Home
last modified time | relevance | path

Searched refs:rtp (Results 1 – 25 of 69) sorted by relevance

123

/external/webrtc/webrtc/modules/video_coding/
Dgeneric_encoder.cc27 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 …]
/external/webrtc/webrtc/video/
Dsend_statistics_proxy_unittest.cc40 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 …]
Dvie_remb_unittest.cc47 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 …]
Dvideo_receive_stream.cc56 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 …]
Dvideo_send_stream.cc94 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 …]
Dend_to_end_tests.cc306 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 …]
Dpayload_router_unittest.cc37 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()
Dsend_statistics_proxy.cc221 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()
Dreplay.cc221 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()
/external/curl/tests/data/
Dtest5717 # 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 …]
/external/webrtc/webrtc/call/
Drampup_tests.cc120 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 …]
Dbitrate_estimator_tests.cc123 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 …]
Drtc_event_log_unittest.cc131 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 …]
Dcall_unittest.cc47 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()
/external/mp4parser/isoparser/src/main/resources/
Disoparser-default.properties203 #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 …]
/external/webrtc/webrtc/test/
Dcall_test.cc190 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 …]
/external/srtp/test/
Drtpw.c314 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()
/external/webrtc/webrtc/audio/
Daudio_receive_stream_unittest.cc104 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()
Daudio_send_stream_unittest.cc97 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()
Daudio_receive_stream.cc37 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()
/external/webrtc/talk/media/webrtc/
Dwebrtcvideoengine2_unittest.cc101 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 …]
Dwebrtcvideoengine2.cc405 !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 …]
/external/webrtc/data/voice_engine/stereo_rtp_files/
DREADME.txt1 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.
/external/curl/lib/
Drtsp.c591 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 …]
/external/webrtc/webrtc/modules/rtp_rtcp/source/
Drtp_rtcp_impl_unittest.cc414 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 …]

123