_.voe_channel_id = kChannelId; |
| stream_config_.rtp.local_ssrc = kLocalSsrc; |
| stream_config_.rtp.remote_ssrc = kRemoteSsrc; |
| stream_config_.rtp.nack.rtp_history_ms = 300; |
| stream_config_.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
| stream_config_.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
| stream_config_.rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); |
| stream_config_.decoder_factory = decoder_factory_; |
| } |
| |
| MockCongestionController* congestion_controller() { |
| return &congestion_controller_; |
| } |
| MockRemoteBitrateEstimator* remote_bitrate_estimator() { |
| return &remote_bitrate_estimator_; |
| } |
| MockRtcEventLog* event_log() { return &event_log_; } |
| AudioReceiveStream::Config& config() { return stream_config_; } |
| rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } |
| MockVoiceEngine& voice_engine() { return voice_engine_; } |
| MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } |
| |
| void SetupMockForBweFeedback(bool send_side_bwe) { |
| EXPECT_CALL(congestion_controller_, |
| GetRemoteBitrateEstimator(send_side_bwe)) |
| .WillOnce(Return(&remote_bitrate_estimator_)); |
| EXPECT_CALL(remote_bitrate_estimator_, |
| RemoveStream(stream_config_.rtp.remote_ssrc)); |
| } |
| |
| void SetupMockForGetStats() { |
| using testing::DoAll; |
| using testing::SetArgReferee; |
| |
| ASSERT_TRUE(channel_proxy_); |
| EXPECT_CALL(*channel_proxy_, GetRTCPStatistics()) |
| .WillOnce(Return(kCallStats)); |
| EXPECT_CALL(*channel_proxy_, GetDelayEstimate()) |
| .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay)); |
| EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange()) |
| .WillOnce(Return(kSpeechOutputLevel)); |
| EXPECT_CALL(*channel_proxy_, GetNetworkStatistics()) |
| .WillOnce(Return(kNetworkStats)); |
| EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics()) |
| .WillOnce(Return(kAudioDecodeStats)); |
| |
| EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _)) |
| .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0))); |
| } |
| |
| private: |
| SimulatedClock simulated_clock_; |
| PacketRouter packet_router_; |
| testing::NiceMock<MockCongestionObserver> bitrate_observer_; |
| testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_; |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
| MockCongestionController congestion_controller_; |
| MockRemoteBitrateEstimator remote_bitrate_estimator_; |
| MockRtcEventLog event_log_; |
| testing::StrictMock<MockVoiceEngine> voice_engine_; |
| rtc::scoped_refptr<AudioState> audio_state_; |
| AudioReceiveStream::Config stream_config_; |
| testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
| }; |
| |
| void BuildOneByteExtension(std::vector<uint8_t>::iterator it, |
| int id, |
| uint32_t extension_value, |
| size_t value_length) { |
| const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE; |
| ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId); |
| it += 2; |
| |
| ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4); |
| it += 2; |
| const size_t kExtensionDataLength = kOneByteExtensionLength - 1; |
| uint32_t shifted_value = extension_value |
| << (8 * (kExtensionDataLength - value_length)); |
| *it = (id << 4) + (static_cast<uint8_t>(value_length) - 1); |
| ++it; |
| ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it), |
| shifted_value); |
| } |
| |
| const std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension( |
| int extension_id, |
| uint32_t extension_value, |
| size_t value_length) { |
| std::vector<uint8_t> header; |
| header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength + |
| kOneByteExtensionLength); |
| header[0] = 0x80; // Version 2. |
| header[0] |= 0x10; // Set extension bit. |
| header[1] = 100; // Payload type. |
| header[1] |= 0x80; // Marker bit is set. |
| ByteWriter<uint16_t>::WriteBigEndian(&header[2], 0x1234); // Sequence number. |
| ByteWriter<uint32_t>::WriteBigEndian(&header[4], 0x5678); // Timestamp. |
| ByteWriter<uint32_t>::WriteBigEndian(&header[8], 0x4321); // SSRC. |
| |
| BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id, |
| extension_value, value_length); |
| return header; |
| } |
| |
| const std::vector<uint8_t> CreateRtcpSenderReport() { |
| std::vector<uint8_t> packet; |
| const size_t kRtcpSrLength = 28; // In bytes. |
| packet.resize(kRtcpSrLength); |
| packet[0] = 0x80; // Version 2. |
| packet[1] = 0xc8; // PT = 200, SR. |
| // Length in number of 32-bit words - 1. |
| ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6); |
| ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc); |
| return packet; |
| } |
| } // namespace |
| |
| TEST(AudioReceiveStreamTest, ConfigToString) { |
| AudioReceiveStream::Config config; |
| config.rtp.remote_ssrc = kRemoteSsrc; |
| config.rtp.local_ssrc = kLocalSsrc; |
| config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
| config.voe_channel_id = kChannelId; |
| EXPECT_EQ( |
| "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, " |
| "nack: {rtp_history_ms: 0}, extensions: [{uri: " |
| "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, " |
| "rtcp_send_transport: nullptr, " |
| "voe_channel_id: 2}", |
| config.ToString()); |
| } |
| |
| TEST(AudioReceiveStreamTest, ConstructDestruct) { |
| ConfigHelper helper; |
| internal::AudioReceiveStream recv_stream( |
| helper.congestion_controller(), helper.config(), helper.audio_state(), |
| helper.event_log()); |
| } |
| |
| MATCHER_P(VerifyHeaderExtension, expected_extension, "") { |
| return arg.extension.hasAbsoluteSendTime == |
| expected_extension.hasAbsoluteSendTime && |
| arg.extension.absoluteSendTime == |
| expected_extension.absoluteSendTime && |
| arg.extension.hasTransportSequenceNumber == |
| expected_extension.hasTransportSequenceNumber && |
| arg.extension.transportSequenceNumber == |
| expected_extension.transportSequenceNumber; |
| } |
| |
| TEST(AudioReceiveStreamTest, ReceiveRtpPacket) { |
| ConfigHelper helper; |
| helper.config().rtp.transport_cc = true; |
| helper.SetupMockForBweFeedback(true); |
| internal::AudioReceiveStream recv_stream( |
| helper.congestion_controller(), helper.config(), helper.audio_state(), |
| helper.event_log()); |
| const int kTransportSequenceNumberValue = 1234; |
| std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( |
| kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); |
| PacketTime packet_time(5678000, 0); |
| const size_t kExpectedHeaderLength = 20; |
| RTPHeaderExtension expected_extension; |
| expected_extension.hasTransportSequenceNumber = true; |
| expected_extension.transportSequenceNumber = kTransportSequenceNumberValue; |
| EXPECT_CALL(*helper.remote_bitrate_estimator(), |
| IncomingPacket(packet_time.timestamp / 1000, |
| rtp_packet.size() - kExpectedHeaderLength, |
| VerifyHeaderExtension(expected_extension))) |
| .Times(1); |
| EXPECT_CALL(*helper.channel_proxy(), |
| ReceivedRTPPacket(&rtp_packet[0], |
| rtp_packet.size(), |
| _)) |
| .WillOnce(Return(true)); |
| EXPECT_TRUE( |
| recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); |
| } |
| |
| TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) { |
| ConfigHelper helper; |
| helper.config().rtp.transport_cc = true; |
| helper.SetupMockForBweFeedback(true); |
| internal::AudioReceiveStream recv_stream( |
| helper.congestion_controller(), helper.config(), helper.audio_state(), |
| helper.event_log()); |
| |
| std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport(); |
| EXPECT_CALL(*helper.channel_proxy(), |
| ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size())) |
| .WillOnce(Return(true)); |
| EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size())); |
| } |
| |
| TEST(AudioReceiveStreamTest, GetStats) { |
| ConfigHelper helper; |
| internal::AudioReceiveStream recv_stream( |
| helper.congestion_controller(), helper.config(), helper.audio_state(), |
| helper.event_log()); |
| helper.SetupMockForGetStats(); |
| AudioReceiveStream::Stats stats = recv_stream.GetStats(); |
| EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); |
| EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd); |
| EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived), |
| stats.packets_rcvd); |
| EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost); |
| EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost); |
| EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name); |
| EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum); |
| EXPECT_EQ(kCallStats.jitterSamples / (kCodecInst.plfreq / 1000), |
| stats.jitter_ms); |
| EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms); |
| EXPECT_EQ(kNetworkStats.preferredBufferSize, |
| stats.jitter_buffer_preferred_ms); |
| EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay), |
| stats.delay_estimate_ms); |
| EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level); |
| EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate); |
| EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate), |
| stats.speech_expand_rate); |
| EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate), |
| stats.secondary_decoded_rate); |
| EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate), |
| stats.accelerate_rate); |
| EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate), |
| stats.preemptive_expand_rate); |
| EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator, |
| stats.decoding_calls_to_silence_generator); |
| EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq); |
| EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal); |
| EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc); |
| EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng); |
| EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng); |
| EXPECT_EQ(kAudioDecodeStats.decoded_muted_output, |
| stats.decoding_muted_output); |
| EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, |
| stats.capture_start_ntp_time_ms); |
| } |
| |
| TEST(AudioReceiveStreamTest, SetGain) { |
| ConfigHelper helper; |
| internal::AudioReceiveStream recv_stream( |
| helper.congestion_controller(), helper.config(), helper.audio_state(), |
| helper.event_log()); |
| EXPECT_CALL(*helper.channel_proxy(), |
| SetChannelOutputVolumeScaling(FloatEq(0.765f))); |
| recv_stream.SetGain(0.765f); |
| } |
| } // namespace test |
| } // namespace webrtc |
