| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | remote_ntp_time_estimator_unittest.cc | 58 uint32_t rtp_timestamp, bool expected_result) { in UpdateRtcpTimestamp()
86 uint32_t rtp_timestamp = GetRemoteTimestamp(); in TEST_F() local
|
| D | remote_ntp_time_estimator.cc | 53 int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) { in Estimate()
|
| D | rtcp_receiver_help.h | 86 uint32_t rtp_timestamp; variable
|
| D | rtcp_packet.h | 168 void WithRtpTimestamp(uint32_t rtp_timestamp) { in WithRtpTimestamp()
|
| D | rtcp_sender.cc | 287 void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp, in SetLastRtpTime()
474 uint32_t rtp_timestamp = in BuildSR() local
|
| /external/webrtc/webrtc/system_wrappers/source/ |
| D | rtp_to_ntp.cc | 59 uint32_t rtp_timestamp, in UpdateRtcpList()
95 bool RtpToNtpMs(int64_t rtp_timestamp, in RtpToNtpMs()
|
| /external/webrtc/webrtc/system_wrappers/include/ |
| D | rtp_to_ntp.h | 25 uint32_t rtp_timestamp; member
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/ |
| D | audio_encoder.cc | 27 uint32_t rtp_timestamp, in Encode()
|
| D | audio_decoder.cc | 63 uint32_t rtp_timestamp, in IncomingPacket()
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | audio_decoder_impl.cc | 55 uint32_t rtp_timestamp, in IncomingPacket()
|
| D | dtmf_buffer.cc | 69 int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp, in ParseEvent()
|
| /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()
139 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
153 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
171 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
219 uint32_t rtp_timestamp = frame_duration_ms * 90; in TEST_F() local
251 uint32_t rtp_timestamp = frame_duration_ms * 90; in TEST_F() local
282 uint32_t rtp_timestamp = frame_duration_ms * 90; in TEST_F() local
|
| D | remote_bitrate_estimator_unittest_helper.h | 52 uint32_t rtp_timestamp; member
|
| D | remote_bitrate_estimator_single_stream.cc | 75 uint32_t rtp_timestamp = header.timestamp + in IncomingPacket() local
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| D | audio_decoder_isac_t_impl.h | 79 uint32_t rtp_timestamp, in IncomingPacket()
|
| D | audio_encoder_isac_t_impl.h | 117 uint32_t rtp_timestamp, in EncodeInternal()
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| D | audio_encoder_ilbc.cc | 93 uint32_t rtp_timestamp, in EncodeInternal()
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| D | audio_encoder_copy_red.cc | 56 uint32_t rtp_timestamp, in EncodeInternal()
|
| /external/webrtc/webrtc/video/ |
| D | vie_sync_module.cc | 33 uint32_t rtp_timestamp = 0; in UpdateMeasurements() local
|
| D | vie_receiver.cc | 435 uint32_t rtp_timestamp = 0; in InsertRTCPPacket() local
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_encoder_pcm.cc | 81 uint32_t rtp_timestamp, in EncodeInternal()
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| D | audio_encoder_g722.cc | 95 uint32_t rtp_timestamp, in EncodeInternal()
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| D | audio_encoder_cng.cc | 99 uint32_t rtp_timestamp, in EncodeInternal()
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_encoder_opus.cc | 134 uint32_t rtp_timestamp, in EncodeInternal()
|
| /external/webrtc/talk/media/base/ |
| D | rtpdump.cc | 261 uint32_t rtp_timestamp = 0; in UpdateStreamStatistics() local
|