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hi all,
what's the proper way to list all alsa-devices, including those
(plugin, slave,...) defined in asoundrc?
e.g. on my desktop i have
$ cat ~/.asoundrc
pcm.jack {
type jack
playback_ports {
0
1
}
capture_ports {
0
1
}
}
$ cat /proc/asound/cards
0 [MID ]: HDA-Intel - HDA Intel MID
HDA Intel MID at 0xf4ff8000 irq 42
1 [Bt878 ]: Bt87x - Brooktree Bt878
Brooktree Bt878 at 0xf3fff000, irq 17
but aplay will show me:
<snip>
$ LANG=C aplay -L -l
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
sysdefault:CARD=MID
HDA Intel MID, VT1828S Analog
Default Audio Device
front:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
Front speakers
surround40:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
hdmi:CARD=MID,DEV=0
HDA Intel MID, VT1828S Digital
HDMI Audio Output
**** List of PLAYBACK Hardware Devices ****
card 0: MID [HDA Intel MID], device 0: VT1828S Analog [VT1828S Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: MID [HDA Intel MID], device 2: VT1828S Alt Analog [VT1828S Alt
Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: MID [HDA Intel MID], device 3: VT1828S Digital [VT1828S Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
</snip>
there is no mention of my 'jack' pcm, *and* there are devices that i
don't know where they come from: namely 'null' and 'pulse' (pulseaudio
is installed but not running, as confirmed by
<snip>
$ pgrep pulse
$
</snip>
so i figure that the extra devices come from system-configuration
files in /usr/share/alsa/alsa.conf(.d/*.conf)
but why does my "jack" PCM not show up?
i know that i can extend my asoundrc to:
<snip>
pcm.jack {
type jack
playback_ports {
0
1
}
capture_ports {
0
1
}
hint {
description "use jack as alsa-device"
}
}
</snip>
but i don't fully understand why i have to manually add the "hint" (of
which i haven't found *any* documentation) in order to make the device
show up with
char**hints=NULL;
snd_device_name_hint(-1, "pcm", (void***)&hints);
any clues?
fgasmdr
IOhannes
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Hi guys,
As those interested know the raspberry pi lacks an audio input. Now, I
came across this some days ago:
http://www.noiseisgood.co.nz/?p=365
Apparently it is possible to wire an i2c adc/dac to the raspberry pi
and utilize some already available alsa driver.
Just thought I would share and ask if anyone has tried anything similar?
Regards,
Robert
Hi there,
I'm looking for a library that I could use to categorise audio.
Relatively simple categories would be sufficient, for example silence,
music, speech, noise.
It would be preferable if I could use it from a general purpose
language over a specialised one (rather java or C than pd).
Do you know or even have experience with such a library?
Thanks in advance,
Philipp
>> problem is get_supplied_options() returns 0 options when called from the UI.
>>
> get_supplied_options() should work fine both in UIs and Plugins. My guess
> is the host only gives the sampleRate option to the Plugin and not the UI.
> Which host are you working with?
I'm using Ingen - I sent a ticket to David Robillard onto
http://dev.drobilla.net/timeline and will wait for his advise.
In the meant time I'll try and debug Ingen itself and see if I can
create a patch.
> Also, you can compile the UI with a -DLVTK_DEBUG=1, and the console will
> print the validation status for any features passed in by the host.
>
> You could also from the UI send a patch get message and make sure the
> Plugin knows how to respond appropriately.
Thanks for the ideas - I'll give those a try.
Aurélien
On Mon, October 14, 2013 4:08 am, Markus Seeber wrote:
> Hi,
> i'm not sure, but you could have a look at what he is doing:
>
> http://mohayonao.github.io/timbre.js/reverb.html
>
> That is quite a large JavaScript framework, but maybe the right place to
> start?
>
Looks interesting but it doesn't have support for ogg afaict. However I
might be able to do something with the examples.
Anyone else have any other suggestions?
>
> Am 10/13/2013 06:58 PM, schrieb Patrick Shirkey:
>> Hi,
>>
>> Can anyone point me to an example for echo/delay/reverb filters using
>> html5 audio?
>>
>>
--
Patrick Shirkey
Boost Hardware Ltd
Hi all,
This is a call for testing my ALSA driver for Fireworks/BeBoB based devices.
Please test 'snd-fireworks' for Fireworks and 'snd-bebob' for BeBoB if
you have some devices listed in the end of this mail.
Status:
- still under development
- Without snd-dice and Clemens' development (I must do this later)
Functionality:
- playback/capturing (full duplex) with PCM/MIDI interface
- hardware metering for some devices with CONTROL interface
- switching clock source/digital interface/digital mode with CONTROL
interface
- print hardware status with PROC interface
Note:
- Don't use simultaneously 'ALSA PCM/MIDI playback/capture' and 'jackd
with Firewire (FFADO) backend'. Both of them try connecting to the
device when another is running.
- I add much modification into snd-firewire-lib for full duplex
synchronization of receive/transmit AMDTP stream.
Requirement:
- Linux kernel 3.11 or later because of Juju (nickname of Firewire
stack) changing its API.
- Dynamic Kernel Module Support (DKMS) is reccomended for safely
installing/uninstalling
(I work with Ubuntu 13.10)
Bug report:
- report with /proc/asound/cardX/#XXX
- please send your experiences to me with the output
How to install (DKMS):
1. $ git clone https://github.com/takaswie/snd-firewire-improve.git
2. $ ln -s $(pwd)/snd-firewire-improve/ /usr/src/alsa-firewire-3.11
(superuser)
3. $ dkms install snd-firewire/3.11 (superuser)
How to uninstall (DKMS):
1. $ modprobe -r snd-bebob snd-fireworks snd-firewire-lib (superuser)
2. $ dkms remove ans-firewire/3.11 --all (superuser)
3. $ rm /usr/src/alsa-firewire-3.11 (superuser)
4. $ rm snd-firewire-improve
How to install (Manual):
1. $ git clone https://github.com/takaswie/snd-firewire-improve.git
2. $ cd snd-firewire-improve
3. $ make
4, backup system snd-firewire-lib/snd-firewire-speakers/snd-isight
(superuser)
5. install
snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob (superuser)
6. depmod -a (superuser)
How to uninstall (Manual)
1. modprobe -r snd-firewire-lib snd-firewire-speakers snd-isight
snd-fireworks snd-bebob (superuser)
2. remove
snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob (superuser)
3. recover snd-firewire-lib/snd-firewire-speakers/snd-isight (superuser)
4. depmod -a (superuser)
Confirmed to work:
- AudioFire4
- AudioFirePre8
- Ozonic
- Firewire Solo
- Firewire Audiophile
- Firewire 410
== Fireworks based devices
[Echo Audio]
AudioFire2
AudioFire4
AudioFirePre8
AudioFire8 (till 2009)
AudioFire8 (since 2009)
AudioFire12
[Gibson]
RIP
[Mackie]
Onyx 400F
Onyx 1200F
== BeBoB based devices
[Yamaha]
GO44
GO46
[M-Audio]
(to control mixer channels please use FFADO upstream)
Ozonic
Firewire 410
Firewire Audiophile
Firewire Solo
NRV10
ProFireLightbridge
[Focusrite]
SaffirePro 26 I/O
SaffirePro 10 I/O
Saffire(LE)
[Edirol]
FA-66
FA-101
[TerraTecElectronic GmbH]
Phase88FW
PhaseX24FW
[PreSonus]
FireBox
FirePod
[Mackie]
OnyxFirewire
[Tascam]
IF-FW/DM
[Behringer]
X32
[ApogeeElectronics]
Rosetta200
[ESI]
Quatafire610
Regards
Takashi Sakamoto
o-takashi(a)sakamocchi.jp
> I would just bypass the LV2 host altogether and use shared memory to get
> the audio data to the GUI.
> ...and use the same mechanism to send the sample rate.
That's through Instance-Access right?
That's exactly what I want to avoid - this plugin is meant to be used
in Ingen and Ingen doesn't support this.
> LV2 UIs can get the sample-rate using the options extension.
> Hosts will set the appropriate option lv2param:sampleRate which the UI
> can get on initialization.
I tried that but without success.
I'm using the LVTK library to develop this plugin. My approach is to:
- use the get_supplied_options() method to access the host supplied options
- iterate until I find SampleRate
problem is get_supplied_options() returns 0 options when called from the UI.
Any advice?
Aurélien
Hello community. First, have to note, that i'm not sure, what fluidsynth
backend is best today, as first time i got it, was several years ago when i
tried ubuntustudio (even simpler - just installed pro-media software to Ubuntu
7.04) and when ubuntu 10.04 released and even 12.04, i still suppose using of
qsynth, since i don't now others with such full amount of settings.
Now proposals.
1 - optional feature (i.e. with switch to disable it) to publish fluidsynth
engines as dbus or other kind of services, exactly like what jack already has.
If such option is enabled by default, then even when engine is loaded via
linked library, as done in MuseScore, it still may be controlled by external
gui like QSynth. Also, another example of such way is linuxsampler way
(limited, because qsampler only allowes to change backend addess in settings,
but hope it is to be fixed).
2 - only about gui at all, not only qsynth: i read one time somewhere, that
fluidsynth supports microtonality (scale tuning), but only few months ago could
try it in action, controlling manually standalone fluidsynth, started in
terminal. It would be great to add in qsynth panel like in zyn/yoshimi for
scale tuning. Also interested, is there some gui, where scale tuning is
implemented (hard even to hope when even most featureful gui doesn't have it).
