| /* |
| * 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 "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/audio/audio_receive_stream.h" |
| #include "webrtc/audio/conversion.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/test/mock_voice_engine.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| const int kChannelId = 2; |
| const uint32_t kRemoteSsrc = 1234; |
| const uint32_t kLocalSsrc = 5678; |
| const size_t kAbsoluteSendTimeLength = 4; |
| |
| void BuildAbsoluteSendTimeExtension(uint8_t* buffer, |
| int id, |
| uint32_t abs_send_time) { |
| const size_t kRtpOneByteHeaderLength = 4; |
| const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE; |
| ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId); |
| |
| const uint32_t kPosLength = 2; |
| ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength, |
| kAbsoluteSendTimeLength / 4); |
| |
| const uint8_t kLengthOfData = 3; |
| buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1); |
| ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian( |
| buffer + kRtpOneByteHeaderLength + 1, abs_send_time); |
| } |
| |
| size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header, |
| int extension_id, |
| uint32_t abs_send_time) { |
| 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. |
| int32_t rtp_header_length = webrtc::kRtpHeaderSize; |
| |
| BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id, |
| abs_send_time); |
| rtp_header_length += kAbsoluteSendTimeLength; |
| return rtp_header_length; |
| } |
| } // namespace |
| |
| TEST(AudioReceiveStreamTest, ConfigToString) { |
| const int kAbsSendTimeId = 3; |
| AudioReceiveStream::Config config; |
| config.rtp.remote_ssrc = kRemoteSsrc; |
| config.rtp.local_ssrc = kLocalSsrc; |
| config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); |
| config.voe_channel_id = kChannelId; |
| config.combined_audio_video_bwe = true; |
| EXPECT_EQ( |
| "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: " |
| "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, " |
| "receive_transport: nullptr, rtcp_send_transport: nullptr, " |
| "voe_channel_id: 2, combined_audio_video_bwe: true}", |
| config.ToString()); |
| } |
| |
| TEST(AudioReceiveStreamTest, ConstructDestruct) { |
| MockRemoteBitrateEstimator remote_bitrate_estimator; |
| MockVoiceEngine voice_engine; |
| AudioReceiveStream::Config config; |
| config.voe_channel_id = kChannelId; |
| internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config, |
| &voice_engine); |
| } |
| |
| TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) { |
| MockRemoteBitrateEstimator remote_bitrate_estimator; |
| MockVoiceEngine voice_engine; |
| AudioReceiveStream::Config config; |
| config.combined_audio_video_bwe = true; |
| config.voe_channel_id = kChannelId; |
| const int kAbsSendTimeId = 3; |
| config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); |
| internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config, |
| &voice_engine); |
| uint8_t rtp_packet[30]; |
| const int kAbsSendTimeValue = 1234; |
| CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue); |
| PacketTime packet_time(5678000, 0); |
| const size_t kExpectedHeaderLength = 20; |
| EXPECT_CALL(remote_bitrate_estimator, |
| IncomingPacket(packet_time.timestamp / 1000, |
| sizeof(rtp_packet) - kExpectedHeaderLength, |
| testing::_, false)) |
| .Times(1); |
| EXPECT_TRUE( |
| recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time)); |
| } |
| |
| TEST(AudioReceiveStreamTest, GetStats) { |
| 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, -198, |
| -103}; |
| |
| const NetworkStatistics kNetworkStats = { |
| 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0}; |
| |
| webrtc::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; |
| } |
| |
| MockRemoteBitrateEstimator remote_bitrate_estimator; |
| MockVoiceEngine voice_engine; |
| AudioReceiveStream::Config config; |
| config.rtp.remote_ssrc = kRemoteSsrc; |
| config.voe_channel_id = kChannelId; |
| internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config, |
| &voice_engine); |
| |
| using testing::_; |
| using testing::DoAll; |
| using testing::Return; |
| using testing::SetArgPointee; |
| using testing::SetArgReferee; |
| EXPECT_CALL(voice_engine, GetRemoteSSRC(kChannelId, _)) |
| .WillOnce(DoAll(SetArgReferee<1>(0), Return(0))); |
| EXPECT_CALL(voice_engine, GetRTCPStatistics(kChannelId, _)) |
| .WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0))); |
| EXPECT_CALL(voice_engine, GetRecCodec(kChannelId, _)) |
| .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0))); |
| EXPECT_CALL(voice_engine, GetDelayEstimate(kChannelId, _, _)) |
| .WillOnce(DoAll(SetArgPointee<1>(kJitterBufferDelay), |
| SetArgPointee<2>(kPlayoutBufferDelay), Return(0))); |
| EXPECT_CALL(voice_engine, GetSpeechOutputLevelFullRange(kChannelId, _)) |
| .WillOnce(DoAll(SetArgReferee<1>(kSpeechOutputLevel), Return(0))); |
| EXPECT_CALL(voice_engine, GetNetworkStatistics(kChannelId, _)) |
| .WillOnce(DoAll(SetArgReferee<1>(kNetworkStats), Return(0))); |
| EXPECT_CALL(voice_engine, GetDecodingCallStatistics(kChannelId, _)) |
| .WillOnce(DoAll(SetArgPointee<1>(audio_decode_stats), Return(0))); |
| |
| 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(audio_decode_stats.calls_to_silence_generator, |
| stats.decoding_calls_to_silence_generator); |
| EXPECT_EQ(audio_decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq); |
| EXPECT_EQ(audio_decode_stats.decoded_normal, stats.decoding_normal); |
| EXPECT_EQ(audio_decode_stats.decoded_plc, stats.decoding_plc); |
| EXPECT_EQ(audio_decode_stats.decoded_cng, stats.decoding_cng); |
| EXPECT_EQ(audio_decode_stats.decoded_plc_cng, stats.decoding_plc_cng); |
| EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, |
| stats.capture_start_ntp_time_ms); |
| } |
| } // namespace test |
| } // namespace webrtc |