Searched refs:rtp_header (Results 1 - 25 of 60) sorted by relevance

123

/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/
H A Drtp_generator.cc20 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
18 GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader* rtp_header) argument
47 GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader* rtp_header) argument
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H A Dneteq_performance_test.cc57 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);
H A Drtp_generator.h39 // Writes the next RTP header to |rtp_header|, which will be of type
44 WebRtcRTPHeader* rtp_header);
73 WebRtcRTPHeader* rtp_header) OVERRIDE;
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtp_receiver_video.cc50 int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, argument
60 rtp_header->header.sequenceNumber,
62 rtp_header->header.timestamp);
63 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
66 payload_length - rtp_header->header.paddingLength;
69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
74 RtpDepacketizer::Create(rtp_header->type.Video.codec, data_callback_));
80 rtp_header->type.Video.isFirstPacket = is_first_packet;
81 return depacketizer->Parse(rtp_header, payload, payload_data_length) ? 0 : -1;
110 int32_t RTPReceiverVideo::BuildRTPheader(const WebRtcRTPHeader* rtp_header, argument
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H A Drtp_format_vp8_unittest.cc414 WebRtcRTPHeader rtp_header; local
415 memset(&rtp_header, 0, sizeof(rtp_header));
418 EXPECT_TRUE(depacketizer_->Parse(&rtp_header, packet, sizeof(packet)));
420 EXPECT_EQ(kVideoFrameDelta, rtp_header.frameType);
421 VerifyBasicHeader(&rtp_header, 0, 1, 4);
423 &rtp_header, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
435 WebRtcRTPHeader rtp_header; local
436 memset(&rtp_header, 0, sizeof(rtp_header));
468 WebRtcRTPHeader rtp_header; local
486 WebRtcRTPHeader rtp_header; local
505 WebRtcRTPHeader rtp_header; local
526 WebRtcRTPHeader rtp_header; local
543 WebRtcRTPHeader rtp_header; local
567 WebRtcRTPHeader rtp_header; local
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H A Drtp_receiver_audio.cc183 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, argument
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
288 ParseAudioCodecSpecific( WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_length, const AudioPayload& audio_specific, bool is_red) argument
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H A Drtp_sender_unittest.cc47 const uint8_t* GetPayloadData(const RTPHeader& rtp_header, argument
49 return packet + rtp_header.headerLength;
52 uint16_t GetPayloadDataLength(const RTPHeader& rtp_header, argument
54 uint16_t length = packet_length - rtp_header.headerLength -
55 rtp_header.paddingLength;
110 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) { argument
111 EXPECT_EQ(kMarkerBit, rtp_header.markerBit);
112 EXPECT_EQ(payload_, rtp_header.payloadType);
113 EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
114 EXPECT_EQ(kTimestamp, rtp_header
209 webrtc::RTPHeader rtp_header; local
240 webrtc::RTPHeader rtp_header; local
281 webrtc::RTPHeader rtp_header; local
311 webrtc::RTPHeader rtp_header; local
349 webrtc::RTPHeader rtp_header; local
399 webrtc::RTPHeader rtp_header; local
477 webrtc::RTPHeader rtp_header; local
539 webrtc::RTPHeader rtp_header; local
582 webrtc::RTPHeader rtp_header; local
752 webrtc::RTPHeader rtp_header; local
1026 webrtc::RTPHeader rtp_header; local
1055 webrtc::RTPHeader rtp_header; local
1113 webrtc::RTPHeader rtp_header; local
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H A Drtp_receiver_impl.cc164 const RTPHeader& rtp_header,
173 CheckSSRCChanged(rtp_header);
179 if (CheckPayloadChanged(rtp_header,
198 webrtc_rtp_header.header = rtp_header;
201 uint16_t payload_data_length = payload_length - rtp_header.paddingLength;
208 last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
209 last_received_timestamp_ != rtp_header.timestamp;
230 if (last_received_timestamp_ != rtp_header.timestamp) {
231 last_received_timestamp_ = rtp_header.timestamp;
234 last_received_sequence_number_ = rtp_header
163 IncomingRtpPacket( const RTPHeader& rtp_header, const uint8_t* payload, int payload_length, PayloadUnion payload_specific, bool in_order) argument
265 CheckSSRCChanged(const RTPHeader& rtp_header) argument
336 CheckPayloadChanged( const RTPHeader& rtp_header, const int8_t first_payload_byte, bool& is_red, PayloadUnion* specific_payload, bool* should_reset_statistics) argument
429 CheckCSRC(const WebRtcRTPHeader& rtp_header) argument
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H A Drtp_receiver_video.h29 virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
61 int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
H A Drtp_format_h264.cc40 void ParseSingleNalu(WebRtcRTPHeader* rtp_header, argument
43 rtp_header->type.Video.codec = kRtpVideoH264;
44 rtp_header->type.Video.isFirstPacket = true;
45 RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264;
59 rtp_header->frameType = kVideoFrameKey;
62 rtp_header->frameType = kVideoFrameDelta;
67 void ParseFuaNalu(WebRtcRTPHeader* rtp_header, argument
85 rtp_header->frameType = kVideoFrameKey;
87 rtp_header->frameType = kVideoFrameDelta;
89 rtp_header
297 Parse(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_data_length) argument
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H A Drtp_format_video_generic.cc93 bool RtpDepacketizerGeneric::Parse(WebRtcRTPHeader* rtp_header, argument
99 rtp_header->frameType =
103 rtp_header->type.Video.isFirstPacket =
107 payload_data, payload_data_length, rtp_header) != 0) {
H A Drtp_receiver_impl.h47 const RTPHeader& rtp_header,
73 void CheckSSRCChanged(const RTPHeader& rtp_header);
74 void CheckCSRC(const WebRtcRTPHeader& rtp_header);
75 int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
H A Dfec_receiver_impl.h31 virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
H A Drtp_format_vp8.cc131 int ParseVP8FrameSize(WebRtcRTPHeader* rtp_header, argument
134 assert(rtp_header != NULL);
135 if (rtp_header->frameType != kVideoFrameKey) {
144 rtp_header->type.Video.width = ((data[7] << 8) + data[6]) & 0x3FFF;
145 rtp_header->type.Video.height = ((data[9] << 8) + data[8]) & 0x3FFF;
173 bool ParseVP8(WebRtcRTPHeader* rtp_header, argument
177 assert(rtp_header != NULL);
183 rtp_header->type.Video.isFirstPacket =
186 rtp_header->type.Video.codecHeader.VP8.nonReference =
188 rtp_header
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H A Drtp_sender.cc478 RTPHeader rtp_header; local
479 rtp_parser.Parse(rtp_header);
480 bytes_left -= length - rtp_header.headerLength;
578 RTPHeader rtp_header; local
579 rtp_parser.Parse(rtp_header);
583 padding_packet, length, rtp_header, now_ms - capture_time_ms);
586 UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
590 UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
820 RTPHeader rtp_header; local
821 rtp_parser.Parse(rtp_header);
916 RTPHeader rtp_header; local
1267 UpdateTransmissionTimeOffset( uint8_t *rtp_packet, const uint16_t rtp_packet_length, const RTPHeader &rtp_header, const int64_t time_diff_ms) const argument
1313 UpdateAudioLevel(uint8_t *rtp_packet, const uint16_t rtp_packet_length, const RTPHeader &rtp_header, const bool is_voiced, const uint8_t dBov) const argument
1356 UpdateAbsoluteSendTime( uint8_t *rtp_packet, const uint16_t rtp_packet_length, const RTPHeader &rtp_header, const int64_t now_ms) const argument
1632 RTPHeader rtp_header; local
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/external/chromium_org/media/cast/net/rtp/
H A Drtp_packetizer_unittest.cc42 void VerifyRtpHeader(const RtpCastTestHeader& rtp_header) { argument
43 VerifyCommonRtpHeader(rtp_header);
44 VerifyCastRtpHeader(rtp_header);
47 void VerifyCommonRtpHeader(const RtpCastTestHeader& rtp_header) { argument
48 EXPECT_EQ(kPayload, rtp_header.payload_type);
49 EXPECT_EQ(sequence_number_, rtp_header.sequence_number);
50 EXPECT_EQ(expectd_rtp_timestamp_, rtp_header.rtp_timestamp);
51 EXPECT_EQ(config_.ssrc, rtp_header.ssrc);
52 EXPECT_EQ(0, rtp_header.num_csrcs);
55 void VerifyCastRtpHeader(const RtpCastTestHeader& rtp_header) { argument
67 RtpCastTestHeader rtp_header; variable
69 VerifyRtpHeader(rtp_header); variable
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H A Dframe_buffer.cc27 const RtpCastHeader& rtp_header) {
30 frame_id_ = rtp_header.frame_id;
31 max_packet_id_ = rtp_header.max_packet_id;
32 is_key_frame_ = rtp_header.is_key_frame;
33 new_playout_delay_ms_ = rtp_header.new_playout_delay_ms;
35 DCHECK_EQ(rtp_header.frame_id, rtp_header.reference_frame_id);
36 last_referenced_frame_id_ = rtp_header.reference_frame_id;
37 rtp_timestamp_ = rtp_header.rtp_timestamp;
40 if (rtp_header
25 InsertPacket(const uint8* payload_data, size_t payload_size, const RtpCastHeader& rtp_header) argument
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H A Drtp_parser.h25 // pointed to by |rtp_header| and sets the |payload_data| pointer and
32 RtpCastHeader* rtp_header,
H A Dframer.cc37 const RtpCastHeader& rtp_header,
40 uint32 frame_id = rtp_header.frame_id;
42 if (rtp_header.is_key_frame && waiting_for_key_) {
48 << " packet:" << static_cast<int>(rtp_header.packet_id)
49 << " max packet:" << static_cast<int>(rtp_header.max_packet_id);
72 if (!it->second->InsertPacket(payload_data, payload_size, rtp_header)) {
74 << static_cast<int>(rtp_header.frame_id) << ", packet "
75 << rtp_header.packet_id;
35 InsertPacket(const uint8* payload_data, size_t payload_size, const RtpCastHeader& rtp_header, bool* duplicate) argument
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Drtcp.cc33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { argument
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;
H A Dneteq_impl_unittest.cc261 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, payloa
372 WebRtcRTPHeader rtp_header; local
414 WebRtcRTPHeader rtp_header; local
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H A Drtcp.h35 void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/
H A Dfec_receiver.h25 virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dreceiver_tests.h33 const webrtc::WebRtcRTPHeader* rtp_header) OVERRIDE {
34 return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
/external/chromium_org/media/cast/receiver/
H A Dframe_receiver.cc79 RtpCastHeader rtp_header; local
84 &rtp_header,
90 ProcessParsedPacket(rtp_header, payload_data, payload_size);
91 stats_.UpdateStatistics(rtp_header);
112 void FrameReceiver::ProcessParsedPacket(const RtpCastHeader& rtp_header, argument
119 frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] =
120 rtp_header.rtp_timestamp;
122 now, PACKET_RECEIVED, event_media_type_, rtp_header.rtp_timestamp,
123 rtp_header.frame_id, rtp_header
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