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post #41 of 65
you're making things complicated for yourself. shifty software algorithms aren't going to make your source sound any better, just get pulse to pass it straight to the ALSA sink then set up the ALSA config to send the source straight to the hardware without doing anything to it (AFAIK ALSA resamples when necessary by default). The whole point of the fancy DAC in your sound card is to take on this work, so the best thing you can do is remove obstacles instead of adding them.
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post #42 of 65
Quote:
Originally Posted by Petrol View Post

you're making things complicated for yourself. shifty software algorithms aren't going to make your source sound any better, just get pulse to pass it straight to the ALSA sink then set up the ALSA config to send the source straight to the hardware without doing anything to it (AFAIK ALSA resamples when necessary by default). The whole point of the fancy DAC in your sound card is to take on this work, so the best thing you can do is remove obstacles instead of adding them.

This; as has been stated on head-fi a million times: Lossless digital audio only retains bit-perfect replication until it is processed. ANY processing adds distortion, even that done by your fancy DAC. The first processing that occurs is decoding (with lossless codecs, this can be kept to perfect quality) from there if pulse resamples it - thats more distortion, if alsa resamples again - further distortion. The most important piece of the audio puzzle is the SOURCE. If you are listening to an 8-bit track on hundreds of thousands of dollars worth of audio equipment it will still never sound as good as a lossless recording of the same track on significantly worse audio equipment. Tampering with the source only makes it worse, especially in the case of lossless audio. Noise cancellation or reduction, virtual surround, up/down sampling, and filtering are all mostly gimmicky and add distortion. There are reasons for some of them, but for music listening and authoring, you can't beat zero distortion.
    
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post #43 of 65
Thread Starter 
Quote:
Originally Posted by Petrol View Post

you're making things complicated for yourself. shifty software algorithms aren't going to make your source sound any better, just get pulse to pass it straight to the ALSA sink then set up the ALSA config to send the source straight to the hardware without doing anything to it (AFAIK ALSA resamples when necessary by default). The whole point of the fancy DAC in your sound card is to take on this work, so the best thing you can do is remove obstacles instead of adding them.

OK...so could you help me out with an example of that whole setup AKA that pulse, then the ALSA configs that should be made or how to make them? I don't think I have a clue where to begin with that. But IF someone didn't want to do all that (their choice) is the config directions in the first post fairly good? In other words...I still want to give people the best they can get out whatever choice they make even if it is a "poor/wrong/bad" one.

I'll also add some finishing touches to the config direction after I'm done with some testing on a few more changes I've made. One is setting the priority to highest (haven't tried realtime).

@Xaero: So if a person doesn't have a fancy DAC but say a basic 24bit 96kHZ onboard chip would you still recommend following Petrol's method of forcing everything straight to the sink for the DAC to handle on it's own?

@Both: Would your proposed change cause any problems with Pulse's abilities like porting audio to different ports (not that I know how to do that), independent sound control, and allowing multiple programs access to the audio output hardware at the same time? Man this audio stuff make me feel horribly ignorant no matter how much I read it and try to grasp it.
     
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post #44 of 65
http://www.hydrogenaudio.org/forums/index.php?showtopic=94080

This covers things well. Third post is very valuable.
    
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post #45 of 65
Thread Starter 
Quote:
Originally Posted by Xaero252 View Post

http://www.hydrogenaudio.org/forums/index.php?showtopic=94080

This covers things well. Third post is very valuable.

Hmm...all I got out of that thread is that if you want the best sound then you have to give up being able to switch program to program wanting access to the sound card and get ready to kill them if they don't let go?

The second part of the third post is exactly what I'm doing though with Pulse and not ALSA. Should I also change something with ALSA? LOL. I hope I don't frustrate you guys I'm really trying here. redface.gif
     
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post #46 of 65
Quote:
Originally Posted by Petrol View Post

(AFAIK ALSA resamples when necessary by default).

Only with dmix - which Pulse is supposed to "replace" anyway. If you send an unsupported sample rate to the card, it will sound distorted and unpleasant.

Setting the bit rate higher than the card can handle only hurts in one way - a waste of CPU cycles. There's a 32-bit option for inter-communications between programs, which can be useful if you have sound that exceeds the 144dB of SNR that's available with 24-bit audio. For example, DAW's generally work in 32-bit float or beyond, to avoid digital distortion before hitting the "master" bus, which is generally 24-bit (or 16-bit, again, depending on your sound card).

For the most part, the stock config has the most optimal sound quality available. Latency is the only thing I would really bother looking into with Pulse. Software resampling can be beneficial if your sound card does not support multiple audio streams - for example, playing back a 44kHz and a 48kHz audio stream at the same time can cause issues with ALSA (especially dmix), which Pulse kind of rectifies.
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post #47 of 65
Quote:
Originally Posted by gonX View Post

Only with dmix - which Pulse is supposed to "replace" anyway. If you send an unsupported sample rate to the card, it will sound distorted and unpleasant.

Setting the bit rate higher than the card can handle only hurts in one way - a waste of CPU cycles. There's a 32-bit option for inter-communications between programs, which can be useful if you have sound that exceeds the 144dB of SNR that's available with 24-bit audio. For example, DAW's generally work in 32-bit float or beyond, to avoid digital distortion before hitting the "master" bus, which is generally 24-bit (or 16-bit, again, depending on your sound card).

For the most part, the stock config has the most optimal sound quality available. Latency is the only thing I would really bother looking into with Pulse. Software resampling can be beneficial if your sound card does not support multiple audio streams - for example, playing back a 44kHz and a 48kHz audio stream at the same time can cause issues with ALSA (especially dmix), which Pulse kind of rectifies.

Could you elaborate a little on your point about hardware not supporting above 21bit. I'd not heard about this before so quite intrigued smile.gif
post #48 of 65
Quote:
Originally Posted by Plan9 View Post

Could you elaborate a little on your point about hardware not supporting above 21bit. I'd not heard about this before so quite intrigued smile.gif

To be fair I don't know a whole a lot about that area specifically. I went by Wikipedia's information. But still, we're talking about a signal-to-noise ratio. Chances are your computer alone is more than likely putting out 35dB. Even with a 16-bit audio stream, we have 96dB of SNR, which means 35+96dB = 131dB. To put that into perspective, a rock concert in a small venue is about 110dB. For every ~3dB, you "double" the effect of the system, exponentially, so 131dB would result in a 2^7 times higher volume, which would damage your hearing in a few seconds.

Of course, that's putting SNR in overly simplified terms. There are obviously frequencies which your system does not output as noise, but if we take the computer as the base noise floor, it will be at 35dB.

A higher bit-depth than 16-bit is generally only desired when you're working with multiple tracks at a time. Think of recording a drum kit - let's put a microphone on the kick drum, one on the snare, 1 for 2 toms, and a stereo overhead, and we suddenly have a lot of noise, even when nothing is playing. Whenever there's a silent part - you will hear the noise of all the tracks added together, essentially lowering your audible bit-depth.
Edited by gonX - 5/22/13 at 1:35pm
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post #49 of 65
Quote:
Originally Posted by gonX View Post

To be fair I don't know a whole a lot about that area specifically. I went by Wikipedia's information. But still, we're talking about a signal-to-noise ratio. Chances are your computer alone is more than likely putting out 35dB. Even with a 16-bit audio stream, we have 96dB of SNR, which means 35+96dB = 131dB. To put that into perspective, a rock concert in a small venue is about 110dB. For every ~3dB, you "double" the effect of the system, exponentially, so 131dB would result in a 2^7 times higher volume, which would damage your hearing in a few seconds.

Of course, that's putting SNR in overly simplified terms. There are obviously frequencies which your system does not output as noise, but if we take the computer as the base noise floor, it will be at 35dB.

A higher bit-depth than 16-bit is generally only desired when you're working with multiple tracks at a time. Think of recording a drum kit - let's put a microphone on the kick drum, one on the snare, 1 for 2 toms, and a stereo overhead, and we suddenly have a lot of noise, even when nothing is playing. Whenever there's a silent part - you will hear the noise of all the tracks added together, essentially lowering your audible bit-depth.
Yeah. I know all that (i used to be a producer and had set up my own studio a few years back). I was just puzzled by your statement that no equipment supports > 21bit.
post #50 of 65
I don't know if it's 21 bit but upon reading (some of us weren't producers, it's hard to remember it all) there was something about even high end digital has issues due to certain parts having to be analogue. I'm sure some of the really high end hardware doesn't have that, probably more towards the consumer end. Was reading it when you brought up distortion comments. I knew about some of it, just thought it would be smart of me to read up on all of it. I don't remember where....

Here's a wiki clip, could find the source but I'm sure it corresponds. [edit: For clarity, first half was written before I double checked online, just so you know why it sounds a bit mixed/broken - the response that is]

http://en.wikipedia.org/wiki/Audio_bit_depth
"as of 2007 digital audio converter technology is limited to a S/N of about 124 dB (21-bit)[4] because of real world limitations in integrated circuit design"

I believe it's due to how the circuitry has to be made. I remember skimming details in another article, but I was just skimming for a refresh. That was new to me as well, so I just kept a note of their being a limitation (just took a mental note of "around 24 bit").
Edited by mushroomboy - 5/23/13 at 6:26pm
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