| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| RTPFile.h | 29 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
40 void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
48 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
54 uint8_t payloadType;
68 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
100 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
|
| Channel.cc | 21 int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType,
33 rtpInfo.header.payloadType = payloadType;
76 rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
126 if ((rtpInfo.header.payloadType != _lastPayloadType)
133 if (_lastPayloadType == _payloadStats[n].payloadType) {
139 _lastPayloadType = rtpInfo.header.payloadType;
144 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
191 currentPayloadStr->payloadType = rtpInfo.header.payloadType
[all...] |
| RTPFile.cc | 31 rtpInfo->header.payloadType = rtpHeader[1];
42 void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
46 rtpHeader[1] = (unsigned char) (payloadType & 0xFF);
62 RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
65 : payloadType(payloadType),
88 void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
91 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
105 rtpInfo->header.payloadType = packet->payloadType;
[all...] |
| Channel.h | 41 int16_t payloadType;
54 const FrameType frameType, const uint8_t payloadType,
67 void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
|
| EncodeDecodeTest.h | 33 const FrameType frameType, const uint8_t payloadType,
39 static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_audio.cc | 88 const int8_t payloadType,
98 _cngNBPayloadType = payloadType;
101 _cngWBPayloadType = payloadType;
104 _cngSWBPayloadType = payloadType;
107 _cngFBPayloadType = payloadType;
115 // we dont want to allow send with a DTMF payloadtype
116 _dtmfPayloadType = payloadType;
132 const int8_t payloadType)
138 if(_lastPayloadType != payloadType)
143 if(_cngNBPayloadType == payloadType)
[all...] |
| rtp_sender_audio.h | 30 const int8_t payloadType,
37 const int8_t payloadType,
62 int32_t SetRED(const int8_t payloadType);
65 int32_t RED(int8_t& payloadType) const;
76 const int8_t payloadType);
|
| rtp_sender_video.h | 41 const int8_t payloadType,
47 const int8_t payloadType,
98 const int8_t payloadType,
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| initial_delay_manager.cc | 28 last_packet_rtp_info_.header.payloadType = kInvalidPayloadType;
45 rtp_info.header.payloadType != audio_payload_type_));
65 last_packet_rtp_info_.header.payloadType == kInvalidPayloadType) {
68 audio_payload_type_ = rtp_info.header.payloadType;
131 sync_stream->rtp_info.header.payloadType = audio_payload_type_;
145 sync_stream->rtp_info.header.payloadType = audio_payload_type_;
207 sync_stream->rtp_info.header.payloadType = audio_payload_type_;
215 last_packet_rtp_info_.header.payloadType = audio_payload_type_;
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| packet.cc | 20 payloadType(0),
40 payloadType(rtpHeader.header.payloadType),
61 payloadType(0),
80 payloadType = 0;
|
| packet.h | 34 uint8_t payloadType;
|
| generic_encoder.h | 52 void SetPayloadType(uint8_t payloadType) { _payloadType = payloadType; };
|
| /external/chromium_org/third_party/webrtc/modules/utility/source/ |
| video_coder.cc | 112 const uint8_t payloadType,
124 _videoEncodedData->payloadType = payloadType;
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| coder.h | 47 uint8_t payloadType,
|
| coder.cc | 61 const uint8_t payloadType = _receiveCodec.pltype;
65 payloadType,
109 uint8_t /* payloadType */,
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| /external/chromium_org/media/cast/net/rtp/ |
| mock_rtp_feedback.h | 17 int32(const int8 payloadType,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/mock/ |
| mock_rtp_payload_strategy.h | 35 const int8_t payloadType,
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| packet_buffer.cc | 86 if (decoder_database.IsComfortNoise(packet->header.payloadType)) {
88 *current_cng_rtp_payload_type != packet->header.payloadType) {
94 *current_cng_rtp_payload_type = packet->header.payloadType;
95 } else if (!decoder_database.IsDtmf(packet->header.payloadType)) {
98 *current_rtp_payload_type != packet->header.payloadType) {
103 *current_rtp_payload_type = packet->header.payloadType;
215 decoder_database->GetDecoder(packet->header.payloadType);
|
| neteq_impl.cc | 124 ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
143 ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
407 if (decoder_database_->IsDtmf(rtp_header.header.payloadType) ||
408 decoder_database_->IsRed(rtp_header.header.payloadType) ||
409 decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) {
411 << rtp_header.header.payloadType;
415 rtp_header.header.payloadType != current_rtp_payload_type_ ||
433 packet->header.payloadType = rtp_header.header.payloadType;
472 current_rtp_payload_type_ = main_header.payloadType;
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_analyze.cc | 108 packet->header().payloadType,
120 if (packet->header().payloadType == FLAGS_red) {
131 red->payloadType);
|
| packet.cc | 99 header->payloadType = payload_ptr[0] & 0x7F;
112 header->payloadType = payload_ptr[0] & 0x7F;
151 destination->payloadType = header_.payloadType;
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
| TestSenderReceiver.h | 65 const int8_t payloadType = 127);
76 const int8_t payloadType,
127 const int8_t payloadType = 127);
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| test_callbacks.cc | 56 const uint8_t payloadType,
90 rtpInfo.header.payloadType = payloadType;
148 const uint8_t payloadType,
160 payloadType,
306 header.payloadType, &payload_specific)) {
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
| NETEQTEST_RTPpacket.cc | 159 if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
199 if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
289 rtp_header->header.payloadType = _rtpInfo.header.payloadType;
353 uint8_t NETEQTEST_RTPpacket::payloadType() const
366 return tempRTPinfo.header.payloadType;
445 _rtpInfo.header.payloadType = pt;
553 RTPinfo->header.payloadType,
627 void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, uint8_t payloadType, uint16_t seqNo, uint32_t timestamp, uint32_t ssrc, uint8_t markerBit) const
638 rtp_data[1]=(unsigned char)(payloadType & 0xFF)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| rtp_rtcp.h | 163 * payloadType - payload type of codec
168 const int8_t payloadType) = 0;
276 int* payloadType) const = 0;
319 * payloadType - payload type of frame to send
330 const int8_t payloadType,
656 const int8_t payloadType) = 0;
664 int8_t& payloadType) const = 0;
|
