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 Post Posted: Fri Apr 24, 2015 5:34 pm 
 

Joined: Thu Jan 01, 2015 1:07 pm
Posts: 30
Hi All,

Recently, I have been observing lip synchronization using GA server and android client. However, I noticed that they are out of sync (around 1-1.5 sec) in either combination of android or windows client. I have been reading about the RTP synchronization after that and found that SR reports are key part to sync server client clocks. Using wire shark I confirmed that SR reports are being received by client and that suggests that the clocks are sync.

The audio packets are of ~80ms duration and that suggest that skew should be around that number isn't it? However, it is not the case. Any help on AV sync would be greatly appreciated :)

Thanks in advance!


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 Post Posted: Fri Jul 10, 2015 10:45 pm 
 

Joined: Tue Sep 16, 2014 4:38 pm
Posts: 9
Have you found a solution for AV sync? I have the same problem: about 1.5s off sync and a very low volume on client.
Thanks


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 Post Posted: Sun Oct 04, 2015 11:20 pm 
 

Joined: Tue Jun 04, 2013 2:07 pm
Posts: 301
This issue is fixed now, and it should be included in the next release. If you would like to test, use the latest version from github.

Chun-Ying


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 Post Posted: Mon Oct 05, 2015 4:28 am 
 

Joined: Tue Sep 16, 2014 4:38 pm
Posts: 9
I'm sorry, but the problem remains with new commits. I have just opened a new issue on github. Please give me some advises.... What should I check for identify the problem?


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 Post Posted: Mon Oct 05, 2015 10:10 am 
 

Joined: Tue Jun 04, 2013 2:07 pm
Posts: 301
Hmm, I think you may provide more information to investigate the issue.

There are many places that could cause the (audio) latency. In general, the places includes 1) capturing; 2) encoding; and 3) transmission; 4) decoding; and 5) playout. In my test environment, which is a Ubuntu Linux x64 virtual desktop + PC client, the delay can be reduced to as short as network latency plus encoding/decoding latency using both MP3 and OPUS codec. The OPUS codec has a even shorter delay than MP3. The performance with a Windows 7 x64 server is similar.

Our modifications now include two parts: 1) use pulseaudio on Linux to capture audio (the original implementation is with ALSA); and 2) buffer-based audio latency reduction on the client. We have not published the pulseaudio module, but the client side modifications has improved a lot. The current experimental codes only apply to PC client and software-based mobile decoder. If you are working with a built-in (hardware) decoder on a mobile device, it is not activated.

For your case, I think the chances could be that the audio capture is too slow. If you would like to check the buffering status at the client, you may activate the debug code in rtspcleint.c (https://github.com/chunying/gaminganywh ... client.cpp, lines 1296--1300) to clarify the problem.

Thanks,
Chun-Ying


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 Post Posted: Mon Feb 13, 2017 9:48 pm 
 

Joined: Mon Feb 13, 2017 9:33 pm
Posts: 1
I also have same problem... Please help me to fix this!

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 Post Posted: Mon Feb 27, 2017 10:04 pm 
 

Joined: Wed Feb 01, 2017 5:26 pm
Posts: 3
sometimes while we watch films or television shows, we found people’s voices earlier than the actual time or after the mouth moving. The audio is out of sync with the video by seconds or more. it is really a very stressful experience, isn’t it? It removes the fun of watching a movie. Anyone tell me the solution? Cisco 300-115 Real Exam Guide

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