21 May
2009
21 May
'09
5:28 p.m.
Heya!
I am currently working on the volume control handling of
PulseAudio. Until now the sliders in the UI mapped their position
linearly to the dB volume scale. i.e. if the slider is at '0%' we get
-90dB, if it is at '100%' we get 0dB, and between that we map linearly
from those percentages to the dB value. To be frank, this sucked big
time because the 'interesting' part is usually way up at -10dB to
0dB. and the remaining part of the volume sliders is pretty boring.
From reading through the ALSA drivers I know that a lot
of sound card
mixer controls usually have a much higher dB resolution near 0dB then
the have near -90dB. Most of the time the steps they choose are very
arbitrary however.
So, I was looking for better suggestions for mapping 'pixel
distances' in the UI to volume factors. I googled a bit and found this:
http://www.robotplanet.dk/audio/audio_gui_design/
Which suggests a cubic relation between 'pixel distances' and the
linear factor.
Any opinions on that? Other suggestions?
Thanks,
Lennart
--
Lennart Poettering Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/ GnuPG 0x1A015CC4
21 May
21 May
5:48 p.m.
On Thu, May 21, 2009 at 8:27 PM, Lennart Poettering <mzynq(a)0pointer.de> wrote:
From reading through the ALSA drivers I know that a
lot of sound card
mixer controls usually have a much higher dB resolution near 0dB then
the have near -90dB. Most of the time the steps they choose are very
arbitrary however.
There is an IEC standard for this mapping in meters (IEC 60-268-18)
which works OK for faders as well if extended in a sensible way down
to -inf, particularly if you don't want to go above 0dB. It divides
the range up into several sections which are individually linear in
dB. There's code for this mapping in several places, including
base/AudioLevel.cpp in the Rosegarden source code.
A cubic mapping is also good.
Chris
6:07 p.m.
On Thu, May 21, 2009 at 8:48 PM, Chris Cannam
<cannam(a)all-day-breakfast.com> wrote:
There is an IEC standard for this mapping in meters
(IEC 60-268-18)
which works OK for faders as well if extended in a sensible way down
to -inf, particularly if you don't want to go above 0dB. It divides
the range up into several sections which are individually linear in
dB. There's code for this mapping in several places, including
base/AudioLevel.cpp in the Rosegarden source code.
Actually I might as well summarise here, not least because I've
noticed the RG code does not in fact implement exactly what's in the
standard.
The standard has the top 25% of the meter covering 0dB down to -9dB;
the next 25% covering -9 to -20; the next 20% covering -20 to -30; the
next 15% covering -30 to -40; the next 7.5% covering -40 to -50; the
next 5% covering -50 to -60; and the last 2.5% covering -60 to -Inf.
Each of those ranges is linear in dB.
The numbered levels are 0, -5, -10, -15, -20, -30, -40, -50, -60.
Ticks are per dB down to -20, then only at numbered positions, with an
additional tick at -45dB (why?)
This is actually quite different from the cubic mapping; it gives you
a lot more space in -10 to -20 relative to 0 to -10. That suggests
that you can get away with quite a few different mappings in practice.
A cubic mapping is also good.
In Rosegarden we actually switched from a cubic mapping to one based
on the IEC meter mapping, many years ago, because "it seemed like a
good idea at the time". I've never really made up my mind whether it
turned out better or worse. Nowadays I'd probably go for cubic, since
it's a one-liner to code.
Chris
6:12 p.m.
Am Donnerstag, den 21.05.2009, 21:07 +0100 schrieb Chris Cannam:
On Thu, May 21, 2009 at 8:48 PM, Chris Cannam
<cannam(a)all-day-breakfast.com> wrote:
Here is a simple discription for a fader taper like that on musikdsp.org
http://www.musicdsp.org/archive.php?classid=5#94
There is an IEC standard for this mapping in
meters (IEC 60-268-18)
which works OK for faders as well if extended in a sensible way down
to -inf, particularly if you don't want to go above 0dB. It divides
the range up into several sections which are individually linear in
dB. There's code for this mapping in several places, including
base/AudioLevel.cpp in the Rosegarden source code.
Actually I might as well summarise here, not least because I've
noticed the RG code does not in fact implement exactly what's in the
standard.
The standard has the top 25% of the meter covering 0dB down to -9dB;
the next 25% covering -9 to -20; the next 20% covering -20 to -30; the
next 15% covering -30 to -40; the next 7.5% covering -40 to -50; the
next 5% covering -50 to -60; and the last 2.5% covering -60 to -Inf.
Each of those ranges is linear in dB.
The numbered levels are 0, -5, -10, -15, -20, -30, -40, -50, -60.
Ticks are per dB down to -20, then only at numbered positions, with an
additional tick at -45dB (why?)
This is actually quite different from the cubic mapping; it gives you
a lot more space in -10 to -20 relative to 0 to -10. That suggests
that you can get away with quite a few different mappings in practice.
A cubic mapping is also good.
In Rosegarden we actually switched from a cubic mapping to one based
on the IEC meter mapping, many years ago, because "it seemed like a
good idea at the time". I've never really made up my mind whether it
turned out better or worse. Nowadays I'd probably go for cubic, since
it's a one-liner to code.
Chris 
6:26 p.m.
Having +10 dB at the top is a good idea.
On Thu, 2009-05-21 at 22:22 +0200, hermann meyer wrote:
Here is a simple discription for a fader taper like
that on musikdsp.org
http://www.musicdsp.org/archive.php?classid=5#94
6:39 p.m.
Chris Cannam wrote:
Jens M Andreasen wrote:
A cubic
mapping is also good.
In Rosegarden we actually switched from a cubic mapping to one based
on the IEC meter mapping, many years ago, because "it seemed like a
good idea at the time". I've never really made up my mind whether it
turned out better or worse. Nowadays I'd probably go for cubic, since
it's a one-liner to code. Having +10 dB at the top is a good idea.
As a user I never felt uncomfortable with the Rosegarden 1.6.1 and 1.7.0
faders and sometimes its fine to have +10 dB too.
7:21 p.m.
On Thu, 2009-05-21 at 22:39 +0200, Ralf Mardorf wrote:
As a user I never felt uncomfortable with the
Rosegarden 1.6.1 and 1.7.0
faders and sometimes its fine to have +10 dB too.
These faders are neither for Rosegarden nor Ardour, I think ... Rather
for application like Rythmbox, Ekiga and various Desktop plings that
needs an easy-to-use short-throw fader for course control of their
respective master volumes.
9:17 p.m.
Jens M Andreasen wrote:
On Thu, 2009-05-21 at 22:39 +0200, Ralf Mardorf
wrote:
If faders should be fine in a wide range for different levels, a trick
can be a pad switch in addition to the faders.
As a user I never felt uncomfortable with the
Rosegarden 1.6.1 and 1.7.0
faders and sometimes its fine to have +10 dB too.
These faders are neither for Rosegarden nor Ardour, I think ... Rather
for application like Rythmbox, Ekiga and various Desktop plings that
needs an easy-to-use short-throw fader for course control of their
respective master volumes.
22 May
22 May
4:17 p.m.
On Fri, 2009-05-22 at 01:17 +0200, Ralf Mardorf wrote:
If faders should be fine in a wide range for different
levels, a trick
can be a pad switch in addition to the faders.
Mmmm ... Not bad at all, since my main problem with Desktop Application
Mixdown is with Ekiga. It has gotten old repeatedly singing: "Put your
sweet lips, a little closer, to the phone ..."
A +20dB button would solve that :)
9:41 p.m.
Jens M Andreasen wrote:
On Fri, 2009-05-22 at 01:17 +0200, Ralf Mardorf
wrote:
Seriously!
I don't use any sound for the desktop, but the computer beep and I don't
run PulseAudio. I don't know how it shall be handled for PulseAudio,
e.g. if there shall be a fader for each application and a master control
or what ever. A +20dB button used for a low level signal can cause
serious damages, while unexpected high level signals. I won't say that
+6dB or +10dB really will double the loudness, but a rule says +6dB =
same acoustic pressure at double distance in the free field, while
another rule says that +10dB will be felt as double loudness.
If an application like Ekiga often needs the signal to be gained, it
should be an option for the application itself and not for the whole
desktop environment. Without a limiter a gain of +6dB or +10dB IMO
shouldn't be overstepped, if it's for all desktop applications.
Another issue is, that if a signal needs to be gained, the side noise
will be gained too. A trick especially for an application like Eikiga
would be, to compress and gain and auto level (optimize) just the
frequencies that are relevant for human speech and to eliminate all
other frequencies by pass filters, but I guess this is something that
needs to be done by the application.
Messing around with +20db gain buttons is nothing for desktop users,
that aren't audio engineers.
Just my 2 cents.
Cheers,
Ralf
If faders should be fine in a wide range for
different levels, a trick
can be a pad switch in addition to the faders.
Mmmm ... Not bad at all, since my main problem with Desktop Application
Mixdown is with Ekiga. It has gotten old repeatedly singing: "Put your
sweet lips, a little closer, to the phone ..."
A +20dB button would solve that :)
9:46 p.m.
Ralf Mardorf wrote:
[snip]
Messing around with +20db gain buttons is nothing for desktop users,
that aren't audio engineers.
Just my 2 cents.
Cheers,
Ralf
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PS:
A pad to make the fader range more comfortable for different input
signal levels, really should be a pad, with a loss function and not a
gain function, still the faders might gain up to +10dB.
23 May
23 May
10:26 a.m.
Just to put some more wood in the fire as we say here...
I think this is a good reason to have pulse audio (naturally for
desktop systems, not high end audio workstations...)
https://launchpad.net/earcandy
Cheers,
Andrés
On Fri, May 22, 2009 at 6:51 PM, Jens M Andreasen
<jens.andreasen(a)comhem.se> wrote:
On Sat, 2009-05-23 at 01:41 +0200, Ralf Mardorf wrote:
--
Andrés
A trick especially for an application like
Eikiga
would be, to compress and gain and auto level (optimize) just the
frequencies that are relevant for human speech
No, please no! I do that in my head with no effort at all ...
_______________________________________________
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Linux-audio-dev(a)lists.linuxaudio.org
http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
25 May
25 May
9:09 p.m.
On Thu, 21.05.09 21:07, Chris Cannam (cannam(a)all-day-breakfast.com) wrote:
On Thu, May 21, 2009 at 8:48 PM, Chris Cannam
<cannam(a)all-day-breakfast.com> wrote:
Thanks a lot for this elaborate information from everyone on this
thread! Very helpful!
I have now changed PulseAudio to use cubic mapping, mostly because I
came to the conclusion that the most interesting attenuation range is
between 0dB and -10dB and hence should be the range with the greatest
resolution, which is not what IEC would give us.
What makes me wonder though is, if I use cubic mapping for < 0dB what
would be a good choice for > 0db? Just continue the cubic mapping?
After having implemented this I can say it doesn't feel that bad in
the range 0..+10dB, but I was wondering if it actually makes sense?
Opinions?
Lennart
--
Lennart Poettering Red Hat, Inc.
lennart [at] poettering [dot] net
http://0pointer.net/lennart/ GnuPG 0x1A015CC4
There is an IEC standard for this mapping in
meters (IEC 60-268-18)
which works OK for faders as well if extended in a sensible way down
to -inf, particularly if you don't want to go above 0dB. It divides
the range up into several sections which are individually linear in
dB. There's code for this mapping in several places, including
base/AudioLevel.cpp in the Rosegarden source code.
Actually I might as well summarise here, not least because I've
noticed the RG code does not in fact implement exactly what's in the
standard.
The standard has the top 25% of the meter covering 0dB down to -9dB;
the next 25% covering -9 to -20; the next 20% covering -20 to -30; the
next 15% covering -30 to -40; the next 7.5% covering -40 to -50; the
next 5% covering -50 to -60; and the last 2.5% covering -60 to -Inf.
Each of those ranges is linear in dB.
The numbered levels are 0, -5, -10, -15, -20, -30, -40, -50, -60.
Ticks are per dB down to -20, then only at numbered positions, with an
additional tick at -45dB (why?)
This is actually quite different from the cubic mapping; it gives you
a lot more space in -10 to -20 relative to 0 to -10. That suggests
that you can get away with quite a few different mappings in practice.
A cubic mapping is also good.
In Rosegarden we actually switched from a cubic mapping to one based
on the IEC meter mapping, many years ago, because "it seemed like a
good idea at the time". I've never really made up my mind whether it
turned out better or worse. Nowadays I'd probably go for cubic, since
it's a one-liner to code.
9:37 p.m.
On Tue, 2009-05-26 at 01:09 +0200, Lennart Poettering wrote:
what
would be a good choice for > 0db? Just continue the cubic mapping?
After having implemented this I can say it doesn't feel that bad in
the range 0..+10dB, but I was wondering if it actually makes sense?
It makes sense. It is the equevalence of putting a bogus 0 dB at +10 and
then attenuating that one instead.
/j
21 May
21 May
5:55 p.m.
On Thu, May 21, 2009 at 3:27 PM, Lennart Poettering <mzynq(a)0pointer.de> wrote:
[ .... non linear faders ... ]
in addition to chris' suggestion, there is ardour's rather arbitrary
but definitely comfortable mapping. these map between a gain
coefficient (not a dB level) and a fader position that varies from 0.0
to 1.0 (gain_t is a double, btw)
--p
-------------------------------------------------------------------
static inline double
gain_to_slider_position (ARDOUR::gain_t g)
{
if (g == 0) return 0;
return pow((6.0*log(g)/log(2.0)+192.0)/198.0, 8.0);
}
static inline ARDOUR::gain_t
slider_position_to_gain (double pos)
{
/* XXX Marcus writes: this doesn't seem right to me. but i don't have
a better answer ... */
if (pos == 0.0) return 0;
return pow (2.0,(sqrt(sqrt(sqrt(pos)))*198.0-192.0)/6.0);
}
6:43 p.m.
On Thu, May 21, 2009 at 03:55:53PM -0400, Paul Davis wrote:
in addition to chris' suggestion, there is
ardour's rather arbitrary
but definitely comfortable mapping.
How comfortable it is depends on what the fader is
used for. I've often wanted more gain than the
available +6dB, and somewhat less 'compression' at
the bottom end.
But what is really confusing is that the fader and
meter scales are different, while they are side by
side - it's quite easy to think that the meter's
scale ticks apply to the fader as well.
Ciao,
--
FA
Io lo dico sempre: l'Italia è troppo stretta e lunga.
8:08 p.m.
On Thu, May 21, 2009 at 4:43 PM, Fons Adriaensen <fons(a)kokkinizita.net> wrote:
On Thu, May 21, 2009 at 03:55:53PM -0400, Paul Davis
wrote:
this is not really a function of the fader directly, but of an
arbitrary decision to have a static range of -inf to +6dB of applied
gain. any such decision will be arbitrary - however, it being software
it would be nice to be able to pick the range.
in addition to chris' suggestion, there is
ardour's rather arbitrary
but definitely comfortable mapping.
How comfortable it is depends on what the fader is
used for. I've often wanted more gain than the
available +6dB, and somewhat less 'compression' at
the bottom end. But what is really confusing is that the fader and
meter scales are different, while they are side by
side - it's quite easy to think that the meter's
scale ticks apply to the fader as well.
i don't see any solution to this. AFAIK the specs for various standard
meters and for various standard faders do not use the same scale. any
time you put them side-by-side, there seems to be potential for
confusion. ardour isn't use standards for either, but the same issue
applies. if you have ideas i'm all ears (but no minutes to code it at
present).
9:27 p.m.
On Thu, May 21, 2009 at 06:08:32PM -0400, Paul Davis wrote:
We are talking about per-strip meters here. In the
HW days most mixers wouldn't even have those - maybe
just an 'overload' led if you were lucky.
In Ardour they are also used as recording level meters,
but apart from that they are mostly useful as 'signal
presence indicators' or to provide a quick visual hint
of from where (from which strip) some part of a mix
originates. There is no need for a 'standard' meter
in these functions.
Meters with a 'standard' scale that enables you to learn
unconsciously a relation between a visual indication
and what you hear (that is to only real reason for having
a meter scale standard), are required for mixing, mastering
and in broadcasting, but not for individual tracks in a DAW.
So in a DAW strip you can just ignore standards and make the
fader and meter scales the same if they are side by side.
What Ardour is lacking is sort of 'monitoring' window,
a place where you can select in a convenient way what
you hear, and how. Anything you do here should never
affect the 'program output' whatever that is, normally
the Master strip out.
*This* is the place to have a 'standard' meter, the
one you use for mixing, mastering, etc.
Such a monitoring module should switch automatically
between
- A signal selected by the user, normally this will
be the Master strip output, but it could also be an
external signal such as the output of a mastering
processor.
- Auditioning when active.
- Solo (not-in-place) when active.
It should also provide switching between different
monitor outputs (each with its own volume control).
It's also the place to integrate things such as
talkback (which requires dimming the monitors).
Ciao,
--
FA
Io lo dico sempre: l'Italia è troppo stretta e lunga.
this is not really a function of the fader directly,
but of an
arbitrary decision to have a static range of -inf to +6dB of applied
gain. any such decision will be arbitrary - however, it being software
it would be nice to be able to pick the range.
Almost all mixers go up to +10dB, some +15, some +20.
But what is
really confusing is that the fader and
meter scales are different, while they are side by
side - it's quite easy to think that the meter's
scale ticks apply to the fader as well.
i don't see any solution to this. AFAIK the specs for various standard
meters and for various standard faders do not use the same scale. any
time you put them side-by-side, there seems to be potential for
confusion. ardour isn't use standards for either, but the same issue
applies. if you have ideas i'm all ears (but no minutes to code it at
present).
9:37 p.m.
On Thu, May 21, 2009 at 7:26 PM, Fons Adriaensen <fons(a)kokkinizita.net> wrote:
We are talking about per-strip meters here. In the
HW days most mixers wouldn't even have those - maybe
just an 'overload' led if you were lucky.
i'm not trying to compete with the race to the bottom.
In Ardour they are also used as recording level
meters,
but apart from that they are mostly useful as 'signal
presence indicators' or to provide a quick visual hint
of from where (from which strip) some part of a mix
originates. There is no need for a 'standard' meter
in these functions.
i didn't try to suggest that ardour or needs standard meters here. my
point was simply that in all likelihood, any scale you choose here
that works for metering is different that the scale you want for the
fader. or do you think otherwise.
What Ardour is lacking is sort of 'monitoring'
window,
a place where you can select in a convenient way what
you hear, and how.
hear where? master outs? a listen bus? control outs bus? some other
bus(ses)? direct outs? there's no simple definition of "what you
hear". there might be N listeners at one time. whose monitoring is
being affected?
anyway, all of this is currently being radically overhauled in 3.0, so
i can't (for now) get too deeply into how 2.X works (or doesn't).
in my experience, many engineers find a distinction between a meter
bridge and a listening/monitoring system to be frequently useful.
tying meters to a specific listening path at any one time seems
potentially counter-productive.
22 May
22 May
5:54 a.m.
On Thu, May 21, 2009 at 07:37:31PM -0400, Paul Davis wrote:
On Thu, May 21, 2009 at 7:26 PM, Fons Adriaensen
<fons(a)kokkinizita.net> wrote:
By 'monitoring' here I mean the main studio monitors.
You should be able to listen to various things (including
monitor sends for musicians), and switch between them
easily and without interrupting the normal program flow.
Setting this up in a convenient way is currently near
impossible in Ardour. And if you emulate it using a lot
of extra strips etc. there is no visual hint of a 'safe'
area where whatever you do will never hurt the 'main'
outputs, and so you end up having to think twice before
pushing any button.
Have a look at what is provided in this area by some
'real' mixers, and ask the people who use them why
these things are what they are, and rather essential.
Ciao,
--
FA
Io lo dico sempre: l'Italia è troppo stretta e lunga.
What Ardour is lacking is sort of
'monitoring' window,
a place where you can select in a convenient way what
you hear, and how.
hear where? master outs? a listen bus? control outs bus? some other
bus(ses)? direct outs? there's no simple definition of "what you
hear". there might be N listeners at one time. whose monitoring is
being affected? 
24 May
24 May
1 a.m.
Check it, yo!: Fons Adriaensen was sayin:
By 'monitoring' here I mean the main studio monitors.
You should be able to listen to various things (including
monitor sends for musicians), and switch between them
easily and without interrupting the normal program flow.
Setting this up in a convenient way is currently near
impossible in Ardour. And if you emulate it using a lot
of extra strips etc. there is no visual hint of a 'safe'
area where whatever you do will never hurt the 'main'
outputs, and so you end up having to think twice before
pushing any button.
Have a look at what is provided in this area by some
'real' mixers, and ask the people who use them why
these things are what they are, and rather essential.
Ciao,
I find it very useful on the digital mixers (at least on ones like MC7L or
PM5D) where you can select a bus and the faders move to show the levels for
each channel being sent to the bus. This is often much more important than the
meter bridges, which only display the input levels off the head amps.
Perhaps you could have it set, either in the channels window or on a separate
meterbridge, where you can select a bus, and get a metering of each channel
reflecting the post send levels being sent to that bus. (This might be tough
given you can route buses to other buses)
I think in mixing that information is more important to me (especially for
live recording) than what standard scale is being used for the meters.
Personally, the meter standard being used doesn't mean much till I start
getting into mastering.
-Reuben
On Thu, May 21, 2009 at 07:37:31PM -0400, Paul Davis
wrote:
> On Thu, May 21, 2009 at 7:26 PM, Fons Adriaensen <fons(a)kokkinizita.net>
wrote:
What Ardour is lacking is sort of 'monitoring'
window,
a place where you can select in a convenient way what
you hear, and how.
hear where? master outs? a listen bus? control outs bus? some other
bus(ses)? direct outs? there's no simple definition of "what you
hear". there might be N listeners at one time. whose monitoring is
being affected?
26 May
26 May
10:16 a.m.
On Fri, May 22, 2009 at 3:53 AM, Fons Adriaensen <fons(a)kokkinizita.net> wrote:
By 'monitoring' here I mean the main studio monitors.
You should be able to listen to various things (including
monitor sends for musicians), and switch between them
easily and without interrupting the normal program flow.
Setting this up in a convenient way is currently near
impossible in Ardour.
Just create your sessions with a control outs bus (its in the advanced
settings section of the new session dialog). You can then switch the
solo model to "solo via bus", and get something extremely close to a
traditional "listen bus".
Have a look at what is provided in this area by some
'real' mixers, and ask the people who use them why
these things are what they are, and rather essential.
and if you have a look at those "real" mixers you will notice that
front-of-house/live-mixing desks are designed in quite different ways
than the ones designed for use in a control room. soloing and muting
in an FOH context can (by choice of the engineer) be very different
from the way it works in a recording studio. there are two entirely
different commercial markets for that stuff, and although its
"trivially easy" to implement both designs in software, figuring out
how to merge them into a coherent single beast is not.
there have been lots of discussions about this on IRC in recents weeks
in the context of 3.0, and things will be a bit different there.
21 May
21 May
7:04 p.m.
On Thu, May 21, 2009 at 09:27:41PM +0200, Lennart Poettering wrote:
From reading through the ALSA drivers I know that a
lot of sound card
mixer controls usually have a much higher dB resolution near 0dB then
the have near -90dB. Most of the time the steps they choose are very
arbitrary however.
With many cards the gain control provided by the HW
mixer is not meant c.q. not usable as volume control,
but just as a means to match the card to the external
hardware - to ensure that the maximum digital level more
or less corresponds to the maximum analog level expected
externally.
Given the arbitrary steps some HW provides, a 'master
volume control' can often be much better implemented in
software, leaving the HW's gain setting for a one-time
adjustment only.
Of course this doesn't work with really low-fi cards
that produce so much noise or interference that you
have no choice but using the analog gain if you want
low listening levels.
Ciao,
--
FA
Io lo dico sempre: l'Italia è troppo stretta e lunga.
5447
days inactive
5452
days old
linux-audio-dev@lists.linuxaudio.org
24 comments
9 participants
participants (9)
-
Andres Cabrera -
Chris Cannam -
Fons Adriaensen -
hermann meyer -
Jens M Andreasen -
Lennart Poettering -
Paul Davis -
Ralf Mardorf -
Reuben Martin