/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | nack_unittest.cc | 11 #include "webrtc/modules/audio_coding/main/acm2/nack.h" 61 scoped_ptr<Nack> nack(Nack::Create(kNackThreshold)); 62 nack->UpdateSampleRate(kSampleRateHz); 69 nack->UpdateLastReceivedPacket(seq_num, timestamp); 70 nack_list = nack->GetNackList(kShortRoundTripTimeMs); 73 nack_list = nack->GetNackList(kShortRoundTripTimeMs); 79 scoped_ptr<Nack> nack(Nack::Create(kNackThreshold)); 80 nack->UpdateSampleRate(kSampleRateHz); 86 nack->UpdateLastReceivedPacket(seq_num, timestamp); 87 nack_list = nack [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_packet_unittest.cc | 329 Nack nack; local 332 nack.From(kSenderSsrc); 333 nack.To(kRemoteSsrc); 334 nack.WithList(kList, kListLength); 335 RawPacket packet = nack.Build(); 338 EXPECT_EQ(1, parser.nack()->num_packets()); 339 EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc()); 340 EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc()); 350 Nack nack; local 353 nack [all...] |
H A D | rtp_rtcp_impl_unittest.cc | 188 rtcp::Nack nack; local 192 nack.From(kReceiverSsrc); 193 nack.To(kSenderSsrc); 194 nack.WithList(list, kListLength); 195 rtcp::RawPacket packet = nack.Build();
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H A D | rtcp_sender.cc | 37 void NACKStringBuilder::PushNACK(uint16_t nack) argument 41 _stream << nack; local 42 } else if (nack == _prevNack + 1) 52 _stream << "," << nack; local 55 _prevNack = nack; 1340 // add nack list 1359 // The nack list should be sorted and not contain duplicates if one 1360 // wants to build the smallest rtcp nack packet. 1367 uint16_t nack = nackList[i++]; local 1370 int shift = static_cast<uint16_t>(nackList[i] - nack) [all...] |
H A D | rtcp_sender.h | 40 void PushNACK(uint16_t nack);
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H A D | rtcp_packet.cc | 393 void CreateNack(const RTCPPacketRTPFBNACK& nack, argument 400 AssignUWord32(buffer, pos, nack.SenderSSRC); 401 AssignUWord32(buffer, pos, nack.MediaSSRC);
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/external/chromium_org/third_party/webrtc/ |
H A D | video_receive_stream.h | 116 NackConfig nack; member in struct:webrtc::VideoReceiveStream::Config::Rtp
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H A D | video_send_stream.h | 94 NackConfig nack; member in struct:webrtc::VideoSendStream::Config::Rtp
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/external/chromium_org/third_party/webrtc/video/ |
H A D | loopback.cc | 118 send_config.rtp.nack.rtp_history_ms = 1000; 158 receive_config.rtp.nack.rtp_history_ms = 1000;
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H A D | video_send_stream.cc | 80 if (nack.rtp_history_ms != 0) 81 ss << ", nack.rtp_history_ms: " << nack.rtp_history_ms; 162 if (config_.rtp.nack.rtp_history_ms > 0) { 176 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
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H A D | end_to_end_tests.cc | 373 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 374 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 469 // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms; 470 // send_config->rtp.nack.rtp_history_ms = rtp_history_ms; 545 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 547 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 748 send_config->rtp.nack.rtp_history_ms = rtp_history_ms_; 749 (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_; 895 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 896 (*receive_configs)[0].rtp.nack [all...] |
H A D | rampup_tests.cc | 418 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 462 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
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H A D | replay.cc | 201 receive_config.rtp.nack.rtp_history_ms = 1000;
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H A D | call_perf_tests.cc | 260 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 267 receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
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H A D | video_receive_stream.cc | 53 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
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/external/chromium_org/third_party/webrtc/test/ |
H A D | rtcp_packet_parser.h | 347 void Set(const RTCPUtility::RTCPPacketRTPFBNACK& nack) { argument 348 nack_ = nack; 652 Nack* nack() { return &nack_; }
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/external/lldb/tools/debugserver/source/ |
H A D | RNBRemote.h | 40 nack, // '-' enumerator in enum:RNBRemote::__anon25440
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcvideoengine2_unittest.cc | 1044 EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0); 1049 EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0); 1052 EXPECT_EQ(send_stream->GetConfig().rtp.nack.rtp_history_ms, 1053 recv_stream->GetConfig().rtp.nack.rtp_history_ms); 1064 EXPECT_EQ(0, send_stream->GetConfig().rtp.nack.rtp_history_ms); 1070 EXPECT_EQ(0, recv_stream->GetConfig().rtp.nack.rtp_history_ms); 1236 EXPECT_EQ(1000, config.rtp.nack.rtp_history_ms);
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H A D | fakewebrtcvoiceengine.h | 191 nack(false), 221 bool nack; member in struct:cricket::FakeWebRtcVoiceEngine::Channel 320 return channels_[channel]->nack; 1011 channels_[channel]->nack = enable;
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/external/chromium_org/third_party/webrtc/modules/ |
H A D | audio_coding_module.target.darwin-arm64.mk | 53 third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
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H A D | audio_coding_module.target.darwin-mips.mk | 53 third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
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H A D | audio_coding_module.target.darwin-mips64.mk | 53 third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
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H A D | audio_coding_module.target.darwin-x86.mk | 53 third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
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H A D | audio_coding_module.target.darwin-x86_64.mk | 53 third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
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H A D | audio_coding_module.target.linux-arm64.mk | 53 third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
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