Ken Restivo:
>> The only tool I've found so for that can handle IEEE FLOAT WAV's is
>> oggenc. Yay Vorbis. But, the FLAC tools, even though they're from the
>> same developers as Ogg Vorbis, dies horribly, with ERROR: unsupported
>> compression type 3
I think IEEE floats can't be used in flac, because its a lossless
compressor and that it gurantees not to alter the data. Floats are
probably too hard (probably impossible?) to handle to sattisfy
requirements.
>>
>> ----------------------------------------
>> Sample Rate : 48000
>> Frames : 14277625
>> Channels : 2
>> Format : 0x00010006
>> Sections : 1
>> Seekable : TRUE
>> Duration : 00:04:57.450
>> Signal Max : 0.724668 (-2.80 dB)
>>
>> This is the format that jack_capture produces. It supposedly has
I chose this format as default because its the most common one for audio
processing.
>> options to save in other bit-depths, which I tried, but they didn't
>> seem to do anything.
>>
Works here:
kjetil@ttleush ~ $ jack_capture -b 16
Recording to "jack_capture_03.wav". Press <Return> or <Ctrl-C> to stop it.
Please wait while writing all data to disk. (shouldn't take long)
disk thread finished
connection thread finished
kjetil@ttleush ~ $ sfinfo jack_capture_03.wav
File Name jack_capture_03.wav
File Format Microsoft RIFF WAVE Format (wave)
Data Format 16-bit integer (2's complement, little endian)
Audio Data 278528 bytes begins at offset 44 (2c hex)
2 channels, 69632 frames
Sampling Rate 44100.00 Hz
Duration 1.579 seconds
>> Granted, I can open these files in Audacity or Rezound or Sweep, and with the GUI convert the file to just about anything. However, I dislike GUI's, and I'd like to be able to do this with the commandline tools. Opening 20 WAV files and click-clacking around on menu options is anathaema: the whole process wants to be a 1-liner bash script really.
>>
>
>
> OK after a bit more experimentation, I solved my own problem.
>
> This is for converting from 32-bit 48000 to redbook CD format:
> sox something-32bit.wav -t wav -w -r 44100 something-cd.wav resample -ql
>
You might want to be careful with sox's resampling... You probably want to
listen to the results, but sndfile-resample or resample are probably
safer program to use instead.
> And, for making FLAC's:
> sox something-32bit.wav -w -t wav - | flac - > something.flac
>
jack_capture actually supports flac very conveniently:
"$jack_capture -f flac"
> A similar thing works for making mp3's; I just pipe it to lame instead of to flac. Et voila.
>
I have a script called makemp3.sh that looks like this:
#!/bin/sh
sndfile-convert -pcm32 $1.wav $1o.wav
lame $1o.wav
mv $1o.wav.mp3 $1.mp3
rm $1o.wav
Hi, I'm using the 64 studio 1.0 distro.
with the command:
startBristol -jack -b3 -midi alsa
I get the following, and was wondering what I might be doing wrong.
larry@64studio:~$ startBristol -jack -b3 -midi alsa
You may want to make bristolengine a suid-root executable
spawning midi thread
Could not reschedule thread to 2
parent going into idle loop
connected to :0.0 (5cc430)
display is 1280 by 1024 pixels
Window is w 1280, h 1024, d 24, 0 0 0
Using TrueColor display
masks are ff0000 ff0000 ff0000
flags are 80000000
Returning socket 3
Opened listening control socket: 5028
ALSA lib rawmidi_hw.c:233:(snd_rawmidi_hw_open) open /dev/snd/midiC0D0
failed: Device or resource busy
Could not open the MIDI interface hw:0,0.
Error opening midi device hw:0,0, exiting midi thread
Initialise the hammondB3 link to bristol: 5d15b0
starting upper manual connection
hostname is localhost, bristol
port is 5028
connfailed
opening link to engine: -1
hostname is localhost, bristol
port is 5028
connfailed
cleanupBristol(-4)
larry@64studio:~$ parent exiting
Larry Troxler
On Thursday 08 February 2007, linux-audio-user-request(a)music.columbia.edu
wrote:
> Have you tried with just the following?
>
> startBristol -jack -b3
>
> It looks like the engine has failed to start due to the midi selection (the
> default midi options are the ALSA seq so I would advise not changing that).
> This causes the GUI to fail as well since the engine is not available. What
> version of bristol is this? I am surprised by the error message regarding
> 'suid root' as I thought this was dropped in the -69 release on sourceforge
> - it should now rely on PAM RT permissions or be started as root, so the
> message is superfluous and you only need suid permissions if you want to
> use bristol dedicated to the audio device with RT scheduling.
Just tried this. Here is the result:
>startBristol -jack -b3
You may want to make bristolengine a suid-root executable
spawning midi thread
Could not reschedule thread to 2
parent going into idle loop
flags are 8a000000
midi sequencer
Returning socket 5
Opened listening control socket: 5028
Client ID = 129
Queue ID = 0
Device name did not parse, defaults 128.0
Cannot subscribe port 0 from client 128: Operation not permitted
Error opening midi device /dev/midi, exiting midi thread
connected to :0.0 (80f90a0)
display is 1280 by 1024 pixels
Window is w 1280, h 1024, d 16, 0 0 0
Using DirectColor display
masks are f800 f800 f800
Initialise the hammondB3 link to bristol: 8101220
starting upper manual connection
hostname is localhost, bristol
port is 5028
Connected to the bristol control socket: 5
bristolengine already active
80d4ea0 80000000 0
parent exiting
return - no data in buffer
cleanupBristol(0
Doing it sudo skips the suid message but the rest is the same!
Hej,
I have a problem with connecting to a running jackd, clients are not
able to connect. They are definitly not running as another user as
jackd itself.
I even tried (only for testing) to run qjackctl as root, with no success
either.
I can start jackd by a shell and by qjackctl, but after starting no
client is able to connect. Even giving the jackd-instance a name by the
-n option, clients are not able to connect while specifying that name.
There are also no other clients running that could have the same
client-id as my test-clients (qjackctl itself, meterbridge, adour2).
OS: Ubuntu 6.10 (Edgy), Kernel 2.6.17-10-386
Hardware: M-Audio Delta 44
jackd: 0.102.20
qjackctl: 0.2.20
I had no problems with jackd before upgrading ubuntu from 6.06 to 6.10.
Adour points to a problem with ulimit and locked memory while starting, so
I give the contents of limits.conf, too (this is not the cause
for my troubles, I believe):
mechko@almeida:~$ cat /etc/security/limits.conf | grep -v "#"
@audio - rtprio 99
@audio - memlock 250000
@audio - nice -10
The message-window after pressing the Start-Button in qjackctl (verbose
mode on). There should be an error-message that gives a reason why
clients are not able to connect, but I can find none.
//snap/////////////
12:05:39.529 Patchbay deactivated.
12:05:39.602 Statistics reset.
12:05:39.973 MIDI connection graph change.
12:05:40.011 MIDI connection change.
QInputContext: no input method context available
QInputContext: no input method context available
12:05:47.289 Startup script...
12:05:47.289 artsshell -q terminate
can't create mcop directory
Creating link /home/mechko/.kde/socket-almeida.
12:05:47.559 Startup script terminated with exit status=256.
12:05:47.561 JACK is starting...
12:05:47.563 jackd -v -R -t2000 -dalsa -dhw:0 -r44100 -p128 -n2 -S
12:05:47.570 JACK was started with PID=25103 (0x620f).
getting driver descriptor from /usr/local/lib/jack/jack_alsa.so
getting driver descriptor from /usr/local/lib/jack/jack_dummy.so
getting driver descriptor from /usr/local/lib/jack/jack_oss.so
jackd 0.102.20
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK compiled with System V SHM support.
server `default' registered
loading driver ..
apparent rate = 44100
creating alsa driver ... hw:0|hw:0|128|2|44100|0|0|nomon|swmeter|-|16bit
control device hw:0
configuring for 44100Hz, period = 128 frames, buffer = 2 periods
ALSA: final selected sample format for capture: 32bit little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit little-endian
ALSA: use 2 periods for playback
25103 waiting for signals
registered builtin port type 32 bit float mono audio
registered builtin port type 8 bit raw midi
clock source = system clock via gettimeofday
new client: alsa_pcm, id = 1 type 1 @ 0x805a738 fd = -1
new buffer size 128
registered port alsa_pcm:capture_1, offset = 512
registered port alsa_pcm:capture_2, offset = 1024
registered port alsa_pcm:capture_3, offset = 1536
registered port alsa_pcm:capture_4, offset = 2048
registered port alsa_pcm:capture_5, offset = 2560
registered port alsa_pcm:capture_6, offset = 3072
registered port alsa_pcm:capture_7, offset = 3584
registered port alsa_pcm:capture_8, offset = 4096
registered port alsa_pcm:capture_9, offset = 4608
registered port alsa_pcm:capture_10, offset = 5120
registered port alsa_pcm:capture_11, offset = 5632
registered port alsa_pcm:capture_12, offset = 6144
registered port alsa_pcm:playback_1, offset = 0
registered port alsa_pcm:playback_2, offset = 0
registered port alsa_pcm:playback_3, offset = 0
registered port alsa_pcm:playback_4, offset = 0
registered port alsa_pcm:playback_5, offset = 0
registered port alsa_pcm:playback_6, offset = 0
registered port alsa_pcm:playback_7, offset = 0
registered port alsa_pcm:playback_8, offset = 0
registered port alsa_pcm:playback_9, offset = 0
registered port alsa_pcm:playback_10, offset = 0
++ jack_rechain_graph():
client alsa_pcm: internal client, execution_order=0.
-- jack_rechain_graph()
load = 0.0861 max usecs: 5.000, spare = 2897.000
12:05:49.611 Could not connect to JACK server as client. Please check
the messages window for more info.
load = 0.3877 max usecs: 20.000, spare = 2882.000
load = 0.2800 max usecs: 5.000, spare = 2897.000
load = 0.2261 max usecs: 5.000, spare = 2897.000
load = 0.2337 max usecs: 7.000, spare = 2895.000
load = 0.2030 max usecs: 5.000, spare = 2897.000
load = 0.1876 max usecs: 5.000, spare = 2897.000
load = 0.1627 max usecs: 4.000, spare = 2898.000
load = 0.3054 max usecs: 13.000, spare = 2889.000
load = 0.3077 max usecs: 9.000, spare = 2893.000
12:05:58.362 Could not connect to JACK server as client. Please check
the messages window for more info.
load = 1.0153 max usecs: 50.000, spare = 2852.000
load = 0.6627 max usecs: 9.000, spare = 2893.000
load = 0.6587 max usecs: 19.000, spare = 2883.000
load = 0.4327 max usecs: 6.000, spare = 2896.000
load = 0.4231 max usecs: 12.000, spare = 2890.000
load = 0.5045 max usecs: 17.000, spare = 2885.000
load = 0.5968 max usecs: 20.000, spare = 2882.000
load = 0.4362 max usecs: 8.000, spare = 2894.000
load = 0.3215 max usecs: 6.000, spare = 2896.000
load = 0.4709 max usecs: 18.000, spare = 2884.000
load = 0.5456 max usecs: 18.000, spare = 2884.000
load = 0.3934 max usecs: 7.000, spare = 2895.000
load = 0.4551 max usecs: 15.000, spare = 2887.000
load = 0.5377 max usecs: 18.000, spare = 2884.000
12:06:12.597 Could not connect to JACK server as client. Please check
the messages window for more info.
load = 0.3722 max usecs: 6.000, spare = 2896.000
[and so on]
\\snip\\\\\\\\\\\\\\\\\\\\\
jackd is running:
mechko@almeida:~$ ps aux | grep jackd
mechko 25185 0.5 2.6 20552 20592 ? SLsl 12:07 0:00
jackd -v -R -t2000 -dalsa -dhw:0 -r44100 -p128 -n2 -S
meterbridge fails:
mechko@almeida:~$ meterbridge -t vu alsa_pcm:capture_1 alsa_pcm:capture_2
jack server not running?
Could you please give me a hint what to do? Thanks in advance,
Mirko Swillus
On Fri, Feb 09, 2007 at 08:35:34AM +0000, Lots of people wrote:
> The attachment?
Whoops!
Attached.
Chris.
-------------------
chris(a)mccormick.cx
http://mccormick.cx
I've been using rhythmbox for listening to my mp3 files on my Ubuntu
laptop. Mostly this was because it supports daap sharing, and gives me
a nice interface for choosing songs. It works well except I've been
having sound quality problems.
I seem to get some minor distortion, and some pops and cracks when
playing a lot of my mp3 files. I took one especially problematic file
and played it with alsaplayer and had no troubles (at a louder volume
even). I'm about 98% sure that somehow gstreamer is peaking/clipping,
even when the volume is way down.
Banshee, I've noticed, does the same thing.
mpg123 and mpg321 seem to have no problems. xine engine of amarok seems
to work.
Has anyone else noticed anything like this? Is this a problem with
gstreamer in general? What's going on here? What's with the levels
that it would clip when it's not even loud?
(and I really wish there was a jack backend for gstreamer)
Any comments, suggestions? Does anyone here follow gstreamer at all?
--
-= =- vroetman(a)myrealbox.com -= =-
"I am working on a tune that needs surf guitar. The CAPS AMP IV does a
great Fender Twin, and the #4 cabinet in CAPS Cabinet II seems to do
pretty well as a 2x12 open-back Fender cab. But the classic surf guitar
sound of course includes the high-pitched plinking of a Fender Twin
reverb spring slapping against its metal case. If you haven't heard it,
it's kind of like entering the V'Ger central core, but really
high-pitched and fast, and in 16th notes while double-picking.
Picking out a muted barre chord into a spring reverb turned up way too
high, you can definitely *hear* the springs."
Despite the way this should be a very non linear and random kind of
reverb, convolution reverbs with a spring reverb sample do this
surprisingly well.
There are plenty of convolvers around, and loads of samples out there
(noisevault is good).
I made my own stereo impulses by recording both ends (both in and out as
outs, impedances way off, but who cares) of a spring reverb tray and
giving it a thwack with a drum stick.
I use SIR running on ardour-vst on a computer dedicated to reverbs and
other heavy DSP. Connect via netjack, chuck artificial latency on the
sends so it comes back in time, and bob's your uncle.
(may not be as easy as it sounds, contents may settle in transit).
I've tried some other non vst native Linux convolvers, but have not
found one as efficient yet. Recommendations welcomed.
are there any facilities on linux to create cart chunks within a wave file?
<http://cartchunk.org:8080/>
i'd love to be able to add cart chunk metadata to a wave file from a
command line. it would save my butt at work. i haven't been able to
find anything via google.
--
daneasley(a)gmail.com
dan(a)towndowner.com dan(a)burntpossum.com
http://towndowner.comhttp://burntpossum.com
Hi everyone,
I'm pleased to announce the release Tranches and Tapeutape.
(my first release !!! )
Tapeutape is a midi-controlled virtual sampler.It is highly
configurable,lets you create instruments/kits/setups and is designed for
live performance.It works on Gnu/Linux with the jack sound server.
There's a command-line and a gui version.
It can be found at http://www.hitmuri.com/Tapeutape
An example of what can be done with it :
http://www.hitmuri.com/Tapeutape/demo_tapeutape.ogg
Tranches is a midi-controlled multi-(inputs|outputs) live beat
repeat|redirect tool for the jack sound server on Gnu/Linux !!! There's
a command-line and a gui version.
website : http://hitmuri.com/Tranches/
audio example : http://www.hitmuri.com/Tranches/demo_tranches.ogg
Regards,
Flo
Hello LAUs,
maybe someone here knows how I can record from an ICE1712's Monitor
Mixer using ALSA? I routed the HW in 1/2 to the digital mixer in
envy24control, and now I want to record that, but I'm perhaps just too
stupid to find out how the monitor mixer is called in ALSA.
That device is R/W for sure:
http://www.alsa-project.org/alsa/ftp/datasheets/icensemble/envy24.pdf
Using 4Front's OSS (not the in-kernel) it is as simple as recording
from /dev/dsp4, which is the monitor on my audiophile 2496. But OSS is
a pain when trying to use jack or any recording app like rezound or
audacity with it. This somehow must be possible using ALSA too, not?
I also tried using ALSA's jack plugin, and record with jack_capture, but
a) I get total sound distortion as soon jack_capture starts, and
b) that's one layer too much anyway, as the card can do "loopback"
itself. So i'd prefer ALSA.
Any help would be appreciated.
Thanks,
Tom