| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| rtc_event_log_source.cc | 40 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); local
41 if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
42 !rtp_packet.has_incoming() || !rtp_packet.incoming() ||
43 !rtp_packet.has_packet_length() || rtp_packet.packet_length() == 0 ||
44 !rtp_packet.has_header() || rtp_packet.header().size() == 0 |
81 const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event); local
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| fec_test_helper.cc | 29 RtpPacket* rtp_packet = new RtpPacket; local
31 rtp_packet->data[i + kRtpHeaderSize] = offset + i;
32 rtp_packet->length = length + kRtpHeaderSize;
33 memset(&rtp_packet->header, 0, sizeof(WebRtcRTPHeader));
34 rtp_packet->header.frameType = kVideoFrameDelta;
35 rtp_packet->header.header.headerLength = kRtpHeaderSize;
36 rtp_packet->header.header.markerBit = (num_packets_ == 1);
37 rtp_packet->header.header.sequenceNumber = seq_num_;
38 rtp_packet->header.header.timestamp = timestamp_;
39 rtp_packet->header.header.payloadType = kVp8PayloadType
[all...] |
| producer_fec_unittest.cc | 120 RtpPacket* rtp_packet = generator_->NextPacket(i, 10); local
121 rtp_packets.push_back(rtp_packet);
122 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data,
123 rtp_packet->length,
125 last_timestamp = rtp_packet->header.header.timestamp;
162 RtpPacket* rtp_packet = generator_->NextPacket(i * kNumPackets + j, 10); local
163 rtp_packets.push_back(rtp_packet);
164 EXPECT_EQ(0, producer_->AddRtpPacketAndGenerateFec(rtp_packet->data,
165 rtp_packet->length,
167 last_timestamp = rtp_packet->header.header.timestamp
[all...] |
| rtp_sender.h | 80 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,
358 // |rtp_packet|. Return false if such extension doesn't exist.
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,
|
| rtp_sender.cc | [all...] |
| /external/webrtc/webrtc/call/ |
| rtc_event_log2rtp_dump.cc | 123 event.rtp_packet().has_header() &&
124 event.rtp_packet().header().size() >= 12 &&
125 event.rtp_packet().has_packet_length() &&
126 event.rtp_packet().has_type()) {
127 const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet(); local
128 if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
130 if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
132 if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
137 reinterpret_cast<const uint8_t*>(rtp_packet.header().data()
[all...] |
| rtc_event_log_unittest.cc | 233 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); local
234 ASSERT_TRUE(rtp_packet.has_incoming());
235 EXPECT_EQ(incoming, rtp_packet.incoming());
236 ASSERT_TRUE(rtp_packet.has_type());
237 EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
238 ASSERT_TRUE(rtp_packet.has_packet_length());
239 EXPECT_EQ(total_size, rtp_packet.packet_length());
240 ASSERT_TRUE(rtp_packet.has_header());
241 ASSERT_EQ(header_size, rtp_packet.header().size())
[all...] |
| /external/webrtc/talk/media/base/ |
| rtpdump_unittest.cc | 45 RtpDumpPacket rtp_packet(rtp_buf.Data(), rtp_buf.Length(), 0, false);
52 EXPECT_FALSE(rtp_packet.is_rtcp());
53 EXPECT_TRUE(rtp_packet.IsValidRtpPacket());
54 EXPECT_FALSE(rtp_packet.IsValidRtcpPacket());
55 EXPECT_TRUE(rtp_packet.GetRtpPayloadType(&payload_type));
57 EXPECT_TRUE(rtp_packet.GetRtpSeqNum(&seq_num));
59 EXPECT_TRUE(rtp_packet.GetRtpTimestamp(&ts));
61 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc));
63 EXPECT_FALSE(rtp_packet.GetRtcpType(&rtcp_type));
|
| testutils.cc | 192 RawRtpPacket rtp_packet; local
193 result &= rtp_packet.ReadFromByteBuffer(&buf);
194 result &= rtp_packet.SameExceptSeqNumTimestampSsrc(
|
| /external/webrtc/webrtc/audio/ |
| audio_receive_stream_unittest.cc | 244 std::vector<uint8_t> rtp_packet = local
253 rtp_packet.size() - kExpectedHeaderLength,
257 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
270 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( local
279 rtp_packet.size() - kExpectedHeaderLength,
283 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
|
| /external/webrtc/webrtc/video/ |
| vie_receiver.cc | 222 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
225 return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet),
250 bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
253 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
258 return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
261 int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
272 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
308 int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order)
|
| vie_receiver.h | 77 int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length,
90 int InsertRTPPacket(const uint8_t* rtp_packet, size_t rtp_packet_length,
|
| vie_channel.h | 206 int32_t ReceivedRTPPacket(const void* rtp_packet,
|
| vie_channel.cc | [all...] |
| /external/webrtc/talk/session/media/ |
| srtpfilter_unittest.cc | 95 char rtp_packet[sizeof(kPcmuFrame) + 10]; local
99 memcpy(rtp_packet, kPcmuFrame, rtp_len);
102 rtc::SetBE16(reinterpret_cast<uint8_t*>(rtp_packet) + 2,
104 memcpy(original_rtp_packet, rtp_packet, rtp_len);
107 EXPECT_TRUE(f1_.ProtectRtp(rtp_packet, rtp_len,
108 sizeof(rtp_packet), &out_len));
110 EXPECT_NE(0, memcmp(rtp_packet, original_rtp_packet, rtp_len));
111 EXPECT_TRUE(f2_.UnprotectRtp(rtp_packet, out_len, &out_len));
113 EXPECT_EQ(0, memcmp(rtp_packet, original_rtp_packet, rtp_len));
115 EXPECT_TRUE(f2_.ProtectRtp(rtp_packet, rtp_len
[all...] |
| /external/webrtc/webrtc/voice_engine/ |
| channel.cc | 508 bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
511 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
520 return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
[all...] |
