Trackers that do 14-bit sound on 68000?

[Wilse]

Well, some argue that the Paula's non-linear output gives it a distinctive sound,

Yeah, those guys have better ears than me.

[bloodline]

Yeah, those guys have better ears than me.

Perhaps not better ears, but probably actively looking for a distinctive quality that can be found in Paula's audio reproduction...

Her hardware does colour the audio quite a bit, and I like to use her for 8bit sample play back (though I prefer record the audio samples on my Mac with it's 24bit FireWire audio box, to ensure a clean recording).

All the old Samplers were quite distinctive, which is why I still use my Roland W-30 for all 12bit work

[bloodline]

Thanks for the replies,

I am actually quite up to snuff on the more *advanced* sound / music-recording aspects on expanded big-box Amigas, as for many years I had a big-box Amiga with AHI compatible 16-bit sound card, 68040, oodles of RAM etc. etc. and used it for MIDI and audio editing work. BUT during that time I did little work with Paula audio or trackers on low-end Amigas, so am a little rusty in that regard, hence my original question.

For the high-quality (i.e. 16 and 24 bit CD-quality work) I've since moved onto PC DAWs as the hardware is far cheaper, faster, and the quality of the software is much better (for DAW work, only, I must emphasize ).  I use Reaper, for instance.  I had spent many years expanding my big box Amiga so that little of the audio work I was doing on it actually used the original motherboard hardware, instead relying on third party CPU cards and AHI audio cards, etc. So I don't see the switch to PC DAWs as a move away from Amigas, as I had essentially already moved away from the hardware that made the Amiga unique when still using my Amiga.

I have decided that when I do audio composing on an Amiga, I am going to go "purely Amiga"; using only the motherboard native chipset and stock 68000 processor.  So while my big box Amiga is gone, I have kept a low-end Amiga (an A500 with Hard-drive side-car and a bit of expanded RAM) as my "pure" Amiga audio workstation.  I figure that if I'm going to use an Amiga - I might as well go for the full experience and use what makes that machine unique - i.e. Paula audio for that distinctive sound, and trackers.  Otherwise, what's the point?

So, thanks for the clarification on 14-bit audio with a stock 680000 system. It seems that a stock A500 cannot do 14-bit audio.  No problem.  Like I said, I want to hear that Paula sound, anyway.

Now, to expand the question.  Were there any trackers that could handle more than 4 channels of audio and would still work on an Amiga 500?

I have found Oktalyzer, but I wonder if there were any more.  

I am most familiar with OctaMED SS 1.03, which would handle more than four channesl, but I was running that on my 68040 A2000. I have read that version of OctaMED needs a 68020.

OctaMED 5 I think was the last preSound Studio version of OctaMED and ran fine on my A500.

A word of warning though, 8 channel mode isn't very nice... I would recommend sticking with 4 channel mode and doing overdubs if you need more channels

[Wilse]

Perhaps not better ears, but probably actively looking for a distinctive quality that can be found in Paula's audio reproduction...

Her hardware does colour the audio quite a bit, and I like to use her for 8bit sample play back (though I prefer record the audio samples on my Mac with it's 24bit FireWire audio box, to ensure a clean recording).

All the old Samplers were quite distinctive, which is why I still use my Roland W-30 for all 12bit work

Well, as far as Paula goes, there was also the added link in the chain of a variety of parallel port, 8-bit samplers. These add their own colour to the mixture.
Even still, to my ears my 8 bit sample collection, some of which I sampled myself, some of which I didn't and some of which probably wasn't even recorded via Paula, sounded very similar when played back via different hardware to what it did on my Amiga.

I know there will be certain, specific sounds which are audibly different when played back via Paula than other 8-bit chips but my ears have never noticed this to any degree where I could say with certainty that I preferred the Paula version. This is despite around a decade of pretty-much-daily sampling and recording on the A1200.

I should also add that I can't tell the difference between a 320kbps mp3 and a 16 bit .wav either. However, there were a couple of youngsters on my sound engineering course who claimed they could. So, in my case, I think it *is* my ears (or at least how I perceive what they hear).  

In such circumstances the pursuit of said, unique sounds would take me into a spiral of diminishing returns that I'm far too old to devote the time too.

[Amiga_Nut]

- i.e. Paula audio for that distinctive sound, and trackers.  Otherwise, what's the point?


Could anyone comment on this ? What makes the Paula audio distinctive ?

Surely its only as good as the sampling hardware and techniques used to capture the sounds?

The limitations of Paula (number of channels, max sampling rat, bitrate etc) can be done with any modern tracker or DAW .

I get the "SID" chip thing being analog in nature its hard to get emulation right - but paula's audio is basically all digital.

I composed music on the Amiga for years , mostly using soundstudio  - using the stock hardware and a midi interface. I got some great sounds out of the machine and knew a fair few techniques for getting the best sound quality. (perhaps someone recognises my nick (polyp) )

But I never once thought the Amiga had a "unique or distinctive sound" (other than the limitations of 8bit samples) - I always thought the sound hardware could have been better - i wanted a successor to SID.

N

I've been saying that for years, Paula is just a simple 4 DAC IC and output through a pretty noisy bus to be quite honest. When people say they attribute the sound of some tune to an Amiga what they really mean is all those sh1t samples that everyone kept using even when they were totally inappropriate. Games companies were the worst at this to be honest, mega demos rarely used those rubbish mid 80s synth samples past, well the mid 80s to be exact. Even the filter is horrible.

And yet in 1985 on the Amiga (AKA Amiga 1000) to have effectively no sound chip and just replace it with a completely flexible 4 channel 8 bit DAC replace it with  basic AM/FM control of playback was pure and utter genius. The only thing wrong with every other Amiga after the 1000 is Commodore never did the very simple trick of making 6 or 8 channel (or dual Paula) motherboards to give you enough for a game. 4 channels was fine in 1985-87 but by 1989/90 we should have had at least a simple dual Paula implementation IMO.

OR the short story....don't shoot Jay Miner and co for putting one in the Amiga (AKA A1000)...... shoot the talentless musicians and mostly the penny pinching software houses that wouldn't stretch to a purchasing sampler and renting some high quality synths from Korg et al to make new exciting 100% appropriate samples for their game soundtrack tunes.

[ChaosLord]

IIRC, 14-bit replay requires cycle exact synchronization between the paired channels that are used to play the MSB/LSB halves of your 16-bit sample. I think that the 14-bit routines bypass Paula DMA sample replay all together and give her (registers) a good poking.

No.
14-bit audio works by using Paula's built-in for free Channel Locking function which locks 2 channels together into 1 channel.  1 of the channels provides 8-bits of data and the other channel provides 6-bits of volume control.  They are fed with the normal DMA mechanism, otherwise it would be quite useless.

Either that, or they pre-fill short buffers for each channel and then trigger her to replay them directly.

Either way, I am not sure that trying the replay as channels in a tracker will work. A simple test would be to play a regular sample on a channel and it's inverse on a channel on the same side. If they cancel each other out completely (or almost completely) then your plan seems plausible. Otherwise, even a sub-millisecond latency between two channels is going to sound dreadful.

If the tracker is programmed 100% correctly then it would, in fact, work perfectly.  I did it myself, not in a tracker but in whatever sample editor I was using that day.  AudioMaster 3  or Audition 4 or whatever.  In my test I sent my sound sample to the left speaker and my inverted sound sample to the right speaker and to my astonishment I heard silence!    IMHO Using 2 8-bit channels together as a 16-bit channel works as long as the tracker replay routine is programmed 100% correctly to start the 2 channels at exactly the same time.

[ChaosLord]

I should also add that I can't tell the difference between a 320kbps mp3 and a 16 bit .wav either. However, there were a couple of youngsters on my sound engineering course who claimed they could. So, in my case, I think it *is* my ears (or at least how I perceive what they hear).  

You can hear the difference if you save the .mp3 out as a sound sample.  Pick your fave sample out of the song... like a 3 second sample.  Cut it out and save it as its own separate sample.  Listen to it on its own.  It will sound like crap.  I have experienced this myself many times.

.mp3s rely on creating an aural illusion.  If you have cast a Dispel Illusion spell or your mom built you with the Immunity to Illusion DNA v3.1 installed to your brain then of course you will hear the difference and .mp3s will sound worse than a normal .shn or .flac or .16sv or .wav file.

[Ral-Clan]

You can hear the difference if you save the .mp3 out as a sound sample.  Pick your fave sample out of the song... like a 3 second sample.  Cut it out and save it as its own separate sample.  Listen to it on its own.  It will sound like crap.  I have experienced this myself many times.

How would this work? An MP3 uncompressed to a WAV should sound exactly the same as an MP3 being uncompressed on the fly.  The should be bit for bit identical.

[ChaosLord]

How would this work? An MP3 uncompressed to a WAV should sound exactly the same as an MP3 being uncompressed on the fly.  The should be bit for bit identical.

It is bit for bit identical.  That is the point.  And it "sounds" perfect.

Now cut a piece out.

Listen to the piece by itself.

It will sound like crap.  All kinds of weird distortion.

When you remove the illusion the distortion becomes obvious to anyone.  Its weird.

[Britelite]

No.
14-bit audio works by using Paula's built-in for free Channel Locking function which locks 2 channels together into 1 channel.  1 of the channels provides 8-bits of data and the other channel provides 6-bits of volume control.  They are fed with the normal DMA mechanism, otherwise it would be quite useless.

That's not actually how it works though, there's no "channel locking" function in the Paula. The trick is to play the highest 8 bits on one channel at full volume and the lower 6 bits and minimum volume on the second channel. The volume control is not touched at any point.

[Ral-Clan]

It is bit for bit identical.  That is the point.  And it "sounds" perfect.

Now cut a piece out.

Listen to the piece by itself.

It will sound like crap.  All kinds of weird distortion.

When you remove the illusion the distortion becomes obvious to anyone.  Its weird.

Sorry, but I don't follow your argument.  MP3 encoding is a lossy encoding format, i.e the frequencies that are removed are done so at the encoding stage.  When you listen to an MP3 or de-encode one to a WAV, they should be identical.  i.e. you cannot "get back" lost frequencies by un-encoding an MP3 back to a WAV.

A 3-second sound clip of an MP3 or the same 3-second clip of this mp3 converted back to a WAV should sound identical.

Granted, an mp3 is always inferior to the ORIGINAL master wav, but the two 3-second test clips both derived post-encoding process should sound the same.

[Wilse]

I thought he meant take a 3 second clip from a 320 kbps mp3 and compare it to the same 3 seconds clipped from a 16 bit wav?

[Ral-Clan]

I thought he meant take a 3 second clip from a 320 kbps mp3 and compare it to the same 3 seconds clipped from a 16 bit wav?

I don't know, he would have to write his original explanation a little more clearly.  Maybe I misunderstood.

[Karlos]

It is bit for bit identical.  That is the point.  And it "sounds" perfect.

Now cut a piece out.

Listen to the piece by itself.

It will sound like crap.  All kinds of weird distortion.

When you remove the illusion the distortion becomes obvious to anyone.  Its weird.

It's not that weird. MP3 isn't just about conversion of amplitude/time domain to frequency/phase domain (and subsequent quantization into bands). Even before that, pretty sophisticated algorithms are applied that account for the way in which sound is perceived and also how the ear reacts to sudden changes in volume. Collectively, this is the Psychoacoustic Modelling part of the encoding and it's used to make quantitative decisions about where to discard information about sound you won't perceive well because of the state your ear and auditory cortex will be in given the most recent sounds you have just heard.

For example, when there's a sudden increase in volume, the ear physically responds by tightening the tympanic membrane, which results in a loss in sensitivity. If a quiet sound immediately follows again, your ear takes some time to relax again and the cells responsible for reporting certain frequencies will be recovering too. You won't hear the quiet part as well as you would without being exposed to the previous sound and it may be that you are especially desensitised to certain frequencies for a moment. All of these phenomena and more are used by the PM stage to work out what it can discard that you won't notice.

So, when you take a sample out of the middle of a MP3 track that's had a good PM algorithm applied, particularly if the sample comes from a region following a sudden transient change in volume, pitch change or whatever, without that preceding cue, you are left with audio encoded in a manner your ear is not expecting and you likely will notice encoding artefacts as a consequence.

[ChaosLord]

It's not that weird. MP3 isn't just about conversion of amplitude/time domain to frequency/phase domain (and subsequent quantization into bands). Even before that, pretty sophisticated algorithms are applied that account for the way in which sound is perceived and also how the ear reacts to sudden changes in volume. Collectively, this is the Psychoacoustic Modelling part of the encoding and it's used to make quantitative decisions about where to discard information about sound you won't perceive well because of the state your ear and auditory cortex will be in given the most recent sounds you have just heard.

For example, when there's a sudden increase in volume, the ear physically responds by tightening the tympanic membrane, which results in a loss in sensitivity. If a quiet sound immediately follows again, your ear takes some time to relax again and the cells responsible for reporting certain frequencies will be recovering too. You won't hear the quiet part as well as you would without being exposed to the previous sound and it may be that you are especially desensitised to certain frequencies for a moment. All of these phenomena and more are used by the PM stage to work out what it can discard that you won't notice.

So, when you take a sample out of the middle of a MP3 track that's had a good PM algorithm applied, particularly if the sample comes from a region following a sudden transient change in volume, pitch change or whatever, without that preceding cue, you are left with audio encoded in a manner your ear is not expecting and you likely will notice encoding artefacts as a consequence.

Well, since you explained it that way... it doesn't sound weird at all.  :razz:  :biglaugh:

[ChaosLord]

I don't know, he would have to write his original explanation a little more clearly.  Maybe I misunderstood.

Let me say it another way:
I don't know how to magically cut sounds out of an mp3.

An mp3 is compressed.  You cannot just arbitrarily cut a piece of sound out of it.  The file is compressed in blocks.  At a minimum you would have to cut at a block boundary.

It is a basic rule of computer science that compressed data must be UNcompressed first before you start manipulating it.


Just like you can't magically cut 3 seconds out of the middle of a .lha file.  First you must UNcompress the .lha file into a normal file.  Then you cut the data.