| /external/webrtc/webrtc/system_wrappers/include/ |
| H A D | 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);
|
| /external/webrtc/webrtc/system_wrappers/source/ |
| H A D | 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, argument
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_ol
57 UpdateRtcpList(uint32_t ntp_secs, uint32_t ntp_frac, uint32_t rtp_timestamp, RtcpList* rtcp_list, bool* new_rtcp_sr) argument
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| H A D | 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);
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| H A D | overuse_detector_unittest.cc | 93 void UpdateDetector(uint32_t rtp_timestamp, int64_t receive_time_ms, argument
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
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/rtp_rtcp/source/ |
| H A D | 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));
57 UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac, uint32_t rtp_timestamp, bool expected_result) argument
|
| H A D | remote_ntp_time_estimator.cc | 53 int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) { argument
59 if (!RtpToNtpMs(rtp_timestamp, rtcp_list_, &sender_capture_ntp_ms)) {
70 LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/ |
| H A D | audio_encoder.cc | 27 uint32_t rtp_timestamp,
35 EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
26 Encode( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
|
| H A D | audio_encoder.h | 94 EncodedInfo Encode(uint32_t rtp_timestamp,
99 virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
| H A D | audio_decoder.h | 75 uint32_t rtp_timestamp,
|
| /external/webrtc/webrtc/video/ |
| H A D | 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/ilbc/ |
| H A D | audio_encoder_ilbc.h | 43 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
| H A D | audio_encoder_ilbc.cc | 93 uint32_t rtp_timestamp,
101 first_timestamp_in_buffer_ = rtp_timestamp;
92 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| H A D | audio_decoder_isac_t.h | 34 uint32_t rtp_timestamp,
|
| H A D | audio_decoder_isac_t_impl.h | 79 uint32_t rtp_timestamp,
82 rtp_sequence_number, rtp_timestamp,
76 IncomingPacket(const uint8_t* payload, size_t payload_len, uint16_t rtp_sequence_number, uint32_t rtp_timestamp, uint32_t arrival_timestamp) argument
|
| H A D | audio_encoder_isac_t.h | 63 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| H A D | audio_decoder_impl.h | 42 uint32_t rtp_timestamp,
|
| H A D | dtmf_buffer.h | 67 // |rtp_timestamp| is simply copied into the struct.
68 static int ParseEvent(uint32_t rtp_timestamp,
|
| H A D | dtmf_buffer.cc | 69 int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp, argument
84 event->timestamp = rtp_timestamp;
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| H A D | audio_encoder_pcm.cc | 81 uint32_t rtp_timestamp,
86 first_timestamp_in_buffer_ = rtp_timestamp;
80 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| H A D | audio_encoder_copy_red.cc | 56 uint32_t rtp_timestamp,
61 speech_encoder_->Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
55 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
|
| H A D | audio_encoder_copy_red.h | 46 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| H A D | audio_encoder_g722.h | 44 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
| H A D | audio_encoder_g722.cc | 95 uint32_t rtp_timestamp,
102 first_timestamp_in_buffer_ = rtp_timestamp;
94 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| H A D | audio_encoder_opus.h | 64 EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
| /external/webrtc/talk/media/base/ |
| H A D | rtpdump.cc | 261 uint32_t rtp_timestamp = 0; local
262 packet.GetRtpTimestamp(&rtp_timestamp);
268 first_rtp_timestamp_ = rtp_timestamp;
271 } else if (rtp_timestamp != prev_rtp_timestamp_) {
277 prev_rtp_timestamp_ = rtp_timestamp;
|