| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| D | overuse_detector_unittest.cc | 93 void UpdateDetector(uint32_t rtp_timestamp, int64_t receive_time_ms, in UpdateDetector() argument
98 if (inter_arrival_->ComputeDeltas(rtp_timestamp, in UpdateDetector()
139 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
143 UpdateDetector(rtp_timestamp, now_ms_, packet_size); in TEST_F()
145 rtp_timestamp += frame_duration_ms * 90; in TEST_F()
153 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
157 UpdateDetector(rtp_timestamp, now_ms_, packet_size); in TEST_F()
158 rtp_timestamp += frame_duration_ms * 90; in TEST_F()
171 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
175 UpdateDetector(rtp_timestamp, now_ms_, packet_size); in TEST_F()
[all …]
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| D | remote_bitrate_estimator_unittest_helper.h | 52 uint32_t rtp_timestamp; member
171 uint32_t rtp_timestamp,
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| D | remote_bitrate_estimator_single_stream.cc | 75 uint32_t rtp_timestamp = header.timestamp + in IncomingPacket() local
99 if (estimator->inter_arrival.ComputeDeltas(rtp_timestamp, arrival_time_ms, in IncomingPacket()
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| /external/webrtc/webrtc/system_wrappers/source/ |
| D | rtp_to_ntp.cc | 20 : ntp_secs(0), ntp_frac(0), rtp_timestamp(0) {} in RtcpMeasurement()
24 : ntp_secs(ntp_secs), ntp_frac(ntp_frac), rtp_timestamp(timestamp) {} in RtcpMeasurement()
59 uint32_t rtp_timestamp, in UpdateRtcpList() argument
70 measurement.rtp_timestamp = rtp_timestamp; in UpdateRtcpList()
95 bool RtpToNtpMs(int64_t rtp_timestamp, in RtpToNtpMs() argument
103 int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp; in RtpToNtpMs()
104 int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp; in RtpToNtpMs()
120 if (!CompensateForWrapAround(rtp_timestamp, rtcp_timestamp_old, in RtpToNtpMs()
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| /external/webrtc/webrtc/system_wrappers/include/ |
| D | rtp_to_ntp.h | 25 uint32_t rtp_timestamp; member
35 uint32_t rtp_timestamp,
41 bool RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp,
46 int CheckForWrapArounds(uint32_t rtp_timestamp, uint32_t rtcp_rtp_timestamp);
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | remote_ntp_time_estimator_unittest.cc | 58 uint32_t rtp_timestamp, bool expected_result) { in UpdateRtcpTimestamp() argument
61 rtp_timestamp)); in UpdateRtcpTimestamp()
86 uint32_t rtp_timestamp = GetRemoteTimestamp(); in TEST_F() local
91 EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp)); in TEST_F()
98 EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp)); in TEST_F()
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| D | remote_ntp_time_estimator.cc | 53 int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) { in Estimate() argument
59 if (!RtpToNtpMs(rtp_timestamp, rtcp_list_, &sender_capture_ntp_ms)) { in Estimate()
70 LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp in Estimate()
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| /external/webrtc/webrtc/video/ |
| D | vie_sync_module.cc | 33 uint32_t rtp_timestamp = 0; in UpdateMeasurements() local
38 &rtp_timestamp)) { in UpdateMeasurements()
44 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) { in UpdateMeasurements()
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| D | vie_receiver.cc | 435 uint32_t rtp_timestamp = 0; in InsertRTCPPacket() local
437 &rtp_timestamp)) { in InsertRTCPPacket()
441 ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); in InsertRTCPPacket()
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| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| D | remote_ntp_time_estimator.h | 35 uint32_t rtp_timestamp);
39 int64_t Estimate(uint32_t rtp_timestamp);
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| /external/webrtc/webrtc/modules/audio_coding/codecs/ |
| D | audio_encoder.cc | 27 uint32_t rtp_timestamp, in Encode() argument
35 EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded); in Encode()
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| D | audio_encoder.h | 94 EncodedInfo Encode(uint32_t rtp_timestamp,
99 virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/talk/media/base/ |
| D | rtpdump.cc | 261 uint32_t rtp_timestamp = 0; in UpdateStreamStatistics() local
262 packet.GetRtpTimestamp(&rtp_timestamp); in UpdateStreamStatistics()
268 first_rtp_timestamp_ = rtp_timestamp; in UpdateStreamStatistics()
271 } else if (rtp_timestamp != prev_rtp_timestamp_) { in UpdateStreamStatistics()
277 prev_rtp_timestamp_ = rtp_timestamp; in UpdateStreamStatistics()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| D | audio_decoder_isac_t_impl.h | 79 uint32_t rtp_timestamp, in IncomingPacket() argument
82 rtp_sequence_number, rtp_timestamp, in IncomingPacket()
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| D | audio_encoder_isac_t_impl.h | 117 uint32_t rtp_timestamp, in EncodeInternal() argument
124 packet_timestamp_ = rtp_timestamp; in EncodeInternal()
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| D | audio_decoder_isac_t.h | 34 uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| D | audio_encoder_ilbc.cc | 93 uint32_t rtp_timestamp, in EncodeInternal() argument
101 first_timestamp_in_buffer_ = rtp_timestamp; in EncodeInternal()
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| D | audio_encoder_ilbc.h | 43 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_encoder_pcm.cc | 81 uint32_t rtp_timestamp, in EncodeInternal() argument
86 first_timestamp_in_buffer_ = rtp_timestamp; in EncodeInternal()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| D | audio_encoder_copy_red.cc | 56 uint32_t rtp_timestamp, in EncodeInternal() argument
61 speech_encoder_->Encode(rtp_timestamp, audio, max_encoded_bytes, encoded); in EncodeInternal()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| D | audio_encoder_g722.cc | 95 uint32_t rtp_timestamp, in EncodeInternal() argument
102 first_timestamp_in_buffer_ = rtp_timestamp; in EncodeInternal()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | dtmf_buffer.cc | 69 int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp, in ParseEvent() argument
84 event->timestamp = rtp_timestamp; in ParseEvent()
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| D | audio_decoder_impl.h | 42 uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_encoder_opus.cc | 134 uint32_t rtp_timestamp, in EncodeInternal() argument
139 first_timestamp_in_buffer_ = rtp_timestamp; in EncodeInternal()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| D | audio_encoder_cng.cc | 99 uint32_t rtp_timestamp, in EncodeInternal() argument
108 rtp_timestamps_.push_back(rtp_timestamp); in EncodeInternal()
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