| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | neteq_stereo_unittest.cc | 173 virtual int GetArrivalTime(int send_time) { in GetArrivalTime() argument
174 int arrival_time = last_arrival_time_ + (send_time - last_send_time_); in GetArrivalTime()
175 last_send_time_ = send_time; in GetArrivalTime()
297 virtual int GetArrivalTime(int send_time) { in GetArrivalTime() argument
299 drift_factor * (send_time - last_send_time_); in GetArrivalTime()
300 last_send_time_ = send_time; in GetArrivalTime()
334 virtual int GetArrivalTime(int send_time) { in GetArrivalTime() argument
336 int arrival_time = std::min(last_arrival_time_, send_time); in GetArrivalTime()
341 last_send_time_ = send_time; in GetArrivalTime()
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| D | neteq_network_stats_unittest.cc | 122 bool Lost(uint32_t send_time) { in Lost() argument
123 if (send_time - last_lost_time_ >= packet_loss_interval_) { in Lost()
124 last_lost_time_ = send_time; in Lost()
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| D | neteq_external_decoder_unittest.cc | 85 int GetArrivalTime(int send_time) { in GetArrivalTime() argument
86 int arrival_time = last_arrival_time_ + (send_time - last_send_time_); in GetArrivalTime()
87 last_send_time_ = send_time; in GetArrivalTime()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_packet_history.cc | 101 stored_packets_[prev_index_].send_time == 0) { in PutRTPPacket()
122 stored_packets_[prev_index_].send_time = 0; // Packet not sent. in PutRTPPacket()
165 if (stored_packets_[index].send_time != 0) { in SetSent()
169 stored_packets_[index].send_time = clock_->TimeInMilliseconds(); in SetSent()
204 ((now - stored_packets_[index].send_time) < min_elapsed_time_ms)) { in GetPacketAndSetSendTime()
216 stored_packets_[index].send_time = clock_->TimeInMilliseconds(); in GetPacketAndSetSendTime()
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| D | rtp_packet_history.h | 94 int64_t send_time = 0; member
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| /external/webrtc/webrtc/test/ |
| D | fake_network_pipe.cc | 41 NetworkPacket(const uint8_t* data, size_t length, int64_t send_time, in NetworkPacket() argument
45 send_time_(send_time), in NetworkPacket()
56 int64_t send_time() const { return send_time_; } in send_time() function in webrtc::NetworkPacket
197 total_packet_delay_ += packet->arrival_time() - packet->send_time(); in Process()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| D | RTPchange.cc | 59 uint32_t send_time; in main() local
62 "%hu %u %u %*i %*i\n", &seq_no, &ts, &send_time) == 3) { in main()
66 packet_stats[temp_pair] = send_time; in main()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| D | unittest.cc | 163 const int send_time = elapsed_time_ms * (sample_rate_hz / 1000); in TestGetSetBandwidthInfo() local
165 encdec, bitstream1.data(), bitstream1.size(), i, send_time, in TestGetSetBandwidthInfo()
166 channel1.Send(send_time, bitstream1.size()))); in TestGetSetBandwidthInfo()
168 dec, bitstream2.data(), bitstream2.size(), i, send_time, in TestGetSetBandwidthInfo()
169 channel2.Send(send_time, bitstream2.size()))); in TestGetSetBandwidthInfo()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
| D | test_iSACfixfloat.c | 54 uint32_t send_time; /* samples */ member
74 BN_data->send_time += current_framesamples; in get_arrival_time()
276 BN_data.send_time = 0; in main()
550 BN_data.send_time, BN_data.arrival_time); in main()
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| D | kenny.cc | 46 uint32_t send_time; /* samples */ member
67 BN_data->send_time += current_framesamples; in get_arrival_time()
484 BN_data.send_time = 0; in main()
708 BN_data.send_time, in main()
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| /external/python/cpython2/Tools/ccbench/ |
| D | ccbench.py | 374 send_time = eval(line)
375 assert isinstance(send_time, float)
376 results.append((send_time, recv_time))
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| /external/python/cpython3/Tools/ccbench/ |
| D | ccbench.py | 376 send_time = eval(line)
377 assert isinstance(send_time, float)
378 results.append((send_time, recv_time))
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_unittest_helper.cc | 65 packet->send_time = time_now_us + kSendSideOffsetUs; in GenerateFrame()
68 ((frequency_ / 1000) * packet->send_time + 500) / 1000); in GenerateFrame()
259 AbsSendTime(packet->send_time, 1000000), true); in GenerateAndProcessFrame()
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| D | remote_bitrate_estimator_unittest_helper.h | 50 int64_t send_time; member
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| /external/webrtc/webrtc/base/ |
| D | virtualsocket_unittest.cc | 103 uint32_t send_time = *reinterpret_cast<const uint32_t*>(data); in OnReadPacket() local
105 uint32_t delay = recv_time - send_time; in OnReadPacket()
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