| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| H A D | neteq_stereo_unittest.cc | 173 virtual int GetArrivalTime(int send_time) { argument
174 int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
175 last_send_time_ = send_time;
297 virtual int GetArrivalTime(int send_time) { argument
299 drift_factor * (send_time - last_send_time_);
300 last_send_time_ = send_time;
334 virtual int GetArrivalTime(int send_time) { argument
336 int arrival_time = std::min(last_arrival_time_, send_time);
341 last_send_time_ = send_time;
|
| H A D | neteq_network_stats_unittest.cc | 122 bool Lost(uint32_t send_time) { argument
123 if (send_time - last_lost_time_ >= packet_loss_interval_) {
124 last_lost_time_ = send_time;
|
| H A D | neteq_external_decoder_unittest.cc | 85 int GetArrivalTime(int send_time) { argument
86 int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
87 last_send_time_ = send_time;
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| H A D | 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/webrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_packet_history.cc | 101 stored_packets_[prev_index_].send_time == 0) {
122 stored_packets_[prev_index_].send_time = 0; // Packet not sent.
165 if (stored_packets_[index].send_time != 0) {
169 stored_packets_[index].send_time = clock_->TimeInMilliseconds();
204 ((now - stored_packets_[index].send_time) < min_elapsed_time_ms)) {
216 stored_packets_[index].send_time = clock_->TimeInMilliseconds();
|
| H A D | rtp_packet_history.h | 94 int64_t send_time = 0; member in struct:webrtc::RTPPacketHistory::StoredPacket
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| H A D | unittest.cc | 163 const int send_time = elapsed_time_ms * (sample_rate_hz / 1000); local
165 encdec, bitstream1.data(), bitstream1.size(), i, send_time,
166 channel1.Send(send_time, bitstream1.size())));
168 dec, bitstream2.data(), bitstream2.size(), i, send_time,
169 channel2.Send(send_time, bitstream2.size())));
|
| /external/webrtc/webrtc/test/ |
| H A D | fake_network_pipe.cc | 41 NetworkPacket(const uint8_t* data, size_t length, int64_t send_time, argument
45 send_time_(send_time),
56 int64_t send_time() const { return send_time_; } function in class:webrtc::NetworkPacket
197 total_packet_delay_ += packet->arrival_time() - packet->send_time();
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
| H A D | test_iSACfixfloat.c | 54 uint32_t send_time; /* samples */ member in struct:__anon21416
74 BN_data->send_time += current_framesamples;
276 BN_data.send_time = 0;
550 BN_data.send_time, BN_data.arrival_time);
|
| H A D | kenny.cc | 46 uint32_t send_time; /* samples */ member in struct:__anon21415
67 BN_data->send_time += current_framesamples;
484 BN_data.send_time = 0;
708 BN_data.send_time,
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| H A D | remote_bitrate_estimator_unittest_helper.h | 50 int64_t send_time; member in struct:webrtc::testing::RtpStream::RtpPacket
|
| H A D | remote_bitrate_estimator_unittest_helper.cc | 65 packet->send_time = time_now_us + kSendSideOffsetUs;
68 ((frequency_ / 1000) * packet->send_time + 500) / 1000);
259 AbsSendTime(packet->send_time, 1000000), true);
|
| /external/netperf/src/ |
| H A D | nettest_unix.c | 2253 struct timeval send_time;
2476 gettimeofday(&send_time,&dummy_zone);
2508 if (recv_time.tv_usec < send_time.tv_usec) {
2515 kept_times[time_index] = (recv_time.tv_sec - send_time.tv_sec) * 1000000;
2516 kept_times[time_index] += (recv_time.tv_usec - send_time.tv_usec);
2244 struct timeval send_time; local
|
| H A D | nettest_dlpi.c | 2674 struct timeval send_time;
2971 gettimeofday(&send_time,&dummy_zone);
3013 if (recv_time.tv_usec < send_time.tv_usec) {
3020 kept_times[time_index] = (recv_time.tv_sec - send_time.tv_sec) * 1000000;
3021 kept_times[time_index] += (recv_time.tv_usec - send_time.tv_usec);
2666 struct timeval send_time; local
|
| /external/webrtc/webrtc/base/ |
| H A D | virtualsocket_unittest.cc | 103 uint32_t send_time = *reinterpret_cast<const uint32_t*>(data); local
105 uint32_t delay = recv_time - send_time;
|