| /development/cmds/monkey/src/com/android/commands/monkey/ |
| MonkeyGetAppFrameRateEvent.java | 43 private static long sEndTime; // in millisecond
144 sEndTime = System.currentTimeMillis();
145 long diff = sEndTime - sStartTime;
|
| /external/iputils/ |
| clockdiff.c | 132 long sendtime, recvtime, histime; local
235 sendtime = ntohl(*(__u32*)(icp+1));
236 diff = recvtime - sendtime;
260 delta1 = histime - sendtime;
309 long sendtime, recvtime, histime, histime1; local
425 sendtime = recvtime = histime = histime1 = 0;
434 sendtime = t;
447 if (!(sendtime&histime&histime1&recvtime)) {
452 diff = recvtime - sendtime;
467 delta1 = histime - sendtime;
[all...] |
| tracepath.c | 41 struct timeval sendtime; member in struct:hhistory
136 rettv = &his[slot].sendtime;
272 his[hisptr].sendtime = hdr->tv;
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| tracepath6.c | 51 struct timeval sendtime; member in struct:hhistory
159 rettv = &his[slot].sendtime;
342 his[hisptr].sendtime = hdr->tv;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| bandwidth_estimator.h | 49 * - sendTime : value in RTP header giving send time in samples
|
| bandwidth_estimator.c | 132 * - sendTime : value in RTP header giving send time in samples
145 const uint32_t sendTime,
261 sendTimeDiff = sendTime - bweStr->prevSendTime;
513 bweStr->prevSendTime = sendTime;
[all...] |
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| bandwidth_estimator.h | 49 * - sendTime : value in RTP header giving send time in samples
|
| bandwidth_estimator.c | 132 * - sendTime : value in RTP header giving send time in samples
145 const WebRtc_UWord32 sendTime,
261 sendTimeDiff = sendTime - bweStr->prevSendTime;
513 bweStr->prevSendTime = sendTime;
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
| RTPencode.cc | 259 double sendtime = 0; local
574 if ( sendtime >= NTone * DTMF_PACKET_INTERVAL ) {
575 if ( sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION ) {
582 enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 0, 4, (int) (sendtime - NTone * DTMF_PACKET_INTERVAL)*(fs/1000) + len);
595 offset = (uint32_t) sendtime; //(timestamp/(fs/1000));
639 if (enc_len > 0 && (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
690 offset = (uint32_t) sendtime;
787 sendtime += (double) len/(fs/1000);
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/base/ |
| socket.h | 191 // if SendTime option is needed at socket level.
|
| /external/chromium_org/third_party/webrtc/base/ |
| socket.h | 174 // if SendTime option is needed at socket level.
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_analyze.cc | 92 fprintf(out_file, "SeqNo TimeStamp SendTime Size PT M SSRC");
|
| /external/chromium_org/content/browser/renderer_host/p2p/ |
| socket_host.cc | 207 // If packet option has non default value (-1) for sendtime extension id,
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| channel.h | 322 // Helper method to get RTP Absoulute SendTime extension header id if
|
| channel.cc | 477 // Socket layer will update rtp sendtime extension header if present in
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| mediachannel.h | 632 // Returns the absoulte sendtime extension id value from media channel.
[all...] |
