Searched refs:rtp_header (Results 26 - 50 of 60) sorted by relevance

123

/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/
H A Dneteq_rtpplay.cc107 WebRtcRTPHeader* rtp_header,
219 WebRtcRTPHeader rtp_header; local
220 packet->ConvertHeader(&rtp_header);
229 &rtp_header,
234 neteq->InsertPacket(rtp_header,
495 WebRtcRTPHeader* rtp_header,
499 if (IsComfortNosie(rtp_header->header.payloadType)) {
511 rtp_header->header.sequenceNumber + 1) {
513 next_packet->header().timestamp - rtp_header->header.timestamp) {
515 next_packet->header().timestamp - rtp_header
490 ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file, webrtc::scoped_ptr<int16_t[]>* replacement_audio, webrtc::scoped_ptr<uint8_t[]>* payload, size_t* payload_mem_size_bytes, size_t* frame_size_samples, WebRtcRTPHeader* rtp_header, const webrtc::test::Packet* next_packet) argument
[all...]
/external/chromium_org/media/cast/net/rtp/
H A Dframe_buffer.h25 const RtpCastHeader& rtp_header);
H A Dframer.h39 const RtpCastHeader& rtp_header,
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtp_format.h60 virtual bool Parse(WebRtcRTPHeader* rtp_header,
H A Drtp_receiver_audio.h56 int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
105 WebRtcRTPHeader* rtp_header,
H A Dproducer_fec.h26 void CreateHeader(const uint8_t* rtp_header, int header_length,
H A Drtp_format_video_generic.h71 virtual bool Parse(WebRtcRTPHeader* rtp_header,
H A Drtp_receiver_strategy.h42 virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
H A Dreceive_statistics_impl.h42 void IncomingPacket(const RTPHeader& rtp_header,
56 void UpdateCounters(const RTPHeader& rtp_header,
H A Drtp_format_h264.h99 virtual bool Parse(WebRtcRTPHeader* rtp_header,
H A Dnack_rtx_unittest.cc44 const webrtc::WebRtcRTPHeader* rtp_header) {
46 EXPECT_EQ(kTestSsrc, rtp_header->header.ssrc);
47 sequence_numbers_.push_back(rtp_header->header.sequenceNumber);
41 OnReceivedPayloadData( const uint8_t* data, const uint16_t size, const webrtc::WebRtcRTPHeader* rtp_header) argument
H A Drtp_sender_audio.cc442 RTPHeader rtp_header; local
443 rtp_parser.Parse(rtp_header);
444 _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/
H A Dreceive_statistics.h54 virtual void IncomingPacket(const RTPHeader& rtp_header,
82 virtual void IncomingPacket(const RTPHeader& rtp_header,
H A Drtp_receiver.h73 virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/
H A Daudio_coding_module_unittest.cc55 void Populate(WebRtcRTPHeader* rtp_header) { argument
56 rtp_header->header.sequenceNumber = 0xABCD;
57 rtp_header->header.timestamp = 0xABCDEF01;
58 rtp_header->header.payloadType = payload_type_;
59 rtp_header->header.markerBit = false;
60 rtp_header->header.ssrc = 0x1234;
61 rtp_header->header.numCSRCs = 0;
62 rtp_header->frameType = kAudioFrameSpeech;
64 rtp_header->header.payload_type_frequency = kSampleRateHz;
65 rtp_header
69 Forward(WebRtcRTPHeader* rtp_header) argument
[all...]
H A Daudio_coding_module_unittest_oldapi.cc54 void Populate(WebRtcRTPHeader* rtp_header) { argument
55 rtp_header->header.sequenceNumber = 0xABCD;
56 rtp_header->header.timestamp = 0xABCDEF01;
57 rtp_header->header.payloadType = payload_type_;
58 rtp_header->header.markerBit = false;
59 rtp_header->header.ssrc = 0x1234;
60 rtp_header->header.numCSRCs = 0;
61 rtp_header->frameType = kAudioFrameSpeech;
63 rtp_header->header.payload_type_frequency = kSampleRateHz;
64 rtp_header
68 Forward(WebRtcRTPHeader* rtp_header) argument
[all...]
H A Dacm_receiver.h59 // - rtp_header : RTP header for the incoming payload containing
68 int InsertPacket(const WebRtcRTPHeader& rtp_header,
323 const RTPHeader& rtp_header, const uint8_t* payload) const;
H A Dacm_receiver.cc256 int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, argument
263 const RTPHeader* header = &rtp_header.header; // Just a shorthand.
322 rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz,
334 if (neteq_->InsertPacket(rtp_header, incoming_payload, length_payload,
800 const RTPHeader &rtp_header, const uint8_t* payload) const {
801 uint8_t payload_type = rtp_header.payloadType;
799 RtpHeaderToCodecIndex( const RTPHeader &rtp_header, const uint8_t* payload) const argument
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/
H A Dneteq.h131 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
145 virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
/external/chromium_org/media/cast/receiver/
H A Dframe_receiver.h74 void ProcessParsedPacket(const RtpCastHeader& rtp_header,
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Dneteq_impl.cc117 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header, argument
122 LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
123 ", sn=" << rtp_header.header.sequenceNumber <<
124 ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
125 ", ssrc=" << rtp_header.header.ssrc <<
127 int error = InsertPacketInternal(rtp_header, payload, length_bytes,
137 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, argument
141 << rtp_header.header.timestamp <<
142 ", sn=" << rtp_header.header.sequenceNumber <<
143 ", pt=" << static_cast<int>(rtp_header
396 InsertPacketInternal(const WebRtcRTPHeader& rtp_header, const uint8_t* payload, int length_bytes, uint32_t receive_timestamp, bool is_sync_packet) argument
626 const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader(); local
[all...]
H A Dneteq_impl.h82 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
96 virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
207 int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/
H A Dtest_api.h110 webrtc::WebRtcRTPHeader rtp_header() const { function in class:webrtc::TestRtpReceiver
/external/chromium_org/third_party/webrtc/test/
H A Drtp_file_reader.cc263 uint8_t pt = packets_[packet_numbers[0]].rtp_header.payloadType;
325 RTPHeader rtp_header; member in struct:webrtc::test::PcapReader::RtpPacketMarker
401 rtp_parser.ParseRtcp(&marker.rtp_header);
404 if (!rtp_parser.Parse(marker.rtp_header, NULL)) {
409 uint32_t ssrc = marker.rtp_header.ssrc;
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/
H A DNETEQTEST_RTPpacket.cc283 void NETEQTEST_RTPpacket::parseHeader(webrtc::WebRtcRTPHeader* rtp_header) { argument
287 if (rtp_header) {
288 rtp_header->header.markerBit = _rtpInfo.header.markerBit;
289 rtp_header->header.payloadType = _rtpInfo.header.payloadType;
290 rtp_header->header.sequenceNumber = _rtpInfo.header.sequenceNumber;
291 rtp_header->header.timestamp = _rtpInfo.header.timestamp;
292 rtp_header->header.ssrc = _rtpInfo.header.ssrc;

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