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Searched refs:rtp_packet (Results 1 – 25 of 62) sorted by relevance

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/external/webrtc/modules/rtp_rtcp/source/
Dulpfec_receiver_impl.cc78 const RtpPacketReceived& rtp_packet, in AddReceivedRedPacket() argument
80 if (rtp_packet.Ssrc() != ssrc_) { in AddReceivedRedPacket()
85 if (rtp_packet.size() > IP_PACKET_SIZE) { in AddReceivedRedPacket()
94 if (rtp_packet.payload_size() == 0) { in AddReceivedRedPacket()
105 uint8_t payload_type = rtp_packet.payload()[0] & 0x7f; in AddReceivedRedPacket()
107 received_packet->is_recovered = rtp_packet.recovered(); in AddReceivedRedPacket()
108 received_packet->ssrc = rtp_packet.Ssrc(); in AddReceivedRedPacket()
109 received_packet->seq_num = rtp_packet.SequenceNumber(); in AddReceivedRedPacket()
111 if (rtp_packet.payload()[0] & 0x80) { in AddReceivedRedPacket()
119 packet_counter_.num_bytes += rtp_packet.size(); in AddReceivedRedPacket()
[all …]
Dulpfec_generator_unittest.cc121 RtpPacketToSend rtp_packet(nullptr); in TEST_F() local
122 EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size())); in TEST_F()
123 ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet); in TEST_F()
159 RtpPacketToSend rtp_packet(nullptr); in TEST_F() local
160 EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size())); in TEST_F()
161 ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet); in TEST_F()
187 RtpPacketToSend rtp_packet(nullptr); in TEST_F() local
188 EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size())); in TEST_F()
189 EXPECT_EQ(rtp_packet.headers_size(), kShortRtpHeaderLength); in TEST_F()
190 ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet); in TEST_F()
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Drtp_format_h264.cc237 bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet) { in NextPacket() argument
238 RTC_DCHECK(rtp_packet); in NextPacket()
247 uint8_t* buffer = rtp_packet->AllocatePayload(bytes_to_send); in NextPacket()
252 NextAggregatePacket(rtp_packet); in NextPacket()
254 NextFragmentPacket(rtp_packet); in NextPacket()
256 rtp_packet->SetMarker(packets_.empty()); in NextPacket()
261 void RtpPacketizerH264::NextAggregatePacket(RtpPacketToSend* rtp_packet) { in NextAggregatePacket() argument
263 size_t payload_capacity = rtp_packet->FreeCapacity(); in NextAggregatePacket()
265 uint8_t* buffer = rtp_packet->AllocatePayload(payload_capacity); in NextAggregatePacket()
290 rtp_packet->SetPayloadSize(index); in NextAggregatePacket()
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Drtp_format_h264.h45 bool NextPacket(RtpPacketToSend* rtp_packet) override;
78 void NextAggregatePacket(RtpPacketToSend* rtp_packet);
79 void NextFragmentPacket(RtpPacketToSend* rtp_packet);
Dflexfec_sender_unittest.cc64 RtpPacketToSend rtp_packet(nullptr); // No header extensions. in GenerateSingleFlexfecPacket() local
65 rtp_packet.Parse(packet->data); in GenerateSingleFlexfecPacket()
66 sender->AddPacketAndGenerateFec(rtp_packet); in GenerateSingleFlexfecPacket()
132 RtpPacketToSend rtp_packet(nullptr); in TEST() local
133 rtp_packet.Parse(packet->data); in TEST()
134 sender.AddPacketAndGenerateFec(rtp_packet); in TEST()
171 RtpPacketToSend rtp_packet(nullptr); in TEST() local
172 rtp_packet.Parse(packet->data); in TEST()
173 sender.AddPacketAndGenerateFec(rtp_packet); in TEST()
/external/webrtc/video/
Drtp_video_stream_receiver2_unittest.cc372 RtpPacketReceived rtp_packet; in TEST_F() local
374 rtp_packet.SetPayloadType(kPayloadType); in TEST_F()
375 rtp_packet.SetSequenceNumber(1); in TEST_F()
381 rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet, in TEST_F()
391 RtpPacketReceived rtp_packet(&extension_map); in TEST_F() local
392 rtp_packet.SetPayloadType(kPayloadType); in TEST_F()
394 rtp_packet.SetSequenceNumber(1); in TEST_F()
395 rtp_packet.SetTimestamp(1); in TEST_F()
396 rtp_packet.SetSsrc(kSsrc); in TEST_F()
397 rtp_packet.SetExtension<AbsoluteCaptureTimeExtension>( in TEST_F()
[all …]
Drtp_video_stream_receiver_unittest.cc367 RtpPacketReceived rtp_packet; in TEST_F() local
369 rtp_packet.SetPayloadType(kPayloadType); in TEST_F()
370 rtp_packet.SetSequenceNumber(1); in TEST_F()
376 rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet, in TEST_F()
386 RtpPacketReceived rtp_packet(&extension_map); in TEST_F() local
387 rtp_packet.SetPayloadType(kPayloadType); in TEST_F()
389 rtp_packet.SetSequenceNumber(1); in TEST_F()
390 rtp_packet.SetTimestamp(1); in TEST_F()
391 rtp_packet.SetSsrc(kSsrc); in TEST_F()
392 rtp_packet.SetExtension<AbsoluteCaptureTimeExtension>( in TEST_F()
[all …]
Dvideo_send_stream_tests.cc198 RtpPacket rtp_packet(&extensions_); in TEST_F() local
199 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
202 EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>()); in TEST_F()
203 EXPECT_TRUE(rtp_packet.GetExtension<AbsoluteSendTime>(&abs_send_time)); in TEST_F()
253 RtpPacket rtp_packet(&extensions_); in TEST_F() local
254 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
257 EXPECT_TRUE(rtp_packet.GetExtension<TransmissionOffset>(&toffset)); in TEST_F()
258 EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>()); in TEST_F()
300 RtpPacket rtp_packet(&extensions_); in TEST_F() local
301 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
[all …]
Drtp_video_stream_receiver2.cc361 const RtpPacketReceived& rtp_packet, in ParseGenericDependenciesExtension() argument
363 if (rtp_packet.HasExtension<RtpDependencyDescriptorExtension>()) { in ParseGenericDependenciesExtension()
365 if (!rtp_packet.GetExtension<RtpDependencyDescriptorExtension>( in ParseGenericDependenciesExtension()
372 RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc() in ParseGenericDependenciesExtension()
378 RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc() in ParseGenericDependenciesExtension()
431 if (!rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>( in ParseGenericDependenciesExtension()
466 const RtpPacketReceived& rtp_packet, in OnReceivedPayloadData() argument
470 rtp_packet, video, ntp_estimator_.Estimate(rtp_packet.Timestamp()), in OnReceivedPayloadData()
487 video_header.is_last_packet_in_frame |= rtp_packet.Marker(); in OnReceivedPayloadData()
495 rtp_packet.GetExtension<VideoOrientation>(&video_header.rotation); in OnReceivedPayloadData()
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Drtp_video_stream_receiver.cc393 const RtpPacketReceived& rtp_packet, in ParseGenericDependenciesExtension() argument
395 if (rtp_packet.HasExtension<RtpDependencyDescriptorExtension>()) { in ParseGenericDependenciesExtension()
397 if (!rtp_packet.GetExtension<RtpDependencyDescriptorExtension>( in ParseGenericDependenciesExtension()
404 RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc() in ParseGenericDependenciesExtension()
410 RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc() in ParseGenericDependenciesExtension()
463 if (!rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>( in ParseGenericDependenciesExtension()
498 const RtpPacketReceived& rtp_packet, in OnReceivedPayloadData() argument
502 rtp_packet, video, ntp_estimator_.Estimate(rtp_packet.Timestamp()), in OnReceivedPayloadData()
519 video_header.is_last_packet_in_frame |= rtp_packet.Marker(); in OnReceivedPayloadData()
527 rtp_packet.GetExtension<VideoOrientation>(&video_header.rotation); in OnReceivedPayloadData()
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Dvideo_analyzer.cc228 RtpPacket rtp_packet; in DeliverPacket() local
229 rtp_packet.Parse(packet); in DeliverPacket()
230 if (!IsFlexfec(rtp_packet.PayloadType()) && in DeliverPacket()
231 (rtp_packet.Ssrc() == ssrc_to_analyze_ || in DeliverPacket()
232 rtp_packet.Ssrc() == rtx_ssrc_to_analyze_)) { in DeliverPacket()
239 wrap_handler_.Unwrap(rtp_packet.Timestamp() - rtp_timestamp_delta_); in DeliverPacket()
269 RtpPacket rtp_packet; in SendRtp() local
270 rtp_packet.Parse(packet, length); in SendRtp()
277 if (rtp_timestamp_delta_ == 0 && rtp_packet.Ssrc() == ssrc_to_analyze_) { in SendRtp()
279 rtp_timestamp_delta_ = rtp_packet.Timestamp() - *first_sent_timestamp_; in SendRtp()
[all …]
Dpicture_id_tests.cc85 RtpPacket rtp_packet; in ParsePayload() local
86 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in ParsePayload()
87 EXPECT_TRUE(rtp_packet.Ssrc() == test::CallTest::kVideoSendSsrcs[0] || in ParsePayload()
88 rtp_packet.Ssrc() == test::CallTest::kVideoSendSsrcs[1] || in ParsePayload()
89 rtp_packet.Ssrc() == test::CallTest::kVideoSendSsrcs[2]) in ParsePayload()
92 if (rtp_packet.payload_size() == 0) { in ParsePayload()
96 parsed->timestamp = rtp_packet.Timestamp(); in ParsePayload()
97 parsed->ssrc = rtp_packet.Ssrc(); in ParsePayload()
100 depacketizer_->Parse(rtp_packet.PayloadBuffer()); in ParsePayload()
/external/webrtc/media/base/
Drtp_utils_unittest.cc287 std::vector<uint8_t> rtp_packet( in TEST() local
291 rtp_packet.insert(rtp_packet.end(), kFakeTag, kFakeTag + sizeof(kFakeTag)); in TEST()
292 EXPECT_TRUE(ApplyPacketOptions(&rtp_packet[0], rtp_packet.size(), in TEST()
297 memcmp(&rtp_packet[sizeof(kRtpMsgWithOneByteAbsSendTimeExtension)], in TEST()
301 EXPECT_EQ(0, memcmp(&rtp_packet[kAstIndexInOneByteRtpMsg], kTestAstValue, in TEST()
312 std::vector<uint8_t> rtp_packet( in TEST() local
316 rtp_packet.insert(rtp_packet.end(), kFakeTag, kFakeTag + sizeof(kFakeTag)); in TEST()
317 EXPECT_TRUE(ApplyPacketOptions(&rtp_packet[0], rtp_packet.size(), in TEST()
322 memcmp(&rtp_packet[sizeof(kRtpMsgWithOneByteAbsSendTimeExtension)], in TEST()
326 EXPECT_EQ(0, memcmp(&rtp_packet[kAstIndexInOneByteRtpMsg], kTestAstValue, in TEST()
[all …]
/external/webrtc/video/end_to_end_tests/
Dfec_tests.cc64 RtpPacket rtp_packet; in TEST_F() local
65 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
67 EXPECT_TRUE(rtp_packet.PayloadType() == kVideoSendPayloadType || in TEST_F()
68 rtp_packet.PayloadType() == kRedPayloadType) in TEST_F()
70 EXPECT_EQ(kVideoSendSsrcs[0], rtp_packet.Ssrc()) in TEST_F()
75 if (rtp_packet.PayloadType() == kRedPayloadType) { in TEST_F()
76 encapsulated_payload_type = rtp_packet.payload()[0]; in TEST_F()
92 dropped_sequence_numbers_.insert(rtp_packet.SequenceNumber()); in TEST_F()
93 dropped_timestamps_.insert(rtp_packet.Timestamp()); in TEST_F()
174 RtpPacket rtp_packet; in OnSendRtp() local
[all …]
Dretransmission_tests.cc63 RtpPacket rtp_packet; in TEST_F() local
64 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
67 if (dropped_packets_.find(rtp_packet.SequenceNumber()) != in TEST_F()
69 retransmitted_packets_.insert(rtp_packet.SequenceNumber()); in TEST_F()
89 if (packets_left_to_drop_ > 0 && rtp_packet.padding_size() == 0) { in TEST_F()
91 dropped_packets_.insert(rtp_packet.SequenceNumber()); in TEST_F()
157 RtpPacket rtp_packet; in TEST_F() local
158 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
161 sequence_number_to_retransmit_ = rtp_packet.SequenceNumber(); in TEST_F()
164 } else if (rtp_packet.SequenceNumber() == in TEST_F()
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Dtransport_feedback_tests.cc75 RtpPacket rtp_packet(&extensions_); in TEST() local
76 EXPECT_TRUE(rtp_packet.Parse(data, length)); in TEST()
80 EXPECT_TRUE(rtp_packet.GetExtension<TransportSequenceNumber>( in TEST()
91 if (rtp_packet.PayloadType() != kSendRtxPayloadType && in TEST()
92 rtp_packet.payload_size() > 0 && in TEST()
94 dropped_seq_[rtp_packet.Ssrc()].insert(rtp_packet.SequenceNumber()); in TEST()
98 if (rtp_packet.payload_size() == 0) { in TEST()
100 } else if (rtp_packet.PayloadType() == kSendRtxPayloadType) { in TEST()
103 rtp_packet.payload().data()); in TEST()
105 rtx_to_media_ssrcs_.find(rtp_packet.Ssrc())->second; in TEST()
[all …]
Dssrc_tests.cc148 RtpPacket rtp_packet; in TestSendsSetSsrcs() local
149 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TestSendsSetSsrcs()
151 EXPECT_TRUE(valid_ssrcs_[rtp_packet.Ssrc()]) in TestSendsSetSsrcs()
152 << "Received unknown SSRC: " << rtp_packet.Ssrc(); in TestSendsSetSsrcs()
154 if (!valid_ssrcs_[rtp_packet.Ssrc()]) in TestSendsSetSsrcs()
157 if (!is_observed_[rtp_packet.Ssrc()]) { in TestSendsSetSsrcs()
158 is_observed_[rtp_packet.Ssrc()] = true; in TestSendsSetSsrcs()
272 RtpPacket rtp_packet; in TEST_F() local
273 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
275 if (!registered_rtx_ssrc_[rtp_packet.Ssrc()]) in TEST_F()
[all …]
Dmulti_codec_receive_tests.cc80 RtpPacket rtp_packet; in OnSendRtp() local
81 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in OnSendRtp()
82 EXPECT_EQ(rtp_packet.Ssrc(), test::CallTest::kVideoSendSsrcs[0]); in OnSendRtp()
83 if (rtp_packet.payload_size() == 0) in OnSendRtp()
87 rtp_packet.PayloadType() != expected_payload_type_.value()) { in OnSendRtp()
91 if (!last_timestamp_ || rtp_packet.Timestamp() != *last_timestamp_) { in OnSendRtp()
98 sent_timestamps_.push_back(rtp_packet.Timestamp()); in OnSendRtp()
101 last_timestamp_ = rtp_packet.Timestamp(); in OnSendRtp()
/external/webrtc/test/
Dlayer_filtering_transport.cc86 RtpPacket rtp_packet; in SendRtp() local
87 rtp_packet.Parse(packet, length); in SendRtp()
89 if (rtp_packet.Ssrc() < ssrc_to_filter_min_ || in SendRtp()
90 rtp_packet.Ssrc() > ssrc_to_filter_max_) { in SendRtp()
95 if (rtp_packet.PayloadType() == vp8_video_payload_type_ || in SendRtp()
96 rtp_packet.PayloadType() == vp9_video_payload_type_) { in SendRtp()
97 const bool is_vp8 = rtp_packet.PayloadType() == vp8_video_payload_type_; in SendRtp()
100 if (auto parsed_payload = depacketizer.Parse(rtp_packet.PayloadBuffer())) { in SendRtp()
138 rtp_packet.SetMarker(true); in SendRtp()
159 rtp_packet.SetPayloadSize(0); in SendRtp()
[all …]
/external/webrtc/audio/
Daudio_send_stream_tests.cc85 RtpPacket rtp_packet; in TEST_F() local
86 EXPECT_TRUE(rtp_packet.Parse(packet, length)); // rtp packet is valid. in TEST_F()
115 RtpPacket rtp_packet(&extensions_); in TEST_F() local
116 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST_F()
120 EXPECT_TRUE(rtp_packet.GetExtension<AudioLevel>(&voice, &audio_level)); in TEST_F()
161 RtpPacket rtp_packet(&extensions_); in OnSendRtp() local
162 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in OnSendRtp()
164 EXPECT_EQ(rtp_packet.HasExtension<TransportSequenceNumber>(), in OnSendRtp()
166 EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>()); in OnSendRtp()
167 EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>()); in OnSendRtp()
[all …]
/external/webrtc/test/fuzzers/
Drtp_packetizer_av1_fuzzer.cc46 RtpPacketToSend rtp_packet(nullptr); in FuzzOneInput() local
49 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput()
50 RTC_CHECK_LE(rtp_packet.payload_size(), in FuzzOneInput()
55 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput()
56 RTC_CHECK_LE(rtp_packet.payload_size(), in FuzzOneInput()
60 RTC_CHECK(packetizer.NextPacket(&rtp_packet)) in FuzzOneInput()
62 RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len) in FuzzOneInput()
66 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput()
67 RTC_CHECK_LE(rtp_packet.payload_size(), in FuzzOneInput()
/external/webrtc/call/
Drtp_video_sender_unittest.cc437 RtpPacket rtp_packet; in TEST() local
438 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST()
439 rtp_sequence_numbers.push_back(rtp_packet.SequenceNumber()); in TEST()
466 RtpPacket rtp_packet; in TEST() local
467 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST()
468 EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1); in TEST()
470 rtc::ArrayView<const uint8_t> payload = rtp_packet.payload(); in TEST()
505 RtpPacket rtp_packet; in TEST() local
506 EXPECT_TRUE(rtp_packet.Parse(packet, length)); in TEST()
507 EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1); in TEST()
[all …]
/external/webrtc/modules/video_coding/
Dpacket_buffer.cc39 PacketBuffer::Packet::Packet(const RtpPacketReceived& rtp_packet, in Packet() argument
43 : marker_bit(rtp_packet.Marker()), in Packet()
44 payload_type(rtp_packet.PayloadType()), in Packet()
45 seq_num(rtp_packet.SequenceNumber()), in Packet()
46 timestamp(rtp_packet.Timestamp()), in Packet()
50 packet_info(rtp_packet.Ssrc(), in Packet()
51 rtp_packet.Csrcs(), in Packet()
52 rtp_packet.Timestamp(), in Packet()
54 rtp_packet.GetExtension<AbsoluteCaptureTimeExtension>(), in Packet()
/external/webrtc/rtc_tools/py_event_log_analyzer/
Dpb_parse.py48 return [DataPoint(event.rtp_packet.header,
49 event.rtp_packet.packet_length,
50 event.timestamp_us, event.rtp_packet.incoming)
/external/webrtc/rtc_tools/rtp_generator/
Drtp_generator.cc271 test::RtpPacket rtp_packet = DataToRtpPacket(packet, length); in SendRtp() local
272 rtp_dump_writer_->WritePacket(&rtp_packet); in SendRtp()
314 webrtc::test::RtpPacket rtp_packet; in DataToRtpPacket() local
315 memcpy(rtp_packet.data, packet, packet_len); in DataToRtpPacket()
316 rtp_packet.length = packet_len; in DataToRtpPacket()
317 rtp_packet.original_length = packet_len; in DataToRtpPacket()
318 rtp_packet.time_ms = in DataToRtpPacket()
320 return rtp_packet; in DataToRtpPacket()

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