| /external/webrtc/webrtc/system_wrappers/include/ |
| rtp_to_ntp.h | 25 uint32_t rtp_timestamp; member in struct:webrtc::RtcpMeasurement
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/system_wrappers/source/ |
| rtp_to_ntp.cc | 20 : ntp_secs(0), ntp_frac(0), rtp_timestamp(0) {}
24 : ntp_secs(ntp_secs), ntp_frac(ntp_frac), rtp_timestamp(timestamp) {}
59 uint32_t rtp_timestamp,
70 measurement.rtp_timestamp = rtp_timestamp;
91 // Converts |rtp_timestamp| to the NTP time base using the NTP and RTP timestamp
95 bool RtpToNtpMs(int64_t rtp_timestamp,
103 int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp;
104 int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp;
120 if (!CompensateForWrapAround(rtp_timestamp, rtcp_timestamp_old
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| remote_ntp_time_estimator.h | 35 uint32_t rtp_timestamp);
37 // Estimates the NTP timestamp in local timebase from |rtp_timestamp|.
39 int64_t Estimate(uint32_t rtp_timestamp);
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator_unittest.cc | 58 uint32_t rtp_timestamp, bool expected_result) {
61 rtp_timestamp));
86 uint32_t rtp_timestamp = GetRemoteTimestamp(); local
91 EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp));
98 EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp));
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| remote_ntp_time_estimator.cc | 53 int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) {
59 if (!RtpToNtpMs(rtp_timestamp, rtcp_list_, &sender_capture_ntp_ms)) {
70 LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
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| rtcp_receiver_help.h | 86 uint32_t rtp_timestamp; member in class:webrtc::RTCPHelp::RTCPPacketInformation
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| overuse_detector_unittest.cc | 93 void UpdateDetector(uint32_t rtp_timestamp, int64_t receive_time_ms,
98 if (inter_arrival_->ComputeDeltas(rtp_timestamp,
139 uint32_t rtp_timestamp = 10 * 90; local
143 UpdateDetector(rtp_timestamp, now_ms_, packet_size);
145 rtp_timestamp += frame_duration_ms * 90;
153 uint32_t rtp_timestamp = 10 * 90; local
157 UpdateDetector(rtp_timestamp, now_ms_, packet_size);
158 rtp_timestamp += frame_duration_ms * 90;
171 uint32_t rtp_timestamp = 10 * 90; local
175 UpdateDetector(rtp_timestamp, now_ms_, packet_size)
219 uint32_t rtp_timestamp = frame_duration_ms * 90; local
251 uint32_t rtp_timestamp = frame_duration_ms * 90; local
282 uint32_t rtp_timestamp = frame_duration_ms * 90; local
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/ |
| audio_encoder.cc | 27 uint32_t rtp_timestamp,
35 EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
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| audio_decoder.cc | 63 uint32_t rtp_timestamp,
|
| /external/webrtc/webrtc/video/ |
| vie_sync_module.cc | 33 uint32_t rtp_timestamp = 0; local
38 &rtp_timestamp)) {
44 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| audio_decoder_isac_t_impl.h | 79 uint32_t rtp_timestamp,
82 rtp_sequence_number, rtp_timestamp,
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| audio_decoder_isac_t.h | 34 uint32_t rtp_timestamp,
|
| audio_encoder_isac_t.h | 63 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| audio_encoder_ilbc.h | 43 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| audio_encoder_ilbc.cc | 93 uint32_t rtp_timestamp,
101 first_timestamp_in_buffer_ = rtp_timestamp;
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.cc | 81 uint32_t rtp_timestamp,
86 first_timestamp_in_buffer_ = rtp_timestamp;
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| audio_encoder_pcm.h | 44 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| audio_encoder_copy_red.cc | 56 uint32_t rtp_timestamp,
61 speech_encoder_->Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
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| audio_encoder_copy_red.h | 46 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| dtmf_buffer.h | 67 // |rtp_timestamp| is simply copied into the struct.
68 static int ParseEvent(uint32_t rtp_timestamp,
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| audio_decoder_impl.h | 42 uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_encoder_g722.h | 44 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| audio_encoder_cng.h | 59 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_encoder_opus.h | 64 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/neteq/mock/ |
| mock_external_decoder_pcm16b.h | 83 uint16_t rtp_sequence_number, uint32_t rtp_timestamp,
|
