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[linux-audio-user] PTFree

by RTaylor

Anyone tried this under wine: http://www.digidesign.com/ptfree/

22 years, 2 months

[linux-audio-user] Upgrading ALSA on Mandrake 9.1

by Daniel James

Hello all, I've got an odd one here - I'm trying to upgrade ALSA on a Mandrake 9.1 box. I did this before on another (upgraded to 0.9.2) and it worked fine, but on this one upgrading the RPMs seems to have no effect. I installed these RPMs with urpmi: libalsa2-0.9.5-1mdk.i586.rpm libalsa-data-0.9.5-1mdk.i586.rpm kernel-multimedia-source-2.4.21-0.16mdk.i586.rpm But if I do: service alsa restart I get: Shutting down ALSA sound driver (version 0.9.0rc8): [ OK ] Starting ALSA version 0.9.0rc8: cs46xx. [ OK ] I've tried rebooting. Any clues? Have I missed something obvious here? Thanks Daniel

22 years, 2 months

[linux-audio-user] JACK Release 0.80.0

by Taybin Rutkin

JACK RELEASE 0.80.0 JACK is a low-latency audio server, written primarily for the GNU/Linux operating system. It can connect a number of different applications to an audio device, as well as allowing them to share audio between themselves. Its clients can run in their own processes (ie. as normal applications), or can they can run within the JACK server (ie. as a "plugin"). JACK is different from other audio server efforts in that it has been designed from the ground up to be suitable for professional audio work. This means that it focuses on two key areas: synchronous execution of all clients, and low latency operation. CHANGES: New transport API (details (of a sort) below). new example client for control the transport. ignoring of first xrun on jackd startup. Much more portable across processors (details below). jackd -v (--verbose) now prints useful transport state change information for debugging JACK and clients. Also reports timeout info in seconds, not microseconds now. new dummy driver (along side the existing alsa and portaudio drivers). Removed incomplete Solaris driver. support for asymmetric soundcards (for example, es1968 chip has interleaved stereo for playback but non-interleaved stereo for capture). Now enforces power of two sized buffer lengths. Many minor bug fixes. DETAILS: The new transport API: It has greatly changed; if you're a developer, please see the documentation. But as a highlight: jack_set_transport_info() and jack_engine_takeover_timebase(), (the old timebase master interfaces) now do nothing. Instead, use jack_set_timebase_callback(). Portability: <jack/types.h> typedefs are now defined using the C99 standard fixed-size integer typedefs. These new typedefs are binary compatible with 32-bit platforms, but not 64-bit machines. Programs using printf on these values will get GCC compiler warnings. To suppress the warnings, use the corresponding C99 printf specifications defined in <inttypes.h>. That header is already implicitly included by <jack/types.h>, but can also be included explicitly to maintain compatibility with older versions of JACK without messy #ifdef's. Adding explicit casts will also work, but may suppress future warnings you might want to see. jack_get_sample_rate() now returns jack_nframes_t rather than unsigned long. These are the same on 32-bit machines, but not on 64-bit platforms. These changes were made to accommodate the increasingly popular 64-bit platforms; specifically, the new Opteron with it's 32-bit compatibility mode. 32-bit mode probably won't work, but if all programs are compiled as 64-bit (which Jack supports), it should work fine. Taybin Rutkin

22 years, 2 months

[linux-audio-user] Which kernel to use for audio work?

by Robert Epprecht

I'm currently setting up an audio system on Debian sarge. The system will be used for audio work only. My sound hardware is RME Multiface and the PCI card. Of course I want ALSA and JACK. I wonder which kernel version I should try first, 2.4.?? (which one) or a 2.6.0 test kernel? I'm fine with kernel source from kernel.org and don't need a debianised kernel source. Please let me know which patches are needed (like lowlat, preemt) for the kernel version in question, as this is not clear to me. I know this question shows that I'm not very well informed about the different kernels, but I *do* like learning ;-) Thanks a lot for your help! Robert Epprecht

22 years, 2 months

[linux-audio-user] ecasound 2.3.0 released

by Kai Vehmanen

1. A short summary of changes A set of severe bugs in audio mixing code have been fixed. Pyecasound build process has been improved. Reporting chainsetup parsing errors has been improved significantly. Support for the JACK 0.80 transport interface has been added. Support for reading and writing aiff, snd and au files has been fixed. Changes have been made to ensure correct operation with the NPTL package recently added to Linux kernel and glibc. A serious bug in metronome timing was fixed. Minor bugs in dynamic sample rate changes, MIDI-server initialization and the ewf file format have been fixed. --- 2. What is Ecasound? Ecasound is a software package designed for multitrack audio processing. It can be used for simple tasks like audio playback, recording and format conversions, as well as for multitrack effect processing, mixing, recording and signal recycling. Ecasound supports a wide range of audio inputs, outputs and effect algorithms. Effects and audio objects can be combined in various ways, and their parameters can be controlled by operator objects like oscillators and MIDI-CCs. A versatile console mode user-interface is included in the package. Ecasound is licensed under the GPL. The Ecasound Control Interface (ECI) is licensed under the LGPL. --- 3. Changes since last release * Support for JACK 0.80 transport interface. The recently released JACK 0.80.0 marks another milestone for Linux audio development. The new transport interface allows concurrent use of multiple independent audio applications with full sample-accurate transport synchronization. This release of Ecasound provides full support for the new transport API. See the following links to ecasound-list postings that give examples of how to use Ecasound with the new transport API: http://eca.cx/ecasound-list/2003/08/0005.html http://eca.cx/ecasound-list/2003/08/0070.html * Metronome timing bug. There was a subtle bug in the pulse gate operator that is used to generate the metronome signal. This bug caused a small (<0.5%) timing error in metronome speed. The bug has been fixed, but the change in speed might cause problems for working with sessions recorded with previous versions of Ecasound. Thanks to Carsten Bauer for finding the bug. Full list of changes is available at <http://www.wakkanet.fi/~kaiv/ecasound/history.html>. --- 4. Interface and configuration file changes Libecasound interface version number was incremented to 11. This release is backwards compatible with interface versions 9 and 10. See 'ecasound/libecasound/ChangeLog' for a detailed list of changes. Libkvutils interface version number was incremented to 6. This release is backwards compatible with versions 5 and 4. See 'ecasound/kvutils/ChangeLog' for a detailed list of changes. Note! Using libecasound and libkvutils outside Ecasound has been discouraged since the release of 2.2.0 in Jan 2003. The interface changes introduced in this 2.3.0 release should only affect a small group of distributors and developers. Apps such as Ecamegapedal need to be recompiled only if you want to take advantage of the new features introduced in the latest Ecasound release. --- 5. Contributors Patches - Accepted code, documentation and build system changes Kai Vehmanen (various) Bug Hunting - Reports that led to bugfixes (items closed) Jeremy Hall (3) -- non-default chainsetup srates, mixdown signal leak bug, pyecasound install Junichi Uekawa (3) -- non-default chainsetup srates, pyecasound install, gcc-3.3 verification Janne Halttunen (2) -- locale-settings broke map-* commands, MIDI-server startup bug Vegard Lima (2) -- feedback bug, ewf srate bug Carsten Bauer (1) -- timing bug in metronome and pulse gate Ismail Donmez (1) -- build errors with sys-readline option Lars Henrik Mai (1) -- aiff/au/snd bug Stephan Niemz (1) -- non-default chainsetup srates Al Oemens (1) -- reporting chainsetup parsing errors Oliver Thuns (1) -- -ei big shift-x values Tommi Uimonen (1) -- problems with ecasignalview channel meter layout --- 6. Links and files Web sites: http://www.eca.cx http://www.eca.cx/ecasound Source packages: http://ecasound.seul.org/download http://ecasound.seul.org/download/ecasound-2.3.0.tar.gz Distributions with maintained Ecasound support: Agnula - http://www.agnula.org Debian - http://packages.debian.org/unstable/sound/ecasound2.2.html DeMuDi - http://www.demudi.org FreeBSD - http://www.freebsd.org/ports/audio.html Gentoo Linux - http://www.gentoo.org PLD Linux - http://www.pld.org.pl PlanetCCRMA - http://www-ccrma.stanford.edu/planetccrma/software SuSE Linux - http://www.suse.de/en Contrib Packages for Distributions: Mandrake - http://rpm.nyvalls.se/ Slackware - http://www.audioslack.com Various distros - http://apps.kde.com/rf/2/info/id/2146 Note! Distributors do not necessarily provide packages for the very latest Ecasound version. -- http://www.eca.cx Audio software for Linux!

22 years, 2 months

[linux-audio-user] Low Latency in 2.4.22 kernel

by Luis Pablo Gasparotto

Hi, When I run the Benno Sennoner's Latencytest utility on an unpatched 2.4.22 kernel I get the same results that I get with a patched 2.4.21. Does the 2.4.22 kernel have a low latency improvement built-in? Regards, Luis Pablo Gasparotto

22 years, 2 months

[linux-audio-user] [ANN] MidiSport firmware loader 0.2

by Clemens Ladisch

The MidiSport firmware loader allows to use the MidiSport USB MIDI interfaces from M-Audio/Midiman with Linux. Version 0.2 fixes a bug in the loader script which tried to load the firmware files from the wrong directory. Supported devices: MidiSport 1x1, 2x2, 4x4, 8x8, KeyStation, Oxygen, Radium, Uno. To download the loader, go to the project home page: http://usb-midi-fw.sf.net/ -- Clemens

22 years, 2 months

[linux-audio-user] Jack 0.80.0 release follow up

by Taybin Rutkin

Just a quick follow up to the release announcement. Jack's website is:http://jackit.sourceforge.net/ and the files are available at: http://sourceforge.net/project/showfiles.php?group_id=39687 Taybin Rutkin

22 years, 2 months

[linux-audio-user] USB audio capture line/mic: A/D quality, how set record levels?

by Mannr＠uwaterloo.ca

I am considering getting a USB audio interface. My goal is to capture audio for making CDs. I will capture from line level, eg., from minidisc or phonograph, and from microphone inputs, eg., for recording live shows to a laptop. It should be completely USB powered for maximum flexibility. Digital I/O would be great, but is not required. I am not looking for professional quality. But I would like something pretty good, eg., at the level of consumer minidisc recorders. That is what I am using to record now. I don't need any mixing capability. Plain capture at 44.1kHz/16bit and optionally 48.0kHz would be fine. I've seen lots of gear, eg., Edirol, M-audio, etc. But I have several questions: - What is the A/D quality of some of the retail units? Retail units: Edirol UA-1A, UA-3D (edirol.com), MAudio Transit (m-audio.com), Soundblaster Extigy, Soundblaster MP3+ Higher quality? Tascam U-122, Edirol UA-20?, MAudio MicPre, Professional level (>500USD) SoundDevices USBPre (http://www.sounddevices.com/products/usbpremaster.htm) - Is the Mic preamp any good? Ie., as good or better than the one in my consumer minidisc? - What is the "native" capture mode of these units? 16bits is OK, but I have heard reports that some, like the Soundblaster, capture at 48 or 96kHz and resample for 44.1kHz. I am looking for true sampling at 44.1kHz. - How can I control record level on these units? Some have level controls, but it is unclear what level these control, the mic preamp or the line level, or both. The only one I can see a clear "block diagram" for is the USBPre. This shows separate mic and line level gain controls, as it should have. (See user guide PDF file.) If there are no knobs, can the level be set with software? Is this what the Linux "xmixer" program does? The point is that I want to: - adjust the line level gain *before* A/D is done - adjust the mic level gain *within the mic preamp*, to avoid clipping So the unit must have internal gain control. Here is what I have learned so far. Please help... Richard Encl: list ------------------------------------------------------------------------------ Product Cost |USB |LineI/O |MicI/O |DigiI/O |Sample |Depth |Quality self level? level? (native) 16/24 Consumer/ (street) power control control exact-bit Audioph/ | | copy?| Pro phantom power?| ------------------------------------------------------------------------------ Edirol UA-1A Y I/O, N none none 44.1k 16 Consumer UA-3D Y I/O, Y I/O, YN I/O, N ?(1) 16 Consumer UA-20 Y I/O, Y none I/O, N 44.1k 16/24 ? Edirol UA-5 9VDC I/O, Y I/O, YY none 44/48/96(2) 16/24 ??? Creative MP3+ Y I/O, N I/O, NN I/O, N 48k(3) 16 Consumer Creative Extigy Y I/O, Y I/O, YN I/O, N 48k(3) 16/24 Consumer AFAICT all above units sample at 48/96K and resample for 44.1K. So they will add artifacts, and they will not make an accruate digital copy either! (1) claims 32/44.1/48k. Is this 48k native sample rate? (2) "real time sample rate conversion", is this 96k native sample rate? (3) rumored to be resampled. MAudio Transit $100 Y I/O, N I/O, NN I/O, ? ? 16/24 Consumer MAudio MicPre $180 Y I/O, ? I/O, YY none 44.1/48 16 ??? (150?) MAudio Duo $350 9VAC! I/O, ? I/O, YY I/O, Y 44.1/48 16/24 Pro Tascam US-122 $270 Y I/O, ? I/O, YY none 44.1/48 16/24 Pro? (200?) USBPre $700 Y I/O, Y I/O, YY I/O, Y 44.1/48 16/24 Pro Possibilities: ------------- MAudio Transit would be great if it had: - reasonable quality A/D - true level controls (from software) - true hardware sampling at 44.1, 48k and 96k. MAudio MobilePre looks good if I am happy with 16bit sample depth and no digital I/O. Tascam US-122 looks good as well, but is more expensive.

22 years, 2 months

[linux-audio-user] RE: Invert And Mix

by Jamie Lutch

what works for me is to generate a sine wave (I used wavelab in windows), then copy and paste it until it is several hours long, then insert 10 seconds of silence at the beginning. Start rolling your sine wav recording from another machine with a known good soundcard, and start your test machine recording in the silent period. Its super easy to sync up to your original wav by deleting all the silence at the beginning.

22 years, 2 months

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