Author
|
Sample Rate question
|
Astralist
Started Topics :
4
Posts :
16
Posted : Feb 6, 2013 01:08
|
Quote:
|
On 2013-02-05 21:46, knocz wrote:
it can affect the signal in many ways. Learn about dsp, understand how frequencies are derived, and you'll see it.
And changing the speed of the recording will affect gravely on the sample rate, at it changes the sample rate. If you take something at 30 kHz and play it at half speed, than the 30 kHz will become 15 kHz, which is audible. Play it at quarter speed and you hear a lot more. |
|
With that in mind, it seems prudent still to work in 44.1 and upsample before doing any bouncing or final mixing. Would you agree? Working in 44.1 during composition will make your workflow a lot smoother I think. |
|
|
Alien Bug
IsraTrance Junior Member
Started Topics :
27
Posts :
682
Posted : Feb 6, 2013 02:53
|
Quote:
|
On 2013-02-05 21:34, Upavas wrote:
Quote:
|
On 2013-02-05 17:30, Alien Bug wrote:
Quote:
|
On 2013-02-05 17:20, maxee wrote:
Quote:
| You cannot hear anything above 20khz frequency, period! |
|
I once read an interview with walter sear and he stated: audio information from sources like a microphone carrys a lot of information above 20 kHz and we are actually able to hear all the subharmonics resulting from that.
what you guys think?
|
|
in maximum short: we dont hear this frequencies above (and below) but they affect the audible frequencies.
this is interesting question for longer debate
|
|
Once again, there is nothing to affect, because what is being affected your ear can't hear. A lowpass filter around 20.000hz frequency and the result is just the same, exactly the same to our ears!!! Someone noted a comparison with mine is bigger than yours. Exactly true, higher sampling rates in sound cards are nothing but marketing strategy!
|
|
i was talking about sound in real world before it is converted to digital world
+ i wrote about 'mine is bigger than yours'
  http://www.beatport.com/release/cross-the-atoms/1042450
http://soundcloud.com/alien-bug
http://www.facebook.com/ali3nBug |
|
|
Dylan89
IsraTrance Junior Member
Started Topics :
11
Posts :
24
Posted : Feb 6, 2013 19:28
|
|
Upavas
Upavas
Started Topics :
150
Posts :
3315
Posted : Feb 7, 2013 03:06
|
Quote:
|
On 2013-02-05 21:46, knocz wrote:
Quote:
|
On 2013-02-05 21:34, Upavas wrote:
Once again, there is nothing to affect, because what is being affected your ear can't hear. A lowpass filter around 20.000hz frequency and the result is just the same, exactly the same to our ears!!! Someone noted a comparison with mine is bigger than yours. Exactly true, higher sampling rates in sound cards are nothing but marketing strategy!
|
|
Sorry man, but it can affect the signal in many ways. Learn about dsp, understand how frequencies are derived, and you'll see it.
And changing the speed of the recording will affect gravely on the sample rate, at it changes the sample rate. If you take something at 30 kHz and play it at half speed, than the 30 kHz will become 15 kHz, which is audible. Play it at quarter speed and you hear a lot more.
|
|
I was not referring to a change of speed in recordings, I mean who does that? I certainly don't. Why record something in one speed and then change it to another? It's completely unnecessary!
The notion that I should learn about dsp is ridiculous, I suggest you do what you propose to others, before you make yourself a laughing stock.
One more time! Unless you forget to set a lowpass filter of at least 1 pole (-6dB per octave, 2 pole is better) around 20khz frequency of your audio file, there is absolutely ZERO difference to different sampling frequencies as far as perception to our ears (human ears) is concerned, unless they are below 44.1khz sampling rate. It is indeed all and I mean ALL marketing strategy, nothing more! Period. I suggest you read the link I posted earlier about the Niquist theorem and learn all there is to know about dsp instead of proposing false information in this forum...
One morte thing, the only reason why 48khz is used in the movie insustry, is simply because it makes it easier to change the sampling rate when the frame rate of the movie (e.g. for cinema format to tv format) is changed, if you change the fps without changing the sampling rate, the audio of the movie gets out of sync. so from 24fps cinematic format, to 29.97 fps which is NTFC format (north american tv) you have to also change the sampling rate accordingly.
I have worked too long in the movie industry to have to listen to crap like yours...
  Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
|
|
knocz
Moderator
Started Topics :
40
Posts :
1151
Posted : Feb 7, 2013 05:34
|
Quote:
|
On 2013-02-07 03:06, Upavas wrote:
I was not referring to a change of speed in recordings, I mean who does that? I certainly don't. Why record something in one speed and then change it to another? It's completely unnecessary!
The notion that I should learn about dsp is ridiculous, I suggest you do what you propose to others, before you make yourself a laughing stock.
One more time! Unless you forget to set a lowpass filter of at least 1 pole (-6dB per octave, 2 pole is better) around 20khz frequency of your audio file, there is absolutely ZERO difference to different sampling frequencies as far as perception to our ears (human ears) is concerned, unless they are below 44.1khz sampling rate. It is indeed all and I mean ALL marketing strategy, nothing more! Period. I suggest you read the link I posted earlier about the Niquist theorem and learn all there is to know about dsp instead of proposing false information in this forum...
|
|
I certainly play with my recordings speed: I slice it up, stretch some stuff out and shorten others.. Probablty not for band recordings, but for experimental stuff yeah. Also, I play a lot with tape delays, automating the delay time. I believe these are valid situations where the sampling rate affects the sound. Also, time variation effects like chorusing, phasing and flanging can be affected by this, as they are only automated delays.
I'm not saying your wrong, I'm just referring that the sampling rate choice can have some impact, depending on whats done to the samples.
Thanks for the info on why the movie industry uses 48 kHz. And thanks for the nice compliments too.
  Super Banana Sauce http://www.soundcloud.com/knocz |
|
|
Colin OOOD
Moderator
Started Topics :
95
Posts :
5380
Posted : Feb 7, 2013 14:37
|
When you change the speed of a recording in your DAW, whether it's with a modulated delay, or a sampler, or any other way of replaying those recorded samples out at a different rate, remember that the samplerate of your DAW does not change, so you're still hearing a stream of samples at 44.1kHz or whatever. Whatever technique you're using to change the speed of your recording has absolutely no effect on the number of samples per second you're actually listening to.
  Mastering - http://mastering.OOOD.net :: www.is.gd/mastering
OOOD 5th album 'You Think You Are' - www.is.gd/tobuyoood :: www.OOOD.net
www.facebook.com/OOOD.music :: www.soundcloud.com/oood
Contact for bookings/mastering - colin@oood.net |
|
|
Upavas
Upavas
Started Topics :
150
Posts :
3315
Posted : Feb 7, 2013 14:47
|
Quote:
|
On 2013-02-07 14:37, Colin OOOD wrote:
When you change the speed of a recording in your DAW, whether it's with a modulated delay, or a sampler, or any other way of replaying those recorded samples out at a different rate, remember that the samplerate of your DAW does not change, so you're still hearing a stream of samples at 44.1kHz or whatever. Whatever technique you're using to change the speed of your recording has absolutely no effect on the number of samples per second you're actually listening to.
|
|
I thought he meant changing the sampling rate (speed???) without conversion.
You are absolutely right of course...
  Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
|
|
demoniac
Demoniac Insomniac
Started Topics :
85
Posts :
1281
Posted : Feb 7, 2013 15:56
|
|
PoM
IsraTrance Full Member
Started Topics :
162
Posts :
8087
Posted : Feb 7, 2013 16:55
|
on some modern plug it shouldn t change much as these may run already at higher rate than 96 khz,or oversample for process than need it.
lot of hardware like virus and nord run at higher sample rate, but lot of plugs too..imo it s probably one area it need improvement.. real time sample rate convertion like on some plugins sound not as good as the best offline application.. and bad sample rate convertion damage the sound.. to the point sometimes it s better to use normal quality insteed of highest on some plugs depending the settings used. |
|
|
demoniac
Demoniac Insomniac
Started Topics :
85
Posts :
1281
Posted : Feb 7, 2013 18:50
|
|
PoM
IsraTrance Full Member
Started Topics :
162
Posts :
8087
Posted : Feb 7, 2013 19:21
|
It don t give more resolution, the increased sampling rate just allows to sample higher inaudible freqs.
if higher sample rate sound better it can be cause of the design in a converter/audio interface. when 44,100 is used everyhting above 22050 Hz need to be removed , but some filter will let aliasing come back or cut the highs in the audible range, so in these interface using a higher sampling rate may increase sound quality. it should not happen in a pro converter/interface
on a synth whith audio rate modulation like fm it can lower artifacts a little, filters may have a better response in the highs too but i think all this depens how it s implemented
|
|
|
demoniac
Demoniac Insomniac
Started Topics :
85
Posts :
1281
Posted : Feb 7, 2013 19:26
|
|
PoM
IsraTrance Full Member
Started Topics :
162
Posts :
8087
Posted : Feb 7, 2013 19:46
|
check the Nyquist Shanon theorem i posted a link on the other page. to have the audible frequency range to be reconstructed right when sampled you need to use at least 44,100 in simple words.why i don t know i have no clue about digital audio/ don t remember lot of stuff i studyed a little..but it s the way it works
sampling rate with itb production is not simple subject as many things are involved, from the plugins to the converters, the clock... |
|
|
astro_traveller
IsraTrance Junior Member
Started Topics :
13
Posts :
65
Posted : Feb 7, 2013 19:54
|
if u use 22.05 khz sample rate u will have alise filter on 11.025 khz. so u wont be able to hear any thing above.
higher the sample rate.. better conversion. If u zoom into ur wave on daw, u will see wave is made of small line connected togehther. On 44.1khz .. it samples the wave 44100 per sec times to represent the wave.
They use alaise filter coz the frequency wave above 22.05 khz becomes smaller than each sample.
some times when u convert from Dvd to cd with out downgrading the sample rate u get an annoying high sound .. which is called aliase frequency, it is the result of not using alaise filter.
the higher sample rate makes difference.. coz it is representing the whole wave. Higer the sample rate .. smaller the samples are.. so they represent better.
when u work on 16 bits 44.1khz.. it makes 44100 samples to represent the wave and 0 and 1 on digital domain are written in 16 variables to represent those 44100 samples. |
|
|
PoM
IsraTrance Full Member
Started Topics :
162
Posts :
8087
Posted : Feb 7, 2013 20:04
|
the problem is most audio aplication show you a visual representation by drawing a line between the points, it s not real representation of the waveform like if it you take a oscilloscope and measure it after the da converter in the analog domain |
|
|