| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_generator.cc | 20 WebRtcRTPHeader* rtp_header) {
21 assert(rtp_header);
22 if (!rtp_header) {
25 rtp_header->header.sequenceNumber = seq_number_++;
26 rtp_header->header.timestamp = timestamp_;
28 rtp_header->header.payloadType = payload_type;
29 rtp_header->header.markerBit = false;
30 rtp_header->header.ssrc = ssrc_;
31 rtp_header->header.numCSRCs = 0;
32 rtp_header->frameType = kAudioFrameSpeech
[all...] |
| neteq_performance_test.cc | 57 WebRtcRTPHeader rtp_header; local
63 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
80 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
85 rtp_header, input_payload, payload_len,
94 &rtp_header);
|
| rtp_generator.h | 37 // Writes the next RTP header to |rtp_header|, which will be of type
41 WebRtcRTPHeader* rtp_header);
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_receiver_video.cc | 50 WebRtcRTPHeader* rtp_header,
58 "seqnum", rtp_header->header.sequenceNumber,
59 "timestamp", rtp_header->header.timestamp);
60 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
63 payload_length - rtp_header->header.paddingLength;
66 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
69 return ParseVideoCodecSpecific(rtp_header,
104 WebRtcRTPHeader* rtp_header,
110 switch (rtp_header->type.Video.codec) {
112 rtp_header->type.Video.isFirstPacket = is_first_packet
[all...] |
| rtp_receiver_audio.cc | 183 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
191 "seqnum", rtp_header->header.sequenceNumber,
192 "timestamp", rtp_header->header.timestamp);
193 rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs;
194 num_energy_ = rtp_header->type.Audio.numEnergy;
195 if (rtp_header->type.Audio.numEnergy > 0 &&
196 rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) {
198 rtp_header->type.Audio.arrOfEnergy,
199 rtp_header->type.Audio.numEnergy)
[all...] |
| rtp_receiver_video.h | 30 WebRtcRTPHeader* rtp_header,
64 int32_t ReceiveGenericCodec(WebRtcRTPHeader* rtp_header,
68 int32_t ReceiveVp8Codec(WebRtcRTPHeader* rtp_header,
72 int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
77 WebRtcRTPHeader* rtp_header,
|
| rtp_sender_unittest.cc | 46 const uint8_t* GetPayloadData(const RTPHeader& rtp_header,
48 return packet + rtp_header.headerLength;
51 uint16_t GetPayloadDataLength(const RTPHeader& rtp_header,
53 uint16_t length = packet_length - rtp_header.headerLength -
54 rtp_header.paddingLength;
109 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
110 EXPECT_EQ(kMarkerBit, rtp_header.markerBit);
111 EXPECT_EQ(payload_, rtp_header.payloadType);
112 EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
113 EXPECT_EQ(kTimestamp, rtp_header.timestamp)
208 webrtc::RTPHeader rtp_header; local
239 webrtc::RTPHeader rtp_header; local
280 webrtc::RTPHeader rtp_header; local
310 webrtc::RTPHeader rtp_header; local
348 webrtc::RTPHeader rtp_header; local
398 webrtc::RTPHeader rtp_header; local
476 webrtc::RTPHeader rtp_header; local
538 webrtc::RTPHeader rtp_header; local
581 webrtc::RTPHeader rtp_header; local
749 webrtc::RTPHeader rtp_header; local
1023 webrtc::RTPHeader rtp_header; local
1052 webrtc::RTPHeader rtp_header; local
[all...] |
| rtp_receiver_impl.cc | 175 const RTPHeader& rtp_header,
184 CheckSSRCChanged(rtp_header);
190 if (CheckPayloadChanged(rtp_header,
209 webrtc_rtp_header.header = rtp_header;
212 uint16_t payload_data_length = payload_length - rtp_header.paddingLength;
219 last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
220 last_received_timestamp_ != rtp_header.timestamp;
241 if (last_received_timestamp_ != rtp_header.timestamp) {
242 last_received_timestamp_ = rtp_header.timestamp;
245 last_received_sequence_number_ = rtp_header.sequenceNumber
[all...] |
| fec_receiver_impl.h | 31 virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
|
| rtp_receiver_impl.h | 49 const RTPHeader& rtp_header,
84 void CheckSSRCChanged(const RTPHeader& rtp_header);
85 void CheckCSRC(const WebRtcRTPHeader& rtp_header);
86 int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
|
| /external/chromium_org/media/cast/framer/ |
| frame_buffer.cc | 25 const RtpCastHeader& rtp_header) {
28 frame_id_ = rtp_header.frame_id;
29 max_packet_id_ = rtp_header.max_packet_id;
30 is_key_frame_ = rtp_header.is_key_frame;
32 DCHECK_EQ(rtp_header.frame_id, rtp_header.reference_frame_id);
33 last_referenced_frame_id_ = rtp_header.reference_frame_id;
34 rtp_timestamp_ = rtp_header.rtp_timestamp;
37 if (rtp_header.frame_id != frame_id_)
41 if (packets_.find(rtp_header.packet_id) != packets_.end())
[all...] |
| framer.cc | 34 const RtpCastHeader& rtp_header,
37 PacketType packet_type = frame_id_map_.InsertPacket(rtp_header);
43 << static_cast<int>(rtp_header.frame_id) << ", packet "
44 << rtp_header.packet_id;
50 FrameList::iterator it = frames_.find(rtp_header.frame_id);
54 frame_buffer->InsertPacket(payload_data, payload_size, rtp_header);
55 frames_.insert(std::make_pair(rtp_header.frame_id, frame_buffer));
58 it->second->InsertPacket(payload_data, payload_size, rtp_header);
|
| frame_id_map.cc | 62 PacketType FrameIdMap::InsertPacket(const RtpCastHeader& rtp_header) {
63 uint32 frame_id = rtp_header.frame_id;
65 reference_frame_id = rtp_header.reference_frame_id;
67 if (rtp_header.is_key_frame && waiting_for_key_) {
73 << " packet:" << static_cast<int>(rtp_header.packet_id)
74 << " max packet:" << static_cast<int>(rtp_header.max_packet_id);
92 rtp_header.max_packet_id,
93 rtp_header.is_key_frame));
97 packet_type = retval.first->second->InsertPacket(rtp_header.packet_id);
100 packet_type = it->second->InsertPacket(rtp_header.packet_id)
[all...] |
| frame_buffer.h | 25 const RtpCastHeader& rtp_header);
|
| framer.h | 40 const RtpCastHeader& rtp_header,
|
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/ |
| rtp_packetizer_unittest.cc | 43 void VerifyRtpHeader(const RtpCastTestHeader& rtp_header) {
44 VerifyCommonRtpHeader(rtp_header);
45 VerifyCastRtpHeader(rtp_header);
48 void VerifyCommonRtpHeader(const RtpCastTestHeader& rtp_header) {
49 EXPECT_EQ(kPayload, rtp_header.payload_type);
50 EXPECT_EQ(sequence_number_, rtp_header.sequence_number);
51 EXPECT_EQ(expectd_rtp_timestamp_, rtp_header.rtp_timestamp);
52 EXPECT_EQ(config_.ssrc, rtp_header.ssrc);
53 EXPECT_EQ(0, rtp_header.num_csrcs);
56 void VerifyCastRtpHeader(const RtpCastTestHeader& rtp_header) {
68 RtpCastTestHeader rtp_header; variable
70 VerifyRtpHeader(rtp_header); variable
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| fec_receiver.h | 25 virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
|
| receive_statistics.h | 54 virtual void IncomingPacket(const RTPHeader& rtp_header,
82 virtual void IncomingPacket(const RTPHeader& rtp_header,
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| rtcp.cc | 33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
36 int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_;
38 if (rtp_header.sequenceNumber < max_seq_no_) {
42 max_seq_no_ = rtp_header.sequenceNumber;
48 int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
54 transit_ = rtp_header.timestamp - receive_timestamp;
|
| neteq_impl_unittest.cc | 261 WebRtcRTPHeader rtp_header; local
262 rtp_header.header.payloadType = kPayloadType;
263 rtp_header.header.sequenceNumber = kFirstSequenceNumber;
264 rtp_header.header.timestamp = kFirstTimestamp;
265 rtp_header.header.ssrc = kSsrc;
320 .WillOnce(Return(&rtp_header.header));
354 neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime);
357 rtp_header.header.timestamp += 160;
358 rtp_header.header.sequenceNumber += 1;
359 neteq_->InsertPacket(rtp_header, payload, kPayloadLength
372 WebRtcRTPHeader rtp_header; local
414 WebRtcRTPHeader rtp_header; local
[all...] |
| rtcp.h | 35 void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| receiver_tests.h | 33 const webrtc::WebRtcRTPHeader* rtp_header) {
34 return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
|
| /external/chromium_org/media/cast/rtp_receiver/rtp_parser/ |
| rtp_parser.h | 25 // pointed to by |rtp_header| and sets the |payload_data| pointer and
32 RtpCastHeader* rtp_header,
|
| /external/chromium_org/media/cast/receiver/ |
| frame_receiver.cc | 83 RtpCastHeader rtp_header; local
88 &rtp_header,
94 ProcessParsedPacket(rtp_header, payload_data, payload_size);
95 stats_.UpdateStatistics(rtp_header);
116 void FrameReceiver::ProcessParsedPacket(const RtpCastHeader& rtp_header,
123 frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] =
124 rtp_header.rtp_timestamp;
126 now, PACKET_RECEIVED, event_media_type_, rtp_header.rtp_timestamp,
127 rtp_header.frame_id, rtp_header.packet_id, rtp_header.max_packet_id
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| audio_coding_module_unittest.cc | 46 void Populate(WebRtcRTPHeader* rtp_header) {
47 rtp_header->header.sequenceNumber = 0xABCD;
48 rtp_header->header.timestamp = 0xABCDEF01;
49 rtp_header->header.payloadType = payload_type_;
50 rtp_header->header.markerBit = false;
51 rtp_header->header.ssrc = 0x1234;
52 rtp_header->header.numCSRCs = 0;
53 rtp_header->frameType = kAudioFrameSpeech;
55 rtp_header->header.payload_type_frequency = kSampleRateHz;
56 rtp_header->type.Audio.channel = 1
[all...] |
