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#144 FrameServer Question
I have the Frameserver up and running for VD2. FFmpeg or such software is having problems. It just crashes when trying to access the audio part. Even programs like Staxrip refuses to load the audio. It loads the video without any issues.
Any idea why only part of the video loads but not the other?
Discussion
Anonymous
How do you access VD frameserver from FFmpeg? Please show command line.
Check Audio->audio conversion and try 16-bit stereo, I never checked before how frameserver works with extended formats.
Audio: AAC | Tried both Direct & Full processing.
Sample Rate: 48,000Hz
Bitrate: 128
ffmpeg Command Line:
Error Find Gain 1 using ffmpeg 3.3.3 (1.7.0.6)
Find Gain 1 using ffmpeg 3.3.3 failed with exit code: 1 (0x1)
The exit code might be a system error code: STATUS_WAIT_1
The exit code might be a system error code: Incorrect function.
-------------------- Find Gain 1 using ffmpeg 3.3.3 --------------------
ffmpeg.exe -i Test1.vdr -sn -vn -hide_banner -af volumedetect -f null NUL
C:\Users\Revan\Desktop\Test1.vdr: Invalid data found when processing input
Find Gain 1 using ffmpeg 3.3.3 failed with exit code: 1 (0x1)
Are you sure FFmpeg has support for vd frameserver? Do you know what is demuxer name for this? I think this it is not implemented at all.
The problem with other programs may be something else. I tried serving video+audio from vd2 to another instance of vd2, and it works. 6 channel float format works too. Again, float samples may be incompatible with some programs, to make sure you need to enable "audio->full processing" and "audio->convert format->16bit".
Do you know of any other software that can demux or read the vdr file on the Audio Side?
Currently I have access to the following software for Audio:
Not sure if some Flags just needs changed for FFMPEG or if can't be done at all. It's hard to find selective things in FFMPEG documation.
No, I don't know. Avisynth AviSource should have no problem so maybe you could setup "Avisynth tunnel".
AVISource is default for vdr format. How would I setup a avisynth Tunnel?
The idea is you make this: vd->avisynth->ffmpeg. Not sure what you want to achieve.
All I want to do is be able to Load the audio in StaxRip. This way I don't have to make dual copies (Ori + Edit versions). Plus Uncompressed data takes up allot of space. This way I can use just one file.
It looks like right now best option is extact the audio file to Wav and then import it into Staxrip along with vdr file.
Not sure if it's VD2 or one of the audio programs that can't decode the vdr file.
I really can't check what happens in StaxRip. Did you try to reach StaxRip developer?
The only issue I could think of is float pcm format, but you did not tell
whether StaxRip can open float pcm audio
what your audio actually is
whether you tried to fix it with convert format
The Devloper of Staxrip hasn't been seen for ages on any forums or main page for StaxRip. I started to make changes to source code. It's just diffcult trying to figure everything out.
When I put the file in mediainfo it list the audio as 7.1 PCM Audio.
When capturing uncompressed audio in VD2. What file format does it capture it in? I'm just having issue figuring out if the file is recorded in .wav or in some other format.
If you don't enable compression (usually not recommended) it will save as pcm. Also all capture sources are 16-bit afaik.
A good way to know what is being frameserved is to open .vdr in vd2 and see file->properties.
I think I got everything working, I just have one last question. Should the Video Frame be the same number as the audio Segments? or are they always different?
Cool, you mean you fixed StaxRip?
Yes, audio is passed independently, not bound to video frames.
Yes, it seems you can't load the audio seperatly which is what Staxrip defaults to. You have to have the script load the audio with the video. It seems ffmpeg doesn't understand what .vdr is and how to process it.
I have included the needed files with latest modded version of Staxrip. Everything should run smoothly now.
Or you could just extact the wave file before starting the service. Everything should sync up correctly.