/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | neteq_rtpplay.cc | 107 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 D | frame_buffer.h | 25 const RtpCastHeader& rtp_header);
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H A D | framer.h | 39 const RtpCastHeader& rtp_header,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_format.h | 60 virtual bool Parse(WebRtcRTPHeader* rtp_header,
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H A D | rtp_receiver_audio.h | 56 int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, 105 WebRtcRTPHeader* rtp_header,
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H A D | producer_fec.h | 26 void CreateHeader(const uint8_t* rtp_header, int header_length,
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H A D | rtp_format_video_generic.h | 71 virtual bool Parse(WebRtcRTPHeader* rtp_header,
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H A D | rtp_receiver_strategy.h | 42 virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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H A D | receive_statistics_impl.h | 42 void IncomingPacket(const RTPHeader& rtp_header, 56 void UpdateCounters(const RTPHeader& rtp_header,
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H A D | rtp_format_h264.h | 99 virtual bool Parse(WebRtcRTPHeader* rtp_header,
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H A D | nack_rtx_unittest.cc | 44 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
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H A D | rtp_sender_audio.cc | 442 RTPHeader rtp_header; local 443 rtp_parser.Parse(rtp_header); 444 _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | receive_statistics.h | 54 virtual void IncomingPacket(const RTPHeader& rtp_header, 82 virtual void IncomingPacket(const RTPHeader& rtp_header,
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H A D | rtp_receiver.h | 73 virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | audio_coding_module_unittest.cc | 55 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 D | audio_coding_module_unittest_oldapi.cc | 54 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 D | acm_receiver.h | 59 // - 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;
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H A D | acm_receiver.cc | 256 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
H A D | neteq.h | 131 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header, 145 virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
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/external/chromium_org/media/cast/receiver/ |
H A D | frame_receiver.h | 74 void ProcessParsedPacket(const RtpCastHeader& rtp_header,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 117 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 D | neteq_impl.h | 82 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header, 96 virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header, 207 int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api.h | 110 webrtc::WebRtcRTPHeader rtp_header() const { function in class:webrtc::TestRtpReceiver
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/external/chromium_org/third_party/webrtc/test/ |
H A D | rtp_file_reader.cc | 263 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;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
H A D | NETEQTEST_RTPpacket.cc | 283 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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