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Searched refs:rtp_packet (Results 1 – 17 of 17) sorted by relevance

/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
Drtc_event_log_source.cc40 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in GetRtpPacket() local
41 if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO || in GetRtpPacket()
42 !rtp_packet.has_incoming() || !rtp_packet.incoming() || in GetRtpPacket()
43 !rtp_packet.has_packet_length() || rtp_packet.packet_length() == 0 || in GetRtpPacket()
44 !rtp_packet.has_header() || rtp_packet.header().size() == 0 || in GetRtpPacket()
45 rtp_packet.packet_length() < rtp_packet.header().size()) in GetRtpPacket()
47 return &rtp_packet; in GetRtpPacket()
81 const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event); in NextPacket() local
83 if (rtp_packet) { in NextPacket()
84 uint8_t* packet_header = new uint8_t[rtp_packet->header().size()]; in NextPacket()
[all …]
/external/webrtc/webrtc/modules/rtp_rtcp/source/
Dfec_test_helper.cc29 RtpPacket* rtp_packet = new RtpPacket; in NextPacket() local
31 rtp_packet->data[i + kRtpHeaderSize] = offset + i; in NextPacket()
32 rtp_packet->length = length + kRtpHeaderSize; in NextPacket()
33 memset(&rtp_packet->header, 0, sizeof(WebRtcRTPHeader)); in NextPacket()
34 rtp_packet->header.frameType = kVideoFrameDelta; in NextPacket()
35 rtp_packet->header.header.headerLength = kRtpHeaderSize; in NextPacket()
36 rtp_packet->header.header.markerBit = (num_packets_ == 1); in NextPacket()
37 rtp_packet->header.header.sequenceNumber = seq_num_; in NextPacket()
38 rtp_packet->header.header.timestamp = timestamp_; in NextPacket()
39 rtp_packet->header.header.payloadType = kVp8PayloadType; in NextPacket()
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Dproducer_fec_unittest.cc120 RtpPacket* rtp_packet = generator_->NextPacket(i, 10); in TEST_F() local
121 rtp_packets.push_back(rtp_packet); in TEST_F()
122 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data, in TEST_F()
123 rtp_packet->length, in TEST_F()
125 last_timestamp = rtp_packet->header.header.timestamp; in TEST_F()
162 RtpPacket* rtp_packet = generator_->NextPacket(i * kNumPackets + j, 10); in TEST_F() local
163 rtp_packets.push_back(rtp_packet); in TEST_F()
164 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data, in TEST_F()
165 rtp_packet->length, in TEST_F()
167 last_timestamp = rtp_packet->header.header.timestamp; in TEST_F()
Drtp_sender.h80 virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
192 uint8_t* rtp_packet,
199 bool UpdateAudioLevel(uint8_t* rtp_packet,
205 bool UpdateVideoRotation(uint8_t* rtp_packet,
360 const uint8_t* rtp_packet,
365 void UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
369 void UpdateAbsoluteSendTime(uint8_t* rtp_packet,
376 uint16_t UpdateTransportSequenceNumber(uint8_t* rtp_packet,
Drtp_sender.cc1444 const uint8_t* rtp_packet, in FindHeaderExtensionPosition() argument
1469 if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) && in FindHeaderExtensionPosition()
1470 (rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) { in FindHeaderExtensionPosition()
1482 uint8_t* rtp_packet, in VerifyExtension() argument
1493 if (!FindHeaderExtensionPosition(extension_type, rtp_packet, in VerifyExtension()
1498 if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) && in VerifyExtension()
1499 (rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) { in VerifyExtension()
1507 if (rtp_packet[block_pos] != first_block_byte) in VerifyExtension()
1514 void RTPSender::UpdateTransmissionTimeOffset(uint8_t* rtp_packet, in UpdateTransmissionTimeOffset() argument
1520 switch (VerifyExtension(kRtpExtensionTransmissionTimeOffset, rtp_packet, in UpdateTransmissionTimeOffset()
[all …]
/external/webrtc/webrtc/call/
Drtc_event_log2rtp_dump.cc123 event.rtp_packet().has_header() && in main()
124 event.rtp_packet().header().size() >= 12 && in main()
125 event.rtp_packet().has_packet_length() && in main()
126 event.rtp_packet().has_type()) { in main()
127 const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in main() local
128 if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO) in main()
130 if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO) in main()
132 if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA) in main()
137 reinterpret_cast<const uint8_t*>(rtp_packet.header().data() + in main()
144 packet.length = rtp_packet.header().size(); in main()
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Drtc_event_log_unittest.cc233 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in VerifyRtpEvent() local
234 ASSERT_TRUE(rtp_packet.has_incoming()); in VerifyRtpEvent()
235 EXPECT_EQ(incoming, rtp_packet.incoming()); in VerifyRtpEvent()
236 ASSERT_TRUE(rtp_packet.has_type()); in VerifyRtpEvent()
237 EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type())); in VerifyRtpEvent()
238 ASSERT_TRUE(rtp_packet.has_packet_length()); in VerifyRtpEvent()
239 EXPECT_EQ(total_size, rtp_packet.packet_length()); in VerifyRtpEvent()
240 ASSERT_TRUE(rtp_packet.has_header()); in VerifyRtpEvent()
241 ASSERT_EQ(header_size, rtp_packet.header().size()); in VerifyRtpEvent()
243 EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i])); in VerifyRtpEvent()
Drtc_event_log.proto49 optional RtpPacket rtp_packet = 3; field
/external/webrtc/talk/media/base/
Drtpdump_unittest.cc45 RtpDumpPacket rtp_packet(rtp_buf.Data(), rtp_buf.Length(), 0, false); in TEST() local
52 EXPECT_FALSE(rtp_packet.is_rtcp()); in TEST()
53 EXPECT_TRUE(rtp_packet.IsValidRtpPacket()); in TEST()
54 EXPECT_FALSE(rtp_packet.IsValidRtcpPacket()); in TEST()
55 EXPECT_TRUE(rtp_packet.GetRtpPayloadType(&payload_type)); in TEST()
57 EXPECT_TRUE(rtp_packet.GetRtpSeqNum(&seq_num)); in TEST()
59 EXPECT_TRUE(rtp_packet.GetRtpTimestamp(&ts)); in TEST()
61 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc)); in TEST()
63 EXPECT_FALSE(rtp_packet.GetRtcpType(&rtcp_type)); in TEST()
Dtestutils.cc192 RawRtpPacket rtp_packet; in VerifyTestPacketsFromStream() local
193 result &= rtp_packet.ReadFromByteBuffer(&buf); in VerifyTestPacketsFromStream()
194 result &= rtp_packet.SameExceptSeqNumTimestampSsrc( in VerifyTestPacketsFromStream()
/external/webrtc/webrtc/video/
Dvie_receiver.cc222 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, in ReceivedRTPPacket() argument
225 return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet), in ReceivedRTPPacket()
250 bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet, in OnRecoveredPacket() argument
253 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { in OnRecoveredPacket()
258 return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); in OnRecoveredPacket()
261 int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet, in InsertRTPPacket() argument
272 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, in InsertRTPPacket()
308 int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order) in InsertRTPPacket()
Dvie_receiver.h77 int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length,
90 int InsertRTPPacket(const uint8_t* rtp_packet, size_t rtp_packet_length,
Dvie_channel.h206 int32_t ReceivedRTPPacket(const void* rtp_packet,
Dvie_channel.cc935 int32_t ViEChannel::ReceivedRTPPacket(const void* rtp_packet, in ReceivedRTPPacket() argument
939 rtp_packet, rtp_packet_length, packet_time); in ReceivedRTPPacket()
/external/webrtc/webrtc/audio/
Daudio_receive_stream_unittest.cc244 std::vector<uint8_t> rtp_packet = in TEST() local
253 rtp_packet.size() - kExpectedHeaderLength, in TEST()
257 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); in TEST()
270 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( in TEST() local
279 rtp_packet.size() - kExpectedHeaderLength, in TEST()
283 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); in TEST()
/external/webrtc/talk/session/media/
Dsrtpfilter_unittest.cc95 char rtp_packet[sizeof(kPcmuFrame) + 10]; in TestProtectUnprotect() local
99 memcpy(rtp_packet, kPcmuFrame, rtp_len); in TestProtectUnprotect()
102 rtc::SetBE16(reinterpret_cast<uint8_t*>(rtp_packet) + 2, in TestProtectUnprotect()
104 memcpy(original_rtp_packet, rtp_packet, rtp_len); in TestProtectUnprotect()
107 EXPECT_TRUE(f1_.ProtectRtp(rtp_packet, rtp_len, in TestProtectUnprotect()
108 sizeof(rtp_packet), &out_len)); in TestProtectUnprotect()
110 EXPECT_NE(0, memcmp(rtp_packet, original_rtp_packet, rtp_len)); in TestProtectUnprotect()
111 EXPECT_TRUE(f2_.UnprotectRtp(rtp_packet, out_len, &out_len)); in TestProtectUnprotect()
113 EXPECT_EQ(0, memcmp(rtp_packet, original_rtp_packet, rtp_len)); in TestProtectUnprotect()
115 EXPECT_TRUE(f2_.ProtectRtp(rtp_packet, rtp_len, in TestProtectUnprotect()
[all …]
/external/webrtc/webrtc/voice_engine/
Dchannel.cc508 bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet, in OnRecoveredPacket() argument
511 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { in OnRecoveredPacket()
520 return ReceivePacket(rtp_packet, rtp_packet_length, header, false); in OnRecoveredPacket()