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/external/webrtc/test/
Dcall_config_utils_unittest.cc29 recv_config.rtp.remote_ssrc = 100; in TEST()
30 recv_config.rtp.local_ssrc = 101; in TEST()
31 recv_config.rtp.rtcp_mode = RtcpMode::kCompound; in TEST()
32 recv_config.rtp.transport_cc = false; in TEST()
33 recv_config.rtp.lntf.enabled = false; in TEST()
34 recv_config.rtp.nack.rtp_history_ms = 150; in TEST()
35 recv_config.rtp.red_payload_type = 50; in TEST()
36 recv_config.rtp.rtx_ssrc = 1000; in TEST()
37 recv_config.rtp.rtx_associated_payload_types[10] = 10; in TEST()
38 recv_config.rtp.extensions.emplace_back("uri", 128, true); in TEST()
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Dcall_config_utils.cc40 receive_config.rtp.remote_ssrc = json["rtp"]["remote_ssrc"].asInt64(); in ParseVideoReceiveStreamJsonConfig()
41 receive_config.rtp.local_ssrc = json["rtp"]["local_ssrc"].asInt64(); in ParseVideoReceiveStreamJsonConfig()
42 receive_config.rtp.rtcp_mode = in ParseVideoReceiveStreamJsonConfig()
46 receive_config.rtp.transport_cc = json["rtp"]["transport_cc"].asBool(); in ParseVideoReceiveStreamJsonConfig()
47 receive_config.rtp.lntf.enabled = json["rtp"]["lntf"]["enabled"].asInt64(); in ParseVideoReceiveStreamJsonConfig()
48 receive_config.rtp.nack.rtp_history_ms = in ParseVideoReceiveStreamJsonConfig()
50 receive_config.rtp.ulpfec_payload_type = in ParseVideoReceiveStreamJsonConfig()
52 receive_config.rtp.red_payload_type = in ParseVideoReceiveStreamJsonConfig()
54 receive_config.rtp.rtx_ssrc = json["rtp"]["rtx_ssrc"].asInt64(); in ParseVideoReceiveStreamJsonConfig()
60 receive_config.rtp.rtx_associated_payload_types[std::stoi(members[0])] = in ParseVideoReceiveStreamJsonConfig()
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Dcall_test.cc251 video_config->rtp.payload_name = "FAKE"; in CreateVideoSendConfig()
252 video_config->rtp.payload_type = kFakeVideoSendPayloadType; in CreateVideoSendConfig()
253 video_config->rtp.extmap_allow_mixed = true; in CreateVideoSendConfig()
255 &video_config->rtp.extensions); in CreateVideoSendConfig()
257 &video_config->rtp.extensions); in CreateVideoSendConfig()
259 &video_config->rtp.extensions); in CreateVideoSendConfig()
261 &video_config->rtp.extensions); in CreateVideoSendConfig()
268 video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]); in CreateVideoSendConfig()
270 &video_config->rtp.extensions); in CreateVideoSendConfig()
272 &video_config->rtp.extensions); in CreateVideoSendConfig()
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/external/webrtc/logging/rtc_event_log/
Dlogged_events.cc14 LoggedPacketInfo::LoggedPacketInfo(const LoggedRtpPacket& rtp, in LoggedPacketInfo() argument
18 : ssrc(rtp.header.ssrc), in LoggedPacketInfo()
19 stream_seq_no(rtp.header.sequenceNumber), in LoggedPacketInfo()
20 size(static_cast<uint16_t>(rtp.total_length)), in LoggedPacketInfo()
21 payload_size(static_cast<uint16_t>(rtp.total_length - in LoggedPacketInfo()
22 rtp.header.paddingLength - in LoggedPacketInfo()
23 rtp.header.headerLength)), in LoggedPacketInfo()
24 padding_size(static_cast<uint16_t>(rtp.header.paddingLength)), in LoggedPacketInfo()
25 payload_type(rtp.header.payloadType), in LoggedPacketInfo()
28 marker_bit(rtp.header.markerBit), in LoggedPacketInfo()
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Drtc_event_log2rtp_dump.cc54 rtp,
117 reconstructed_packet.SetMarker(incoming.rtp.header.markerBit); in ConvertRtpPacket()
118 reconstructed_packet.SetPayloadType(incoming.rtp.header.payloadType); in ConvertRtpPacket()
119 reconstructed_packet.SetSequenceNumber(incoming.rtp.header.sequenceNumber); in ConvertRtpPacket()
120 reconstructed_packet.SetTimestamp(incoming.rtp.header.timestamp); in ConvertRtpPacket()
121 reconstructed_packet.SetSsrc(incoming.rtp.header.ssrc); in ConvertRtpPacket()
122 if (incoming.rtp.header.numCSRCs > 0) { in ConvertRtpPacket()
124 incoming.rtp.header.arrOfCSRCs, incoming.rtp.header.numCSRCs)); in ConvertRtpPacket()
128 if (incoming.rtp.header.extension.hasTransmissionTimeOffset) in ConvertRtpPacket()
130 incoming.rtp.header.extension.transmissionTimeOffset); in ConvertRtpPacket()
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/external/webrtc/video/end_to_end_tests/
Dconfig_tests.cc55 EXPECT_FALSE(default_send_config.rtp.lntf.enabled) in TEST_F()
57 EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms) in TEST_F()
59 EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty()) in TEST_F()
61 EXPECT_TRUE(default_send_config.rtp.extensions.empty()) in TEST_F()
70 VerifyEmptyNackConfig(default_send_config.rtp.nack); in TEST_F()
71 VerifyEmptyUlpfecConfig(default_send_config.rtp.ulpfec); in TEST_F()
72 VerifyEmptyFlexfecConfig(default_send_config.rtp.flexfec); in TEST_F()
77 EXPECT_EQ(RtcpMode::kCompound, default_receive_config.rtp.rtcp_mode) in TEST_F()
79 EXPECT_FALSE(default_receive_config.rtp.lntf.enabled) in TEST_F()
82 default_receive_config.rtp.rtcp_xr.receiver_reference_time_report) in TEST_F()
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Dfec_tests.cc118 send_config->rtp.payload_name = "VP8"; in TEST_F()
119 send_config->rtp.payload_type = kVideoSendPayloadType; in TEST_F()
128 send_config->rtp.ulpfec.red_payload_type = kRedPayloadType; in TEST_F()
129 send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType; in TEST_F()
130 (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType; in TEST_F()
131 (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType; in TEST_F()
292 (*receive_configs)[0].rtp.local_ssrc = kVideoLocalSsrc; in ModifyVideoConfigs()
296 send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs; in ModifyVideoConfigs()
297 send_config->rtp.rtx.ssrcs.push_back(test::CallTest::kSendRtxSsrcs[0]); in ModifyVideoConfigs()
298 send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType; in ModifyVideoConfigs()
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Dretransmission_tests.cc110 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST_F()
111 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST_F()
185 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST_F()
186 local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc; in TEST_F()
187 remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc; in TEST_F()
336 send_config->rtp.nack.rtp_history_ms = rtp_history_ms_; in ReceivesPliAndRecovers()
337 (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_; in ReceivesPliAndRecovers()
443 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in DecodesRetransmittedFrame()
453 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in DecodesRetransmittedFrame()
456 send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType; in DecodesRetransmittedFrame()
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Dtransport_feedback_tests.cc181 send_config->rtp.extensions.clear(); in TEST()
182 send_config->rtp.extensions.push_back( in TEST()
195 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST()
196 send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[stream_index]); in TEST()
197 send_config->rtp.rtx.payload_type = kSendRtxPayloadType; in TEST()
199 send_config->rtp.ssrcs[0]; in TEST()
205 receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in TEST()
206 receive_config->rtp.extensions.clear(); in TEST()
207 receive_config->rtp.extensions.push_back( in TEST()
297 (*receive_configs)[0].rtp.transport_cc = feedback_enabled_; in ModifyVideoConfigs()
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Dhistogram_tests.cc95 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in VerifyHistogramStats()
96 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; in VerifyHistogramStats()
100 send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType; in VerifyHistogramStats()
101 send_config->rtp.ulpfec.red_payload_type = kRedPayloadType; in VerifyHistogramStats()
103 send_config->rtp.payload_name = "VP8"; in VerifyHistogramStats()
106 (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType; in VerifyHistogramStats()
107 (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType; in VerifyHistogramStats()
111 send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); in VerifyHistogramStats()
112 send_config->rtp.rtx.payload_type = kSendRtxPayloadType; in VerifyHistogramStats()
113 (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0]; in VerifyHistogramStats()
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/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
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/external/webrtc/media/base/
Drtp_utils.cc79 void UpdateRtpAuthTag(uint8_t* rtp, in UpdateRtpAuthTag() argument
96 uint8_t* auth_tag = rtp + (length - tag_length); in UpdateRtpAuthTag()
109 packet_time_params.srtp_auth_key.size(), rtp, in UpdateRtpAuthTag()
339 bool ValidateRtpHeader(const uint8_t* rtp, in ValidateRtpHeader() argument
350 size_t cc_count = rtp[0] & 0x0F; in ValidateRtpHeader()
358 if (!(rtp[0] & 0x10)) { in ValidateRtpHeader()
365 rtp += header_length_without_extension; in ValidateRtpHeader()
373 uint16_t extension_length_in_32bits = rtc::GetBE16(rtp + 2); in ValidateRtpHeader()
393 bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp, in UpdateRtpAbsSendTimeExtension() argument
411 if (!(rtp[0] & 0x10)) { in UpdateRtpAbsSendTimeExtension()
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/external/webrtc/call/
Drampup_tests.cc169 send_config->rtp.payload_name = "VP8"; in ModifyVideoConfigs()
183 send_config->rtp.extensions.clear(); in ModifyVideoConfigs()
190 send_config->rtp.extensions.push_back( in ModifyVideoConfigs()
195 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs()
200 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs()
204 send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs; in ModifyVideoConfigs()
205 send_config->rtp.ssrcs = video_ssrcs_; in ModifyVideoConfigs()
207 send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType; in ModifyVideoConfigs()
208 send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_; in ModifyVideoConfigs()
211 send_config->rtp.ulpfec.ulpfec_payload_type = in ModifyVideoConfigs()
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Dbitrate_estimator_tests.cc127 video_send_config.rtp.ssrcs.push_back(kVideoSendSsrcs[0]); in SetUp()
132 video_send_config.rtp.payload_name = "FAKE"; in SetUp()
133 video_send_config.rtp.payload_type = kFakeVideoSendPayloadType; in SetUp()
141 receive_config_.rtp.remote_ssrc = GetVideoSendConfig()->rtp.ssrcs[0]; in SetUp()
142 receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc; in SetUp()
143 receive_config_.rtp.extensions.push_back( in SetUp()
145 receive_config_.rtp.extensions.push_back( in SetUp()
177 test_->GetVideoSendConfig()->rtp.ssrcs[0]++;
195 decoder.payload_type = test_->GetVideoSendConfig()->rtp.payload_type;
197 SdpVideoFormat(test_->GetVideoSendConfig()->rtp.payload_name);
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Dcall.cc93 return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc); in UseSendSideBwe()
97 return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc); in UseSendSideBwe()
115 rtclog_config->remote_ssrc = config.rtp.remote_ssrc; in CreateRtcLogStreamConfig()
116 rtclog_config->local_ssrc = config.rtp.local_ssrc; in CreateRtcLogStreamConfig()
117 rtclog_config->rtx_ssrc = config.rtp.rtx_ssrc; in CreateRtcLogStreamConfig()
118 rtclog_config->rtcp_mode = config.rtp.rtcp_mode; in CreateRtcLogStreamConfig()
119 rtclog_config->rtp_extensions = config.rtp.extensions; in CreateRtcLogStreamConfig()
123 FindKeyByValue(config.rtp.rtx_associated_payload_types, d.payload_type); in CreateRtcLogStreamConfig()
134 rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index]; in CreateRtcLogStreamConfig()
135 if (ssrc_index < config.rtp.rtx.ssrcs.size()) { in CreateRtcLogStreamConfig()
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/external/libsrtp2/test/
Drtp_decoder.c358 srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp); in main()
362 srtp_crypto_policy_set_aes_gcm_128_8_auth(&policy.rtp); in main()
368 srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp); in main()
372 srtp_crypto_policy_set_aes_gcm_256_8_auth(&policy.rtp); in main()
387 &policy.rtp); in main()
392 &policy.rtp); in main()
401 &policy.rtp); in main()
406 &policy.rtp); in main()
421 &policy.rtp); in main()
426 &policy.rtp); in main()
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Drtpw.c371 srtp_crypto_policy_set_aes_gcm_128_8_auth(&policy.rtp); in main()
375 srtp_crypto_policy_set_aes_gcm_256_8_auth(&policy.rtp); in main()
387 srtp_crypto_policy_set_rtp_default(&policy.rtp); in main()
391 srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(&policy.rtp); in main()
405 srtp_crypto_policy_set_aes_cm_128_null_auth(&policy.rtp); in main()
409 srtp_crypto_policy_set_aes_cm_256_null_auth(&policy.rtp); in main()
420 srtp_crypto_policy_set_aes_gcm_128_8_only_auth(&policy.rtp); in main()
425 srtp_crypto_policy_set_aes_gcm_256_8_only_auth(&policy.rtp); in main()
436 srtp_crypto_policy_set_null_cipher_hmac_sha1_80(&policy.rtp); in main()
451 policy.rtp.sec_serv = sec_servs; in main()
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/external/webrtc/audio/voip/test/
Daudio_egress_unittest.cc120 RtpPacketReceived rtp; in TEST_F() local
122 rtp.Parse(packet, length); in TEST_F()
144 rtp.GetHeader(&header); in TEST_F()
145 size_t packet_length = rtp.size(); in TEST_F()
148 const uint8_t* payload = rtp.data() + header.headerLength; in TEST_F()
158 RtpPacketReceived rtp; in TEST_F() local
160 rtp.Parse(packet, length); in TEST_F()
180 rtp.GetHeader(&header); in TEST_F()
181 size_t packet_length = rtp.size(); in TEST_F()
184 const uint8_t* payload = rtp.data() + header.headerLength; in TEST_F()
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/external/webrtc/test/fuzzers/corpora/sdp-corpus/
Dsimulcast.2.sdp16 a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
17 a=extmap:13 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
20 a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
21 a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
22 a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
24 a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
25 a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
26 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
27 a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
Dsimulcast.1.sdp16 a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
17 a=extmap:13 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
20 a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
21 a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
22 a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
24 a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
25 a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
26 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
27 a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
/external/webrtc/rtc_tools/rtp_generator/
Drtp_generator.cc91 config.rtp.ssrcs.push_back(kDefaultSsrc); in ParseVideoSendStreamConfig()
98 &config.rtp.payload_name)) { in ParseVideoSendStreamConfig()
102 if (!IsValidCodecType(config.rtp.payload_name)) { in ParseVideoSendStreamConfig()
107 config.rtp.payload_type = in ParseVideoSendStreamConfig()
108 GetDefaultTypeForPayloadName(config.rtp.payload_name); in ParseVideoSendStreamConfig()
110 &config.rtp.payload_type)) { in ParseVideoSendStreamConfig()
113 << config.rtp.payload_type; in ParseVideoSendStreamConfig()
178 video_config.rtp = send_config.rtp; in RtpGenerator()
181 video_config.rtp.mid = "mid-" + std::to_string(stream_count); in RtpGenerator()
188 PayloadStringToCodecType(video_config.rtp.payload_name); in RtpGenerator()
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/external/webrtc/test/fuzzers/
Dvp8_replay_fuzzer.cc29 vp8_config.rtp.local_ssrc = 7731; in FuzzOneInput()
30 vp8_config.rtp.remote_ssrc = 1337; in FuzzOneInput()
31 vp8_config.rtp.rtx_ssrc = 100; in FuzzOneInput()
32 vp8_config.rtp.transport_cc = true; in FuzzOneInput()
33 vp8_config.rtp.nack.rtp_history_ms = 1000; in FuzzOneInput()
34 vp8_config.rtp.lntf.enabled = true; in FuzzOneInput()
Dvp9_replay_fuzzer.cc29 vp9_config.rtp.local_ssrc = 7731; in FuzzOneInput()
30 vp9_config.rtp.remote_ssrc = 1337; in FuzzOneInput()
31 vp9_config.rtp.rtx_ssrc = 100; in FuzzOneInput()
32 vp9_config.rtp.transport_cc = true; in FuzzOneInput()
33 vp9_config.rtp.nack.rtp_history_ms = 1000; in FuzzOneInput()
/external/webrtc/audio/
Daudio_send_stream.cc61 if (old_config && config.rtp.ssrc == old_config->rtp.ssrc && in UpdateEventLogStreamConfig()
62 config.rtp.extensions == old_config->rtp.extensions && in UpdateEventLogStreamConfig()
69 rtclog_config->local_ssrc = config.rtp.ssrc; in UpdateEventLogStreamConfig()
70 rtclog_config->rtp_extensions = config.rtp.extensions; in UpdateEventLogStreamConfig()
128 config.rtp.extmap_allow_mixed, in AudioSendStream()
130 config.rtp.ssrc, in AudioSendStream()
163 RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc; in AudioSendStream()
179 RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc; in ~AudioSendStream()
224 return FindExtensionIds(config.rtp.extensions).transport_sequence_number; in TransportSeqNumId()
240 RTC_DCHECK(first_time || old_config.rtp.ssrc == new_config.rtp.ssrc); in ConfigureStream()
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Daudio_receive_stream.cc56 ss << "{rtp: " << rtp.ToString(); in ToString()
81 event_log, config.rtp.local_ssrc, config.rtp.remote_ssrc, in CreateChannelReceive()
123 RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc; in AudioReceiveStream()
138 config.rtp.remote_ssrc, channel_receive_.get()); in AudioReceiveStream()
144 RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc; in ~AudioReceiveStream()
179 stats.remote_ssrc = config_.rtp.remote_ssrc; in GetStats()
295 return config_.rtp.remote_ssrc; in Ssrc()
304 return config_.rtp.remote_ssrc; in id()
388 old_config.rtp.remote_ssrc == new_config.rtp.remote_ssrc); in ConfigureStream()
398 RTC_DCHECK_EQ(old_config.rtp.local_ssrc, new_config.rtp.local_ssrc); in ConfigureStream()
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