Searched refs:_rtpSender (Results 1 - 4 of 4) sorted by relevance
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_sender_video.cc | 35 : _rtpSender(*rtpSender),
121 _rtpSender.SequenceNumber());
124 _rtpSender.SendToNetwork(red_packet->data(),
150 _rtpSender.IncrementSequenceNumber(),
161 _rtpSender.SequenceNumber());
164 _rtpSender.SendToNetwork(red_packet->data(),
188 _rtpSender.SequenceNumber());
189 int ret = _rtpSender.SendToNetwork(data_buffer,
212 RtpUtility::AssignUWord32ToBuffer(data + 4, _rtpSender.SSRC());
217 _rtpSender
[all...] |
| H A D | rtp_sender_audio.cc | 23 _rtpSender(rtpSender),
240 uint16_t maxPayloadLength = _rtpSender->MaxPayloadLength();
354 uint32_t oldTimeStamp = _rtpSender->Timestamp();
355 rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, _REDPayloadType,
359 timestampOffset = uint16_t(_rtpSender->Timestamp() - oldTimeStamp);
361 rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, payloadType,
444 _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
450 "timestamp", _rtpSender->Timestamp(),
451 "seqnum", _rtpSender->SequenceNumber());
452 return _rtpSender
[all...] |
| H A D | rtp_sender_audio.h | 81 RTPSender* _rtpSender; member in class:webrtc::RTPSenderAudio
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| H A D | rtp_sender_video.h | 107 RTPSenderInterface& _rtpSender; member in class:webrtc::RTPSenderVideo
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