| /external/webrtc/webrtc/modules/pacing/ |
| H A D | packet_router.cc | 61 size_t bytes_sent = local
63 total_bytes_sent += bytes_sent;
|
| H A D | paced_sender.cc | 447 size_t bytes_sent = callback_->TimeToSendPadding(padding_needed); local
450 if (bytes_sent > 0) {
451 prober_->PacketSent(clock_->TimeInMilliseconds(), bytes_sent);
452 media_budget_->UseBudget(bytes_sent);
453 padding_budget_->UseBudget(bytes_sent);
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| /external/ltp/utils/sctp/func_tests/ |
| H A D | test_basic.c | 81 int error, bytes_sent; local
304 bytes_sent = sendmsg(sk1, &outmessage, MSG_NOSIGNAL);
305 if ((bytes_sent > 0) || (EPIPE != errno))
313 bytes_sent = sendmsg(sk1, &outmessage, MSG_NOSIGNAL);
314 if ((bytes_sent > 0) || (EPIPE != errno))
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| H A D | test_fragments.c | 88 int error, bytes_sent; local
252 bytes_sent = test_sendmsg(sk1, &outmessage, 0, msg_len);
254 tst_resm(TINFO, "Sent %d byte message", bytes_sent);
260 test_check_msg_data(&inmessage, error, bytes_sent,
274 bytes_sent - error,
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| /external/webrtc/webrtc/ |
| H A D | audio_send_stream.h | 36 int64_t bytes_sent = 0; member in struct:webrtc::AudioSendStream::Stats
|
| /external/python/cpython2/Lib/wsgiref/ |
| H A D | handlers.py | 74 bytes_sent = 0 variable in class:BaseHandler
147 self.headers['Content-Length'] = str(self.bytes_sent)
211 self.bytes_sent = len(data) # make sure we know content-length
214 self.bytes_sent += len(data)
262 self.bytes_sent = 0; self.headers_sent = False
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| /external/python/cpython3/Lib/wsgiref/ |
| H A D | handlers.py | 126 bytes_sent = 0 variable in class:BaseHandler
199 self.headers['Content-Length'] = str(self.bytes_sent)
273 self.bytes_sent = len(data) # make sure we know content-length
276 self.bytes_sent += len(data)
324 self.bytes_sent = 0; self.headers_sent = False
|
| /external/webrtc/talk/app/webrtc/test/ |
| H A D | mockpeerconnectionobservers.h | 140 &stats_.bytes_sent);
176 return stats_.bytes_sent;
223 bytes_sent = 0;
234 int bytes_sent; member in struct:webrtc::MockStatsObserver::__anon30513
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtcp_sender.cc | 845 size_t bytes_sent = container.SendPackets(); local
846 return bytes_sent == 0 ? -1 : 0;
|
| H A D | rtp_rtcp_impl.cc | 580 size_t* bytes_sent,
586 if (bytes_sent) {
587 *bytes_sent = rtp_stats.transmitted.payload_bytes +
579 DataCountersRTP( size_t* bytes_sent, uint32_t* packets_sent) const argument
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| H A D | rtp_sender.cc | 613 size_t bytes_sent = 0; local
695 bytes_sent += padding_bytes_in_packet;
699 return bytes_sent;
753 int bytes_sent = -1; local
755 bytes_sent = transport_->SendRtp(packet, size, options)
761 bytes_sent);
763 if (bytes_sent <= 0) {
801 const int32_t bytes_sent = ReSendPacket(*it, 5 + avg_rtt); local
802 if (bytes_sent > 0) {
803 bytes_re_sent += bytes_sent;
1018 size_t bytes_sent = TrySendRedundantPayloads(bytes); local
[all...] |
| /external/webrtc/talk/media/base/ |
| H A D | mediachannel.h | 590 : bytes_sent(0),
625 int64_t bytes_sent; member in struct:cricket::MediaSenderInfo
|