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Offline Karlos

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Re: Amiga audio early lead lost..
« on: December 25, 2012, 12:33:17 PM »
Quote from: haywirepc;720181
4 tracks 8 bit.... Mod heaven for a time, but the world moved on...

Why no answer even to sb 8 bit, that did 32 tracks at 8 bit with noisetracker?

You clearly never used OctaMED sound studio. With a fast enough CPU It could play up to 64 tracks* through Paula at 14-bit (up to 56kHz on RTG or doublescan) or through a number of directly supported 16-bit soundcards. Even on slower machines, you could compose at lower fidelity and then render to disk at full quality.

*each track could play notes as either 8 or 16-bit mono or stereo samples, a synthsound or a MIDI device.
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Offline Karlos

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Re: Amiga audio early lead lost..
« Reply #1 on: December 26, 2012, 01:52:32 AM »
Quote from: ChaosLord;720318
Yes that is exactly what I meant by a cheap onboard sound chip.

When you play a mod on that sound chip, 4 channels or 64 channels, it doesn't matter:  The CPU has to do all the work of mixing the soundchannels together which is exactly what Amiga Oktalyzer and TFMX have been doing since the 1980s and Octamed Sound Studio and Digibooster and others have been doing since 1990s.

At least Paula can mix 4 channels together for free on its own.  Your cheap onboard sound chip can't even do that.

It has 10 24-bit DACs capable of 192kHz playback that can be multiplexed down into a single stereo stream (for instance, when using headphones). In what sense is it not capable of mixing channels?

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At least Paula accepts a wide variety of sample rates.  Your cheap onboard soundchip can't.  It just laughs at you and lazily refuses to play any sound that has not been laboriously resampled by the CPU to one of the few frequencies that it can handle.

Well, there you may have a point, but plenty of on sound chips support hardware mixing of sounds at different playback rates. It was very common for older DirectX drivers to support hardware sample replay/mixing on the soundcard before Windows Vista came along with it's entirely new floating-point based audio system (Universal Audio Architecture).

Mixing audio on the CPU, however, uses an absolutely pitiful amount of CPU time, even on any remotely accelerated Amiga (I've done 32-channel stuff in OctaMED SS on an 040), let alone a present day PC.

I can play back two dozen MP3 streams simultaneously on my current (linux) PC before I see pulseaudio lingering in the top 10 processes, and even then it's never more than a few percent of the available capacity of one core.
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Offline Karlos

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Re: Amiga audio early lead lost..
« Reply #2 on: December 27, 2012, 03:57:31 PM »
Quote from: ChaosLord;720443
Because it has 10 wires.
Because it does not mix the 10 audio streams.


Yes it can, actually, but never mind. I suggest you read the technical data sheets for these parts before making such statements.

Look, I love Paula's unique sound more than most and still compose for it. However, the stuff you come out with at times is just mindbogglingly messed up.

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Because it does not support any kind of useable sampling rates.


In the audio industry, 11025, 22050, 24000, 32000, 44100, 48000 and so on are all standard frequencies. Any dedicated sampler (back in the days when you bought dedicated samplers) even used these frequencies. Entire professionally captured sample libraries were published in them.

However, since you raise the issue, as a musician and coder I must point out that neither does Paula. Paula doesn't even have the concept of a sampling rate; it has a sample period which is the inverse function, a physical duration of time measured in clocks between loading the sample registers. Being a 1/x function and x measured as an integer number of clocks, the error in this approximation gets worse for higher playback rates. I've tested this. Given a synthetic looped sinewave of say 64 samples, you can hear the off-pitch playback, particularly for higher notes when comparing to the same notes rendered by software into a buffer.

Also, Paula doesn't mix the four audio outputs it has, either. At least not digitally. The four channels are configured L R R L at the analogue stage and the audio filter happens after that.

Paula would have been far more useful if each of the 4 channels could be panned and if the audio filter could have been toggled per channel. Better still if the filter had more than one cut off frequency that could be selected.

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Because nobody writes 10 channel mods.
Because nobody writes mods using 100% 44.1Khz sample speed.


Irrelevant. Who cares? All the Amiga mod based music I've written in the last 15 years have been OctaMED Mix Mode because as a musician it's *vastly* more useful than the old 4 channel mode. 4 channel mode is handy if all you care about is CPU time, but if you want to make music for it's own sake, no musician is going to turn down more channels and more control over them.

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Of all the sample speeds used in a random mod, this sound chip supports between 0% and 1% of the required sample rates.


So what? You get a set of industry standard replay rates versus an ad-hoc set of non-standard rates that are the result of dividing your PAL/NTSC clock by some fixed integer value.

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It only outputs what the CPU has already mixed itself into 7.1 format or into 2 channel stereo format.
Either way the sound chip does almost nothing.  Its just 10 dumb DACs sitting there waiting to be spoonfed data from the CPU.


Er, no. It supports all sorts of hardware DSP for channel expansion, equalisation and many other things besides. These are standard features on all modern sound chips, not just the DACs.

The driver has the responsibility of playing a potentially limitless number of input audio streams through whatever hardware configuration the chip has been programmed for, and in that it has to do whatever mixing is required. However, the chip does far more than basic DAC duty. The incoming streams can be at any of the standard audio rates. They are resampled to the configured output rate using nth-order digital filtering, equalized, companded, maybe even a bit of DSP effects depending on whatever goofy preset I select, and formatted for whatever output configuration is selected.

On all but the most unfeatured chipsets, the CPU just does upstream mixing of samples into input streams. Yes that does use CPU time, but it's absolutely insignificant on a modern CPU.

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Its not a Paula or anything of that technology level.


No, it isn't.
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Offline Karlos

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Re: Amiga audio early lead lost..
« Reply #3 on: December 27, 2012, 04:22:35 PM »
Quote from: ChaosLord;720453
Guess what?  Paula can do fixed frequency too.


No, it can do fixed period. They aren't quite the same. You can't play a 22050Hz stream because there's no integer value you can divide either the PAL or NTSC clock by to get that value. The closest you can get with PAL is ~22030Hz.
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Offline Karlos

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Re: Amiga audio early lead lost..
« Reply #4 on: December 27, 2012, 04:58:37 PM »
Quote from: ChaosLord;720458
Paula does fixed frequency of 22030hz in PAL or any of a wide range of other values.


None of which are correctly tuned for musical notes. Given that your argument was that paula is "playing" the music by using variable sample rate replay, this is pretty important. Humans with good hearing can typically resolve different pitches to within a few Hz. As you need shorter periods for higher frequency note play back, the error in 1/x for integer x becomes more apparent.

That said, Paula's imperfect reproduction (in many ways - quantisation noise, aliasing etc) is what makes it unique and I really do love it as an instrument in it's own right. However, for *production*, that is, the end result of combining individual instrument performances into a finished recording, all of those cool thing suddenly become a handicap.

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And btw: I tried to convince Gunnar that Natami Paula needs a special fix to be able to generate 44100Hz exactly.  But he repeatedly rejected the idea as unnecessary and silly and pointless.  What is your opinion on that?


It's fine if you don't mind having, for instance, to replay video at the wrong rate to synchronize the audio or resample the latter to avoid issues.
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Offline Karlos

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Re: Amiga audio early lead lost..
« Reply #5 on: December 27, 2012, 05:02:09 PM »
Quote from: ChaosLord;720459
True that it mixes 2 channels to the left and 2 to the right.  But it does not require you to use the CPU to resample all the sounds to a specific frequency.  It supports a giant range of frequencies and just mixes the different samples of different frequencies together like magic.  And it does the mixing on its own with DMA.  One DMA channel per DAC.


It's not really mixing (at least in any data processing perspective). The analogue outputs are simply summed.

I would have much preferred if there were some software controllable left/right volumes (or just volume + pan) for each of the 4 channels. That would have been awesome.
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Offline Karlos

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Re: Amiga audio early lead lost..
« Reply #6 on: December 28, 2012, 12:56:00 PM »
Quote from: Thorham;720498
I find that when playing back CDs in 44Khz 14 bit stereo with calibration, there's not much coloring at all.


Well, the whole point of CyberSound's calibration (the driver for which is what Paula 14-bit mode in AHI uses) is to mitigate the non-linearity of Paula's DACs, so if you have calibrated it well, it should sound relatively clean. It does depend on what you listen to, however. I can still hear quantization noise and aliasing however, particularly on gentle volume ramps in quiet parts of music, even having painstakingly calibrated my machine. These days, it's hidden behind digital noise. I suspect some caps have dried out wherever the analogue stage is decoupled.

It doesn't bother me particularly however, since any multichannel mods I write end up rendered to disk and when I do use Paula as a sample player, I usually want all that colouring.
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Offline Karlos

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Re: Amiga audio early lead lost..
« Reply #7 on: December 28, 2012, 03:38:12 PM »
Quote from: paul1981;720531
Personally, I find even the uncalibrated 14 bit output sounds fantastic. Maybe I'm biased. ;)


Again, it all depends what you use it for. Really, the intent of the 14-bit mode was to overcome the fidelity issues imposed by the existing 8-bit mode. Viewed in that light, you really ought to calibrate your CyberSound driver and use that calibrated output to get the best fidelity from your system.
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