| /aosp_15_r20/external/webrtc/modules/rtp_rtcp/source/ |
| H A D | absolute_capture_time_interpolator.cc | 41 uint32_t rtp_timestamp, in OnReceivePacket()
79 uint32_t rtp_timestamp, in InterpolateAbsoluteCaptureTimestamp()
95 uint32_t rtp_timestamp, in ShouldInterpolateExtension()
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| H A D | absolute_capture_time_sender.cc | 44 uint32_t rtp_timestamp, in OnSendPacket()
75 uint32_t rtp_timestamp, in ShouldSendExtension()
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| H A D | remote_ntp_time_estimator.cc | 55 uint32_t rtp_timestamp) { in UpdateRtcpTimestamp()
78 NtpTime RemoteNtpTimeEstimator::EstimateNtp(uint32_t rtp_timestamp) { in EstimateNtp()
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| H A D | remote_ntp_time_estimator_unittest.cc | 75 uint32_t rtp_timestamp = GetRemoteTimestamp(); in TEST_F() local
101 uint32_t rtp_timestamp = GetRemoteTimestamp(); in TEST_F() local
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| H A D | rtp_sender_audio.cc | 151 uint32_t rtp_timestamp, in SendAudio()
162 uint32_t rtp_timestamp, in SendAudio()
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| H A D | rtp_sender_video_frame_transformer_delegate.cc | 32 uint32_t rtp_timestamp, in TransformableVideoSenderFrame()
118 uint32_t rtp_timestamp, in TransformFrame()
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| H A D | rtp_sender_video.cc | 473 uint32_t rtp_timestamp, in SendVideo()
760 uint32_t rtp_timestamp, in SendEncodedImage()
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| /aosp_15_r20/external/webrtc/api/video_codecs/ |
| H A D | vp8_temporal_layers.cc | 67 uint32_t rtp_timestamp) { in NextFrameConfig()
73 uint32_t rtp_timestamp, in OnEncodeDone()
84 uint32_t rtp_timestamp) { in OnFrameDropped()
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| /aosp_15_r20/external/webrtc/modules/remote_bitrate_estimator/ |
| H A D | overuse_detector_unittest.cc | 96 void UpdateDetector(uint32_t rtp_timestamp, in UpdateDetector()
141 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
155 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
173 uint32_t rtp_timestamp = 10 * 90; in TEST_F() local
223 uint32_t rtp_timestamp = frame_duration_ms * 90; in TEST_F() local
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| /aosp_15_r20/external/webrtc/modules/audio_coding/neteq/ |
| H A D | packet_arrival_history.cc | 23 void PacketArrivalHistory::Insert(uint32_t rtp_timestamp, in Insert()
69 int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp, in GetDelayMs()
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| /aosp_15_r20/external/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| H A D | sender_report.h | 40 void SetRtpTimestamp(uint32_t rtp_timestamp) { in SetRtpTimestamp()
54 uint32_t rtp_timestamp() const { return rtp_timestamp_; } in rtp_timestamp() function
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| /aosp_15_r20/external/webrtc/api/transport/rtp/ |
| H A D | rtp_source.h | 50 uint32_t rtp_timestamp, in RtpSource()
82 uint32_t rtp_timestamp() const { return rtp_timestamp_; } in rtp_timestamp() function
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| /aosp_15_r20/external/webrtc/audio/ |
| H A D | channel_send_frame_transformer_delegate.cc | 22 uint32_t rtp_timestamp, in TransformableOutgoingAudioFrame()
88 uint32_t rtp_timestamp, in Transform()
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| H A D | channel_send.cc | 347 uint32_t rtp_timestamp, in SendData()
368 uint32_t rtp_timestamp, in SendRtpAudio()
925 int64_t absolute_capture_timestamp_ms) { in InitFrameTransformerDelegate()
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| /aosp_15_r20/external/webrtc/video/ |
| H A D | video_stream_buffer_controller.cc | 64 const uint32_t rtp_timestamp; member
276 void VideoStreamBufferController::FrameReadyForDecode(uint32_t rtp_timestamp, in FrameReadyForDecode()
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| H A D | decode_synchronizer.h | 83 uint32_t rtp_timestamp() const { return rtp_timestamp_; } in rtp_timestamp() function
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| /aosp_15_r20/external/webrtc/modules/video_coding/codecs/vp8/ |
| H A D | screenshare_layers.cc | 292 uint32_t rtp_timestamp, in OnEncodeDone()
414 uint32_t rtp_timestamp) { in OnFrameDropped()
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| H A D | default_temporal_layers.cc | 517 uint32_t rtp_timestamp, in OnEncodeDone()
626 uint32_t rtp_timestamp) { in OnFrameDropped()
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| /aosp_15_r20/external/webrtc/modules/audio_coding/codecs/opus/ |
| H A D | audio_encoder_opus_unittest.cc | 526 uint32_t rtp_timestamp = 12345; // Just a number not important to this test. in TEST_P() local
787 uint32_t rtp_timestamp = 0u; in TEST_P() local
842 uint32_t rtp_timestamp = 0; in TEST() local
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| /aosp_15_r20/external/webrtc/modules/rtp_rtcp/include/ |
| H A D | remote_ntp_time_estimator.h | 45 int64_t Estimate(uint32_t rtp_timestamp) { in Estimate()
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| /aosp_15_r20/external/webrtc/modules/video_coding/timing/ |
| H A D | inter_frame_delay.cc | 37 uint32_t rtp_timestamp, in CalculateDelay()
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| /aosp_15_r20/external/openscreen/cast/streaming/ |
| H A D | packet_receive_stats_tracker.cc | 19 RtpTimeTicks rtp_timestamp, in OnReceivedValidRtpPacket()
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| H A D | encoded_frame.h | 68 RtpTimeTicks rtp_timestamp; member
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| /aosp_15_r20/external/webrtc/api/ |
| H A D | rtp_packet_info.cc | 23 uint32_t rtp_timestamp, in RtpPacketInfo()
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| /aosp_15_r20/external/webrtc/api/test/ |
| H A D | videocodec_test_stats.cc | 19 size_t rtp_timestamp, in FrameStatistics()
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