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-   -   Sound cards (Was: no sound (http://www.linux-archive.org/ubuntu-studio-user/528902-sound-cards-no-sound.html)

Thomas Orgis 05-20-2011 10:30 PM

Sound cards (Was: no sound
 
Am Fri, 20 May 2011 15:33:39 +0200
schrieb Ralf Mardorf <ralf.mardorf@alice-dsl.net>:

> No! But I suspect a bug for Jack and btw. there already was a rounding
> bug that was fixed, perhaps years ago.

Ah, OK. I'll ignore the discussion about different sound cards influencing this as a heap of confusion and settle for this: If JACK would not have bugs, by design it would do bit-exact copies, or at least copies with floating point computation errors in non-audible amplitude. I don't claim that this ideal case is indeed the reality.

When you find those bugs, I hope they'll get reported and fixed, for the benefit to us all. In the meantime, those of us with some time on their hands can just try to verify the issue of bit-exact copying of data through the pipeline ... I just don't want to go down that path right now, since I'm already not able to do my actual work because of debugging some gcc 4.6 breakage on code of mine.


Alrighty then,

Thomas.


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Ralf 05-20-2011 11:52 PM

Sound cards (Was: no sound
 
On Fri, 2011-05-20 at 16:53 +0200, Hartmut Noack wrote:
> Am 20.05.2011 15:15, schrieb Ralf Mardorf:
> > On Fri, 2011-05-20 at 15:00 +0200, Hartmut Noack wrote:
> >> Am 20.05.2011 14:37, schrieb Ralf Mardorf:
> >>> On Fri, 2011-05-20 at 14:04 +0200, Thomas Orgis wrote:
> >>>> Am Fri, 20 May 2011 13:54:57 +0200
> >>>> schrieb Ralf Mardorf<ralf.mardorf@alice-dsl.net>:
> >>>>
> >>>>> When recording soft synth just by
> >>>>> JACK, without the sound cards being involved, there's a loss for the
> >>>>> sound quality too!
> >>>>
> >>>> Wait a minute... could you explain that? You have a loss of quality compared to live playback of the soft synths (using JACK?) when playing back a recording taken from JACK? A recording that preserves 32 bit floating point sample format (heck, or 24 bit integer) and the sample rate, of course?
> >>>
> >>> Yes and other people who can't hear it, do have it too.
> >>
> >> I do not.
> >>
> >>> You can see it
> >>> by watching the waves spectral by Audacity. I did this regarding to a
> >>> zero-copy issue, that appears if a Jack client is connected directly to
> >>> itself, e.g. to do the mastering. 48 and 96 KHz, 32-bit wav 32-bit
> >>> float.
> >>
> >> If a synth has dynamic filters it will never produce the exactly same
> >> stream twice. But if you think about yourself you will find out, that
> >> given you use the same settings for Jack on a HDA or a HDSP you will get
> >> exactly the same quality.
> >
> > I'm an expert for audio engineering. I did work for Brauner microphones
> > development and others, hence I know a little bit about how to do
> > tests ;).
> > No dynamic filters are involved!
> >
> > It's very simple, there's a natural sounding drum set as example drum
> > kit for Hydrogen. Play a rhythm, record this Rhythm and then record this
> > recording and compare both recordings. They should be equal, but they
> > aren't equal. I can here a !clear! loss and it's visible by spectral
> > waves.
> >
> >>
> >> Simply because a synth-software only delivers, what it renders to Jack
> >> and Jack does *not* change anything in that rendered data. There is
> >> simply not soundcard and not even a driver involved in the rendering
> >> itself. DSPs only do the very same thing faster as cheap chips.
> >>
> >> All difference in sound quality is related to DAC/ADC period
> >
> > No! Before any converter is involved, there at least could be rounding
> > errors, if you don't use 32-bit float all the times.
>
> And why should I not use 32bit float all the time?
>
> Of course there are differences, if format-conversion is involved. But
> you did not mention such conversions, you only talked about sound cards
> causing mysterious differences when Jack delivers a stream from a
> synth-application directly to a recorder.

No, a misunderstanding.

1. Yes my sound card is bad, but ...
2. I was writing about using 32-bit float only and use audio streams
internal Jack only, without the sound cards being involved.

>
> The normal, sane setup fpr recording a synth directly with Jack is, that
> the synth, Jack and the recorder all run with the same samplerate and
> 32bit float or at least all 3 with 16bit Int. And if that is set up like
> this, there is zero influence of the soundcard on the recording.

Correct, there's no influence of the sound cards, but on my machine
there's loss, even when the sound cards aren't involved, just by
recording a soft synth. Everything is set to the same sample rate, 96
KHz or 48 KHz and 32-bit float ... as far as I know ... I don't know if
e.g. Yoshimi does use 32-bit float.

>
> >
> > And by the way, the sound card will effect the original and the digital
> > copy in the same way, even with a bad sound card both recordings
> > shouldn't differ.
> >
> > Hey, do a recording of a recording
>
> You mean, like recording something from ams via jack then play the
> wav-file with mhw and record this with ardour. Then compare the two
> recorded streams?
>
> If in such a process recording A would differ from recording B then MHW
> or Ardour *could* cause such a difference. Jack itself could only be
> charged, if resampling and/or dithering would be involved. That is: if
> the synth-engine would work with 44.1Hz while Jack runs with 48Hz. And
> you will not want to set up a synth like this.
>
> But you said, that there would be a difference between a 1st-level
> recording from a synth via Jack with sound card A and another recording
> of the same synth with soundcard B.
>
> And that is not the case.
>
> > and then run the diff command to
> > compare them ;)!
> >
> >>
> >>>
> >>>> I have to wonder what you did there to alter the data from the soft synth.
>
> Format conversion?
> Some synths (like ZynaddSubFX) can be configured to produce streams in a
> certain format (such as 44.1 Hz / 16bit Int) and still run with Jack
> that runs with different settings.
> Every body sets Zynadd to run with the same SR as Jack and Zynadd
> recommends that if started from the command line.
>
>
> >>> I mean ... we're talking bit-exact copy here, aren't we? Can you present a test setup to observe that issue?
> >>>
> >>> Any Linux install I know, e.g. 64 Studio 64-bit 3.0, 3.3, Suse 64-bit
> >>> 11.2 and Edubuntu 32-bit Maverick + Ubuntu Studio meta packages and
> >>> others! If you can't here it, try to see it. If you don't have this
> >>> issue too, some people claim that they get 100% correct digital copies,
> >>> then something on my machine might cause a software issue, but I don't
> >>> think so.
> >>>
> >>> Ralf
> >>>
> >>>>
> >>>> Alrighty then,
> >>>>
> >>>> Thomas.
> >>>
> >>>
> >>
> >>
> >
> >
> >
>
>



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Ralf 05-21-2011 12:04 AM

Sound cards (Was: no sound
 
On Fri, 2011-05-20 at 10:54 -0400, Mike Holstein wrote:
> Ralph,
>
>
> the sound card *cant* have anything to do with the audio that is
> generated by software synths.

Again, a misunderstanding, perhaps regarding to my broken English.
There're two issues, one is caused by my sound cards, they do produce a
muddy bass, the other issue has nothing to do with my sound cards. A
recording internal Linux, without the sound cards involved does cause
loss here. Audible loss, you don't need trained or good ears to hear it.

> this is not a grey area. and expert such as yourself might want to
> look at the source code for the software in question

I can't read C and much more worse, never coded for modern PCs.

> , and see what about the hardware is being utilized for rendering the
> audio. heres a test scenario: take all the sound devices out of the
> machine (or disable them). take a MIDI file and render it using JACK
> utilizing the 'dummy' driver. you can render the same file with a
> sound card in use, and share both of those here if you would like. *i
> am not talking about monitoring those sounds, OR recording them analog
> from the main outs of the sound card. by render, im thinking recording
> in ardour and exporting or exporting from something else.

No need to do this, because ... again ... the sound cards already aren't
involved. This was a misunderstanding.

Btw. doing the mastering for a "real song" might not work by an
exporting option. At least Qtractor can't do this for "faster than
real-time" included apps, in other words, DSSI plugins and especially
not for inserts, e.g. jconvolver.

That I wish to buy a good sound card has nothing to do with the loss
internal Linux on my machine (and I suspect on many other machines too,
at least I know some people who do have the same issue).
>
>
>
> --
> MH
>
> http://opensourcemusician.libsyn.com/
> http://wnclug.ourproject.org/
>
>



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Ralf 05-21-2011 01:09 AM

Sound cards (Was: no sound
 
On Sat, 2011-05-21 at 00:30 +0200, Thomas Orgis wrote:
> Am Fri, 20 May 2011 15:33:39 +0200
> schrieb Ralf Mardorf <ralf.mardorf@alice-dsl.net>:
>
> > No! But I suspect a bug for Jack and btw. there already was a rounding
> > bug that was fixed, perhaps years ago.
>
> Ah, OK. I'll ignore the discussion about different sound cards influencing this as a heap of confusion and settle for this: If JACK would not have bugs, by design it would do bit-exact copies, or at least copies with floating point computation errors in non-audible amplitude. I don't claim that this ideal case is indeed the reality.
>
> When you find those bugs, I hope they'll get reported and fixed, for the benefit to us all. In the meantime, those of us with some time on their hands can just try to verify the issue of bit-exact copying of data through the pipeline ... I just don't want to go down that path right now, since I'm already not able to do my actual work because of debugging some gcc 4.6 breakage on code of mine.
>
>
> Alrighty then,
>
> Thomas.
>
>

Is there a way of monitoring the data that is captured and read by Jack
and Jack clients in a human readable way, e.g. to see if there would be
rounding errors? I always used Audacity's spectral view (to show
recordings of square and sine waves), when writing with Rui. For
Qtractor there really was and is an issue when connecting it's outputs
to the inputs, that is caused by Jack.

Phew, a lot of OT blah blah, you don't need to read does follow, pardon:

(That's why I wish to stop the discussion about the audio quality loss
issue.)

I guess it's because a vector points to the original buffer, instead of
coping it, but I might be completely wrong. Anyway, if a client's input
is connected to it's output this only should work correctly, if there's
a special order.
Here I did avoid to do this and there still is loss, similar to cheap
4-Track cassette recorders. If I directly connect Hydrogen or Yoshimi to
Qtractor or Ardour2, that does mean one client to _another_ client,
there's this kind of audible loss and this shouldn't happen :(.

I can't help with reading the source code. When I programmed in C 20
years ago, I just did it for one program and then switched back to
Assembler. On Linux I tried to learn C again, but I wasn't able to write
even simplest make files, IIRC 20 years ago it was the work, the
compiler had to do ;), I could take a look on my Atari ST's 80286
hardware emulation, there still should be the old editor and compiler.
So, I'm not a coder ;). I'm an audio/video engineer and the computers I
privately programmed have less in common with current PCs, e.g. the
Atari's TOS and the emulation running DR DOS, are sharing a 42MB hard
disk. Especially regarding to music software, there was the advantage
that those machines don't do real multitasking and the hardware, e.g.
for the C64 MIDI interface, was directly accessible, that's why there's
no MIDI jitter for old hardware. Turn of the interrupt, check the
ACIA/UART and send in real-time, that's how it did work years ago, today
we do have USB protocols etc. that at all events do cause MIDI jitter.
Some smart guys, I guess Stéphane is one of them, do work on this issue
and they/he already had success, at least for my machine. If I use PCI
MIDI and

edubuntu@edubuntu:~$ jackd -V
jackdmp 1.9.8

using the -Xalsarawmidi switch + a2jmidi_bridge I'm able to use external
MIDI equipment without audible jitter, unfortunately I need to disable
-Xalsarawmidi, if I wish to record MIDI events with a Linux sequencer,
because there's no bridge vice versa and as far as I know no good
sequencer using Jack MIDI, resp. I guess Ardour3 might use Jack MIDI.
Btw. if I run latency tests I also get best results for Alsa, but at
least I'm able to hear that there's jitter, a lot of people with less
good results aren't able to hear jitter.

There're a lot of issues for Linux audio, some are already solved by svn
versions, other issues aren't solved. It makes me wonder that there are
so less people in the Linux community who notice those issues.

I can't program myself, but I could run tests and report issues.

Linux audio and MIDI IMO do need a lot of upgrades and some isolation
from the non-audio community.



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