AVI Writer - Sound should have same samplerate as Console

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When recording a game from whatever console, it's currently using it's native 
resolution (which is all and well), but Audio is in 44.1khz.

This is something that should be changed, or at least adjustable.
Many systems use 48khz as it's pretty Standard in TV, but older consoles use 
something like 32khz.

And as the emulator is feeding the AVI file directly without going through 
Windows own resampling (as far as i know?) this really should be acknowledged.

Bit depth i guess is locked at 16bit, as i guess it's made in float and then 
truncated, so that doesn't really matter as long as the noise floor is down.

Thanks

Original issue reported on code.google.com by 24levit...@gmail.com on 9 Oct 2014 at 1:50

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Front end accepts only 44.1khz 16 bit, which all of the cores produce directly. 
 Supporting anything else would be a bit of a project.

Most of the older consoles produce audio directly at very high frequencies 
(1MHZ+).  In most cases, it's sampled down using blargg's blip buffer.  You can 
find out more about the theory and practice of that here:
https://code.google.com/p/blip-buf/
http://www.slack.net/~ant/libs/audio.html#Blip_Buffer
https://code.google.com/p/blip-buffer/

In my mind, the highest priority thing for AVI output would be to get rid of 
that damn timestretcher (dup\drop worthless video frames instead).

Original comment by goyu...@gmail.com on 10 Oct 2014 at 1:22

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Oh, that's to bad. As it resamples everything as i don't think consoles use 
44.1, but 48 (if they even are their, most are probably at 32 or below).

And of course as you say, those that create their own sound, that's another 
story and resampling there is a must either way.

But well how does the timestretcher currently work, cause i notice that the FPS 
is not 60 for me, it's above.

This tends to happen when allowing a master stream of Audio control the Video, 
so the FPS will increase/decrease in order to fit the Audio. This happens only 
if Video and Audio follow 2 different clocks.

Original comment by 24levit...@gmail.com on 10 Oct 2014 at 1:53

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As I said, many of the consoles we emulate natively produce audio at rates much 
higher than 48khz, often times 1mhz or higher.  Other consoles like the genesis 
involve multiple chips at different frequencies that are mixed in analog, so it 
has to be resampled immediately anyway.  The only consoles I can think of 
offhand that support a native 48khz output are a gamecube\wii and a ps2.  On 
the other hand, the ps1 and the PCE support native 44.1khz output.

The timestretcher works by dumping video at its stated rate and then stretching 
audio to match.  How much of a change that makes depends on the console; some 
of ours produce more accurate audio than others.

Original comment by goyu...@gmail.com on 10 Oct 2014 at 3:49

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Thought the N64 did that, guess it resampled to that as it's the TV standard 
then.
Yeah Gamecube PS2 etc had a DAC and files that were all 48khz i guess. So 
pretty much like CDs are 44.1khz.

Hmm, i see thought it acted like Directhow capturing would.

But yeah well Audio is never in sync with the PC (Video in this case) cause 
they have 2 different clocks, it's impossible for them to be in sync and keep 
the same pace. You either stretch or change the pace.

So how are you wanting to tackle that?

There are many ways, Sync to PC, Sync to Audio, Dynamically Resample Audio to 
Fit the PC.

I would say Syncing to the PC is the most correct way in terms of you playing 
it. as 1000 seconds on your PC may be 999 seconds to your Audio.

And that's what you currently do.
As for making that at a rate of 60fps (or whatever) instead of "nearly 60fps", 
you would need to simply pour frames into the file and have some kind of check 
on them.

So if it looks at the PC Timer and it says 0.900 seconds has passed and you 
have 59 frames, you would be getting to many frames, so you let it hit 60 and 
skipp the rest till the Timer Resets(goes to the next second).

If it's to low, you of course duplicate.

But you have to keep track on the frames, as they don't fit into the figure at 
all times, as you can be behind with 3.53 frames, so you have to keep track on 
that, and when it reaches and Int (4 in this case) you add/skip 4 frames  
(Though it would probably always be at 1 as you don't really need to wait on 
several frames).

Still i guess that's what you are trying to do. And it's easier said then done. 
I only knows how to Sync audio to the PC by discarding/adding data during 
recording, and Stretching is better for that (If of course it can keep track if 
Audio really gets missing cause of CPU load and can fill that gap, else it will 
go way off).

How do you Stretch if i may ask, i am guessing dynamic resampling, as you can't 
mean stretching afterwards and you can also not mean a dynamic samplerate (like 
a dynamic fps that changes to fit what it's at) as sample rates are far off to 
be good at that as it doesn't work with Float.

Damn much text of waste there, sorry and Thanks;)

Original comment by 24levit...@gmail.com on 10 Oct 2014 at 7:49

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N64 is programmable adjustable sound rate I think, although I'm not sure of 
that one.

The AVI writer only has to worry about discrepancies between the emulated 
core's sound and video clocks, thankfully; it doesn't need to worry about 
realtime problems.

Anyway, I'm filing this whole audio upgrade thing as low priority enhancement.  

Original comment by goyu...@gmail.com on 10 Oct 2014 at 1:38

• Added labels: Priority-Low, Type-Enhancement

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Okay thanks for having it in your mind:)

Original comment by 24levit...@gmail.com on 10 Oct 2014 at 3:08

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As an experimental option, I've added the ability to dump avi without 
stretching audio (video is stretched instead).  Support for anything other than 
44.1khz isn't any time soon though.

Original comment by goyu...@gmail.com on 11 Oct 2014 at 5:54

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Interesting, will try it out when i find a build later;)

Thanks

Original comment by 24levit...@gmail.com on 11 Oct 2014 at 8:40