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A client-driven media synchronization mechanism for RTP ...
https://www.researchgate.net/publication/275223650_A_client-driven_media_synchronization_mechanism_for_RTP_packet-based_video_streaming#:~:text=Media%20synchronization%20is%20used%20to%20correctly%20playback%20a,networks%2C%20an%20RTP%2FRTCP%20protocol%20suite%20is%20usually%20employed.
Synchronization in RTP A Made Easy Tutorial
http://www.siptutorial.net/RTP/synchro.html
So the sequence number is not enough for synchronization. You already know that in a audio/video session audio and video data are transmitted using separate channels (if you don't know this, please go through applications of RTP). The receiver matches the video data with corresponding audio data using timestamp.
RTP - Synchronizing media streams · GitHub
https://gist.github.com/simonkim/a9a3aa4f1ca04448c64212b3b079d107
The audio and video may even be transmitted by different hosts if the reference clocks on the two hosts are synchronized by some means such as NTP. A receiver can then synchronize presentation of the audio and video packets by relating their RTP timestamps using the timestamp pairs in RTCP SR packets . -- 5.1 RTP Fixed Header Fields, RFC 3550 ...
7. Lip Synchronization - RTP: Audio and Video for the ...
https://www.oreilly.com/library/view/rtp-audio-and/0672322498/ch07.html
Lip Synchronization - RTP: Audio and Video for the Internet [Book] Chapter 7. Lip Synchronization. A multimedia session comprises several media streams, and in RTP each is transported via a separate RTP session. Because the delays associated with different encoding formats vary greatly, and because the streams are transported separately across ...
Chapter 7. Lip Synchronization | RTP: Audio and Video for ...
https://flylib.com/books/en/4.245.1.56/1/
This chapter describes how RTP provides the information needed to facilitate the synchronization of multiple media streams. The typical use for this technique is to align audio and video streams to provide lip synchronization, although the techniques described may be applied to the synchronization of any set of media streams. Figure 7.1.
Transmitting Audio and Video using RTP - Oracle
https://www.oracle.com/java/technologies/javase/transmitting-audio-video-rtp.html
The default (first available) RTP format is set for each track. For video, special attention is taken to ensure that the input sizes are usable for RTP transmission. Real-time scaling is applied when necessary. Note that due to limitations of the JMF 2.1 implementation, audio and video are not in tight synchronization.
rtp to webrtc: How to handle audio/video synchronization ...
https://github.com/pion/webrtc/discussions/1825
(Except in their case they record incoming audio/video WebRTC streams by sending the RTP/RTCP to gstreamer. In my case of using Pion to send RTP to WebRTC, I would have to do the opposite: send the RTP like you already have done to the WebRTC tracks, and also get the RTCP from GStreamer to Pion.
audio - AAC RTP timestamps and synchronization - Stack ...
https://stackoverflow.com/questions/13726282/aac-rtp-timestamps-and-synchronization
I am currently streaming audio (AAC-HBR at 8kHz) and video (H264) using RTP. Both feeds works fine individually, but when put together they get out of sync pretty fast (lass than 15 sec). I am not sure how to increment the time stamp on the audio RTP header, I thought it should be the time difference between two RTP packets (around 127ms) or a ...
Synchronizing Audio and Video in RTSP Source Filter
https://social.msdn.microsoft.com/Forums/en-US/32e20270-5f4e-49f7-a915-5b22d5596c62/synchronizing-audio-and-video-in-rtsp-source-filter
Hello, I've been developing an RTSP source filter and can decode both audio and video correctly but I'm having problems synchronizing them. The audio is currently 1-1.5 seconds behind the video. My source filter is subclassed from CSource, implements IAMFilterMiscFlags ands sets the MiscFlag · How do you set the start times on the samples, are you ...
Streaming audio and video in sync for mp4 container …
https://stackoverflow.com/questions/35843178/streaming-audio-and-video-in-sync-for-mp4-container-using-gstreamer-framework
There seem to be issues with AAC in RTP as well as other RTP payloaders in gstreamer. As mentioned in the other answer, it is anyway the best strategy to not demux and split video and audio if it is desired to have a synchronized playback on the receiver side. Additionally, a container format that is streamable can also improve overall ...
e2b05126bcc9b79d5ccd897bfae3b240dfb2bccd - chromiumos ...
https://chromium.googlesource.com/chromiumos/third_party/webrtc-apm/+/e2b05126bcc9b79d5ccd897bfae3b240dfb2bccd
WebRTC APM. APM is the audio processing module of WebRTC project in charge of effects like echo cancellation, noise suppression, etc. The purpose of this project is to build a standalone library for Chrome OS system side audio processing.
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