| /external/chromium_org/content/browser/renderer_host/p2p/ |
| D | socket_host_unittest.cc | 307 std::vector<char> rtp_packet; in TEST() local
308 rtp_packet.resize(sizeof(kRtpMsgWithAbsSendTimeExtension) + 4); // tag length in TEST()
309 memcpy(&rtp_packet[0], kRtpMsgWithAbsSendTimeExtension, in TEST()
311 memcpy(&rtp_packet[sizeof(kRtpMsgWithAbsSendTimeExtension)], fake_tag, 4); in TEST()
314 &rtp_packet[0], rtp_packet.size(), options, 0)); in TEST()
316 EXPECT_EQ(0, memcmp(&rtp_packet[sizeof(kRtpMsgWithAbsSendTimeExtension)], in TEST()
320 EXPECT_EQ(0, memcmp(&rtp_packet[kAstIndexInRtpMsg], in TEST()
332 std::vector<char> rtp_packet; in TEST() local
333 rtp_packet.resize(sizeof(kRtpMsgWithAbsSendTimeExtension) + 4); // tag length in TEST()
334 memcpy(&rtp_packet[0], kRtpMsgWithAbsSendTimeExtension, in TEST()
[all …]
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | fec_test_helper.cc | 28 RtpPacket* rtp_packet = new RtpPacket; in NextPacket() local
30 rtp_packet->data[i + kRtpHeaderSize] = offset + i; in NextPacket()
31 rtp_packet->length = length + kRtpHeaderSize; in NextPacket()
32 memset(&rtp_packet->header, 0, sizeof(WebRtcRTPHeader)); in NextPacket()
33 rtp_packet->header.frameType = kVideoFrameDelta; in NextPacket()
34 rtp_packet->header.header.headerLength = kRtpHeaderSize; in NextPacket()
35 rtp_packet->header.header.markerBit = (num_packets_ == 1); in NextPacket()
36 rtp_packet->header.header.sequenceNumber = seq_num_; in NextPacket()
37 rtp_packet->header.header.timestamp = timestamp_; in NextPacket()
38 rtp_packet->header.header.payloadType = kVp8PayloadType; in NextPacket()
[all …]
|
| D | producer_fec_unittest.cc | 71 RtpPacket* rtp_packet = generator_->NextPacket(i, 10); in TEST_F() local
72 rtp_packets.push_back(rtp_packet); in TEST_F()
73 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data, in TEST_F()
74 rtp_packet->length, in TEST_F()
76 last_timestamp = rtp_packet->header.header.timestamp; in TEST_F()
113 RtpPacket* rtp_packet = generator_->NextPacket(i * kNumPackets + j, 10); in TEST_F() local
114 rtp_packets.push_back(rtp_packet); in TEST_F()
115 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data, in TEST_F()
116 rtp_packet->length, in TEST_F()
118 last_timestamp = rtp_packet->header.header.timestamp; in TEST_F()
|
| D | rtp_sender.cc | 1268 uint8_t *rtp_packet, const uint16_t rtp_packet_length, in UpdateTransmissionTimeOffset() argument
1296 if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) && in UpdateTransmissionTimeOffset()
1297 (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) { in UpdateTransmissionTimeOffset()
1304 if (rtp_packet[block_pos] != first_block_byte) { in UpdateTransmissionTimeOffset()
1309 RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, in UpdateTransmissionTimeOffset()
1313 bool RTPSender::UpdateAudioLevel(uint8_t *rtp_packet, in UpdateAudioLevel() argument
1341 if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) && in UpdateAudioLevel()
1342 (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) { in UpdateAudioLevel()
1348 if (rtp_packet[block_pos] != first_block_byte) { in UpdateAudioLevel()
1352 rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f); in UpdateAudioLevel()
[all …]
|
| D | 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/third_party/libjingle/source/talk/media/base/ |
| D | rtpdump_unittest.cc | 45 RtpDumpPacket rtp_packet(rtp_buf.Data(), rtp_buf.Length(), 0, false); in TEST() local
51 EXPECT_FALSE(rtp_packet.is_rtcp()); in TEST()
52 EXPECT_TRUE(rtp_packet.IsValidRtpPacket()); in TEST()
53 EXPECT_FALSE(rtp_packet.IsValidRtcpPacket()); in TEST()
54 EXPECT_TRUE(rtp_packet.GetRtpPayloadType(&type)); in TEST()
56 EXPECT_TRUE(rtp_packet.GetRtpSeqNum(&seq_num)); in TEST()
58 EXPECT_TRUE(rtp_packet.GetRtpTimestamp(&ts)); in TEST()
60 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc)); in TEST()
62 EXPECT_FALSE(rtp_packet.GetRtcpType(&type)); in TEST()
|
| D | testutils.cc | 187 RawRtpPacket rtp_packet; in VerifyTestPacketsFromStream() local
188 result &= rtp_packet.ReadFromByteBuffer(&buf); in VerifyTestPacketsFromStream()
189 result &= rtp_packet.SameExceptSeqNumTimestampSsrc( in VerifyTestPacketsFromStream()
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| D | vie_receiver.cc | 161 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, in ReceivedRTPPacket() argument
164 return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet), in ReceivedRTPPacket()
189 bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet, in OnRecoveredPacket() argument
192 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { in OnRecoveredPacket()
197 return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); in OnRecoveredPacket()
211 int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet, in InsertRTPPacket() argument
220 rtp_dump_->DumpPacket(rtp_packet, in InsertRTPPacket()
226 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, in InsertRTPPacket()
243 int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order) in InsertRTPPacket()
|
| D | 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,
|
| D | vie_channel.h | 284 int32_t ReceivedRTPPacket(const void* rtp_packet,
|
| D | vie_channel.cc | 1374 const void* rtp_packet, const int32_t rtp_packet_length, in ReceivedRTPPacket() argument
1383 rtp_packet, rtp_packet_length, packet_time); in ReceivedRTPPacket()
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| D | srtpfilter_unittest.cc | 90 char rtp_packet[sizeof(kPcmuFrame) + 10]; in TestProtectUnprotect() local
94 memcpy(rtp_packet, kPcmuFrame, rtp_len); in TestProtectUnprotect()
97 rtc::SetBE16(reinterpret_cast<uint8*>(rtp_packet) + 2, in TestProtectUnprotect()
99 memcpy(original_rtp_packet, rtp_packet, rtp_len); in TestProtectUnprotect()
102 EXPECT_TRUE(f1_.ProtectRtp(rtp_packet, rtp_len, in TestProtectUnprotect()
103 sizeof(rtp_packet), &out_len)); in TestProtectUnprotect()
105 EXPECT_NE(0, memcmp(rtp_packet, original_rtp_packet, rtp_len)); in TestProtectUnprotect()
106 EXPECT_TRUE(f2_.UnprotectRtp(rtp_packet, out_len, &out_len)); in TestProtectUnprotect()
108 EXPECT_EQ(0, memcmp(rtp_packet, original_rtp_packet, rtp_len)); in TestProtectUnprotect()
110 EXPECT_TRUE(f2_.ProtectRtp(rtp_packet, rtp_len, in TestProtectUnprotect()
[all …]
|
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| D | channel.cc | 561 bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet, in OnRecoveredPacket() argument
564 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { in OnRecoveredPacket()
573 return ReceivePacket(rtp_packet, rtp_packet_length, header, false); in OnRecoveredPacket()
|