| /* |
| * 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/fake_voice_engine.h" |
| |
| namespace { |
| |
| using webrtc::ByteWriter; |
| |
| 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 |
| |
| namespace webrtc { |
| namespace test { |
| |
| TEST(AudioReceiveStreamTest, ConfigToString) { |
| const int kAbsSendTimeId = 3; |
| AudioReceiveStream::Config config; |
| config.rtp.remote_ssrc = 1234; |
| config.rtp.local_ssrc = 5678; |
| config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); |
| config.voe_channel_id = 1; |
| 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: 1, combined_audio_video_bwe: true}", config.ToString()); |
| } |
| |
| TEST(AudioReceiveStreamTest, ConstructDestruct) { |
| MockRemoteBitrateEstimator remote_bitrate_estimator; |
| FakeVoiceEngine voice_engine; |
| AudioReceiveStream::Config config; |
| config.voe_channel_id = 1; |
| internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config, |
| &voice_engine); |
| } |
| |
| TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) { |
| MockRemoteBitrateEstimator remote_bitrate_estimator; |
| FakeVoiceEngine voice_engine; |
| AudioReceiveStream::Config config; |
| config.combined_audio_video_bwe = true; |
| config.voe_channel_id = FakeVoiceEngine::kRecvChannelId; |
| 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) { |
| MockRemoteBitrateEstimator remote_bitrate_estimator; |
| FakeVoiceEngine voice_engine; |
| AudioReceiveStream::Config config; |
| config.rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc; |
| config.voe_channel_id = FakeVoiceEngine::kRecvChannelId; |
| internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config, |
| &voice_engine); |
| |
| AudioReceiveStream::Stats stats = recv_stream.GetStats(); |
| const CallStatistics& call_stats = FakeVoiceEngine::kRecvCallStats; |
| const CodecInst& codec_inst = FakeVoiceEngine::kRecvCodecInst; |
| const NetworkStatistics& net_stats = FakeVoiceEngine::kRecvNetworkStats; |
| const AudioDecodingCallStats& decode_stats = |
| FakeVoiceEngine::kRecvAudioDecodingCallStats; |
| EXPECT_EQ(FakeVoiceEngine::kRecvSsrc, stats.remote_ssrc); |
| EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd); |
| EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived), |
| stats.packets_rcvd); |
| EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost); |
| EXPECT_EQ(Q8ToFloat(call_stats.fractionLost), stats.fraction_lost); |
| EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name); |
| EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum); |
| EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000), |
| stats.jitter_ms); |
| EXPECT_EQ(net_stats.currentBufferSize, stats.jitter_buffer_ms); |
| EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms); |
| EXPECT_EQ(static_cast<uint32_t>(FakeVoiceEngine::kRecvJitterBufferDelay + |
| FakeVoiceEngine::kRecvPlayoutBufferDelay), stats.delay_estimate_ms); |
| EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kRecvSpeechOutputLevel), |
| stats.audio_level); |
| EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate); |
| EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate), |
| stats.speech_expand_rate); |
| EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate), |
| stats.secondary_decoded_rate); |
| EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate); |
| EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate), |
| stats.preemptive_expand_rate); |
| EXPECT_EQ(decode_stats.calls_to_silence_generator, |
| stats.decoding_calls_to_silence_generator); |
| EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq); |
| EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal); |
| EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc); |
| EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng); |
| EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng); |
| EXPECT_EQ(call_stats.capture_start_ntp_time_ms_, |
| stats.capture_start_ntp_time_ms); |
| } |
| } // namespace test |
| } // namespace webrtc |