Hello,
Let me point out in advance that my problem is pretty trivial compared to
most of the postings on that list. If a journey in sound engineering on
Linux is 1000 steps, I'm at step 1 here. I basically tried configuring sound
into the kernel but it didn't seem to work. If you think you can help,
please read on. Otherwise, stop here.
And thanks to all who continue from here...
I want to build sound support into my embedded Linux platform (2.6.28); the
CPU supports I2S in silicon, but I haven't got a driver for that yet. My
plan is to build the kernel with SOUND/SND configurations enabled and start
working on the driver later. In my configurations, I've left all the sound
card-specific options out, keeping just the CONFIG_SND_SOC. I thought it
would be sufficient to install snd-dummy so that any cross-compiled audio
applications could be fooled into working. l rebuilt the kernel after *make
clean* and noted several new SND files being compiled. So far so good. The
final zImage was only a little larger. Overall, the configuration and build
seemed to be going through all the right motions, but when I loaded zImage
onto the target I didn't see any bootup logging that suggested audio was now
working. As an experiment, I tried to insmod snd_dummy.ko but got a long
list of errors. I guess I'm missing something basic.
Here's my configuration choices:
> CONFIG_SOUND=m
> CONFIG_SOUND_OSS_CORE=y
> CONFIG_SND=m
> CONFIG_SND_TIMER=m
> CONFIG_SND_PCM=m
> # CONFIG_SND_SEQUENCER is not set
> CONFIG_SND_OSSEMUL=y
> CONFIG_SND_MIXER_OSS=m
> CONFIG_SND_PCM_OSS=m
> CONFIG_SND_PCM_OSS_PLUGINS=y
> # CONFIG_SND_DYNAMIC_MINORS is not set
> CONFIG_SND_SUPPORT_OLD_API=y
> CONFIG_SND_VERBOSE_PROCFS=y
> # CONFIG_SND_VERBOSE_PRINTK is not set
> # CONFIG_SND_DEBUG is not set
> CONFIG_SND_DRIVERS=y
> # CONFIG_SND_PCSP is not set
> CONFIG_SND_DUMMY=m
> # CONFIG_SND_MTPAV is not set
> # CONFIG_SND_MTS64 is not set
> # CONFIG_SND_SERIAL_U16550 is not set
> # CONFIG_SND_MPU401 is not set
> # CONFIG_SND_PORTMAN2X4 is not set
> # CONFIG_SND_ISA is not set
> # CONFIG_SND_PCI is not set
> # CONFIG_SND_SPI is not set
> # CONFIG_SND_USB is not set
> # CONFIG_SND_PCMCIA is not set
> CONFIG_SND_SOC=m
> CONFIG_SND_SOC_ALL_CODECS=m
> CONFIG_SND_SOC_AD73311=m
> CONFIG_SND_SOC_AK4535=m
> CONFIG_SND_SOC_CS4270=m
> CONFIG_SND_SOC_SSM2602=m
> CONFIG_SND_SOC_TLV320AIC23=m
> CONFIG_SND_SOC_TLV320AIC26=m
> CONFIG_SND_SOC_TLV320AIC3X=m
> CONFIG_SND_SOC_UDA1380=m
> CONFIG_SND_SOC_WM8510=m
> CONFIG_SND_SOC_WM8580=m
> CONFIG_SND_SOC_WM8731=m
> CONFIG_SND_SOC_WM8750=m
> CONFIG_SND_SOC_WM8753=m
> CONFIG_SND_SOC_WM8900=m
> CONFIG_SND_SOC_WM8903=m
> CONFIG_SND_SOC_WM8971=m
> CONFIG_SND_SOC_WM8990=m
> # CONFIG_SOUND_PRIME is not set
>
And here is the list of errors from installing snd-dummy.ko
[root@127 drivers]#insmod snd-dummy.ko
Using snd-dummy.ko
snd_dummy: Unknown symbol snd_pcm_lib_free_pages
snd_dummy: Unknown symbol snd_pcm_set_ops
snd_dummy: Unknown symbol snd_ctl_boolean_stereo_info
snd_dummy: Unknown symbol snd_card_new
snd_dummy: Unknown symbol snd_pcm_lib_ioctl
snd_dummy: Unknown symbol snd_pcm_lib_malloc_pages
snd_dummy: Unknown symbol snd_pcm_lib_preallocate_pages_for_all
snd_dummy: Unknown symbol snd_card_free
snd_dummy: Unknown symbol snd_ctl_add
snd_dummy: Unknown symbol snd_pcm_new
snd_dummy: Unknown symbol snd_pcm_suspend_all
snd_dummy: Unknown symbol snd_pcm_period_elapsed
snd_dummy: Unknown symbol snd_card_register
snd_dummy: Unknown symbol snd_pcm_format_set_silence
snd_dummy: Unknown symbol snd_pcm_format_width
snd_dummy: Unknown symbol snd_ctl_new1
insmod: cannot insert `snd-dummy.ko': Unknown symbol in module (-1): No such
fil
e or directory
Best Regards,
Rory
Drumstick is a C++ wrapper around the ALSA library sequencer interface using
Qt4 objects, idioms and style. ALSA sequencer provides software support for
MIDI technology on Linux. Complementary classes for SMF, WRK and OVE file
processing are also included. This library is used in KMetronome, KMidimon and
KMid2, and was formerly known as "aseqmm".
Changes:
* OVE file format support, contributed by Rui Fan
* Optional RealtimeKit support for MIDI input thread
* guiplayer simplified and optimized, with OVE format playback
* Build system fixes: using visibility=hidden if it is available,
exceptions, static build.
Copyright (C) 2009-2010, Pedro Lopez-Cabanillas
License: GPL v2 or later
Project web site
http://sourceforge.net/projects/drumstick
Online documentation
http://drumstick.sourceforge.net/docs/
Downloads
http://sourceforge.net/projects/drumstick/files/
I was going through Paul Davis's excellent tutorial on the
ALSA API:
http://www.equalarea.com/paul/alsa-audio.html#interruptex
I've set alsa to wake me up every N frames. However, when I
awake, I find that I often have fewer than N frames
available:
int err;
uint32_t N = 2048, frames_to_deliver;
assert( 0 == ((N-1)&N) ); /* power of 2 */
/*...*/
snd_pcm_sw_params_set_avail_min (playback_handle, sw_params, N)
/* ... */
while(1) {
err = snd_pcm_wait(playback_handle, 1000);
assert(err >= 0); /* OK */
frames_to_deliver = snd_pcm_avail_update(playback_handle);
assert(frames_to_deliver >= 0); /* OK */
/* this fails about every other cycle: */
assert(frames_to_deliver >= N);
}
E.g. with N = 2048, I'll typically get a stream that looks
like 1996, 2048, 2048, 1990, 2048, 1999, 2048...
Any ideas?
-gabriel
On Thursday 09 September 2010 00:32:18 Olivier Guilyardi wrote:
> On 09/08/2010 11:37 PM, Arnold Krille wrote:
> > http://positron.physik.uni-halle.de/~arnold/fadercolors-system_colors-won
> > ton.png
> > http://positron.physik.uni-halle.de/~arnold/fadercolors-system_colors-ob
> > sidian.png
>
> Sorry but wouldn't it be clearer with tiny level bars, instead of colors?
>
> http://testing.samalyse.com/lad/widgets/numbers.png
> http://testing.samalyse.com/lad/widgets/numbers.svg
Hm.
No. The left-to-right level-bars imply a left-to-right movement to change the
values.
And one of the reasons to do the color-thing is space. If I hide the text and
only show it when the mouse hovers, I can fit a lot of mixing-nodes on screen.
A horizontal (or vertical) bars needs more space to have a comparable
resolution.
I like the idea of hiding the mouse during value-changing moves. I already
have these "sliders" react to the mouse-movement logarithmic. That also helps
a lot when both coarse and fine changes should be possible with similar mouse-
movement.
And somehow I dream of a toolkit where you select a knob/slider/fader and it
then reacts to the turn-knob you have on your desk. But given the variety in
toolkits, I think this is a dream of the distant future unless that big knob
sends out mouse-movement signals...
Have fun,
Arnold
So far, nobody's mentioned how to get to the aforesaid knob in the
first place, or alternatives
to manipulating it.
Navigating with the mouse is a poor second to precise locating, and
using qwerty actions to set values, be it a knob or slider, is more
accurate.
So, tab, or an associated action to get to the knob in the first
place, and something like ALT+ up/down arrows (as an example) to change the knob
value. Then it's irrelevant whether the knob responds to mouse
up/down/left/right, or mouse wheel, for users who find those options
second best, when a more efficient and precise option already exists.
For graduated speed in the case of a spin knob (starting slow and
getting faster), the user only needs to
keep the actions pressed.
Alex.
--
www.openoctave.org
midi-subscribe(a)openoctave.org
development-subscribe(a)openoctave.org
Hello
EMutrix is a simple, easy-to-use graphical mixer application for EMU
1010 based cards like the E-MU 1212m, E-Mu 1616m and E-MU 1820 models.
I am releasing the first source package of EMutrix, version 0.1 at
http://emutrix.googlecode.com
It is mostly functional, allowing arbitrary routing between the card's
multiple inputs, outputs and ALSA, setting clock rate and pads. More
features to come in future versions. Try it out if you have an E-Mu
card!
Any comments welcome.
Greetings,
Camilo
After many month of development, we are proud to announce the version
0.6 beta of jMax Phoenix.
The major highlights for this version are:
- A first version of the puredata source compatibility kit, including
the build system and a full example of recompiled object library.
- A large set of usability bug fixed; all the bugs preventing a
smooth work flow have been fixed.
- Error handling improvements: most of the bugs and configuration
errors now results in error messages, and not unexplained freezes.
- A set of examples and tutorials has been recovered from old ISPW
archives; they are not updated to include all the major jMax
functionalities, but it is better than nothing.
Full release notes are available in the release notes section of the
projet site.
This release has been tested on Ubuntu and Ubuntu Studio 10.04, Debian
5, Fedora 13 and Mandriva Spring 2010; check the installation
instruction on the projet site for specific caveats for Debian and Fedora.
The Puredata compatibility sub-project has been *very* time consuming;
in order to better manage my scarce time resources,
we need some user feedback (and possibly help) to be able to evaluate
the actual interest of pursuing this development direction.
For more information and download and installation instructions go to
http://www.jmax-phoenix.org/.
For contacting the project team: contact(a)jmax-phoenix.org
The jMax Phoenix team
Thank you very much for your advice everyone.
As mentioned I have already made the LADSPA plugin and tested it with PulseAudio, so I will take a look at the PulseAudio LADSPA plugin module loader and see if I can add the functionality I need (multichannel audio and detection of when headphones are plugged in). Then I will see how it goes from there.
/Kim
Hi,
I want to make is a DSP module that improves the sound quality
of the sound coming from the laptop builtin speakers by applying DSP
(FIR/IIR filters).
I want to route all audio played back (from for
instance an offline WAV or MP3 file, from a movie, or streamed from the internet) to
pass through the
DSP plugin whenever headphones are NOT connected. When headphones are connected no processing should be applied.
I
want to place the DSP module as close to the hardware as possible to
make sure all audio is really routed through the DSP plugin so I always
can enjoy
the DSP improvements made to the sound.
Where should I place this module/DSP code?
- As a plugin to PulseAudio?
- As a plugin to ALSA?
- As its own virtual sound card?
- In the audio driver for my built-in sound card?
- As a kernel module?
I have tried loading the LADSPA-module using the module-ladspa-sink in PulseAudio but I am not sure this is the best solution.
I need to remove some limitations in module-ladspa-sink to get it working properly ( multichannel audio, only DSP processing for internal speakers).
My requirements:
1) Must be able to detect if headphones are connected
2) Must be able to process stereo and multichannel (5.1/7.1) formats. I
need the multichannel formats to perform binaural downmixing to
stereo.
So it is important that I can receive multichannel audio and mix it down to stereo and output it to a stereo soundcard.
/Kim
Alex, I hope you don't mind quoting part of your message on the list,
as the subject is of general interest.
Alex stone wrote offlist:
> Fons, i've read in the mailing list in the thread concerning mixers,
> that you have something you're working on.
> ...
> More than any other facet of working in software for audio/midi/vid,
> i've been frustrated and tired out by poor navigation options, and i
> don't think i'm on my own.
> The mouse is insufficent for ease, speed, and pain free day in day out
> use, and i ask you to at least consider (or not, as you decide)
> alternatives such as the one above, to make a difference in the way
> users commonly interact with this type of software.
I more than agree with this. Defining the way a user interacts with a
large software mixer is 90% of the work of writing one. Common GUI
methods just don't work for this sort of thing.
The thing that is slowly taking some form here will observe some
GUI guidelines quite strictly, in particular there will be *NO*
repeat *NO* resizing, scrolling or moving of windows, menus will
be used sparingly and be small, and popup dialogs will be used
only for operations that can be assumed to be infrequent (mainly
setting up things, selecting and connecting plugins, etc.).
When setting up a new session, you define the resources:
- number of channel strips
- number of real groups
- number of control groups
- number of visible sections
- number of strips per section
- number of jack/alsa inputs
- number of jack/alsa outputs
These are more or less fixed afterwards, in the sense that you
can't remove any, but probably I can allow to increase these
numbers (the point here is that all snapshots taken for such
a configuration must remain valid without requiring changes
to the resources). Anyway there is no penalty (apart from a
small amount of memory used) for providing more than you need.
A _section_ is a group of strips, typically 8, 10 or 12.
Normally you should select the section size so you can have
two of them side by side, as this facilitates many operations.
For each visible section there is a control strip, which
contains e.g. buttons to select which strips are shown.
You will have default sections, e.g. inputs 1-8, 9-16, 17-24,
etc, same for real groups and control groups. There are also
user-defined sections: these can contain copies of any strip
you want in any order you want. For example you could group
all inputs used for drums in a secion, or have a 'master'
section that contains mainly groups and maybe some solo mics.
There is some 'intelligence' in the section selection buttons,
for example, if you select to see inputs 17-24 in the right
section, you click that button (or use a kb shortcut), then
clicking again restores the previous view. Or selecting a
section that is already visible will swap the two, etc.etc.
Each visible section is vertically divided in two, the bottom
half showing the faders, channel ID, mute, solo etc. most of
the time, while the top half can show parts of the strips, e.g.
EQ, sends or dynamics. Again this layout is configurable to some
extent, you can e.g. select to have some aux sends visible all
the time. The top part of each visible section can also be used
for plugins, metering, 'visual' panning, or whatever extra
functions that would need to be displayed.
All Jack/Alsa ports are generic, each of them can be assigned
to any function inside the mixer. That means e.g. that Jack
connections can remain fixed no matter how you 'rewire' the
mixer, and this in turn means that nothing is lost if you
connect directly to a multichannel Alsa device, e.g. a MADI
card.
The mixer will have its own plugin interface, and it will not
accept any of the existing standards. If someone wants to use
a LADSPA or LV2 I will consider porting it. Condition for this
will be the *quality* of the plugin, and nothing else.
Ciao from very hot Crete (I'll be cooling down 20m below the
water surface in a short time).
--
FA
There are three of them, and Alleline.