Hi
I would save a internal audio buffer to file on exit, to reuse it after
e new start.
Currently I use stdio fopen/fwrite/fread and save the binary data from
the array.
That works well.
Now I've started to play with libsndfile, first I use SF_FORMAT_WAV |
SF_FORMAT_FLOAT which work as nice as stdio. The file size is the same
then with plain binary data.
To save some bytes on the disk I tried SF_FORMAT_FLAC | SF_FORMAT_PCM_24
which reduce the size nearly to the half. The drawback is that the
floats in the internal buffer could go out of the range from -1.0 <->
1.0 which leads to crackles when the buffer get refiled. As long the
values are in range, flac works very well.
I tried it with the sf_command (sndfile, SFC_SET_NORM_FLOAT, NULL, SF_TRUE)
for write and read, but that didn't help.
Yes, I use sf_write_float to write/read the flac file, even if flac
didn't support floats. When I understand the libsndfile api right,
libsndfile will handle that.
So my quetion will be, which is the common format to write flac file
with libsndfile, and is there a way to write values out of range into a
flac file?
regards
hermann
Re-posting. Maybe I can get some responses here :)
---------- Forwarded message ----------
From: Rafael Vega <email.rafa(a)gmail.com>
Date: Sun, Nov 3, 2013 at 3:31 PM
Subject: Measuring phase and frequency response of a filter.
To: linux-audio-user <linux-audio-user(a)lists.linuxaudio.org>
Hi.
I'm building a little ear training application for eqing for which I'm
building some filter banks in puredata. Can someone point me to a practical
way of measuring and plotting my filter's frequency and phase responses?
Jack apps, pd patches or code should be fine.
Thanks!
--
Rafael Vega
email.rafa(a)gmail.com
--
Rafael Vega
email.rafa(a)gmail.com
Hi all,
Due to excessive posts - in particular on the linux-audio-user email
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yours truly,
robin
Hi all.
Ive been working on a linux-port of IRCAMs OpenMusic lately, and think
its approaching a useful state now.
I would very much appreciate it if some of the brave ones out there did
some testing and sent me feedback:
http://forge.ircam.fr/p/OM6/downloads/label/421/
The tarball you'll find there includes the OM-6.7 image, source code, a
README and the various resources it uses (.so's, libs, fonts, icons,
help-files, tutorials, other).
This beta is not prepared to be installed anywhere, only to be run from
inside the directory where you unpack it (type ./OM_6.7_beta_4 ... ).
Plan is to make an installable version once the dust settles some.
Theres one thing which needs installing to work however, the music-fonts
used in the various editors and factories etc. Theres an .rpm here:
http://forge.ircam.fr/p/OM6/downloads/get/omfonts-1.0-1.noarch.rpm.zip
- which hopefully takes care of the fonts (theyre also available for
manual install in the tarball). The .rpm installs them (4 .ttf-files)
inside its own folder /usr/share/fonts/omfonts/*
The present version uses JACK for audio + midi I/O, and will try to
start a JACK server if its not already running.
To get any sound from the usual midi-ports, the linux-port uses
libfluidsynth, and attempts to load one of the standard soundfonts into
it. However you should be able to route midi-output to any client you
prefer (with jack-midi support).
Thanks for any feedback.
Cheers,
-anders
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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
