Hi Developers.
I have more as one sound players on my system mplayer, vlc, qmmp ...
Each of em uses jack output , for example 2 independent instances of
mplayer
mplayer -ao jack  -srate 16000
'/home/alf/mp3/mettwoch-de-doof-nuss.mpa.mp2'
mplayer -ao jack  -srate 16000 '/home/alf/mp3/effeckt006.mp3'
So far everything is excellent.
How i do write my own DSP plugin for jack ?
I wanna use my DSP plugin at this point, where all input channels are
mixed together .
How my plugin can ioctl() detect current samplerate,
endianess, nr channels,
samples interleaved or not, ... and maybe some more stream
parameters.
How do compile jack plugins ?
What #include-s  must be used ?
If my plugin will grow, and eat more and more CPU, how i prevent
x-runs ?
Is somewhere _simple_ C example how i can write jack plugins ?
In simplest case -
what is minimal program where i do each sample divide by 2
( or for 16bit short type just  shift  sample>>1 )
Tnx in advance.
Alf
----
FYI, here's an example of the kind of app that needs to have good audio
performance on a handset:
http://www.youtube.com/watch?v=bwqflVX5oNohttp://www.warmplace.ru/soft/sunvox/
( http://lists.linuxaudio.org/pipermail/linux-audio-user/2010-December/074828…
)
It has decent performance on maemo and appears to use pulseaudio,
which eats 1/3 of the CPU of the 'sunvox' process. The sunvox
application appears to have a UI thread and a worker thread each
consuming about 1/2 of the 35% CPU load of the app.
Mem: 238432K used, 7108K free, 0K shrd, 3396K buff, 67580K cached
CPU: 52.2% usr 7.8% sys 0.0% nice 39.8% idle 0.0% io 0.0% irq 0.0% softirq
Load average: 0.99 0.45 0.16
PID PPID USER STAT RSS %MEM %CPU COMMAND
1916 1162 user S 6388 2.5 35.1 /usr/bin/sunvox
825 1 pulse R < 3812 1.5 12.1 /usr/bin/pulseaudio --system
--high-priority
897 730 root S < 16524 6.7 9.6 /usr/bin/Xorg -logfile
/tmp/Xorg.0.log -logverbose 1 -nolisten tcp -noreset -s 0 -core
Doing an "ls -lR /" in a remote xterm (over SSH) results in some audio
glitching, but no "desynchronization" where the audio just stops
playing.
-- Niels.
On 2/3/2011 1:14 PM, Stefan Kost wrote:
> On 02/03/2011 09:27 PM, Bearcat M. Sandor wrote:
>> On 2/2/2011 2:43 PM, Stefan Kost wrote:
>>> Am 16.01.2011 17:42, schrieb Harry Van Haaren:
>>>> Hey guys,
>>>>
>>>> I'm looking for the "lowest-common-denominator" of audio file
>>>> formats that
>>>> handle BPM info.
>>> mp3, wav, vorbis, mp4, mkv files can have BPM metadata (according to
>>> my grep in
>>> the gstreamer source code). GStreamer has a bpm detector as well.
>>>
>>> Stefan
>>>
>> What? No love for my favorite, wavpack? Wavpack never gets any
>> respect! :"(
>>
>> Bearcat M. Sandor
> Erm, it should work already. From the wavpack homepage:
> Uses ID3v1 and APEv2 tags for metadata (including ReplayGain)
> Both are well supported by gstreamer. :)
>
> Stefan
I got it working. I didn't realize that the gst-plugins-soundtouch
plug-in package did not exist in Gentoo as part of the gst-plugins-bad
package. One that was installed i was able to get it all working.
Bearcat M. Sandor
On 2/3/2011 1:14 PM, Stefan Kost wrote:
> On 02/03/2011 09:27 PM, Bearcat M. Sandor wrote:
>> On 2/2/2011 2:43 PM, Stefan Kost wrote:
>>> Am 16.01.2011 17:42, schrieb Harry Van Haaren:
>>>> Hey guys,
>>>>
>>>> I'm looking for the "lowest-common-denominator" of audio file
>>>> formats that
>>>> handle BPM info.
>>> mp3, wav, vorbis, mp4, mkv files can have BPM metadata (according to
>>> my grep in
>>> the gstreamer source code). GStreamer has a bpm detector as well.
>>>
>>> Stefan
>>>
>> What? No love for my favorite, wavpack? Wavpack never gets any
>> respect! :"(
>>
>> Bearcat M. Sandor
> Erm, it should work already. From the wavpack homepage:
> Uses ID3v1 and APEv2 tags for metadata (including ReplayGain)
> Both are well supported by gstreamer. :)
>
> Stefan
Thank you Stefan,
The bpm detection, tagging is not working in banshee. I'll do some
sniffing around and see why that might be.
Thanks,
Bearcat M. Sandor
So I start jackd and it runs fine for a while. Then ALSA crashes and
triggers an xrun. Jackd continues to do its merry thing - minus the part
about spitting info out to my soundcard...
I've tried multiple usb intefaces. All work fine with other computers. I've
tried multiple version of Ubuntu with various versions of ALSA. Only this
machine is affected. I have 3 other machines that run these interfaces just
fine... So it seems to be a hardware compatibility issue not a software
glitch in one version.
I've tried many boot flags but to no avail. Am at a loss on how to
troubleshoot.
The machine is M4N75TD motherboard with a Phenom II 1055T and the latest
bios flash.
...Rods ;-)
Hey guys,
I'm looking for the "lowest-common-denominator" of audio file formats that
handle BPM info.
Do programs like Ardour / Audacity / Mixxx write BPM info when exporting /
analysing a file?
Cheers, -Harry
Hi All,
I am trying to read data from a usb microphone and using the pretty standard
method of using ioctl's to setup the sampling rate, channels, bits and block
size . This all works so the device is correctly setup. I then use "read" to
read samples from the device which shows up as /dev/dsp1. I get a lot more
samples from this read command in one second of recording than the set
sample rate. E.g. if i set 10Khz on one run i got 269312 samples. Looking at
the raw data it looks like there is a lot of duplication of data? is this
common for the audio input device? if so what kind of encoding is it (e.g
with some specific redundancy built in)?
thanks
farhan
Hi all,
I am working on a system which includes a connexant AD1989A HDA codec
connected to a ATOM processor.
I have four microphones connected to the B and C ports of this codec.
If I change one of the 3 capture gains present in the alsamixer (Capture,
Capture 1 or Capture 3), I can't have any more signal on the B and C ports
It seems to be due to the fact that ALSA breaks (for example for the Capture
gain) the link between the ADC selector 0 and ADC_0 widgets of the codec,
what can be easily seen with codecgraph.
Have you ever had this kind of problem?
Do you know how can I resolve it ?
Many thanks for your asnwers and best regards
Vincent
Could the community please review the attribution so we may continue
with our journey? This attribution appears on the website, github,
README and anywhere else you guys need to see it.
OOM2 is developed from the base code of MusE (Muse Sequencer) written by
the mighty Werner Schweer, and maintained and modified to the present
day, by the current Muse2 maintainer Robert Jonsson and his team.
We not only willingly credit Werner for his code, but add our deepest
respect and admiration as well.
Thanks for all your kind words and support.
P.S. Chris Cannam, the offer stands if you would like me to sign over
all my copyrighted oom code to you. As you explained open source
developers have very little but attribution. Well except those that make
big bucks off it.
--
Christopher Cherrett
ccherrett(a)openoctave.org
http://www.openoctave.org