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  #11  
Old 2013-02-09, 11:22
joshuajohn89 joshuajohn89 is offline
 
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Quote:
Originally Posted by SteveDiverse View Post
No offense intended...just saying...when you run out of processing power, you're done.

...How would you push a processor past its ability to process?
I am not talking about pushing the processor beyond its limit, but instead about preventing it from reaching that limit at all.

What is unfeasible about this proposal? If I change my sample rate during playback the audio will momentarily drop out, which is obviously undesirable, so it seems to me that for the transition to be seamless the audio must be sampled at two or more different rates simultaneously, while only the stream of the highest resolution that can be played without ill effect will be output at any given time as a performance safeguard.

Am I terribly unaware of something significant here?
  #12  
Old 2013-02-09, 12:22
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Quote:
Originally Posted by joshuajohn89 View Post

Am I terribly unaware of something significant here?
You've obviously never had the CPU max out then tried to change the latency in Prefs and the your whole computer freezing. Yeah, that'd work well in a live shituation

There is simply no way Props can automate potentially hundreds of third-party sound driver processes!

It's an idiot idea.


"I am not talking about pushing the processor beyond its limit, but instead about preventing it from reaching that limit at all."

Right, so then don't work at high sample rates to start with and don't use devices/Combinators that stress the system?
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Last edited by JiggeryPokery; 2013-02-09 at 12:29.
  #13  
Old 2013-02-09, 13:05
joshuajohn89 joshuajohn89 is offline
 
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Quote:
Originally Posted by JiggeryPokery View Post
You've obviously never had the CPU max out then tried to change the latency in Prefs and the your whole computer freezing. Yeah, that'd work well in a live shituation

There is simply no way Props can automate potentially hundreds of third-party sound driver processes!
Perhaps, then, such a feature would be better implemented in the driver rather than the DAW itself, while Reason would merely provide the option to make use of the feature when it is detected as being available.

Also, waiting for the processor to max out before attempting to lighten its load is clearly not a good idea, so a threshold would have to be employed instead, I think.

Anyway, I am simply brainstorming here. I am not yet convinced of the suggestion's supposedly fatal shortcomings.
  #14  
Old 2013-02-09, 13:19
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JiggeryPokery JiggeryPokery is online now
 
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Quote:
Originally Posted by joshuajohn89 View Post
Perhaps, then, such a feature would be better implemented in the driver rather than the DAW itself, while Reason would merely provide the option to make use of the feature when it is detected as being available.

Also, waiting for the processor to max out before attempting to lighten its load is clearly not a good idea, so a threshold would have to be employed instead, I think.

Anyway, I am simply brainstorming here. I am not yet convinced of the suggestion's supposedly fatal shortcomings.
Yeah, it's not you I'm having a dig at, you're looking at both sides of the argument, which is fair enough.

cyclic frequently posts far-fetched ideas, and often ideas that are already quite achievable by actually thinking through a problem and working around it. It's a symptom of the "spoonfeed me" generation where a few people want every little thing done for them. The only feature he hasn't suggested yet is the "Create #1 Hit" button.

It's like 96k in live performance. Seriously? I'd be quite surprised if there was any other artist on the planet who uses a computer on stage running a 96k sample-rate. Most of us still can't manage 96k full-time on high end desktop rigs!
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  #15  
Old 2013-02-09, 13:49
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dub08 dub08 is offline
 
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Quote:
Originally Posted by JiggeryPokery View Post
cyclic frequently posts far-fetched ideas, and often ideas that are already quite achievable by actually thinking through a problem and working around it. It's a symptom of the "spoonfeed me" generation where a few people want every little thing done for them. The only feature he hasn't suggested yet is the "Create #1 Hit" button.
No matter how many times he is wrong or, posts a feature that makes no sense... He will be back to post 5 feat. suggestions at once. What does it accomplish?

Are you paying attention cyclic? It is a terrible habit you've got here. It only proves that you aren't RTFM or, using the program correctly. (See rpg8 thread below this one)
You should hang out with us on the user forum more and, forget all about features for a while.
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  #16  
Old 2013-02-09, 14:00
joshuajohn89 joshuajohn89 is offline
 
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Quote:
Originally Posted by JiggeryPokery View Post
Yeah, it's not you I'm having a dig at, you're looking at both sides of the argument, which is fair enough.

cyclic frequently posts far-fetched ideas, and often ideas that are already quite achievable by actually thinking through a problem and working around it. It's a symptom of the "spoonfeed me" generation where a few people want every little thing done for them. The only feature he hasn't suggested yet is the "Create #1 Hit" button.

It's like 96k in live performance. Seriously? I'd be quite surprised if there was any other artist on the planet who uses a computer on stage running a 96k sample-rate. Most of us still can't manage 96k full-time on high end desktop rigs!
That is funny.. I was actually just digging through cyclic's posting history (all ten pages, 200+ feature suggestions) when I saw that you had replied. He doesn't seem to possess any sort of filter, which I do admire, even if it makes for a great many suggestions that quickly fade into the depths of the forum.

I suppose there is a fine line between a product that provides a proper set of features (which Reason certainly does, but there is yet room for improvement) and one which simply gives the user instant gratification, and while we Reasoners tend to be a rather self-sufficient DIY lot, I hate to see potential newcomers dissuaded by a perceived inaccessibility of the program.


P.S.: I wonder how long we must wait for 96kHz to become the new standard for every producer and his tablet-based workstation.. :}
  #17  
Old 2013-02-09, 19:21
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SteveDiverse SteveDiverse is offline
 
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Quote:
Originally Posted by joshuajohn89 View Post
the audio must be sampled at two or more different rates simultaneously, while only the stream of the highest resolution that can be played without ill effect will be output at any given time as a performance safeguard.
Lets use rates of 96k and 88k as the examples...

Running at 96k, computer is calculating 96k samples per second (each sample represents 1/96k second).

Running at 88k, computer is calculating 88k samples per second (each sample represents 1/88k second).

88k rate is not just taking 96k and dropping 8k samples each second - it's a whole different alignment of sample points.

In order to run at both rates the computer has to simultaneously calculate the samples for 96k and 88k, which means calculating 184k samples per second.

How's it going to do that if it's already having problems calculating 96k samples per second?

And the problem is a matter of maxing out the CPU - or at least maxing out the ability of the CPU to process sound in real time.

In order to prevent the sound from stopping and automatically stepping down the rate (from say 96k to 88k) the computer would have to 'look ahead', which would mean calculating a buffer of some time duration (perhaps only a few milliseconds) and then playing the buffer back 'behind' the calculations.

if the buffer length shrinks, it means that playback is catching up with the calculations, which means the calculation is starting to fall behind (or already has fallen behind) which would trigger the step down in sample rate.

Note, however, that this buffer would mean built-in latency that you couldn't get rid of.

The look-ahead/buffer concept is actually how most anti-skip features work on CD players. the sound for several seconds is placed in a buffer and then the D/A conversion (playback) starts.

If the player is moved in a way that causes the play-head to skip, playback of the buffer continues and the play-head is 'recovered' and returned to normal read. Since calculations are a little faster than playback, the buffer quickly 'refills' to its full length.

Sound only stops if the time required to recover the play-head exceeds the duration of time held in the buffer.

This works for CD playback because the listener doesn't notice the 'latency' because once the song starts to play, it continues in real time. the only 'lag' is that from pushing play to the start of the song playing.

With a live or interactive situation like Reason, you would notice it because you'd start triggering sounds with your controller, the calculations would be done in real time, but the playback would lag by that delay that is the buffer.
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Last edited by SteveDiverse; 2013-02-09 at 19:45.
  #18  
Old 2013-02-09, 23:09
joshuajohn89 joshuajohn89 is offline
 
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Quote:
Originally Posted by SteveDiverse View Post
...

In order to prevent the sound from stopping and automatically stepping down the rate (from say 96k to 88k) the computer would have to 'look ahead', which would mean calculating a buffer of some time duration (perhaps only a few milliseconds) and then playing the buffer back 'behind' the calculations.

if the buffer length shrinks, it means that playback is catching up with the calculations, which means the calculation is starting to fall behind (or already has fallen behind) which would trigger the step down in sample rate.

...

With a live or interactive situation like Reason, you would notice it because you'd start triggering sounds with your controller, the calculations would be done in real time, but the playback would lag by that delay that is the buffer.
Would this buffer be required in addition to the buffer already in place? Could not a system work wherein the processor steps down the sampling rate (or switches 'playheads', rather) when it detects that the (current) buffer is dangerously depleted and/or is being depleted at an unsustainable pace?

I imagine the setup being something akin to playing the same project on two different computers running at two different rates while identical messages from the controller are sent to both rigs, while playback from only one of the pair is heard at any time. The default player would be that of the higher sample rate, but would switch to the second whenever it is detected that the primary is running too hot. The only difference, then, between such a setup & my interpretation of cyclic's suggestion is that the latter would be managed internally by a single computer. The multiple cores of a modern workstation are essentially multiple computers working together, are they not?

It seems to me, though, you are saying that if I possess processing power sufficient to run my system at 96 & 88kHz simultaneously -effectively a sample rate of 184kHz- then I might as well just run at 184k alone, making the fail-safe needlessly redundant. Whether I use dual rates or a single rate less than or equal to the sum of the two makes no difference. Is this right?
  #19  
Old 2013-02-10, 00:37
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Steve makes a very good point too - audio buffers aren't really dynamic, they are a fixed size for any given sample rate: the higher the sample rate, the bigger each audio buffer needs to be, and the slower the tempo the bigger the audio buffer gets. Changing the samplerate during playback would change the buffer size (so they are dynamic in that sense) but this would affect a whole load of shizzle, such as information held in buffers for anything requiring delay (delay, chorus, reverb etc) as reducing the samplerate would have to either flush or truncate the buffer. Either way, it can't do it a) gracefully, b) without the listerner noticing it, and most likely c) without BSODding the audio driver.

It's not merely a case of 96 downgrading to 88, either, as actually that wouldn't make a whole heap of difference in DSP, so such a fallback would need at least 48 as well to actually get any significant benefit. It would need to be able to run at least three static buffer sizes at those three samplerates concurrently.

So, which audio cards support three samplerates simultaneously?
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  #20  
Old 2013-02-10, 01:29
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SteveDiverse SteveDiverse is offline
 
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Quote:
Originally Posted by joshuajohn89 View Post
setup being something akin to playing the same project on two different computers running at two different rates while identical messages from the controller are sent to both rigs, while playback from only one of the pair is heard at any time. The default player would be that of the higher sample rate, but would switch to the second whenever it is detected that the primary is running too hot. The only difference, then, between such a setup & my interpretation of cyclic's suggestion is that the latter would be managed internally by a single computer.
OK. the 2-computer setup "works" - issues aside on how to automatically switch to the lower-rate computer when the higher-rate one fails.

Hence, I get the the idea of that scenario running on a single computer.

However...

Quote:
Originally Posted by joshuajohn89 View Post
The multiple cores of a modern workstation are essentially multiple computers working together, are they not?
Yes.

Problem is, one instance of Reason running on a multi-core CPU utilizes all cores - that is, the various processing threads in one instance of reason are spread across all of the cores.

So it's not like the 96k computer-too-slow issues is happening on half the computer while the other half sits there idle - the other half thus being available to run the 88k scenario.

The 96k scenario is taking advantage of all available cores/memory/etc - and is none-the-less running out of processing power...so there is nothing left to run another instance at 88k.
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Last edited by SteveDiverse; 2013-02-10 at 03:17.
 

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