23 Nov
2015
23 Nov
'15
4:31 p.m.
Hello all,
When having multiple mono guitar tracks in Ardour and wanting to
regroup them in a single bus, would that bus be preferably mono or
stereo ? How is this in relation to eventually placing the guitar bus
in a stereo field ?
Also, is it possible in Ardour to raise the sampling rate of single
tracks ? I just followed the part of the Brian Bollman music production
(free) course about sampling rates. The higher the rate the more
fidelity to the original signal and also the more material to work with
when bringing it down to CD or ogg/mp3 level. I would like to try with
the guitar and bass tracks as I don't care so much about synths for
now. Is it possible to tell Ardour to work with a higher sampling rate
for a track ? I don't seem to have found the option.
Cheers.
23 Nov
23 Nov
4:50 p.m.
On Mon, 23 Nov 2015 10:31:12 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
When having multiple mono guitar tracks in Ardour and
wanting to
regroup them in a single bus, would that bus be preferably mono or
stereo ? How is this in relation to eventually placing the guitar bus
in a stereo field ?
Ardour's default is to provide 2-channel output, so I've always made my
busses stereo.
Also, is it possible in Ardour to raise the sampling
rate of single
tracks ?
You won't be able to change the sample rate on the fly. The sample
rate is set by JACK, and if your session is at 44.1k and you want to
change to using 48k, you'd have to start JACK with the newer rate. There
will be an issue loading the old session, though, and I don't know what
Ardour does in that situation, but I don't believe it will automagically
change the session.
You may need to start a new session and import the tracks from the old session's
interchange/<session_name>/audiofiles directory.
If you import the old tracks to your region list, I believe Ardour will
resample as needed. I do everything at 48k because of my mixer/interface,
and have pulled in tracks recorded at different rates with no issues.
--
======================================================================
Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com
Without deviation from the norm, "progress" is not possible. - FZappa
5:01 p.m.
On Mon, 23 Nov 2015 10:50:01 -0500
Joe Hartley <jh(a)brainiac.com> wrote:
You may need to start a new session and import the
tracks from the
old session's interchange/<session_name>/audiofiles directory.
If you import the old tracks to your region list, I
believe Ardour
will resample as needed. I do everything at 48k because of my
mixer/interface, and have pulled in tracks recorded at different
rates with no issues.
Importing will defeat in general the purpose of higher quality because
the source material has less to start with. Importing the rest of the
project and redoing the guitar and bass tracks, which is probably what
you meant, creates new sources with higher sampling points.
I will try the stereo bus for the mono guitar and bass tracks then.
I have tried a demo plugin that has 8 outputs. In this case Ardour
will show (see attached) a 'spatial' round control to move the audio
around, at least left to right in stereo. Once this control is there I
haven't found the way to remove it but moreover, now I would like to
use the same control on other regalar stereo tracks - is it possible ?
5:18 p.m.
On Mon, 23 Nov 2015 11:01:15 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
On Mon, 23 Nov 2015 10:50:01 -0500
Joe Hartley <jh(a)brainiac.com> wrote:
I believe you completely misunderstand me.
Your session runs at a sample rate set by JACK. All the files that have
audio in them have to match that value. If the audio you want to use has
a higher sample rate, it will have to be re-sampled at the lower rate for
Ardour to use it in that session.
If you want to have JACK and Ardour run at the higher sample rate, assuming
your interface supports it, you'd have to create a new session and import the
tracks. The files with the audio that match that new, higher rate will be
imported with minimal processing, while the other tracks will be up-sampled.
There is no way for Ardour to use files of a different sample rate without
them being resampled to the rate of the session, as determined by JACK's
setting.
--
======================================================================
Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com
Without deviation from the norm, "progress" is not possible. - FZappa
You may need to start a new session and import
the tracks from the
old session's interchange/<session_name>/audiofiles directory.
If you import the old tracks to your region list,
I believe Ardour
will resample as needed. I do everything at 48k because of my
mixer/interface, and have pulled in tracks recorded at different
rates with no issues.
Importing will defeat in general the purpose of higher quality because
the source material has less to start with. Importing the rest of the
project and redoing the guitar and bass tracks, which is probably what
you meant, creates new sources with higher sampling points. 
5:20 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, Nov 23, 2015 at 11:18 AM, Joe Hartley <jh(a)brainiac.com> wrote:
I believe you completely misunderstand me.
Your session runs at a sample rate set by JACK. All the files that have
audio in them have to match that value. If the audio you want to use has
a higher sample rate, it will have to be re-sampled at the lower rate for
Ardour to use it in that session.
Not strictly true any more: Since 4.0, Ardour has no longer required JACK.
However, the basic point remains: changing the SR is a non-trivial
operation and should not be considered lightly. Get the SR correct before
you start working, not later.
5:33 p.m.
On Mon, 23 Nov 2015 11:18:32 -0500
Joe Hartley <jh(a)brainiac.com> wrote:
Your session runs at a sample rate set by JACK. All
the files that
have audio in them have to match that value. If the audio you want
to use has a higher sample rate, it will have to be re-sampled at the
lower rate for Ardour to use it in that session.
Sure. Thing is, I sample, eg. I record in Ardour only. So I only
speak of Ardour.
If you want to have JACK and Ardour run at the higher
sample rate,
assuming your interface supports it, you'd have to create a new
session and import the tracks. The files with the audio that match
that new, higher rate will be imported with minimal processing, while
the other tracks will be up-sampled.
This "up-sample" as far as I understood from the course, does not exist
in reality as far as the quality of audio is concerned. A
demonstration can be easily made using Audacity and zooming up to see
the actual sampling points of both files, the original and the
"up-sampled": the file that has been resampled from lower sampling rate
will not add anything at all but more sampling points on straight
segments of the audio. It cannot add quality. It cannot create curves.
What I mean by using a higher sampling rate is for recording purposes.
To have more points being sampled when a guitar is played. So that the
points can represent closer sections of the wave instead of jumping and
making straight lines.
There is no way for Ardour to use files of a different
sample rate
without them being resampled to the rate of the session, as
determined by JACK's setting.
Which would not be right as it will be better to re-record the tracks
that actually need a better quality of sampling.
Out of say 15 tracks, there would be 4 that need higher quality
sampling because they are acoustic instruments. But the 15 tracks,
even the synths and synth drums, will take so many more megabytes of
space, because the sampling rate is not set on a per track basis.
Maybe sampling rates on a per-track basis does not make sense in FAWs
in general.
5:44 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, Nov 23, 2015 at 11:33 AM, jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com
wrote:
This "up-sample" as far as I understood from the course, does not exist
in reality as far as the quality of audio is concerned. A
demonstration can be easily made using Audacity and zooming up to see
the actual sampling points of both files, the original and the
"up-sampled": the file that has been resampled from lower sampling rate
will not add anything at all but more sampling points on straight
segments of the audio. It cannot add quality. It cannot create curves.
Your understanding is incorrect. Please watch this:
https://xiph.org/video/vid2.shtml
If you have time (40mins to an hour), I'd also suggest you watch this):
https://www.youtube.com/watch?v=BYTlN6wjcvQ
The meat of this one starts at 16:48 https://youtu.be/BYTlN6wjcvQ?t=16m48s
but the preamble is worth watching for other reasons too.
Out of say 15 tracks, there would be 4 that need higher quality
sampling because they are acoustic instruments. But
the 15 tracks,
even the synths and synth drums, will take so many more megabytes of
space, because the sampling rate is not set on a per track basis.
Maybe sampling rates on a per-track basis does not make sense in FAWs
in general.
your audio interface runs at one sample rate. all tracks not at the same SR
would have to be resampled. Not a good idea.
5:58 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 11:44:00 -0500
Paul Davis <paul(a)linuxaudiosystems.com> wrote:
On Mon, Nov 23, 2015 at 11:33 AM,
jonetsu(a)teksavvy.com
<jonetsu(a)teksavvy.com
> wrote:
> This "up-sample" as far as I understood
from the course, does not
> exist in reality as far as the quality of audio is concerned. A
> demonstration can be easily made using Audacity and zooming up to
> see the actual sampling points of both files, the original and the
> "up-sampled": the file that has been resampled from lower sampling
> rate will not add anything at all but more sampling points on
> straight segments of the audio. It cannot add quality. It cannot
> create curves.
Your understanding is incorrect. Please watch this:
https://xiph.org/video/vid2.shtml
If you have time (40mins to an hour), I'd also
suggest you watch
this): https://www.youtube.com/watch?v=BYTlN6wjcvQ
The meat of this one starts at 16:48
https://youtu.be/BYTlN6wjcvQ?t=16m48s but the preamble is worth
watching for other reasons too.
Thanks for the links I will watch them later on this week.
There's one thing clear though. If you have a sine wave with say 16
sampling points, much of the curves will be lost. If this is saved to
a file and the file is resampled with 128 sampling points, it will not
add the curves. The original is gone, the resampling does not know about
the original source, it cannot and it will not make it up.
So if the sine wave ended up sounding like a robot in the file,
a linked collection of edges, direct resampling at a higher rate will
not add a more natural expression. This is what I described above.
There might be special processing that extrapolates, creates
points between existing sampling points thus adding slopes and curves,
although w/o knowing the source only fiction can be created,
your audio interface runs at one sample rate. all
tracks not at the
same SR would have to be resampled. Not a good idea.
OK, that's the limiting factor then. It is not possible to have various
sampling rates inside a session for hardware interface reasons.
6 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 11:58:09 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
There's one thing clear though. If you have a
sine wave with say 16
sampling points, much of the curves will be lost. If this is saved to
a file and the file is resampled with 128 sampling points, it will not
add the curves. The original is gone, the resampling does not know about
the original source, it cannot and it will not make it up.
Although it seems counter-intuitive that actually is quite wrong!
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
6:08 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 17:00:47 +0000
Will Godfrey <willgodfrey(a)musically.me.uk> wrote:
On Mon, 23 Nov 2015 11:58:09 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
Hmmmm. Not sure about intuition. A sine wave with 16 sampled points
will end up like a linked list of edges. Saving that to file and
resampling at 128 will only add points to the straight lines. It will
not create curves as per the original. It cannot. How would it know it
was a sine wave and not a guitar tone when it had to process basically
what was a robotic tone ?
There's one thing clear though. If you have
a sine wave with say 16
sampling points, much of the curves will be lost. If this is saved
to a file and the file is resampled with 128 sampling points, it
will not add the curves. The original is gone, the resampling does
not know about the original source, it cannot and it will not make
it up.
Although it seems counter-intuitive that actually is quite wrong! 
6:19 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On 23 November 2015 at 17:08, jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com> wrote:
On Mon, 23 Nov 2015 17:00:47 +0000
Will Godfrey <willgodfrey(a)musically.me.uk> wrote:
Interpolation?!
--
Neil C Smith
Artist : Technologist : Adviser
http://neilcsmith.net
Praxis LIVE - hybrid visual IDE for creative coding - www.praxislive.org
Digital Prisoners - interactive spaces and projections -
www.digitalprisoners.co.uk
On Mon, 23 Nov 2015 11:58:09 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
Hmmmm. Not sure about intuition. A sine wave with 16 sampled points
will end up like a linked list of edges. Saving that to file and
resampling at 128 will only add points to the straight lines. It will
not create curves as per the original. It cannot. How would it know it
was a sine wave and not a guitar tone when it had to process basically
what was a robotic tone ? There's one thing clear though. If you have
a sine wave with say 16
sampling points, much of the curves will be lost. If this is saved
to a file and the file is resampled with 128 sampling points, it
will not add the curves. The original is gone, the resampling does
not know about the original source, it cannot and it will not make
it up.
Although it seems counter-intuitive that actually is quite wrong!
6:28 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 17:18:48 +0000
Neil C Smith <neil(a)neilcsmith.net> wrote:
On 23 November 2015 at 17:08, jonetsu(a)teksavvy.com
<jonetsu(a)teksavvy.com> wrote:
> Hmmmm. Not sure about intuition. A sine wave
with 16 sampled points
> will end up like a linked list of edges. Saving that to file and
> resampling at 128 will only add points to the straight lines. It
> will not create curves as per the original. It cannot. How would
> it know it was a sine wave and not a guitar tone when it had to
> process basically what was a robotic tone ?
Interpolation?!
OK, fair enough. I will read about it. The example that was shown in
the course using the Mediahuman software for conversion surely did not
interpolate anything from the original when sampling at a higher rate.
Are sound waves so predictable that a crude wave from a guitar will be
resampled to the warmth tones of wood as well as a wave from a synth ?
6:36 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, Nov 23, 2015 at 12:28 PM, jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com
wrote:
Are sound waves so predictable that a crude wave from a guitar will be
resampled to the warmth tones of wood as well as a wave from a synth ?
this is an area where a little knowledge can be exceptionally dangerous to
your understanding. Combined with the mistaken drawing of waveforms as
stairsteps, it can be disastrous. try to forget whatever you think you
know, and just watch monty's video.
then sit back with a glass of your preferred beverage, and watch ethan
wiener's longer one. you will then be innoculated against most errors of
"intuition". your thinking/knowledge about this stuff is utterly incorrect
(no fault of yours - this isn't trivial stuff), and if you keep guessing
you'll keep guessing wrong.
6:47 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
Paul Davis <paul(a)linuxaudiosystems.com> schrieb am Mo., 23. Nov. 2015 um
18:36 Uhr:
then sit back with a glass of your preferred beverage, and watch ethan
wiener's longer one.
Do you know of a different source for that video. It is blocked in Germany
for the usual GEMA reasons.
Kind regards,
Felix
7:11 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 17:47:35 +0000, Felix Homann wrote:
Paul Davis <paul(a)linuxaudiosystems.com> schrieb
am Mo., 23. Nov. 2015
um 18:36 Uhr:
Fake an IP. I guess for Firefox are add-ons available.
--
http://www.grundgesetz-gratis.de/
then sit back with a glass of your preferred beverage, and watch
ethan wiener's longer one.
Do you know of a different source for that video. It is blocked in
Germany for the usual GEMA reasons. 
8:10 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, Nov 23, 2015 at 10:11 AM, Ralf Mardorf <ralf.mardorf(a)alice-dsl.net>
wrote:
On Mon, 23 Nov 2015 17:47:35 +0000, Felix Homann
wrote:
"Hola" fre VPN for Firefox does the job.
Paul Davis <paul(a)linuxaudiosystems.com>
schrieb am Mo., 23. Nov. 2015
um 18:36 Uhr:
Fake an IP. I guess for Firefox are add-ons available.
then sit back with a glass of your preferred beverage, and watch
ethan wiener's longer one.
Do you know of a different source for that video. It is blocked in
Germany for the usual GEMA reasons.
6:22 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com> schrieb am Mo., 23. Nov. 2015
um 18:08 Uhr:
Hmmmm. Not sure about intuition...
Please, do as Paul has suggested and watch Monty's video (the video on
xiph.org). This will help you adjust your intuition.
Kind regards,
Felix
6:47 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015, jonetsu(a)teksavvy.com wrote:
On Mon, 23 Nov 2015 17:00:47 +0000
Will Godfrey <willgodfrey(a)musically.me.uk> wrote:
Very simplified.... Filtering. Really, watch the video a xiph.org before
you say any more. They compare (on analog equipment) input and output wave
forms at 44.1k.
--
Len Ovens
www.ovenwerks.net
Although it seems counter-intuitive that actually is quite wrong!
Hmmmm. Not sure about intuition. A sine wave with 16 sampled points
will end up like a linked list of edges. Saving that to file and
resampling at 128 will only add points to the straight lines. It will
not create curves as per the original. It cannot. How would it know it
was a sine wave and not a guitar tone when it had to process basically
what was a robotic tone ?
8:55 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 09:46:55 -0800 (PST)
Len Ovens <len(a)ovenwerks.net> wrote:
I watched the video. I still remain with what I said, really. As a
parallel, a picture - parallel also made in this video - if you take a
very low resolution of the Mona Lisa it will not be possible to
reconstruct the quality of the original. The information is not
there. This is quite obvious with pictures. Same with sampling.
Attached is a quick drawing to illustrate. It is coarse as there are
very few sampling points, but the principle applies. The limit is the
finesse of the source material. If there are extremely minute changes
in the wave then the sampling must be done at an according rate if one
wishes to preserve the original as much as possible. Parameters of
interpolation can be applied, but how would you interpolate a 4-bit
Mona Lisa to show the subtleties of Da Vinci's painting ?
Now, I would like to know that in reality things are not like that and
that high, true-to-orignal, quality can be made from low quality sources
so I remain open to all comments and ideas.
On Mon, 23 Nov 2015, jonetsu(a)teksavvy.com wrote:
> On Mon, 23 Nov 2015 17:00:47 +0000
> Will Godfrey <willgodfrey(a)musically.me.uk> wrote:
>> Although it seems counter-intuitive that
actually is quite wrong!
Hmmmm. Not
sure about intuition. A sine wave with 16 sampled points
will end up like a linked list of edges. Saving that to file and
resampling at 128 will only add points to the straight lines. It
will not create curves as per the original. It cannot. How would
it know it was a sine wave and not a guitar tone when it had to
process basically what was a robotic tone ?
Very simplified.... Filtering. Really, watch the video a xiph.org
before you say any more. They compare (on analog equipment) input and
output wave forms at 44.1k. 
25 Nov
25 Nov
1:34 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
Again, you're missing several obvious points, all of which have been made
before.
Nyquist says that to **perfectly** capture all frequencies up to N kHz, you
need to sample at N*2.
Human hearing ends even in the best of us at about 20kHz. We sample at
typically 40kHz plus.
The implication: this "low resolution" example you're talking about
isn't
remotely close to what is happening with real world digital audio. Sure, if
you sample at 8kHz like early digital telephony (thus only capturing
frequencies up to 4kHz, the result after D->A differs substantially from
the original. But that's why we don't do that.
Also, your drawings ignore the basic point that Felix made. Your "original"
curve includes frequencies that are above the cutoff for the "sample rate"
your "digital" version uses.
On Mon, Nov 23, 2015 at 2:55 PM, jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com>
wrote:
On Mon, 23 Nov 2015 09:46:55 -0800 (PST)
Len Ovens <len(a)ovenwerks.net> wrote:
I watched the video. I still remain with what I said, really. As a
parallel, a picture - parallel also made in this video - if you take a
very low resolution of the Mona Lisa it will not be possible to
reconstruct the quality of the original. The information is not
there. This is quite obvious with pictures. Same with sampling.
Attached is a quick drawing to illustrate. It is coarse as there are
very few sampling points, but the principle applies. The limit is the
finesse of the source material. If there are extremely minute changes
in the wave then the sampling must be done at an according rate if one
wishes to preserve the original as much as possible. Parameters of
interpolation can be applied, but how would you interpolate a 4-bit
Mona Lisa to show the subtleties of Da Vinci's painting ?
Now, I would like to know that in reality things are not like that and
that high, true-to-orignal, quality can be made from low quality sources
so I remain open to all comments and ideas.
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
On Mon, 23 Nov 2015, jonetsu(a)teksavvy.com wrote:
> On Mon, 23 Nov 2015 17:00:47 +0000
> Will Godfrey <willgodfrey(a)musically.me.uk> wrote:
>> Although it seems counter-intuitive that
actually is quite wrong!
Hmmmm.
Not sure about intuition. A sine wave with 16 sampled points
will end up like a linked list of edges. Saving that to file and
resampling at 128 will only add points to the straight lines. It
will not create curves as per the original. It cannot. How would
it know it was a sine wave and not a guitar tone when it had to
process basically what was a robotic tone ?
Very simplified.... Filtering. Really, watch the video a xiph.org
before you say any more. They compare (on analog equipment) input and
output wave forms at 44.1k.
2:56 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Wed, 25 Nov 2015 07:34:46 -0500, Paul Davis wrote:
We sample at typically 40kHz plus.
Sometimes guitars, vocals and synth can be recorded without loss, when
using 32 KHz 12 bit, but it also could lead to strange results and most
of the times there's still audible loss at 44,1 kHz 16 bit.
48 KHz 16 bit (or maybe 24 bit) is good enough to never get audible
loss. Getting audible loss at 48 KHz 16 bit happens often, but this
isn't caused by the sample rate and bit depth, a cause could be the used
hardware or software, some hardware might sound better at higher sample
rates, but again then it's the hardware, not the sample rate.
To work with audio data we should use 32-bit floating point.
To care if, when and how to use what kind of dither might be worth
reading.
If digital audio processing nowadays is able to simulate analog gear is
another issue and has nothing to do with this thread ;), but for a
professional audio recording without loss you need 48 kHz only.
44,1 kHz is not enough, more then 48 kHz is not needed.
--
http://www.grundgesetz-gratis.de/
1:55 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 14:55:31 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
(as you will have seen, this is a repeat with the high-res drawing that
was held by the mailing list system)
Now, I would like to know that in reality things are
not like that and
that high, true-to-orignal, quality can be made from low quality
sources so I remain open to all comments and ideas.
23 Nov
23 Nov
10:02 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 09:46:55 -0800 (PST)
Len Ovens <len(a)ovenwerks.net> wrote:
On Mon, 23 Nov 2015, jonetsu(a)teksavvy.com wrote:
> Hmmmm. Not sure about intuition. A sine wave
with 16 sampled points
> will end up like a linked list of edges. Saving that to file and
> resampling at 128 will only add points to the straight lines. It
> will not create curves as per the original. It cannot. How would
> it know it was a sine wave and not a guitar tone when it had to
> process basically what was a robotic tone ?
Very simplified.... Filtering. Really, watch the video
a xiph.org
before you say any more. They compare (on analog equipment) input and
output wave forms at 44.1k.
I have replied an hour ago, but the reply did not make it yet. Maybe
due to the 48K attachment. So let's try with a low-res one :)
I watched the video. I still remain with what I said, really. As a
parallel, a picture - parallel also made in this video - if you take a
very low resolution of the Mona Lisa it will not be possible to
reconstruct the quality of the original. The information is not
there. This is quite obvious with pictures. Same with sampling.
Attached is a quick drawing to illustrate. It is coarse as there are
very few sampling points, but the principle applies. The limit is the
finesse of the source material. If there are extremely minute changes
in the wave then the sampling must be done at an according rate if one
wishes to preserve the original as much as possible. Parameters of
interpolation can be applied, but how would you interpolate a 4-bit
Mona Lisa to show the subtleties of Da Vinci's painting ?
Now, I would like to know that in reality things are not like that and
that high, true-to-orignal, quality can be made from low quality sources
so I remain open to all comments and ideas.
10:15 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 16:02:40 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
I have replied an hour ago, but the reply did not make
it yet. Maybe
due to the 48K attachment. So let's try with a low-res one :)
As the low-res version of the drawing may induce, reconstructing the
original does not seem too hard. You have to know how the letters
of the alphabet of that type of font are, and a few details about
shapes so it is possible for the algorithm to reconstruct the tip of
the arrow. And then based on this to exclude all other noise. Not too
complicated.
Then for audio it would be possible. Simply enter the parameters.
Martin guitar (model type) made in summer of 1994, brand new nanoweb
strings, Dunlop flat pick .73 mm, velocity, and a few other parameters
(using Finger Ease spray or not). Then feed a coarse wave shape and
voila, you get a rich Martin 440Hz (precisely :) 5th string output. :)
The possible technology is interesting, though.
11:28 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
This is just wrong. I wish you would pay more attention to what you don't
know than what you do know.
You are mssing several key things:
1) all waveforms can be represented as the sum of a series of sinusoids.
The more sinusoids in the series, the more accurate the model of the
original waveform (even if it was not composed of any sinusoids to begin
with. This is the basis of both Fourier analysis and Fourier synthesis,
which are key concepts in the abstract representation of wave data. There
are a few complications with this model, but for the purposes we are
discussing here, it is 100% complete and accurate.
2) Nyquist's theorem proves (and note that I said *proves*, not
"asserts") that sampling at a given sample rate provides enough data to
reconstruct **PERFECTLY** any signal made up frequencies up to the sample
rate divided by two.
2) the interpolation done by digital-to-anlog converters is based on a
combination of points (1) and (2), and results in **PERFECT** reproduction
of the original waveform (if (and only) the clocking is identical). In
reality, there is no "low-res". See Monty's example of 8kHz sampling rate
in the video.
Your comparisons with letter shapes is actually somewhat apropos, but only
for letter shapes formed by band-limited waveforms. You can, in fact,
synthesize letter shapes (or any other shapes) using fourier synthesis, but
it is slow and generally not how typography works. If you do know how it
actually works (e.g. the specific spline algorithms that will be used by a
font renderer) then you can, in fact perfectly "sample" a font with a
handful of points per letter that will feed the spline rendering algorithm.
The data is generally called a TTF (TrueType Font) file, and is used by
your computer all the time.
Speculating on interesting ways to abuse/use points (1), (2) and (3) has
been a rich source of ideas for audio engineering and processing for about
a hundred years. It doesn't do a lot, however, to carry out this
speculation without a basic background in signal processing in general and
digital signal processing in particular.
On Mon, Nov 23, 2015 at 4:15 PM, jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com>
wrote:
On Mon, 23 Nov 2015 16:02:40 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
I have replied an hour ago, but the reply did not
make it yet. Maybe
due to the 48K attachment. So let's try with a low-res one :)
As the low-res version of the drawing may induce, reconstructing the
original does not seem too hard. You have to know how the letters
of the alphabet of that type of font are, and a few details about
shapes so it is possible for the algorithm to reconstruct the tip of
the arrow. And then based on this to exclude all other noise. Not too
complicated.
Then for audio it would be possible. Simply enter the parameters.
Martin guitar (model type) made in summer of 1994, brand new nanoweb
strings, Dunlop flat pick .73 mm, velocity, and a few other parameters
(using Finger Ease spray or not). Then feed a coarse wave shape and
voila, you get a rich Martin 440Hz (precisely :) 5th string output. :)
The possible technology is interesting, though.
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
24 Nov
24 Nov
3:29 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015 17:28:02 -0500
Paul Davis <paul(a)linuxaudiosystems.com> wrote:
You are mssing several key things:
1) all waveforms can be represented as the sum of a
series of
sinusoids. The more sinusoids in the series, the more accurate the
model of the original waveform (even if it was not composed of any
sinusoids to begin with.
Ah. This is actually quite an important notion.
2) Nyquist's theorem proves (and note that I
said *proves*, not
"asserts") that sampling at a given sample rate provides enough data
to reconstruct **PERFECTLY** any signal made up frequencies up to the
sample rate divided by two.
I'll look up these two.
One thing that is clear, and this what got me thinking that way, is
that in Audacity, when zooming is done to see the actual sampling
points on a wave from a human voice sound, the line between two
sampling points is straight. Now, with thousands of sampling points a
curve can be represented if one zooms out enough to see it as a curve.
Although this level of detail, to examine straight lines between two
sampling points, might very well be in practical terms /not/ useful
since it foregoes meta-notions such as 1) and 2) which are more
important in day-to-day applications.
3:53 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Tue, Nov 24, 2015 at 9:29 AM, jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com>
wrote:
One thing that is clear, and this what got me thinking that way, is
that in Audacity, when zooming is done to see the actual sampling
points on a wave from a human voice sound, the line between two
sampling points is straight.
In Monty's video he *specifically* talked about how this representation of
digital samples is wrong and has confused so many people.
Now, with thousands of sampling points a
curve can be represented if one zooms out enough to see it as a curve.
The connecting straight line segment is an illusion and does not exist. You
are getting confused in the same way that Monty discussed.
I will say that I don't that he did a very good job of explaining this part
though. Sampled waveforms should never be drawn as "staircases", but should
be pointillistic. However to understand how this works, you need to
understand how the reconstruction filter works in a digital to analog
converter, and he didn't really describe this, and most people don't know
it either.
4:27 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Tue, 24 Nov 2015 09:53:40 -0500
Paul Davis <paul(a)linuxaudiosystems.com> wrote:
In Monty's video he *specifically* talked about
how this
representation of digital samples is wrong and has confused so many
people.
This may be naive, but I thought it did not apply because in
Audacity, at that zoom in level, it really does not look like a
staircase at all, here for an acoustic solo guitar waveform. Even
though the lines between points are straight, they are following
curves. See attached. So I thought his staircase was a gross
representation that did not apply.
The connecting straight line segment is an illusion
and does not
exist. You are getting confused in the same way that Monty discussed.
I will say that I don't that he did a very good
job of explaining
this part though. Sampled waveforms should never be drawn as
"staircases", but should be pointillistic. However to understand how
this works, you need to understand how the reconstruction filter
works in a digital to analog converter, and he didn't really describe
this, and most people don't know it either.
I was wondering about this. In extremely simple terms I think that
capacitors smooth out curves.
4:17 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Tue, 24 Nov 2015, jonetsu(a)teksavvy.com wrote:
One thing that is clear, and this what got me thinking
that way, is
that in Audacity, when zooming is done to see the actual sampling
points on a wave from a human voice sound, the line between two
sampling points is straight. Now, with thousands of sampling points a
That is a graphic representation chosen by the Audacity devs. Nothing to
do with reality, just close enough to be useful.
--
Len Ovens
www.ovenwerks.net
4:21 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com> schrieb am Di., 24. Nov. 2015
um 15:29 Uhr:
On Mon, 23 Nov 2015 17:28:02 -0500
Paul Davis <paul(a)linuxaudiosystems.com> wrote:
Here's a question that might help your understanding a bit:
For a given set of samples, how many possible signals (waveforms) do exist
that
1. are made up of sinusoids with frequencies up to the sample rate divided
by two, and
2. pass through all the sample points?
Answer: One.
2) Nyquist's theorem proves (and note that
I said *proves*, not
"asserts") that sampling at a given sample rate provides enough data
to reconstruct **PERFECTLY** any signal made up frequencies up to the
sample rate divided by two.
4:34 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Tue, Nov 24, 2015 at 10:20 AM, Felix Homann <linuxaudio(a)showlabor.de>
wrote:
Here's a question that might help your understanding a bit:
For a given set of samples, how many possible signals (waveforms) do exist
that
1. are made up of sinusoids with frequencies up to the sample rate divided
by two, and
2. pass through all the sample points?
Answer: One.
That is excellent. It is also the missing piece in Monty's presentation.
Whenever he gets to that part, I know I always have the gut feeling "there
are any number of possible waveforms I could draw that satisfy those two
requirements". But the gut feeling is wrong, and the correct answer
("one")
is deeply significant. Thanks for making this so clear.
23 Nov
23 Nov
11:28 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, 23 Nov 2015, jonetsu(a)teksavvy.com wrote:
On Mon, 23 Nov 2015 09:46:55 -0800 (PST)
Len Ovens <len(a)ovenwerks.net> wrote:
Bad parallel already. "very low" and "beyond the eyes ability to see"
are
two different things. 24khz - whatever the lowpass filter throws at things
is still higher than the ear can hear. It is not "very low res", It is
higher than tape or vinyl.
In order for a picture to make sense it must be viewed from enough
distance for the whole to be seen at a time. Most art galleries with
valuable paintings will not even let you closer than beyond arms reach
anyway. Being able to see something with a microscope does not prove
anything as at any magnification the context for our eyes to see a picture
is lost.
Being able to use a scope to see that bandwidth is limmited to what we can
hear is also out of context... we use our ears to listen, not a scope.
Anyway, it hardly matters, each is free to record at whatever sample rate
they wish. If recording at 48000 makes me happy, Great! and if recording
at 96000 makes you happy, Great!
I am not in any way arguing for low res anything, but rather appropriate
resolution for the use. Higher than detectable with the intended receiver.
Not ultra high, just cause it's possible (and sellable/profitable).
--
Len Ovens
www.ovenwerks.net
On Mon, 23 Nov 2015, jonetsu(a)teksavvy.com wrote:
> Hmmmm. Not sure about intuition. A sine wave
with 16 sampled points
> will end up like a linked list of edges. Saving that to file and
> resampling at 128 will only add points to the straight lines. It
> will not create curves as per the original. It cannot. How would
> it know it was a sine wave and not a guitar tone when it had to
> process basically what was a robotic tone ?
Very simplified.... Filtering. Really, watch the
video a xiph.org
before you say any more. They compare (on analog equipment) input and
output wave forms at 44.1k.
I have replied an hour ago, but the reply did not make it yet. Maybe
due to the 48K attachment. So let's try with a low-res one :)
I watched the video. I still remain with what I said, really. As a
parallel, a picture - parallel also made in this video - if you take a
very low resolution of the Mona Lisa it will not be possible to
reconstruct the quality of the original. The information is not 
24 Nov
24 Nov
12:57 a.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, November 23, 2015 3:02 pm, jonetsu(a)teksavvy.com wrote:
Attached is a quick drawing to illustrate. It is
coarse as there are
very few sampling points, but the principle applies.
The picture you have drawn violates the very first requirement of
sampling, namely that the input is bandwidth limited to less than half of
the sampling rate.
how would you interpolate a 4-bit
Mona Lisa to show the subtleties of Da Vinci's painting ?
Switching arguments between sample rate and bit depth, along with
providing an example which violates the first requirement of discrete time
sampling indicates that you did not understand the presentation. Perhaps
watch it again.
--
Chris Caudle
23 Nov
23 Nov
6:39 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
Am 23.11.2015 um 17:44 schrieb Paul Davis:
Your understanding is incorrect. Please watch this:
https://xiph.org/video/vid2.shtml
Yep, that's definitely recommended watching.
I'd also like to add: if you want to achieve better audio quality in
your recording process, the first *technical* measure is use better
microphones, preamps and audio interfaces.
(Things like acoustic treatment, microphone setup, proper cabling and
shielding etc. are other avenues to explore.)
Chris
5:48 p.m.
On Mon, 23 Nov 2015 11:33:44 -0500, jonetsu(a)teksavvy.com wrote:
But the 15 tracks, even the synths and synth drums,
will take so many
more megabytes of space, because the sampling rate is not set on a per
track basis.
About what values of sample rates are you talking? How less disk space
do you have?
--
http://www.grundgesetz-gratis.de/
6:04 p.m.
On Mon, 23 Nov 2015 17:48:02 +0100
Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote:
On Mon, 23 Nov 2015 11:33:44 -0500,
jonetsu(a)teksavvy.com wrote:
Yes, this could be a good question when wanting to actually go to
/much/ higher sampling rates. The math should be straightforward but I
haven't done it. The example given shows a significant amount of disk
space used when sampling at 174,000 something for only a few seconds,
hundreds of thousands of KB.
But the 15 tracks, even the synths and synth
drums, will take so many
more megabytes of space, because the sampling rate is not set on a
per track basis.
About what values of sample rates are you talking? How less disk space
do you have?
6 p.m.
On Mon, 23 Nov 2015 11:33:44 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
Of course not. Resampling a 48k sample to 96k won't add fidelity, but it'll
get them to match the sample rate of the higher-quality tracks, which I
assumed already existed.
On Mon, 23 Nov 2015 11:18:32 -0500
Joe Hartley <jh(a)brainiac.com> wrote:
As Paul mentioned, even if you're not running JACK, this is still non-trivial.
Your session runs at a sample rate set by JACK.
All the files that
have audio in them have to match that value. If the audio you want
to use has a higher sample rate, it will have to be re-sampled at the
lower rate for Ardour to use it in that session.
Sure. Thing is, I sample, eg. I record in Ardour only. So I only
speak of Ardour. If you want to have JACK and Ardour run at the
higher sample rate,
assuming your interface supports it, you'd have to create a new
session and import the tracks. The files with the audio that match
that new, higher rate will be imported with minimal processing, while
the other tracks will be up-sampled.
This "up-sample" as far as I understood from the course, does not exist
in reality as far as the quality of audio is concerned. > There is no way for Ardour to use files of a
different sample rate
> without them being resampled to the rate of the session, as
> determined by JACK's setting.
This holds true for recording new tracks as well - you can't have a session
at 48k and then add tracks at 96k.
Out of say 15 tracks, there would be 4 that need
higher quality
sampling because they are acoustic instruments. But the 15 tracks,
even the synths and synth drums, will take so many more megabytes of
space, because the sampling rate is not set on a per track basis.
Storage is cheap.
Maybe sampling rates on a per-track basis does not
make sense in FAWs
in general.
With the timing involved, I can't see where multiple sample rates within a
session could ever work well.
--
======================================================================
Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com
Without deviation from the norm, "progress" is not possible. - FZappa
5:19 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, Nov 23, 2015 at 11:01 AM, jonetsu(a)teksavvy.com <jonetsu(a)teksavvy.com
wrote:
I will try the stereo bus for the mono guitar and bass tracks then.
I have tried a demo plugin that has 8 outputs. In this case Ardour
will show (see attached) a 'spatial' round control to move the audio
around, at least left to right in stereo. Once this control is there I
haven't found the way to remove it but moreover, now I would like to
use the same control on other regalar stereo tracks - is it possible ?
we are working on ways to control the number of outputs from "multichannel"
(>2) plugins.
the control you've seen is Ardour's VBAP panner. it exists because you have
multiple channels and so a normal panner cannot be used. You can't use it
with tracks/busses with <3 outputs, and you wouldn't want to. It is
generally best to forget that the VBAP panner exists, especially when it
comes to multi-output plugins (which are themselves a horrible abberation.
____
5:19 p.m.
On Mon, 23 Nov 2015 11:01:15 -0500, jonetsu(a)teksavvy.com wrote:
Importing will defeat in general the purpose of higher
quality because
the source material has less to start with.
There are anti-alias lowpass filters in your hardware's input that fit
to a sample rate. Converting (and importing) the existing material to
this sample rate could be useful before recording the other instruments.
--
http://www.grundgesetz-gratis.de/
5:16 p.m.
New subject: Mono tracks to mono or stereo bus ? And sampling rate
On Mon, Nov 23, 2015 at 10:50 AM, Joe Hartley <jh(a)brainiac.com> wrote:
On Mon, 23 Nov 2015 10:31:12 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
Ardour's default is to provide the same number of outputs as the master bus
has inputs.
The number of channels in the master bus defaults to 2 but can be anything.
I've seen it used with 40.
When having multiple mono guitar tracks in Ardour
and wanting to
regroup them in a single bus, would that bus be preferably mono or
stereo ? How is this in relation to eventually placing the guitar bus
in a stereo field ?
Ardour's default is to provide 2-channel output, so I've always made my
busses stereo.
5:07 p.m.
Regarding the sample rate consider to use 48 kHz usually there's no
advantage to use a higher sampling rate and most of the times a lower
sample rate does cause audible loss.
"The Audio Engineering Society recommends 48 kHz sampling rate for most
applications"
"Most professional audio gear uses 48 kHz sampling, including mixing
consoles, and digital recording devices."
https://en.wikipedia.org/wiki/Sampling_%28signal_processing%29
Regards,
Ralf
--
http://www.grundgesetz-gratis.de/
5:45 p.m.
On Mon, 23 Nov 2015 17:07:22 +0100
Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote:
Regarding the sample rate consider to use 48 kHz
usually there's no
advantage to use a higher sampling rate and most of the times a lower
sample rate does cause audible loss.
"The Audio Engineering Society recommends 48 kHz sampling rate for
most applications"
"Most professional audio gear uses 48 kHz sampling, including mixing
consoles, and digital recording devices."
https://en.wikipedia.org/wiki/Sampling_%28signal_processing%29
That can be a much valid point. Everything boils down to what can be
distributed easily anyways. Is there a significant difference when
sampling at 192,000 (jack max on this machine) and then having a final
mastering for CD purposes ? Well, there will be more points to work
with, this is certain. Does it make a difference ? Will more
complexities of the guitar playing end up being actually felt, tidbits
that audiophiles can argue about for hours ? Will it bring a somewhat
sense of a richer sound to the listener ? Not sure that it is entirely
mathematics. The notion of psychoacoustics was used in developing the
MP3 format as inherently there's more than math to music. Will this
have an influence on the projects I make - most probably not, although
it is an interesting topic.
https://en.wikipedia.org/wiki/Psychoacoustics
5:55 p.m.
On Mon, 23 Nov 2015 11:45:48 -0500, jonetsu(a)teksavvy.com wrote:
Is there a significant difference when sampling at
192,000 (jack max
on this machine) and then having a final mastering for CD purposes ?
192 KHz unlikely will gain anything useful, use 48 KHz.
192 KHz unlikely will sound better for human ears, than 48 KHz already
do. At 48 KHz the bit depth still might be impotant, don't chose 2
bits ;).
48KHz has got a better sound quality than CD quality, so you can do a
better mix before you get loss, when down sampling it to CD quality.
--
http://www.grundgesetz-gratis.de/
5:58 p.m.
On Mon, 23 Nov 2015 11:45:48 -0500
"jonetsu(a)teksavvy.com" <jonetsu(a)teksavvy.com> wrote:
On Mon, 23 Nov 2015 17:07:22 +0100
Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote:
I believe the general advice is for small projects with little going on inside
the box 48k is ideal. If you intend to do a lot of editing, processing and
general effing about go for 96k and down sample the final master.
Don't bother with 192k at all unless your room is on the North wall of the
building and needs extra heating.
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Regarding the sample rate consider to use 48 kHz
usually there's no
advantage to use a higher sampling rate and most of the times a lower
sample rate does cause audible loss.
"The Audio Engineering Society recommends 48 kHz sampling rate for
most applications"
"Most professional audio gear uses 48 kHz sampling, including mixing
consoles, and digital recording devices."
https://en.wikipedia.org/wiki/Sampling_%28signal_processing%29
That can be a much valid point. Everything boils down to what can be
distributed easily anyways. Is there a significant difference when
sampling at 192,000 (jack max on this machine) and then having a final
mastering for CD purposes ? Well, there will be more points to work
with, this is certain. Does it make a difference ? Will more
complexities of the guitar playing end up being actually felt, tidbits
that audiophiles can argue about for hours ? Will it bring a somewhat
sense of a richer sound to the listener ? Not sure that it is entirely
mathematics. The notion of psychoacoustics was used in developing the
MP3 format as inherently there's more than math to music. Will this
have an influence on the projects I make - most probably not, although
it is an interesting topic.
https://en.wikipedia.org/wiki/Psychoacoustics
6:11 p.m.
On Mon, 23 Nov 2015 16:58:35 +0000
Will Godfrey <willgodfrey(a)musically.me.uk> wrote:
Don't bother with 192k at all unless your room is
on the North wall
of the building and needs extra heating.
Ah ah. Actually there was a point in the course about this. Not the
heating source. WAV files are easier to process than compressed files.
So since Ardour uses WAV files, there might be a need for an extra
heating source in any case.
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linux-audio-user@lists.linuxaudio.org
45 comments
11 participants
participants (11)
-
Chris Caudle -
Christopher Arndt -
Felix Homann -
Joe Hartley -
jonetsu＠teksavvy.com -
Len Ovens -
Leonardo Palomares -
Neil C Smith -
Paul Davis -
Ralf Mardorf -
Will Godfrey