All this about sound quality

[fishmonkey]

48kHz is in my eyes a marketing gag. "we are movie and you are sound..." okay, it's technically doable but the difference is not that big im compartment to 88,1. it was a argument to sell more new DVDs/-players. And a DVD will die using 24/88,1 +Movie.

this makes no sense. to play DVDs you need a DVD player, so the choice of sampling rates has no relationship to selling "more new DVD players". as a later standard than CD, it makes sense that they also included the highest commonly used sampling rate in pro audio, which at the time was 48 kHz. in any case, the DVD standard also includes 44.1 kHz audio.

You may can imagine why the 44100Hz/16bit standard was invented? It was not only the state of the art technology at it'S time. It was also a psyhoacoustical term. better always goes, but you always have to get a balance between terms. Here it is audio quality/ filesize/ medium. everything is a compromise- even in real live .

although the rough figure is to do with the Nyquist theorem and filtering compromises, the somewhat odd choice of 44.1 kHz was originally made for recording digital audio on video tape in both NTSC and PAL formats.

actually 48 kHz and its multiples make more sense in the digital world because they are divisible by 8.

[Angstrom]

48kHz is in my eyes a marketing gag. "we are movie and you are sound..." okay, it's technically doable but the difference is not that big im compartment to 88,1. it was a argument to sell more new DVDs/-players. And a DVD will die using 24/88,1 +Movie.

this makes no sense. to play DVDs you need a DVD player, so the choice of sampling rates has no relationship to selling "more new DVD players". as a later standard than CD, it makes sense that they also included the highest commonly used sampling rate in pro audio, which at the time was 48 kHz. in any case, the DVD standard also includes 44.1 kHz audio.

You may can imagine why the 44100Hz/16bit standard was invented? It was not only the state of the art technology at it'S time. It was also a psyhoacoustical term. better always goes, but you always have to get a balance between terms. Here it is audio quality/ filesize/ medium. everything is a compromise- even in real live .

although the rough figure is to do with the Nyquist theorem and filtering compromises, the somewhat odd choice of 44.1 kHz was originally made for recording digital audio on video tape in both NTSC and PAL formats.

actually 48 kHz and its multiples make more sense in the digital world because they are divisible by 8.

Because I am on a phone I'll quote the net to expand on your point

*1981: AES adopts 48kHz as professional digital audio sampling rate standard. 48kHz was proposed by Alastair Heaslett of Ampex. Heaslett's rationale was based on the following considerations:

1) Data "block" (integral samples) boundaries should coincide with frame boundaries of NTSC/PAL/SECAM/Film sound track (for ease of editing).

2) System clock frequency should be multiple of NTSC/PAL/SECAM/Film horizontal frequency (for lock).

3) System clock frequency should be multiple of sampling frequency.

4) Sampling frequency should be twice that (Nyquist theorem) of 22.5kHz (20kHz human hearing range + 2.5kHz, so anti-aliasing filter has required attenuation to prevent aliasing components from appearing in audio passband, ie > 45kHz).

Note that sample boundaries to frames correlation is #1, and those frames aren't necessarilly digital

[Angstrom]

Double post

[NF]

Because I am on a phone I'll quote the net to expand on your point

*1981: AES adopts 48kHz as professional digital audio sampling rate standard. 48kHz was proposed by Alastair Heaslett of Ampex. Heaslett's rationale was based on the following considerations:

1) Data "block" (integral samples) boundaries should coincide with frame boundaries of NTSC/PAL/SECAM/Film sound track (for ease of editing).

2) System clock frequency should be multiple of NTSC/PAL/SECAM/Film horizontal frequency (for lock).

3) System clock frequency should be multiple of sampling frequency.

4) Sampling frequency should be twice that (Nyquist theorem) of 22.5kHz (20kHz human hearing range + 2.5kHz, so anti-aliasing filter has required attenuation to prevent aliasing components from appearing in audio passband, ie > 45kHz).

Note that sample boundaries to frames correlation is #1, and those frames aren't necessarilly digital

so 1) to 3)are pratically and not necessary. it's not an argument for any quality augmentation.
point 4 is more important. our ears can hear the difference between 24/24 and 88/24 and the technique is difficult to handle to our good...

btw: "I'm in" in control theory

[NF]

My final assumtion why live is often beaten by other DAW guys is, that lives impulse-response is not linar in phase.

I hope that this will be done in Live 9 and everyone can sleep better at night.
But in the moment I do not get any disadvantages from it.

So I'm very excited if and what they change...

[Palmer Eldritch]

... that lives impulse-response is not linar in phase.

??? Sorry, but could you explain more precise what you mean with that or why this should happen in live?

Cheers, palmer

[NF]

my assumption is based on the SRC in live. the SRC is nor linear in phase. so I assume that the summing engine is based on nearly the same way of calculation.
now I would like to have a look into the source code (or gettin' a official statement from a ableton DSPmen), but I could imagine that the DSP summing core is based on nearly the same algorithm/part of code. In the end I never produce at 88kHz at my interface options, only bouncin' cause of good render results. So the tests went through the master at 16/44 + 24/44 and on that, SRC and engine is nearly the same in my eyes.

A non-linear low pass, one as you need to match the shannon nyquist criterium, moves the phase at the cut-off about -90°. So its not a linear phase filter implemented to supress the aliasing. And with an eye on the performance: it could be that a hi-res EQ8 code module is doing this.
I don't know, but it sounds former to me. The EQ8 in hi-res is a very preceise and accurate EQ! Only a analogue emulation can top this in my ears- and it's perfect integrated

If not, the DSP core is nicely modularized and ableton did their homework very well.
Otherwise I'll cook in my own earwax

compare the phase of the impulse responses of the SRC here:
http://src.infinitewave.ca/
All big DAWs are listed there.

Every result is very well from the point of live, but the phase shows that small difference.
btw: The response answer time is very fast indeed, faster than izotope with intermediate phase
And the transition is also done very well.
And to call WiKi: "Brick-wall filters(com. by aut.=lin.pase) that run in realtime are not physically realizable as they have infinite latency...".
I name the performace again.

I have to say that this fact does not influence my results in music!
It's only my personal opinion and my technical interest- as growing engineer
I'm very happy with Live and will not change to other DAWs in the near future- okay, maybe for mixing an album in PT...
workflow saves time, we all know.

I think it's interesting to discuss such facts. I give not one cent on posts of this thread like this "one DAW-user sayd about onother..."-dizzing is exactly for...toddler groups and not more. A bit healthy pessimism on everything should never be missed.

*put my devil back into my pocket *

anyone some adds?
Greetz

[fishmonkey]

whilst the information is useful, it's nothing new. avoiding SRC conversions is absolutely necessary if you want to compare Live head-to-head with other DAWs. as has been said before, there are many operations that Live performs that necessarily degrade the audio, and many people's opinions about the sound of Live are the result of not understanding how digital audio works, and the pros and cons of various tools and options.

even the Live manual suggests doing SRC conversions using a higher quality (more CPU intensive) third party application:


Sample rate conversion (during both real-time playback and rendering) is a non-neutral operation. Playback of audio files at a sample rate that is different from the rate set in Live's Preferences window will cause signal degradation. Transposition is also a form of sample-rate conversion, and thus also results in non-neutral behavior.

To minimize potential negative results, it is recommended to do sample rate conversion as an offline process in another application. Once the samples have been converted to the sample rate that you plan to use in Live, the files can be imported without any loss of quality.

Rendering audio from Live with a sampling rate other than the one that was used while working on the project is also a non-neutral operation, and may result in a loss of sound quality. It is recommended to always render using the original sampling rate, and then convert the rendered file using a dedicated mastering application that is optimized for these kinds of CPU-intensive, offline tasks.

While we recommend that you use a high-quality offline tool for sample rate conversion, we recognize that one of Live's core features is its ability to pitch-shift and warp audio in real time. For this situation, it is necessary to make a trade-off between CPU performance and precision. We recommend the use of the Hi-Q button for any clips which undergo transposition in a given Set. The algorithm behind the Hi-Q switch was rewritten for Live 7, and now results in considerably lower distortion than in previous versions.

[H20nly]

^ he said it was in the manual. do you want it on the splash screen or taking up the info panel?

how often do you change sample rates? and why?

[lunabass]

I see. It seems to me though that this bit of information regarding src within Live needs to be more immediately visible so as to be clear to new and inexperienced users. But just my view.

i'm not really sure how they could get this info any more visible as it's in the manual.

beginners generally have more important things to worry about like writing a better drum beat.

when they get to the point of writing good tunes then you'd hope that they have enough awareness to hear any issues regarding src and then find a solution. if they cant hear it in the first place then there is no problem

[NF]

^ he said it was in the manual. do you want it on the splash screen or taking up the info panel?

how often do you change sample rates? and why?

as levimoniz said:
I dont want but I have to. Sample packs are sold in near every sort of quality.
You have to take what you get.

I got projects bounced in 44/48/88/96/192 - 16/24/32. keep it relaxed and smart when getting other projects and have a look into your files.

the worst case is when a e.g. 44/16 sample was used and for a mix/remix/project transfer the project was bounced to 44/32. what do you do when hearing aliasing
now you know: ask for the originals and so on...

[H20nly]

so Ableton should let you know that when you change a sample to a different sample rate that the sample rate is changing.

maybe that should be one of the test questions you must answer to run the installer.

[Palmer Eldritch]

whilst the information is useful, it's nothing new. avoiding SRC conversions is absolutely necessary if you want to compare Live head-to-head with other DAWs. as has been said before, there are many operations that Live performs that necessarily degrade the audio, and many people's opinions about the sound of Live are the result of not understanding how digital audio works, and the pros and cons of various tools and options.

Totally agreed!!! ^^^

@NF >
Summing and SRC are technically totally different things. (and SRC is much more complex than pure summing).
But regardless of that please welcome to the more rational side

@levimoniz >
In Live the SRC is a realtime-process in comparison to - for example - Pro-Tools. There you can convert sample rates only as an offline process.
If you need to convert sample-rates offline you can use for example "audacity" in addition to live.
It is os-x, windows and linux and it is free.
If you follow the link above (http://src.infinitewave.ca/) you will see that it seems to lead to proper results.

And, btw, if I do pure summing with a bunch of tracks and bounce it to a 24 bit file (no dithering) , Pro-Tools, Reaper and Live null to minus Infinite dB (not -80dB, not -144dB, not -500dB, I mean really -infinite dB)

cheers, palmer

[Palmer Eldritch]

^ do you want it on the splash screen?

[Tone Deft]

levimoniz is just a troll at this point. the guy doesn't have a rudimentary grasp on audio concepts. check his posts from the last 24 hours, he hasn't a clue. nothing wrong with that in itself but he flat out lies about what he knows and does. nobody's born knowing this stuff but he's not even trying to learn.

pleasant enough fellow, I might listen to a tune of his someday. levi, where are they again?