| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
| RTPchange.cc | 59 uint32_t send_time; local
62 "%hu %u %u %*i %*i\n", &seq_no, &ts, &send_time) == 3) {
66 packet_stats[temp_pair] = send_time;
|
| /external/chromium_org/net/tools/quic/test_tools/ |
| packet_dropping_test_writer.cc | 110 QuicTime send_time = clock_->ApproximateNow().Add(fake_packet_delay_); local
116 send_time = delayed_packets_.empty() ?
117 send_time.Add(bandwidth_delay) :
118 delayed_packets_.back().send_time.Add(bandwidth_delay);
121 peer_address, send_time));
126 delay_alarm_->Set(send_time);
163 delayed_packets_.begin()->send_time = iter->send_time;
180 return delayed_packets_.begin()->send_time;
185 QuicTime next_send_time = delayed_packets_.front().send_time;
[all...] |
| packet_dropping_test_writer.h | 116 // A single packet which will be sent at the supplied send_time.
123 QuicTime send_time);
129 QuicTime send_time; member in struct:net::tools::test::PacketDroppingTestWriter::DelayedWrite
|
| /external/openssl/ssl/ |
| s23_clnt.c | 276 int send_time = 0; local
281 send_time = (s->mode & SSL_MODE_SEND_SERVERHELLO_TIME) != 0;
283 send_time = (s->mode & SSL_MODE_SEND_CLIENTHELLO_TIME) != 0;
284 if (send_time)
|
| /external/chromium_org/media/cast/receiver/ |
| frame_receiver.cc | 283 base::TimeTicks send_time; local
284 framer_.TimeToSendNextCastMessage(&send_time);
286 send_time - cast_environment_->Clock()->NowTicks();
|
| /external/chromium_org/net/quic/congestion_control/ |
| send_algorithm_simulator.h | 25 QuicTime send_time,
28 send_time(send_time),
31 QuicTime send_time; member in struct:net::SendAlgorithmSimulator::SentPacket
|
| /external/chromium_org/third_party/webrtc/test/ |
| fake_network_pipe.cc | 38 NetworkPacket(const uint8_t* data, size_t length, int64_t send_time,
42 send_time_(send_time),
53 int64_t send_time() const { return send_time_; } function in class:webrtc::NetworkPacket
188 total_packet_delay_ += packet->arrival_time() - packet->send_time();
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
| test_iSACfixfloat.c | 60 uint32_t send_time; /* samples */ member in struct:__anon19678
80 BN_data->send_time += current_framesamples;
310 BN_data.send_time = 0;
605 BN_data.send_time,
|
| kenny.cc | 46 uint32_t send_time; /* samples */ member in struct:__anon19677
66 BN_data->send_time += current_framesamples;
480 BN_data.send_time = 0;
704 BN_data.send_time,
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.h | 49 int64_t send_time; member in struct:webrtc::testing::RtpStream::RtpPacket
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
| test_iSACfixfloat.c | 60 WebRtc_UWord32 send_time; /* samples */ member in struct:__anon37134
80 BN_data->send_time += current_framesamples;
310 BN_data.send_time = 0;
605 BN_data.send_time,
|
| kenny.c | 45 WebRtc_UWord32 send_time; /* samples */ member in struct:__anon37133
65 BN_data->send_time += current_framesamples;
489 BN_data.send_time = 0;
713 BN_data.send_time,
|
| /external/chromium_org/third_party/libjingle/source/talk/base/ |
| virtualsocket_unittest.cc | 109 uint32 send_time = *reinterpret_cast<const uint32*>(data); local
111 uint32 delay = recv_time - send_time;
[all...] |
| /external/chromium_org/third_party/webrtc/base/ |
| virtualsocket_unittest.cc | 92 uint32 send_time = *reinterpret_cast<const uint32*>(data); local
94 uint32 delay = recv_time - send_time;
[all...] |
| /external/netperf/ |
| nettest_dlpi.c | 2426 struct timeval send_time; local
[all...] |
| nettest_unix.c | 2241 struct timeval send_time; local
[all...] |
