Sound reverting at the last second when Frameserving from Vegas

[lonrot]

I have no idea what that is happening. It shouldn't matter how many times you open/close access it, as long as vegas is opened and the frameserver is serving - especially if you have PCM samples written (they are physically written to disc)

Is only the audio affected ? you can open video any number of times without having to restart the frameserver ?

You can't hear the audio after the second playback but the video will play correctly.

[poisondeathray]

I have no idea... something is really messed up on your system. So this doesn't occur if you uncheckmark the write PCM in debugmode configuration ? (but then you get the audio glitch)

[lonrot]

I have no idea... something is really messed up on your system. So this doesn't occur if you uncheckmark the write PCM in debugmode configuration ? (but then you get the audio glitch)

Yes, If PCM is checked in Frameserver the audio will always reproduce as many times as needed, but alas the audio glitch.

Looks like I'll have to return to x264fw encoding in Vegas along with AC3 ACM...

[_Al_]

I have this reported from a college also, frame server working only one time, so it has perhaps the same bug, that gives this other problem (glitch at the end of audio or audio from frame server working only one time). Just want to say that op is not alone with this problem. Unchecking that PCM pre-render solves this problem.

I myself had a problem with this audio glitch before for some reason, but I cannot reproduce it now again.

[_Al_]

Looks like I'll have to return to x264fw encoding in Vegas along with AC3 ACM...

So just unchecking that PCM pre-render is not working for you ? Using avisynth script with avisource loading that signpost.avi should work.

[poisondeathray]

Unchecking PCM gives him glitch, checking it removes glitch, but only one access

Actually you only need to access it once , if you use CLI tools (unless you're doing multipass encoding - but if you were doing that, a lossless intermediate would be better in most cases)


This is the first time I've heard of a one access problem, with this or any program . The only thing semi related I can think of is if multiple programs are attempting accessing the dummy AVI at the same time - that can cause problems

[_Al_]

I was under impression that that glitch and one time audio working problem is caused by checking that box ...

In my batch file I have basically these lines (%~d1%~p1%~n1%~x1 is basically dropped signpost.avi):

SET aviname_input=%~d1%~p1%~n1%~x1
SET delay="%~dp0tools\subroutines\delay.BAT"
( type nul >> "%aviname_input%" ) 2>nul || ( call %delay% )

where if that signpost.avi is in use it jumps into delay subroutine that checks every 30 seconds if it is still busy , if not, it lets script to continue, yes, that signpost.avi cannot be busy more than one time at the same time , maybe this is good direction why it is not functioning ...

[lonrot]

I was under impression that that glitch and one time audio working problem is caused by checking that box ...

As poison mentioned, unchecking that PCM box causes the one time audio issue while having it checked makes the sound glitch at the end.

My knowledge is very limited regarding scripting languages, I would probably die from a stroke trying to figure things out on my own

[_Al_]

unchecking that PCM box causes the one time audio issue while having it checked makes the sound glitch at the end

oh, yes, sorry, my bad ... something switched in my head disputing it, but not sure why ...

you do not have to know anything just post Vegas project properties and you can get batch file that does it all, I would provide it, if it is not in that zip already, we can actually test it, just for the heck of it, if that script would fix it, I personally think it would not, but, who knows ... you would just drop that signpost.avi on that icon in windows explorer to encode video ...

[lonrot]

Good news!

I'm able to render the audio with Ripbot via frameserver doing the following:

1) Start frameserver .avi (PCM unchecked).
2) Open avs script in Ripbot
3) Cancel and restart frameserver .avi to recover the first audio access.
3) Render in Ripbot.

The audio renders correctly.

you do not have to know anything just post Vegas project properties and you can get batch file that does it all

Your script works like a charm with HD Progressive from HD .bat. And it renders the audio correctly during the first read, a second time will kill the audio, but it's more efficient than Ripbot at the current state, AAC Nero is also a plus!

There's only one concern: the audio is delivered as 44100 while Ripbot does 48000 using the avs script. Can you change that?

[poisondeathray]

I downloaded _Al_'s batch file and there is nothing there that is changing the sampling rate. That suggests that you are sending 44100 from vegas , and ripbot is upsampling the audio




Can you do a quick test?

Start your frameserver as normal
Create modified avs script

AVISource("signpost.avi")
Info()

Open that in vdub, what does it say on the overlay about sampling rate ?



Then close ,restart the frameserver , load the "fake" AVI directly into vdub, and look at file=>file information. What does it say about the sampling rate ?

[lonrot]

Yes, Ripbot is upscaling the sampling rate, I have no idea why, the project properties in Vegas are set to 48000 even the audio sources are 48000.

Video:
Frame size, fps (µs per frame): 1920x1080, 29.970 fps (33367 µs)
Length: 2404 frames (1:20.21)
Decompressor: DebugMode FSVFWC (internal ... (DFSC)
Number of key frames: 2404
Min/avg/max/total key frame size: 8/8/8 (19K)
Min/avg/max/total delta size: (no delta frames)
Data rate: 2 kbps (75.00% overhead)

Audio:
Sampling rate: 44100Hz
Channels: 2 (Stereo)
Sample precision: N/A
Compression: DFSC ACM Codec (0xdfac)
Layout: 1 chunks (80.21s preload)
Length: 64168 samples (1:20.21)
Min/avg/max/total frame size: 64168/64168/64168 (63K)
Data rate: 6 kbps (0.04% overhead)

[poisondeathray]

That's the only explanation I can think of, project properties set to 44100 but you said you checked that. Make sure you pushed apply or whatever

It doesn't matter if write PCM is checked or unchecked, frameserver should serve whatever project properties are set to

[lonrot]

That's the only explanation I can think of, project properties set to 44100 but you said you checked that. Make sure you pushed apply or whatever

Computer life is full of mysteries to unravel or bugs.

[lonrot]

Returning to the "one access audio" issue. Virtualdub does return an error after pausing and resuming playback (the audio dies). Maybe you can shed some light.

VirtualDub Error

The audio codec reported an error while decompressing audio data.

Error code: 1 (MMSYSERR_ERROR)

[poisondeathray]

Doesn't make any sense, if write PCM samples is check marked , because they are written to disc. There is no audio codec, it's uncompressed PCM wave

In fact if you have samples written , you can even open up the "fake" AVI with mediainfo and it will report audio, sample rate , because it's physically present

[lonrot]

Doesn't make any sense, if write PCM samples is check marked , because they are written to disc. There is no audio codec, it's uncompressed PCM wave

PCM is unchecked in Frameserver and 48000 is selected in project properties (to avoid confusion) :P

[poisondeathray]

Yeah its tough to keep track

If they aren't written then it will be using custom DMFS codec , it's like a VFW/ACM wrapper. Then it's possible to get errors

But eitherway I still have no idea why this is happening on your setup, either the once access, or 44100

[_Al_]

only "DVD from HD interlace.BAT" or "DVD from HD 30p,25p.BAT" that make DVD, changes audio to 48000 to follow DVD specs, otherwise it should have frequency that is set in Vegas project properties ...

[lonrot]

Thank you guys for all your amazing help, the solutions are either rendering from Vegas with x264fw or merging video and audio with avidemux later, this is definitely an isolated case caused by my computer beyond the programs that are being used. Thanks again

[racer-x]

I know I'm a little late on this conversation. I can confirm that DMFS will write a 44100 wave file regardless of project settings. I never check Write pcm on the Frameserver window. I always export audio separately as .wav out of Vegas, then I frameserve out the video only. I then encode the wav into whatever I need such as ac3 or aac and mux it with the elemental video stream.

You could also frameserve into Virtualdub, import your wav and encode via external encoder. I usually frameserve into MeGUI or Simplex264 encoder.

[_Al_]

I can confirm that DMFS will write a 44100 wave file regardless of project settings.

not here, project is set to 48000 Hz, original is progressive m2t, 1440x1080, 48000 Hz, exporting through dmfs, "unchecking write PCM ...." and I get 48000Hz,
Nero encoder is fed by pipe from bepipe that loads avisynth script, there is no conversion involved, because if project in Vegas I have set to 44,100Hz I get 44100 Hz AAC out of it.
BePipe.exe --script "Import(^input.avs^)" | neroAacEnc.exe -lc -cbr 256000 -if - -of "out.m4a"

so that dmfs can give us different results, someting is going on behind scenes as well?

[lonrot]

so that dmfs can give us different results, someting is going on behind scenes as well?

Perhaps racer-x's computer is affected with a similar problem. After I updated from Win 8 to 8.1 some sound issues have been plaguing my computer, it should not be related but who knows, I'm going to try Monday or Tuesday with a laptop at the office (which is also 8.1) to see if I get different results.