/external/webrtc/webrtc/modules/video_coding/ |
D | generic_encoder.cc | 27 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) { in CopyCodecSpecific() argument 31 rtp->codec = kRtpVideoVp8; in CopyCodecSpecific() 32 rtp->codecHeader.VP8.InitRTPVideoHeaderVP8(); in CopyCodecSpecific() 33 rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId; in CopyCodecSpecific() 34 rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference; in CopyCodecSpecific() 35 rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx; in CopyCodecSpecific() 36 rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync; in CopyCodecSpecific() 37 rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx; in CopyCodecSpecific() 38 rtp->codecHeader.VP8.keyIdx = info->codecSpecific.VP8.keyIdx; in CopyCodecSpecific() 39 rtp->simulcastIdx = info->codecSpecific.VP8.simulcastIdx; in CopyCodecSpecific() [all …]
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/external/webrtc/webrtc/video/ |
D | send_statistics_proxy_unittest.cc | 40 config.rtp.ssrcs.push_back(17); in GetTestConfig() 41 config.rtp.ssrcs.push_back(42); in GetTestConfig() 42 config.rtp.rtx.ssrcs.push_back(18); in GetTestConfig() 43 config.rtp.rtx.ssrcs.push_back(43); in GetTestConfig() 101 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); in TEST_F() 102 it != config_.rtp.ssrcs.end(); in TEST_F() 115 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); in TEST_F() 116 it != config_.rtp.rtx.ssrcs.end(); in TEST_F() 159 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); in TEST_F() 160 it != config_.rtp.ssrcs.end(); in TEST_F() [all …]
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D | vie_remb_unittest.cc | 47 MockRtpRtcp rtp; in TEST_F() local 48 vie_remb_->AddReceiveChannel(&rtp); in TEST_F() 49 vie_remb_->AddRembSender(&rtp); in TEST_F() 58 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs)) in TEST_F() 63 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, ssrcs)) in TEST_F() 67 vie_remb_->RemoveReceiveChannel(&rtp); in TEST_F() 68 vie_remb_->RemoveRembSender(&rtp); in TEST_F() 72 MockRtpRtcp rtp; in TEST_F() local 73 vie_remb_->AddReceiveChannel(&rtp); in TEST_F() 74 vie_remb_->AddRembSender(&rtp); in TEST_F() [all …]
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D | video_receive_stream.cc | 56 ss << ", rtp: " << rtp.ToString(); in ToString() 157 config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions); in VideoReceiveStream() 174 vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, in VideoReceiveStream() 176 RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff) in VideoReceiveStream() 179 vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode); in VideoReceiveStream() 181 RTC_DCHECK(config_.rtp.remote_ssrc != 0); in VideoReceiveStream() 183 RTC_DCHECK(config_.rtp.local_ssrc != 0); in VideoReceiveStream() 184 RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc); in VideoReceiveStream() 186 vie_channel_->SetSSRC(config_.rtp.local_ssrc, kViEStreamTypeNormal, 0); in VideoReceiveStream() 188 Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin(); in VideoReceiveStream() [all …]
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D | video_send_stream.cc | 94 ss << ", rtp: " << rtp.ToString(); in ToString() 133 RTC_DCHECK(!config_.rtp.ssrcs.empty()); in VideoSendStream() 137 for (const RtpExtension& extension : config.rtp.extensions) { in VideoSendStream() 145 const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs; in VideoSendStream() 173 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { in VideoSendStream() 174 const std::string& extension = config_.rtp.extensions[i].name; in VideoSendStream() 175 int id = config_.rtp.extensions[i].id; in VideoSendStream() 196 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; in VideoSendStream() 197 const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; in VideoSendStream() 200 config_.rtp.fec.red_payload_type, in VideoSendStream() [all …]
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D | end_to_end_tests.cc | 306 send_config->rtp.nack.rtp_history_ms = in TEST_F() 307 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST_F() 439 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST_F() 440 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST_F() 540 send_config->rtp.fec.red_payload_type = kRedPayloadType; in TEST_F() 541 send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; in TEST_F() 543 (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType; in TEST_F() 544 (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; in TEST_F() 664 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST_F() 665 send_config->rtp.fec.red_payload_type = kRedPayloadType; in TEST_F() [all …]
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D | payload_router_unittest.cc | 37 MockRtpRtcp rtp; in TEST_F() local 38 std::list<RtpRtcp*> modules(1, &rtp); in TEST_F() 46 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, in TEST_F() 53 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, in TEST_F() 60 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, in TEST_F() 67 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, in TEST_F() 75 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, in TEST_F()
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D | send_statistics_proxy.cc | 221 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) == in GetStatsEntry() 222 config_.rtp.ssrcs.end() && in GetStatsEntry() 223 std::find(config_.rtp.rtx.ssrcs.begin(), in GetStatsEntry() 224 config_.rtp.rtx.ssrcs.end(), in GetStatsEntry() 225 ssrc) == config_.rtp.rtx.ssrcs.end()) { in GetStatsEntry() 254 if (simulcast_idx >= config_.rtp.ssrcs.size()) { in OnSendEncodedImage() 256 << " >= " << config_.rtp.ssrcs.size() << ")."; in OnSendEncodedImage() 259 uint32_t ssrc = config_.rtp.ssrcs[simulcast_idx]; in OnSendEncodedImage()
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D | replay.cc | 221 receive_config.rtp.remote_ssrc = flags::Ssrc(); in RtpReplay() 222 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; in RtpReplay() 223 receive_config.rtp.fec.ulpfec_payload_type = flags::FecPayloadType(); in RtpReplay() 224 receive_config.rtp.fec.red_payload_type = flags::RedPayloadType(); in RtpReplay() 225 receive_config.rtp.nack.rtp_history_ms = 1000; in RtpReplay() 227 receive_config.rtp.extensions.push_back( in RtpReplay() 231 receive_config.rtp.extensions.push_back( in RtpReplay()
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/external/curl/tests/data/ |
D | test571 | 7 # 3) packing rtp after headers, after content, and at the start 52 rtp: part 2 channel 1 size 10 53 rtp: part 2 channel 0 size 500 54 rtp: part 2 channel 0 size 196 55 rtp: part 2 channel 0 size 124 56 rtp: part 2 channel 0 size 824 57 rtp: part 3 channel 1 size 10 58 rtp: part 3 channel 0 size 50 59 rtp: part 4 channel 0 size 798 60 rtp: part 4 channel 0 size 42 [all …]
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/external/webrtc/webrtc/call/ |
D | rampup_tests.cc | 120 send_config->rtp.extensions.clear(); in ModifyVideoConfigs() 127 send_config->rtp.extensions.push_back( in ModifyVideoConfigs() 132 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs() 137 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs() 141 send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs; in ModifyVideoConfigs() 142 send_config->rtp.ssrcs = video_ssrcs_; in ModifyVideoConfigs() 144 send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType; in ModifyVideoConfigs() 145 send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_; in ModifyVideoConfigs() 148 send_config->rtp.fec.ulpfec_payload_type = in ModifyVideoConfigs() 150 send_config->rtp.fec.red_payload_type = test::CallTest::kRedPayloadType; in ModifyVideoConfigs() [all …]
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D | bitrate_estimator_tests.cc | 123 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]); in SetUp() 133 receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0]; in SetUp() 134 receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc; in SetUp() 135 receive_config_.rtp.remb = true; in SetUp() 136 receive_config_.rtp.extensions.push_back( in SetUp() 138 receive_config_.rtp.extensions.push_back( in SetUp() 172 test_->video_send_config_.rtp.ssrcs[0]++; in Stream() 186 receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0]; in Stream() 191 receive_config.rtp.extensions.push_back( in Stream() 205 test_->receive_config_.rtp.remote_ssrc = in Stream() [all …]
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D | rtc_event_log_unittest.cc | 131 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc()); in VerifyReceiveStreamConfig() 133 EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc()); in VerifyReceiveStreamConfig() 136 if (config.rtp.rtcp_mode == RtcpMode::kCompound) in VerifyReceiveStreamConfig() 143 EXPECT_EQ(config.rtp.remb, receiver_config.remb()); in VerifyReceiveStreamConfig() 145 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()), in VerifyReceiveStreamConfig() 150 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type())); in VerifyReceiveStreamConfig() 153 config.rtp.rtx.at(rtx_map.payload_type()); in VerifyReceiveStreamConfig() 160 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()), in VerifyReceiveStreamConfig() 167 EXPECT_EQ(config.rtp.extensions[i].id, id); in VerifyReceiveStreamConfig() 168 EXPECT_EQ(config.rtp.extensions[i].name, name); in VerifyReceiveStreamConfig() [all …]
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D | call_unittest.cc | 47 config.rtp.ssrc = 42; in TEST() 57 config.rtp.remote_ssrc = 42; in TEST() 71 config.rtp.ssrc = ssrc; in TEST() 94 config.rtp.remote_ssrc = ssrc; in TEST()
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/external/mp4parser/isoparser/src/main/resources/ |
D | isoparser-default.properties | 203 #stsd-rtp\ =com.coremedia.iso.boxes.rtp.RtpHintSampleEntry(type) 204 #udta-hnti=com.coremedia.iso.boxes.rtp.HintInformationBox() 205 #udta-hinf=com.coremedia.iso.boxes.rtp.HintStatisticsBox() 206 #hnti-sdp\ =com.coremedia.iso.boxes.rtp.RtpTrackSdpHintInformationBox() 207 #hnti-rtp\ =com.coremedia.iso.boxes.rtp.RtpMovieHintInformationBox() 208 #hinf-pmax=com.coremedia.iso.boxes.rtp.LargestHintPacketBox() 209 #hinf-payt=com.coremedia.iso.boxes.rtp.PayloadTypeBox() 210 #hinf-tmin=com.coremedia.iso.boxes.rtp.SmallestRelativeTransmissionTimeBox() 211 #hinf-tmax=com.coremedia.iso.boxes.rtp.LargestRelativeTransmissionTimeBox() 212 #hinf-maxr=com.coremedia.iso.boxes.rtp.MaximumDataRateBox() [all …]
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/external/webrtc/webrtc/test/ |
D | call_test.cc | 190 video_send_config_.rtp.extensions.push_back( in CreateSendConfig() 194 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); in CreateSendConfig() 195 video_send_config_.rtp.extensions.push_back(RtpExtension( in CreateSendConfig() 202 audio_send_config_.rtp.ssrc = kAudioSendSsrc; in CreateSendConfig() 210 RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty()); in CreateMatchingReceiveConfigs() 212 video_config.rtp.remb = true; in CreateMatchingReceiveConfigs() 213 video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc; in CreateMatchingReceiveConfigs() 214 for (const RtpExtension& extension : video_send_config_.rtp.extensions) in CreateMatchingReceiveConfigs() 215 video_config.rtp.extensions.push_back(extension); in CreateMatchingReceiveConfigs() 216 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) { in CreateMatchingReceiveConfigs() [all …]
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/external/srtp/test/ |
D | rtpw.c | 314 crypto_policy_set_rtp_default(&policy.rtp); in main() 318 crypto_policy_set_aes_cm_128_null_auth(&policy.rtp); in main() 322 crypto_policy_set_null_cipher_hmac_sha1_80(&policy.rtp); in main() 335 policy.rtp.sec_serv = sec_servs; in main() 375 policy.rtp.cipher_type = NULL_CIPHER; in main() 376 policy.rtp.cipher_key_len = 0; in main() 377 policy.rtp.auth_type = NULL_AUTH; in main() 378 policy.rtp.auth_key_len = 0; in main() 379 policy.rtp.auth_tag_len = 0; in main() 380 policy.rtp.sec_serv = sec_serv_none; in main()
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/external/webrtc/webrtc/audio/ |
D | audio_receive_stream_unittest.cc | 104 stream_config_.rtp.local_ssrc = kLocalSsrc; in ConfigHelper() 105 stream_config_.rtp.remote_ssrc = kRemoteSsrc; in ConfigHelper() 106 stream_config_.rtp.extensions.push_back( in ConfigHelper() 108 stream_config_.rtp.extensions.push_back( in ConfigHelper() 127 RemoveStream(stream_config_.rtp.remote_ssrc)); in SetupMockForBweFeedback() 206 config.rtp.remote_ssrc = kRemoteSsrc; in TEST() 207 config.rtp.local_ssrc = kLocalSsrc; in TEST() 208 config.rtp.extensions.push_back( in TEST() 263 helper.config().rtp.transport_cc = true; in TEST() 264 helper.config().rtp.extensions.push_back(RtpExtension( in TEST()
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D | audio_send_stream_unittest.cc | 97 stream_config_.rtp.ssrc = kSsrc; in ConfigHelper() 98 stream_config_.rtp.c_name = kCName; in ConfigHelper() 99 stream_config_.rtp.extensions.push_back( in ConfigHelper() 101 stream_config_.rtp.extensions.push_back( in ConfigHelper() 103 stream_config_.rtp.extensions.push_back(RtpExtension( in ConfigHelper() 170 config.rtp.ssrc = kSsrc; in TEST() 171 config.rtp.extensions.push_back( in TEST() 173 config.rtp.c_name = kCName; in TEST()
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D | audio_receive_stream.cc | 37 if (!config.rtp.transport_cc) { in UseSendSideBwe() 40 for (const auto& extension : config.rtp.extensions) { in UseSendSideBwe() 67 ss << "{rtp: " << rtp.ToString(); in ToString() 98 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); in AudioReceiveStream() 99 for (const auto& extension : config.rtp.extensions) { in AudioReceiveStream() 138 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); in ~AudioReceiveStream() 193 stats.remote_ssrc = config_.rtp.remote_ssrc; in GetStats()
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvideoengine2_unittest.cc | 101 it->second == config.rtp.rtx.payload_type); in VerifySendStreamHasRtxTypes() 103 if (config.rtp.fec.red_rtx_payload_type != -1) { in VerifySendStreamHasRtxTypes() 104 it = rtx_types.find(config.rtp.fec.red_payload_type); in VerifySendStreamHasRtxTypes() 106 it->second == config.rtp.fec.red_rtx_payload_type); in VerifySendStreamHasRtxTypes() 976 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); in TestSetSendRtpHeaderExtensions() 977 EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id); in TestSetSendRtpHeaderExtensions() 978 EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name); in TestSetSendRtpHeaderExtensions() 985 .rtp.extensions.empty()); in TestSetSendRtpHeaderExtensions() 991 EXPECT_TRUE(send_stream->GetConfig().rtp.extensions.empty()); in TestSetSendRtpHeaderExtensions() 997 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); in TestSetSendRtpHeaderExtensions() [all …]
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D | webrtcvideoengine2.cc | 405 !is_screencast && parameters_.config.rtp.ssrcs.size() == 1; in ConfigureVideoEncoderSettings() 1096 config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false; in AddRecvStream() 1097 config.rtp.transport_cc = in AddRecvStream() 1112 config->rtp.remote_ssrc = ssrc; in ConfigureReceiverRtp() 1113 config->rtp.local_ssrc = rtcp_receiver_report_ssrc_; in ConfigureReceiverRtp() 1115 config->rtp.extensions = recv_rtp_extensions_; in ConfigureReceiverRtp() 1116 config->rtp.rtcp_mode = recv_params_.rtcp.reduced_size in ConfigureReceiverRtp() 1124 if (config->rtp.remote_ssrc == config->rtp.local_ssrc) { in ConfigureReceiverRtp() 1125 if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) { in ConfigureReceiverRtp() 1126 config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc; in ConfigureReceiverRtp() [all …]
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/external/webrtc/data/voice_engine/stereo_rtp_files/ |
D | README.txt | 1 Use RTP Play tool with command 'rtpplay.exe -v -T -f <path>\<file.rtp> 127.0.0.1/1236' 2 Example: rtpplay.exe -v -T -f hrtf_g722_1C_48.rtp 127.0.0.1/1236. 3 This sends the stereo rtp file to port 1236.
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/external/curl/lib/ |
D | rtsp.c | 591 char *rtp; /* moving pointer to rtp data */ in rtsp_rtp_readwrite() local 608 rtp = rtspc->rtp_buf; in rtsp_rtp_readwrite() 613 rtp = k->str; in rtsp_rtp_readwrite() 618 (rtp[0] == '$')) { in rtsp_rtp_readwrite() 624 rtspc->rtp_channel = RTP_PKT_CHANNEL(rtp); in rtsp_rtp_readwrite() 627 rtp_length = RTP_PKT_LENGTH(rtp); in rtsp_rtp_readwrite() 639 result = rtp_client_write(conn, &rtp[0], rtp_length + 4); in rtsp_rtp_readwrite() 651 rtp += rtp_length + 4; in rtsp_rtp_readwrite() 668 if(rtp_dataleft != 0 && rtp[0] == '$') { in rtsp_rtp_readwrite() 680 memcpy(scratch, rtp, rtp_dataleft); in rtsp_rtp_readwrite() [all …]
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtp_rtcp_impl_unittest.cc | 414 StreamDataCounters rtp; in TEST_F() local 416 rtp.first_packet_time_ms = kStartTimeMs; in TEST_F() 417 rtp.transmitted.packets = 1; in TEST_F() 418 rtp.transmitted.payload_bytes = 1; in TEST_F() 419 rtp.transmitted.header_bytes = 2; in TEST_F() 420 rtp.transmitted.padding_bytes = 3; in TEST_F() 421 EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes + in TEST_F() 422 rtp.transmitted.header_bytes + in TEST_F() 423 rtp.transmitted.padding_bytes); in TEST_F() 435 StreamDataCounters sum = rtp; in TEST_F() [all …]
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