Linux-audio-user January 2012

linux-audio-user@lists.linuxaudio.org

133 participants
123 discussions

by Moshe Werner

Hi everyone, after many years of studio work using the openSuse distro with the kernel-rt from Jan Engelhard it seems that he no longer continues his great work on rt kernels. Being more on the recording engineer side of things and not a Linux expert (user yes, expert no) I really fret at the thought of patching and compiling my own kernel package. I would like to hear your opinions on what distro is solid for audio work and has a reliable rt kernel. Also I would appreciate if you could explain the degree of difficulty and learning curve of the specific distro. My system: Intel i7 950 Gigabyte motherboard 6 Gb ram Rme HDSP 9652 audio interface Appreciate your answers. Moshe P.S. I tried to use Ubuntu on the same machine I use openSuse 11.2 on and got pretty bad results regarding latency and x runs on jack 2.

13 years, 11 months

[LAU] What audio interface to use for a Linux-powered surround preamp?

by Johan Herland

Hi, (TL;DR: skip to "The main question...") I'm planning a hobby project to build an all-digital surround/hi-fi system. Regular surround/hi-fi systems convert digital input sources to analog in the surround processor/preamp, but I want to keep the signal in the digital domain as far as possible (by using all-digital class D power amps, aka. "PowerDACs" [*] that convert a PCM signal to an amplified PWM signal that is feeds the speaker directly through an LPF. The heart of this system is an all-digital surround processor/preamp that should do the following: 1. Switch between multiple digital inputs. (typically from HDMI or SPDIF) 2. Decode and/or upsample the input (if necessary) into a suitable format, typically 96kHz or 192kHz 24-bit PCM in 8 channels (for a 7.1 system). 3. Perform signal processing on the PCM signal, like room correction and volume control. 4. Output each of the 8 channels on separate digital outputs suitable for connection to the power amp associated with each speaker. AFAIK there are currently (very few or) zero preamps for sale that meet the above requirements. Requirement #4 seems especially hard to satisfy. I'm therefore planning to build my own preamp based on the following components: - A HDMI switch with audio split (SPDIF) and RS-232 control. - A Linux computer that controls the HDMI switch, and contains the audio processing software (e.g. a gstreamer pipeline doing the decoding and manipulation). - A suitable audio interface with at least 8 digital outputs. I believe this rig should be able to meet the above requirements. The main question at this point is what audio interface I should choose for the rig. Some points to consider: - Must work very well with Linux; bonus points if the vendor is Linux-friendly. - Must be able to output 8 channels of 96kHz/24bit PCM, preferably using AES/EBU or SPDIF (as that's what most digital power amps seem to accept as input). - USB, Firewire, PCI or PCI express? Which is more stable/supported? - Bonus points if it also has 8 analog outputs, as I want to prototype this on an analog power amp before investing in digital power amps. - Since this is still in a proof-of-concept phase, I'd like to spend not more than about $1000 on the interface. So far I've been browsing interfaces like: - MOTU 828mk3 (capable, but seems to be poorly supported by Linux) - RME HDSPe AES (very capable, lacks analog outputs, a bit pricey) - RME Multiface II + HDSPe interface card (has analog outputs, not sure if 1 SPDIF plus 1 ADAT can give me 8 channels of digital output in 96/24, pricey) - Focusrite Saffire PRO 40 (seems similar to RME Multiface II, but cheaper. Unsure how well it is supported by Linux) What other audio interfaces should I check out? Are there other things I should keep in mind when picking an audio interface? Have fun! :) ...Johan [*]: Examples of all-digital power amps include the NAD Direct Digital amps (M2 and C390DD), the Tact True Digital Amplification series, etc. -- Johan Herland, <jherland(a)gmail.com> www.herland.net

13 years, 11 months

[LAU] DC offset

by Rustom Mody

Thanks to Al Thomson pointer to look at 'normalize' I have found that my equipment (just a vanilla sound card) has/introduces some DC offset. How to remove it? Why does it come?

13 years, 11 months

[LAU] Have some free music!

by alexander

Time for some harmless selfpromosion ;) five new songs since I last posted. http://sharpattack.bandcamp.com/

13 years, 11 months

[LAU] Hardware: Drum pad

by rosea.grammostola

Hi, Not that I am a drummer, but not a real fan of midi editing either ;) So I need a hardware interface for the drums (something like the Roland octapad). Moreover I have thought about it and this is how I like to use the computer, as server of sounds, midi/audio record and mixing. For workflow an hardware interface is better I think. So searching for a drum pad now to use with LinuxSampler and probably Hydrogen (if it finally get stable with JACK). Any recommendations as it comes to touch, features, connection possibilities, brands etc.? Like I said, I am not a drummer, so it's gonna be pretty low budget I think. Regards, \r

13 years, 12 months

[LAU] announcing chino-0.10

by David Adler

chino-0.10 is out, another re-write. Chino is a session management framework written in Bash. Pretty much alpha, I suspect there might be some bugs left occurring in corner cases I haven't encountered yet. http://chino.tuxfamily.org/ Previous versions generated session scripts for a very limited variety of setups. The current version differs in three main points, in that: (1) no script is written, sessions can now be recovered from a 'session definition file'; (2) the possible variety of setups is much larger since the main script is separated from implementation, i.e. a user can quite freely define applications and their behaviour; (3) it provides a runtime user interface, for adding and removing applications while the session is running. The purpose is not to add another session management system to the LASH/LADISH/JackSession ecosystem, chino is somewhat different in scope. I neither claim that it's exactly simple nor that it is of any particular elegance, but it serves my needs very well -- I can hopefully now start making music. :) An excerpt from the features list on the website: a 'load session' command and a 'save session' keybinding (the latter saves only the session, the state of any involved application—if applicable—needs to be manually saved to the appropriate file from within the application); presets and template sessions for simple creation or forking of sessions; a text user interface to add or remove applications while running, with no manual connection-making involved; adding support for an application amounts to adding a file containing some variables and functions, thus no support on the application's side is required (apart from the ability to recover a state via command line and/or file loading, without manual user interaction); mono-stereo-agnostic audio connecting and Jack-Alsa- agnostic Midi connecting; a hierarchical session layout with dependency check, suggestions and the option to view an image displaying the session graph (the passive "GUI"). best, d

13 years, 12 months

[LAU] Multiple ICE1712 cards setup with jack2

by mn0

Hi, I have two M-Audio Delta1010LT cards on Ubuntustudio 10.04 with FalkTx's repos. I double checked, this is happening also on Kubuntu 11.10. Also it's not a matter of the kernel, currently using 2.6.33-29-realtime. I use Jack2, where can I find the exact version? Only thing I know, it's jackdmp 1.9.7. The cards are hardware synced via S-PDIF. In envy24control, a mixer program that knows about these features, hw:0 is set to internal clock and hw:1 is set to S-PDIF. I know of three options to get both cards working simultaneously, but none is working quite perfectly. a)use .asoundrc I have a .asoundrc from the web. After a few seconds, it crashes saying this: Unknown request 4294967295 Destination port in attempted (dis)connection of system:monitor_6 and system:monitor_6 is not an input port Unknown request 4294967295 Unknown request 0 Unknown request 0 Unknown request 0 Unknown request 4294967295 Unknown request 0 Unknown request 4294967295 Unknown request 4294967295 jackd: ../common/JackGraphManager.cpp:45: void Jack::JackGraphManager::AssertPort(jack_port_id_t): Assertion `port_index < fPortMax' failed. While running, it produces tons of xruns, each of them with -v option generates a message like Jack: fPollTable i = 1 fd = 11 Jack: JackRequest::Notification Jack: JackEngine::NotifyClient: no callback for event = 3 Jack: JackEngine::NotifyClient: no callback for event = 3 The last thing I get before crashing is: Jack: JackRequest::ConnectPorts Jack: fPollTable i = 1 fd = 11 Jack: fPollTable i = 1 fd = 11 Jack: fPollTable i = 1 fd = 11 Jack: fPollTable i = 1 fd = 11 Jack: JackRequest::ConnectPorts Jack: JackEngine::PortConnect src = -1 dst = 3 Jack: JackGraphManager::AssertPort port_index = 4294967295 But there is no delay of the second card. b) use jack_load this is working best atm, only it introduces a delay to the channels of the second card when recording with ardour. I measured the delay... it's exactly (periods/buffer)*(frames/period) frames. However, I'm not able to delay the first card with the -O or -I options. Probably the second is delayed as well, or this option is not working. In ardour I can move the regions to be in sync, but it gets quite confusing in bigger sessions. Btw does jack_load resample? c)use alsa_in produces the same delay but additionally uses a lot of cpu. Resampling is not what I want, and also shouldn't be necessary. Does anyone have an idea on how to fix this? I'd prefer a solution without resampling, as these cards are already synced. Also there should be no delay and it should keep running for hours. Thx! Any help is appreciated. /mn0

14 years

Re: [LAU] sndfile-waveform PNG generator ?

by Robin Gareus

[re-arranged top-posted] On 01/10/2012 02:59 PM, Vytautas Jancauskas wrote: > On Tue, Jan 10, 2012 at 2:47 PM, Robin Gareus <robin(a)gareus.org> wrote: > >> Hi *, >> >> Is there a command-line tool akin to `sndfile-spectrogram` that >> generates an image (preferably PNG) of an audio wave-form? >> >> I found endless GUI apps, but command-line tools are scarce. >> >> TIA >> robin >> > It's fairly trivial to write one your own with python using scipy. > Should be in the ballpark of 20 lines or so. Thanks, indeed. Python's also the only solution I found: https://github.com/endolith/freesound-thumbnailer/blob/master/wav2png.py It's also pretty straight-forward in C; but before whipping up `sndfile-waveform`, I'd thought I ask around. robin

14 years

[LAU] audiophile 2496

by nepal

Hello Everyone. I am a mere lurker, sadly no musician, but do really love listening. I run linux mint and am thinking about treating myself to the above sound card to improve my listening experience. I'll be running the output to my Kenwood hifi amp using the phonos I expect. Is the above sound? as in sensible and workable. I'd rather not get into manually messing with my distro settings if that is possible. Are there any gotchas to be considered? Can I just go ahead, buy, and plug and play? many thanks for any help offered. n ______________________________________________ This email has been scanned by Westcoastcloud. http://www.westcoastcloud.com/

14 years

[LAU] paulstretch portaudio conflict

by James Mckernon

Hi all, I recently tried compiling paulstretch on ubuntu 11.04, and found that it ran into some problems compiling. Here's the output from running compile_linux_fftw.sh: PAaudiooutput.cpp:29:9: error: ‘PaStreamCallbackTimeInfo’ does not name a type PAaudiooutput.cpp:29:35: error: ISO C++ forbids declaration of ‘outTime’ with no type PAaudiooutput.cpp:29:43: error: ‘PaStreamCallbackFlags’ has not been declared PAaudiooutput.cpp: In function ‘void PAaudiooutputinit(Player*, int)’: PAaudiooutput.cpp:40:94: error: invalid conversion from ‘int (*)(const void*, void*, long unsigned int, const int*, int, void*)’ to ‘long unsigned int’ PAaudiooutput.cpp:40:94: error: too few arguments to function ‘PaError Pa_OpenDefaultStream(PortAudioStream**, int, int, PaSampleFormat, double, long unsigned int, long unsigned int, int (*)(void*, void*, long unsigned int, PaTimestamp, void*), void*)’ /usr/include/portaudio.h:355:9: note: declared here I did some googling, and it seems that paulstretch is written to use, and depends on, portaudio version 1.9 (package portaudio19-dev on ubuntu). However, this conflicts with the more recent version of portaudio, which Jack (and all my Jack apps) seem to depend on. Does anyone know of a way around this? Failing that, does anyone know what the likelihood is of paulstretch being patched to use the newer version of portaudio? (Is development on it still active?) Thanks for your help, James

14 years

2026

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-user January 2012