| /external/webrtc/webrtc/modules/audio_coding/codecs/ |
| audio_encoder.h | 94 EncodedInfo Encode(uint32_t rtp_timestamp,
99 virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
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| audio_decoder.h | 75 uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| audio_encoder_cng.cc | 99 uint32_t rtp_timestamp,
108 rtp_timestamps_.push_back(rtp_timestamp);
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_encoder_g722.cc | 95 uint32_t rtp_timestamp,
102 first_timestamp_in_buffer_ = rtp_timestamp;
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| audio_encoder_isac_t_impl.h | 117 uint32_t rtp_timestamp,
124 packet_timestamp_ = rtp_timestamp;
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_single_stream.cc | 75 uint32_t rtp_timestamp = header.timestamp + local
99 if (estimator->inter_arrival.ComputeDeltas(rtp_timestamp, arrival_time_ms,
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| remote_bitrate_estimator_unittest_helper.h | 52 uint32_t rtp_timestamp; member in struct:webrtc::testing::RtpStream::RtpPacket
171 uint32_t rtp_timestamp,
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| remote_bitrate_estimator_unittest_helper.cc | 67 packet->rtp_timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
224 uint32_t rtp_timestamp,
230 header.timestamp = rtp_timestamp;
258 (packet->arrival_time + 500) / 1000, packet->rtp_timestamp,
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| /external/webrtc/talk/media/base/ |
| 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;
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| /external/webrtc/webrtc/video/ |
| vie_receiver.cc | 435 uint32_t rtp_timestamp = 0; local
437 &rtp_timestamp)) {
441 ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
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| stream_synchronization_unittest.cc | 40 rtcp.rtp_timestamp = NowRtp(frequency, offset);
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtcp_receiver_help.cc | 37 rtp_timestamp(0),
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| rtcp_sender.cc | 287 void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
290 last_rtp_timestamp_ = rtp_timestamp;
474 uint32_t rtp_timestamp = local
483 report->WithRtpTimestamp(rtp_timestamp);
[all...] |
| rtcp_sender.h | 94 void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
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| rtcp_packet.h | 168 void WithRtpTimestamp(uint32_t rtp_timestamp) {
169 sr_.RTPTimestamp = rtp_timestamp;
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| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_encoder_opus.cc | 134 uint32_t rtp_timestamp,
139 first_timestamp_in_buffer_ = rtp_timestamp;
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| dtmf_buffer.cc | 69 int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
84 event->timestamp = rtp_timestamp;
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| audio_decoder_impl.cc | 55 uint32_t rtp_timestamp,
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| /external/webrtc/webrtc/system_wrappers/source/ |
| rtp_to_ntp_unittest.cc | 67 EXPECT_TRUE(RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp, ×tamp_in_ms));
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| /system/bt/stack/a2dp/ |
| a2dp_vendor_aptx_encoder.cc | 437 uint32_t rtp_timestamp = local
440 a2dp_aptx_encoder_cb.timestamp += rtp_timestamp;
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| a2dp_vendor_aptx_hd_encoder.cc | 425 uint32_t rtp_timestamp = local
429 a2dp_aptx_hd_encoder_cb.timestamp += rtp_timestamp;
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| /external/webrtc/webrtc/voice_engine/include/ |
| voe_rtp_rtcp.h | 82 uint32_t RTP_timestamp;
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| audio_coding_module_impl.cc | 138 uint32_t rtp_timestamp = local
147 last_rtp_timestamp_ = rtp_timestamp;
152 rtp_timestamp, rtc::ArrayView<const int16_t>(
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| bwe_test_framework_unittest.cc | 793 uint32_t rtp_timestamp = 0; local
810 if (rtp_timestamp > media_packet->header().timestamp) {
813 rtp_timestamp = media_packet->header().timestamp;
[all...] |
| /external/webrtc/webrtc/voice_engine/ |
| channel.cc | 1822 uint32_t rtp_timestamp = 0; local
[all...] |
