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Searched refs:rtp_timestamp (Results 1 – 25 of 139) sorted by relevance

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/external/webrtc/modules/remote_bitrate_estimator/
Doveruse_detector_unittest.cc97 void UpdateDetector(uint32_t rtp_timestamp, in UpdateDetector() argument
104 rtp_timestamp, receive_time_ms, receive_time_ms, packet_size, in UpdateDetector()
142 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
146 UpdateDetector(rtp_timestamp, now_ms_, packet_size); in TEST_F()
148 rtp_timestamp += frame_duration_ms * 90; in TEST_F()
156 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
160 UpdateDetector(rtp_timestamp, now_ms_, packet_size); in TEST_F()
161 rtp_timestamp += frame_duration_ms * 90; in TEST_F()
174 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
178 UpdateDetector(rtp_timestamp, now_ms_, packet_size); in TEST_F()
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/external/webrtc/modules/rtp_rtcp/source/
Dremote_ntp_time_estimator_unittest.cc73 uint32_t rtp_timestamp, in UpdateRtcpTimestamp() argument
76 rtt, ntp_secs, ntp_frac, rtp_timestamp)); in UpdateRtcpTimestamp()
101 uint32_t rtp_timestamp = GetRemoteTimestamp(); in TEST_F() local
106 EXPECT_EQ(kNotEnoughRtcpSr, estimator_->Estimate(rtp_timestamp)); in TEST_F()
114 EXPECT_EQ(capture_ntp_time_ms, estimator_->Estimate(rtp_timestamp)); in TEST_F()
127 uint32_t rtp_timestamp = GetRemoteTimestamp(); in TEST_F() local
130 EXPECT_EQ(capture_ntp_time_ms, estimator_->Estimate(rtp_timestamp)); in TEST_F()
142 rtp_timestamp = GetRemoteTimestamp(); in TEST_F()
146 EXPECT_EQ(capture_ntp_time_ms, estimator_->Estimate(rtp_timestamp)); in TEST_F()
Dabsolute_capture_time_receiver.cc50 uint32_t rtp_timestamp, in OnReceivePacket() argument
59 if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp, in OnReceivePacket()
66 rtp_timestamp, rtp_clock_frequency, last_rtp_timestamp_, in OnReceivePacket()
72 last_rtp_timestamp_ = rtp_timestamp; in OnReceivePacket()
91 uint32_t rtp_timestamp, in InterpolateAbsoluteCaptureTimestamp() argument
99 rtc::dchecked_cast<uint64_t>(rtp_timestamp - last_rtp_timestamp) in InterpolateAbsoluteCaptureTimestamp()
107 uint32_t rtp_timestamp, in ShouldInterpolateExtension() argument
Drtp_sender_audio.cc147 uint32_t rtp_timestamp, in SendAudio() argument
150 return SendAudio(frame_type, payload_type, rtp_timestamp, payload_data, in SendAudio()
158 uint32_t rtp_timestamp, in SendAudio() argument
163 TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type", in SendAudio()
188 dtmf_timestamp_ = rtp_timestamp; in SendAudio()
207 if ((rtp_timestamp - dtmf_timestamp_last_sent_) < in SendAudio()
213 dtmf_timestamp_last_sent_ = rtp_timestamp; in SendAudio()
214 uint32_t dtmf_duration_samples = rtp_timestamp - dtmf_timestamp_; in SendAudio()
235 dtmf_timestamp_ = rtp_timestamp; in SendAudio()
267 packet->SetTimestamp(rtp_timestamp); in SendAudio()
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Dremote_ntp_time_estimator.cc41 uint32_t rtp_timestamp) { in UpdateRtcpTimestamp() argument
43 if (!rtp_to_ntp_.UpdateMeasurements(ntp_secs, ntp_frac, rtp_timestamp, in UpdateRtcpTimestamp()
64 int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) { in Estimate() argument
66 if (!rtp_to_ntp_.Estimate(rtp_timestamp, &sender_capture_ntp_ms)) { in Estimate()
86 RTC_LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp in Estimate()
Dabsolute_capture_time_sender.cc44 uint32_t rtp_timestamp, in OnSendPacket() argument
52 if (!ShouldSendExtension(send_time, source, rtp_timestamp, in OnSendPacket()
59 last_rtp_timestamp_ = rtp_timestamp; in OnSendPacket()
75 uint32_t rtp_timestamp, in ShouldSendExtension() argument
112 rtp_timestamp, rtp_clock_frequency, last_rtp_timestamp_, in ShouldSendExtension()
Dsource_tracker.cc38 entry.rtp_timestamp = packet_info.rtp_timestamp(); in OnFrameDelivered()
47 entry.rtp_timestamp = packet_info.rtp_timestamp(); in OnFrameDelivered()
66 entry.timestamp_ms, key.source, key.source_type, entry.rtp_timestamp, in GetSources()
Dabsolute_capture_time_receiver.h59 uint32_t rtp_timestamp,
67 uint32_t rtp_timestamp,
74 uint32_t rtp_timestamp,
Drtp_sender_video_frame_transformer_delegate.cc31 uint32_t rtp_timestamp, in TransformableVideoSenderFrame() argument
40 timestamp_(rtp_timestamp), in TransformableVideoSenderFrame()
110 uint32_t rtp_timestamp, in TransformFrame() argument
123 encoded_image, video_header, payload_type, codec_type, rtp_timestamp, in TransformFrame()
Drtp_sender_video.h97 uint32_t rtp_timestamp, in SendVideo() argument
103 return SendVideo(payload_type, codec_type, rtp_timestamp, capture_time_ms, in SendVideo()
111 uint32_t rtp_timestamp,
120 uint32_t rtp_timestamp,
/external/webrtc/api/transport/rtp/
Drtp_source.h42 uint32_t rtp_timestamp) in RtpSource() argument
46 rtp_timestamp, in RtpSource()
52 uint32_t rtp_timestamp, in RtpSource() argument
58 rtp_timestamp_(rtp_timestamp) {} in RtpSource()
84 uint32_t rtp_timestamp() const { return rtp_timestamp_; } in rtp_timestamp() function
96 rtp_timestamp_ == o.rtp_timestamp();
/external/webrtc/api/video_codecs/
Dvp8_temporal_layers.cc67 uint32_t rtp_timestamp) { in NextFrameConfig() argument
69 return controllers_[stream_index]->NextFrameConfig(0, rtp_timestamp); in NextFrameConfig()
73 uint32_t rtp_timestamp, in OnEncodeDone() argument
79 return controllers_[stream_index]->OnEncodeDone(0, rtp_timestamp, size_bytes, in OnEncodeDone()
84 uint32_t rtp_timestamp) { in OnFrameDropped() argument
86 controllers_[stream_index]->OnFrameDropped(stream_index, rtp_timestamp); in OnFrameDropped()
Dvp8_temporal_layers.h53 uint32_t rtp_timestamp) override;
56 uint32_t rtp_timestamp,
62 void OnFrameDropped(size_t stream_index, uint32_t rtp_timestamp) override;
Dvp8_frame_buffer_controller.h143 uint32_t rtp_timestamp) = 0;
154 uint32_t rtp_timestamp,
161 virtual void OnFrameDropped(size_t stream_index, uint32_t rtp_timestamp) = 0;
/external/webrtc/api/
Drtp_packet_info.cc24 uint32_t rtp_timestamp, in RtpPacketInfo() argument
30 rtp_timestamp_(rtp_timestamp), in RtpPacketInfo()
54 (lhs.rtp_timestamp() == rhs.rtp_timestamp()) && in operator ==()
/external/webrtc/api/test/
Dvideocodec_test_stats.cc19 size_t rtp_timestamp, in FrameStatistics() argument
22 rtp_timestamp(rtp_timestamp), in FrameStatistics()
44 ss << " rtp_timestamp " << rtp_timestamp; in ToString()
/external/webrtc/modules/video_coding/codecs/test/
Dvideocodec_test_stats_impl_unittest.cc38 EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp); in TEST()
48 EXPECT_EQ(kTimestamp + i, frame_stat->rtp_timestamp); in TEST()
55 EXPECT_EQ(kTimestamp + i, frame_stat->rtp_timestamp); in TEST()
64 EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp - i); in TEST()
/external/openscreen/cast/standalone_sender/
Dstreaming_vp8_encoder.cc147 work_unit.rtp_timestamp = RtpTimeTicks(); in EncodeAndSend()
149 work_unit.rtp_timestamp = RtpTimeTicks::FromTimeSinceOrigin( in EncodeAndSend()
151 if (work_unit.rtp_timestamp <= last_enqueued_rtp_timestamp_) { in EncodeAndSend()
158 if (sender_->GetInFlightMediaDuration(work_unit.rtp_timestamp) > in EncodeAndSend()
176 (work_unit.rtp_timestamp - last_enqueued_rtp_timestamp_) in EncodeAndSend()
183 last_enqueued_rtp_timestamp_ = work_unit.rtp_timestamp; in EncodeAndSend()
366 stats.rtp_timestamp = work_unit->rtp_timestamp; in ComputeFrameEncodeStats()
430 frame.rtp_timestamp = results.rtp_timestamp; in SendEncodedFrame()
/external/openscreen/cast/streaming/
Dframe_collector_unittest.cc51 part.rtp_timestamp = kSomeRtpTimestamp + (RtpTimeDelta::FromTicks(200) * i); in TEST()
86 EXPECT_EQ(part.rtp_timestamp, frame.rtp_timestamp); in TEST()
119 part.rtp_timestamp = kSomeRtpTimestamp; in TEST()
156 EXPECT_EQ(kSomeRtpTimestamp, frame.rtp_timestamp); in TEST()
174 part.rtp_timestamp = kSomeRtpTimestamp; in TEST()
Dpacket_receive_stats_tracker.cc19 RtpTimeTicks rtp_timestamp, in OnReceivedValidRtpPacket() argument
42 (rtp_timestamp - last_rtp_packet_timestamp_) in OnReceivedValidRtpPacket()
52 last_rtp_packet_timestamp_ = rtp_timestamp; in OnReceivedValidRtpPacket()
/external/webrtc/audio/
Dchannel_send_frame_transformer_delegate.cc22 uint32_t rtp_timestamp, in TransformableAudioFrame() argument
30 rtp_timestamp_(rtp_timestamp), in TransformableAudioFrame()
87 uint32_t rtp_timestamp, in Transform() argument
94 frame_type, payload_type, rtp_timestamp, rtp_start_timestamp, in Transform()
/external/webrtc/modules/rtp_rtcp/source/rtcp_packet/
Dsender_report.h40 void SetRtpTimestamp(uint32_t rtp_timestamp) { in SetRtpTimestamp() argument
41 rtp_timestamp_ = rtp_timestamp; in SetRtpTimestamp()
54 uint32_t rtp_timestamp() const { return rtp_timestamp_; } in rtp_timestamp() function
/external/webrtc/modules/audio_coding/codecs/opus/
Dopus_complexity_unittest.cc42 uint32_t rtp_timestamp = 0u; in RunComplexityTest() local
45 info = encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded); in RunComplexityTest()
46 rtp_timestamp += kInputBlockSizeSamples; in RunComplexityTest()
/external/webrtc/video/
Dstream_synchronization_unittest.cc54 uint32_t rtp_timestamp = in DelayedStreams() local
57 ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); in DelayedStreams()
61 rtp_timestamp = clock_sender_.CurrentTime().ms() * video_frequency / 1000; in DelayedStreams()
63 ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); in DelayedStreams()
67 rtp_timestamp = clock_sender_.CurrentTime().ms() * audio_frequency / 1000; in DelayedStreams()
69 ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); in DelayedStreams()
73 rtp_timestamp = clock_sender_.CurrentTime().ms() * video_frequency / 1000; in DelayedStreams()
75 ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); in DelayedStreams()
/external/webrtc/system_wrappers/source/
Drtp_to_ntp_estimator.cc124 uint32_t rtp_timestamp, in UpdateMeasurements() argument
128 int64_t unwrapped_rtp_timestamp = unwrapper_.Unwrap(rtp_timestamp); in UpdateMeasurements()
182 bool RtpToNtpEstimator::Estimate(int64_t rtp_timestamp, in Estimate() argument
187 int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp); in Estimate()

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