| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| fec_test_helper.cc | 28 RtpPacket* rtp_packet = new RtpPacket; local
30 rtp_packet->data[i + kRtpHeaderSize] = offset + i;
31 rtp_packet->length = length + kRtpHeaderSize;
32 memset(&rtp_packet->header, 0, sizeof(WebRtcRTPHeader));
33 rtp_packet->header.frameType = kVideoFrameDelta;
34 rtp_packet->header.header.headerLength = kRtpHeaderSize;
35 rtp_packet->header.header.markerBit = (num_packets_ == 1);
36 rtp_packet->header.header.sequenceNumber = seq_num_;
37 rtp_packet->header.header.timestamp = timestamp_;
38 rtp_packet->header.header.payloadType = kVp8PayloadType
[all...] |
| producer_fec_unittest.cc | 71 RtpPacket* rtp_packet = generator_->NextPacket(i, 10); local
72 rtp_packets.push_back(rtp_packet);
73 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data,
74 rtp_packet->length,
76 last_timestamp = rtp_packet->header.header.timestamp;
113 RtpPacket* rtp_packet = generator_->NextPacket(i * kNumPackets + j, 10); local
114 rtp_packets.push_back(rtp_packet);
115 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data,
116 rtp_packet->length,
118 last_timestamp = rtp_packet->header.header.timestamp
[all...] |
| rtp_sender.cc | [all...] |
| rtp_sender.h | 167 bool UpdateAudioLevel(uint8_t *rtp_packet,
322 void UpdateTransmissionTimeOffset(uint8_t *rtp_packet,
326 void UpdateAbsoluteSendTime(uint8_t *rtp_packet,
|
| /external/chromium_org/content/browser/renderer_host/p2p/ |
| socket_host_unittest.cc | 307 std::vector<char> rtp_packet; local
308 rtp_packet.resize(sizeof(kRtpMsgWithAbsSendTimeExtension) + 4); // tag length
309 memcpy(&rtp_packet[0], kRtpMsgWithAbsSendTimeExtension,
311 memcpy(&rtp_packet[sizeof(kRtpMsgWithAbsSendTimeExtension)], fake_tag, 4);
314 &rtp_packet[0], rtp_packet.size(), options, 0));
316 EXPECT_EQ(0, memcmp(&rtp_packet[sizeof(kRtpMsgWithAbsSendTimeExtension)],
320 EXPECT_EQ(0, memcmp(&rtp_packet[kAstIndexInRtpMsg],
332 std::vector<char> rtp_packet; local
333 rtp_packet.resize(sizeof(kRtpMsgWithAbsSendTimeExtension) + 4); // tag lengt
367 std::vector<char> rtp_packet; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdump_unittest.cc | 45 RtpDumpPacket rtp_packet(rtp_buf.Data(), rtp_buf.Length(), 0, false);
51 EXPECT_FALSE(rtp_packet.is_rtcp());
52 EXPECT_TRUE(rtp_packet.IsValidRtpPacket());
53 EXPECT_FALSE(rtp_packet.IsValidRtcpPacket());
54 EXPECT_TRUE(rtp_packet.GetRtpPayloadType(&type));
56 EXPECT_TRUE(rtp_packet.GetRtpSeqNum(&seq_num));
58 EXPECT_TRUE(rtp_packet.GetRtpTimestamp(&ts));
60 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc));
62 EXPECT_FALSE(rtp_packet.GetRtcpType(&type));
|
| testutils.cc | 187 RawRtpPacket rtp_packet; local
188 result &= rtp_packet.ReadFromByteBuffer(&buf);
189 result &= rtp_packet.SameExceptSeqNumTimestampSsrc(
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_receiver.cc | 161 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
164 return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet),
189 bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
192 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
197 return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
211 int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
220 rtp_dump_->DumpPacket(rtp_packet,
226 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
243 int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order)
|
| vie_receiver.h | 72 int ReceivedRTPPacket(const void* rtp_packet, int rtp_packet_length,
92 int InsertRTPPacket(const uint8_t* rtp_packet, int rtp_packet_length,
|
| vie_channel.h | 284 int32_t ReceivedRTPPacket(const void* rtp_packet,
|
| vie_channel.cc | [all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| srtpfilter_unittest.cc | 90 char rtp_packet[sizeof(kPcmuFrame) + 10]; local
94 memcpy(rtp_packet, kPcmuFrame, rtp_len);
97 rtc::SetBE16(reinterpret_cast<uint8*>(rtp_packet) + 2,
99 memcpy(original_rtp_packet, rtp_packet, rtp_len);
102 EXPECT_TRUE(f1_.ProtectRtp(rtp_packet, rtp_len,
103 sizeof(rtp_packet), &out_len));
105 EXPECT_NE(0, memcmp(rtp_packet, original_rtp_packet, rtp_len));
106 EXPECT_TRUE(f2_.UnprotectRtp(rtp_packet, out_len, &out_len));
108 EXPECT_EQ(0, memcmp(rtp_packet, original_rtp_packet, rtp_len));
110 EXPECT_TRUE(f2_.ProtectRtp(rtp_packet, rtp_len
[all...] |
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| channel.cc | 561 bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
564 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
573 return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
[all...] |
