| /external/chromium_org/third_party/webrtc/system_wrappers/interface/ |
| 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/chromium_org/media/cast/rtp_receiver/ |
| rtp_receiver_defines.h | 22 uint32 rtp_timestamp; member in struct:media::cast::RtpCastHeader
|
| /external/chromium_org/media/cast/logging/ |
| logging_impl.cc | 24 uint32 rtp_timestamp,
28 rtp_timestamp, frame_id);
34 uint32 rtp_timestamp,
40 rtp_timestamp, frame_id, frame_size, key_frame, target_bitrate);
45 EventMediaType event_media_type, uint32 rtp_timestamp, uint32 frame_id,
49 rtp_timestamp, frame_id, delay);
59 uint32 rtp_timestamp; local
64 big_endian_reader.ReadU32(&rtp_timestamp);
70 // rtp_timestamp is enough - no need for frame_id as well.
74 rtp_timestamp,
[all...] |
| logging_raw_unittest.cc | 31 RtpTimestamp rtp_timestamp = 123u; local
34 rtp_timestamp, frame_id);
41 EXPECT_EQ(rtp_timestamp, frame_events_[0].rtp_timestamp);
54 RtpTimestamp rtp_timestamp = 123u; local
60 rtp_timestamp, frame_id, size, key_frame, target_bitrate);
67 EXPECT_EQ(rtp_timestamp, frame_events_[0].rtp_timestamp);
82 RtpTimestamp rtp_timestamp = 123u; local
86 rtp_timestamp, frame_id, delay)
108 RtpTimestamp rtp_timestamp = 123u; local
139 RtpTimestamp rtp_timestamp = 123u; local
[all...] |
| receiver_time_offset_estimator_impl_unittest.cc | 61 RtpTimestamp rtp_timestamp = 0; local
69 rtp_timestamp,
80 rtp_timestamp, frame_id);
87 rtp_timestamp, frame_id);
110 RtpTimestamp rtp_timestamp = 0; local
118 rtp_timestamp,
132 event_c_time, FRAME_ACK_RECEIVED, VIDEO_EVENT, rtp_timestamp, frame_id);
137 event_b_time, FRAME_ACK_SENT, VIDEO_EVENT, rtp_timestamp, frame_id);
|
| encoding_event_subscriber_unittest.cc | 109 /*rtp_timestamp*/ i * 100,
127 RtpTimestamp rtp_timestamp = 100; local
131 rtp_timestamp,
138 rtp_timestamp,
160 RtpTimestamp rtp_timestamp = 100; local
163 rtp_timestamp,
170 RtpTimestamp relative_rtp_timestamp = rtp_timestamp - first_rtp_timestamp_;
192 RtpTimestamp rtp_timestamp = 100; local
195 now, FRAME_PLAYOUT, AUDIO_EVENT, rtp_timestamp,
202 RtpTimestamp relative_rtp_timestamp = rtp_timestamp - first_rtp_timestamp_
222 RtpTimestamp rtp_timestamp = 100; local
314 RtpTimestamp rtp_timestamp = 100; local
350 RtpTimestamp rtp_timestamp = 100; local
400 RtpTimestamp rtp_timestamp = 100; local
520 RtpTimestamp rtp_timestamp = 12345; local
560 RtpTimestamp rtp_timestamp = 0xffffffff - 20; local
589 RtpTimestamp rtp_timestamp = 100; local
[all...] |
| logging_defines.h | 55 RtpTimestamp rtp_timestamp; member in struct:media::cast::FrameEvent
83 RtpTimestamp rtp_timestamp; member in struct:media::cast::PacketEvent
|
| logging_impl_unittest.cc | 50 uint32 rtp_timestamp = 0; local
56 now, FRAME_CAPTURE_BEGIN, VIDEO_EVENT, rtp_timestamp, frame_id);
59 rtp_timestamp += kFrameIntervalMs * 90;
78 uint32 rtp_timestamp = 0; local
87 FRAME_ENCODED, VIDEO_EVENT, rtp_timestamp,
90 rtp_timestamp += kFrameIntervalMs * 90;
108 uint32 rtp_timestamp = 0; local
117 rtp_timestamp,
121 rtp_timestamp += kFrameIntervalMs * 90;
135 uint32 rtp_timestamp = 0u local
180 RtpTimestamp rtp_timestamp = 0; local
[all...] |
| stats_event_subscriber_unittest.cc | 71 uint32 rtp_timestamp = 0; local
79 rtp_timestamp,
83 rtp_timestamp += 90;
105 uint32 rtp_timestamp = 0; local
116 rtp_timestamp,
123 rtp_timestamp += 90;
151 uint32 rtp_timestamp = 0; local
158 rtp_timestamp,
162 rtp_timestamp += 90;
184 uint32 rtp_timestamp = 0 local
219 uint32 rtp_timestamp = 0; local
264 uint32 rtp_timestamp = 0; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator.cc | 41 uint32_t rtp_timestamp = 0; local
46 &rtp_timestamp)) {
52 ntp_secs, ntp_frac, rtp_timestamp, &rtcp_list_, &new_rtcp_sr)) {
68 int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) {
74 if (!RtpToNtpMs(rtp_timestamp, rtcp_list_, &sender_capture_ntp_ms)) {
|
| remote_ntp_time_estimator_unittest.cc | 97 uint32_t rtp_timestamp = GetRemoteTimestamp(); local
102 EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp));
109 EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp));
|
| rtcp_receiver_help.h | 81 uint32_t rtp_timestamp; member in class:webrtc::RTCPHelp::RTCPPacketInformation
|
| /external/chromium_org/media/cast/receiver/ |
| cast_receiver_impl.cc | 139 const uint32 rtp_timestamp = encoded_frame->rtp_timestamp; local
147 rtp_timestamp,
165 "rtp_timestamp", encoded_frame->rtp_timestamp,
171 const uint32 rtp_timestamp = encoded_frame->rtp_timestamp; local
179 rtp_timestamp,
188 uint32 rtp_timestamp,
196 now, FRAME_DECODED, AUDIO_EVENT, rtp_timestamp, frame_id)
[all...] |
| frame_receiver.cc | 124 rtp_header.rtp_timestamp;
126 now, PACKET_RECEIVED, event_media_type_, rtp_header.rtp_timestamp,
149 fresh_sync_rtp = rtp_header.rtp_timestamp;
178 RtpTimestamp rtp_timestamp = local
182 rtp_timestamp, cast_message.ack_frame_id_);
209 GetPlayoutTime(encoded_frame->rtp_timestamp);
272 base::TimeTicks FrameReceiver::GetPlayoutTime(uint32 rtp_timestamp) const {
276 static_cast<int32>(rtp_timestamp - lip_sync_rtp_timestamp_),
|
| frame_receiver_unittest.cc | 91 rtp_header_.rtp_timestamp = 0;
121 const int64 rtp_timestamp = (now - start_time_) * local
123 CHECK_LE(0, rtp_timestamp);
130 static_cast<uint32>(rtp_timestamp));
205 EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp);
241 rtp_header_.rtp_timestamp = 0;
255 rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame;
281 rtp_header_.rtp_timestamp += rtp_advance_per_frame;
307 frame_events[i].rtp_timestamp);
[all...] |
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/test/ |
| rtp_header_parser.cc | 29 rtp_timestamp(0),
66 uint32 rtp_timestamp, ssrc; local
67 big_endian_reader.ReadU32(&rtp_timestamp);
75 parsed_packet->rtp_timestamp = rtp_timestamp;
|
| rtp_header_parser.h | 32 uint32 rtp_timestamp; member in struct:media::cast::transport::RtpCastTestHeader
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_sync_module.cc | 36 uint32_t rtp_timestamp = 0; local
41 &rtp_timestamp)) {
47 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
|
| /external/chromium_org/media/cast/audio_sender/ |
| audio_sender.cc | 131 last_send_time_, FRAME_ENCODED, AUDIO_EVENT, encoded_frame->rtp_timestamp,
136 frame_id_to_rtp_timestamp_[frame_id & 0xff] = encoded_frame->rtp_timestamp;
140 encoded_frame->rtp_timestamp);
277 const RtpTimestamp rtp_timestamp = local
282 rtp_timestamp,
|
| /external/chromium_org/media/cast/video_sender/ |
| video_sender.cc | 102 RtpTimestamp rtp_timestamp = GetVideoRtpTimestamp(capture_time); local
105 rtp_timestamp, kFrameIdUnknown);
109 rtp_timestamp,
117 "rtp_timestamp", rtp_timestamp);
166 last_send_time_, FRAME_ENCODED, VIDEO_EVENT, encoded_frame->rtp_timestamp,
171 frame_id_to_rtp_timestamp_[frame_id & 0xff] = encoded_frame->rtp_timestamp;
177 "rtp_timestamp", encoded_frame->rtp_timestamp);
181 encoded_frame->rtp_timestamp);
330 RtpTimestamp rtp_timestamp = local
[all...] |
| /external/chromium_org/media/cast/rtcp/ |
| rtcp.cc | 50 rtcp_->OnReceivedLipSyncInfo(remote_sender_info.rtp_timestamp,
263 void Rtcp::OnReceivedLipSyncInfo(uint32 rtp_timestamp, uint32 ntp_seconds,
269 lip_sync_rtp_timestamp_ = rtp_timestamp;
274 bool Rtcp::GetLatestLipSyncTimes(uint32* rtp_timestamp,
288 *rtp_timestamp = lip_sync_rtp_timestamp_;
399 uint32 rtp_timestamp = it->rtp_timestamp_; local
408 event_media_type_, rtp_timestamp,
415 rtp_timestamp, kFrameIdUnknown);
420 rtp_timestamp, kFrameIdUnknown, event_it->delay_delta);
|
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdump.cc | 261 uint32 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;
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_single_stream.cc | 113 uint32_t rtp_timestamp = header.timestamp + local
134 overuse_detector->Update(payload_size, -1, rtp_timestamp, arrival_time_ms);
|
| remote_bitrate_estimator_unittest_helper.h | 51 uint32_t rtp_timestamp; member in struct:webrtc::testing::RtpStream::RtpPacket
170 uint32_t rtp_timestamp,
|
| /external/chromium_org/third_party/webrtc/video_engine/test/libvietest/testbed/ |
| tb_external_transport.cc | 115 uint32_t rtp_timestamp = ptr[4] << 24; local
116 rtp_timestamp += ptr[5] << 16;
117 rtp_timestamp += ptr[6] << 8;
118 rtp_timestamp += ptr[7];
121 _firstRTPTimestamp = rtp_timestamp;
125 _lastSendRTPTimestamp != rtp_timestamp) {
126 _send_frame_callback->FrameSent(rtp_timestamp);
129 _lastSendRTPTimestamp = rtp_timestamp;
225 if (previous_drop_ && _firstRTPTimestamp != rtp_timestamp)
441 uint32_t rtp_timestamp = ptr[4] << 24 local
[all...] |
