Linux-audio-dev

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7501 discussions

by Andreas Degert

On behalf of the guitarix team I'm proud to announce Guitarix Version 0.11 Guitarix is a simple Linux Rock Guitar amplifier and is designed to achieve nice thrash/metal/rock/blues guitar sounds. Guitarix uses the Jack Audio Connection Kit as its audio backend and brings to the jack audio graph a mono amplifier input/output port, and a FX mono input with two (stereo) output ports. Guitarix provides a jack midi input port to connect a midi controller (midi learn) and a (3 channel) jack midi output port, feed by a (scalable) mix of the tuner and a beat-detector. Release 0.11.0 comes with following changes : Users visible changes: - new version of the impulse response parameter editor. You can now graphically define a gain line to be applied to the IR data, e.g. emphasize a part of the early reflections or damping the reverb tail to make it fit to your guitar sound. - the multi-line distortion effect now has 4 frequency bands and is better optimized - fix regression from version 0.10.0: now the default preset file is created again if it doesn't exist (e.g. in new installations). Changes that might be interesting for Developers: We put the Guitarix widgets into a library, with the goal of making them usable independently from Guitarix. You can build it as shared library and there's a c++ (gtkmm) wrapper, a python wrapper and glade support. Check it out and look for examples in those directories, or just build a nice looking display with the glade editor, and of course ask in our Guitarix forum (it's still alpha). As a side note, Guitarix is now in debian (unstable), of course at the moment still the previous version 0.10. have fun _________________________________________________________________________ guitarix is licensed under the GPL. Project page with screenshots: http://guitarix.sourceforge.net/ download: http://sourceforge.net/projects/guitarix/ please report bugs and suggestions in our forum here: http://sourceforge.net/apps/phpbb/guitarix/ ________________________________________________________________________ For capture, guitarix uses the great 'jack_capture' (version >= 0.9.30) written by Kjetil S. Matheussen. If you don't have it installed, you can look here: http://old.notam02.no/arkiv/src/?M=D For extra Impulse Responses, guitarix uses the zita-convolver library, and, for up/down sampling we use zita-resampler, both written by Fons Adriaensen. If you don't have it installed, get it here: http://www.kokkinizita.net/linuxaudio/index.html We use the marvellous faust compiler to build the amp and effects and will say thanks to : Julius Smith http://ccrma.stanford.edu/realsimple/faust/ : Albert Graef http://q-lang.sourceforge.net/examples.html#Faust : Yann Orlary http://faust.grame.fr/ ________________________________________________________________________ For faust users : All used Faust dsp files are included in /guitarix/src/faust, the resulting cc files are in /guitarix/src/faust-generated The tools we use to convert (post-processing and plot) the resulting faust cpp files to the needed include format, stay in the /guitarix/tools directory. ________________________________________________________________________ regards Hermann Meyer, James Warden, Andreas Degert

14 years, 11 months

Document on using Jack Apps

by Arvind Venkatasubramanian

Hello: I am very much new to Linux and Linux audio. I am trying to measure audio IO latency for my system. Jdelay seems to be the right tool but when I run it on the terminal, I am getting message "Signal below threshold..." which probably might be because Jack is trying to capture audio and ends up getting the low noise floor because I do not have meaningful signal source connected. But then I tried to patch the this App to the qJACKctl app but the settings console is not straight forward to interpret. as it involved many parameters. I guess there are # of frames per period that may eventually be used to calculate the target latency but *is there a step-by step document description for Jdelay and any other JACK tools and also using JACK audio server in an effective manner*? Also, qJACKctl console does not offer options for very low sampling rates like 8 KHz. With ALSA, this should be possible but may be this particular tool does not support it. Can anyone help here? Thanks, -F

14 years, 11 months

Re: [LAD] [LAA] announcing envy24control, mudita (*) edition.

by Niels Mayer

[trimmed down reply-to linux-audio-dev(a)lists.linuxaudio.org] On Tue, Jul 27, 2010 at 8:25 AM, john ffitch <jpff> wrote: > Any chance of an option to envy24control to allow colour blind people > to see three zones? I had been using it for years before I was told > that there is a red section at the top and green below > ==John ffitch John -- Thank you for writing and you'll be happy to know that the new "mutida24" (to avoid confusion with the old and overly colorful envy24contol) does not use coloration in the same way, and yes, I'd imagine the old choice of colors would be particularly bad for http://en.wikipedia.org/wiki/Color_blindness#Red-green_color_blindness . Although there's been some recent minor changes, this is the look of the new meters: http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorInputs.… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorPCM.png Hopefully, at least for any colors at issue, there is enough contrast against the background to make it properly visible. Please let me know of any specific issues w/ the simplified, and more X-efficient meters in this version, which try to be less eye-candy, and more useful for making a visual level determination of inputs, outputs, or digital mixer inputs. The peak metering features a green/white/orange/red progression of the peak color (the thin band that stays fixed while the meter below bounces around). But in addition, and more helpful to you, are he peak-level readouts in dBFS from -48 to 0dBFS for all inputs and outputs, including in the "Analog Volume" panel. The 0dBFS label is red with white background, whereas the other levels remain black with white background. That should provide enough redundancy in the metering to not hide important information from you. However, I'm not sure of the perceptibility, of the red "0dBFS" marking. On the other hand, it does say "0dBFS" (in red) which is quite different-looking than, say "-5.02" (in black) In http://nielsmayer.com/envy24control/Screenshot-Envy24Control-AnalogVolume.p… the green (at 0dB) and red (+6 +12 +18) markings on the ADC would not be very useful for you; fortunately, the "+" should be visually distinctive from all the other "-" markings in blue. Other alternatives would be Bolding amplification values, and underlining unity gain. Thanks, Niels http://nielsmayer.com

14 years, 11 months

Re: [LAD] what does "full range" mean with regard to ambisonics and speakers?

by Martin Leese

"Bearcat M. ?andor" <hometheater(a)feline-soul.com> wrote: > Thank you. I think i understand all that, but let me take this apart to > make sure. What you're [J?rn Nettingsmeier is] saying is that having full range speakers only > effects the playback quality of the music not the ambisonics and > that ambisonics itself does not *need* full range speakers, but that > having full range speakers is better than not. Is that correct? There are (at least) two things to reproduce: the frequency range and the localisation. Ambisonics satisfies localisation cues up to about 5 kHz, and down to very low. (There is evidence that humans can localise frequencies below 20 Hz using things like chest cavities. Ambisonics can reproduce such localisation cues if these frequencies are in the source.) So, for localisation, the speakers can top out at 5 kHz. However, music played though such speakers is unlikely to sound very good. At the low end, lower is better. However, it is a law of diminishing returns, limited by the standing waves in your listening room which will interfere with the localisation of subwoofers. > I think part of my confusion is that i'm still thinking of it as having > 6 or 8 or more *channels* when that is not the case. The speakers are > not steered, they are driven. So you are not going to have a situation > where the speakers behind you are only reproducing high to midrange > information as was the case with Dolby pro-logic or something. Right? Right. With Ambisonics, all the speakers cooperate to localise a sound. While the speakers on the left push, those on the right pull. (This is less pronounced with frequencies above 400 Hz and less important with higher-order Ambisonics, but ignore this.) Regards, Martin -- Martin J Leese E-mail: martin.leese stanfordalumni.org Web: http://members.tripod.com/martin_leese/

14 years, 11 months

Re: [LAD] what does "full range" mean with regard to ambisonics and speakers?

by Martin Leese

Jörn Nettingsmeier wrote: ... >> it's no problem (and in fact customary) to use subwoofers with >> ambisonics for low-range extension. you can either forego localisation >> in that band and just use a mono sub (perfectly ok), use 2 and have some >> l/r signal in ithem, or you can use 4 subs in the corners of the room >> and drive them with a special horizontal decode. For subwoofers (but not main speakers) you can also use three. To see the various options discussed in more detail, visit: http://www.ambisonia.com/wiki/index.php/Bass_management Regards, Martin -- Martin J Leese E-mail: martin.leese stanfordalumni.org Web: http://members.tripod.com/martin_leese/

14 years, 11 months

(Off-topic) Audigy 4 and Surround

by JohnLM

Hi! This might be a complete offtopic here, but there hardly is another place to ask, where there are serious audio-tech savvy people. :) So I have Creative Audigy 4 sound card on WinXP x64. It's been a pain to set-up a working 5.1 surround system. In fact I've never done it. Has anyone got it right? I had no luck with Linux' ALSA either. I currently run it with the bloody Creative's CMSS, which is pretty good for stereo music recordings and most movies. I knew my 5.1 wasn't right, but until I tried ambisonics I was quite oblivious to the fact. So the actual problem. There is no real surround. The back speakers actually put out the same thing as front, with, what appears to be, some phase offset. This is most likely CMSS doing, but If I disable CMSS I get ouput only of two front speakers. Regardless of CMSS setting anything that is recorded as in "directly behind" doesn't play at all. With Linux and ALSA, there is a bit different problem. ALSA appears to cap the maximum amplitude below capability of the sound card, so I get "clipping artefacts" at *much* lower volumes than WinXP. So neither ALSA driver, nor music players were satisfactory to me and I never went though pains to get surround on it. I had not tried my luck with OSS4 yet. If it was up to me I'd use stripped down OSS4, which would provide direct access to 6-channel output. On top of it I'd put JACK and then route what I want, where I want. This is no small task however, and I'm not even sure if it can be done at all. I ask of you to share any experiences and ideas you have with Audigy, on any of these platforms. Thanks in advance! -- John

14 years, 11 months

Floating point processing and high dynamic range audio

by JohnLM

Hey, fellow devs! I'm a newcomer to this mailing list and I'm sure this has been discussed here at some point, but I'll ask anyway. I'm a bit better on programming side than on audio-tech side so... - Does all floating point formats mandate valid values in [-1.0, 1.0] range? - How does floating point (and perhaps fixed point/integer) values relate to dB notation? - If I muliply float by 2.0, do I get twice the sound pressure or twice the sound intensity, or something else? Well as for internal processing. If I mostly use integer formats for inputs, it doesn't appear I gain anything from "normalizing" them (scale to [-1.0, 1.0]) during internal processing. I just output integer part directly to integer formats (adjusted for different bit resolutions when necessary), and normalize only when saving to float format. What do you think? Proprietary formats such as AC3 and DTS market themselves as high dynamic range. Do they use values over 0dBFS or do they just have higher bit precision than "standard" formats? Thanks! -- JohnLM

14 years, 11 months

Re: [LAD] [LAA] announcing envy24control, mudita (*) edition.

by Jeff McClintock

> > Real detents require force feedback. A mouse doesn't > > provide that. This GTK thing is completely broken. Logitech iFeel mouses..would be cool if supported .. http://www.dansdata.com/ifeel.htm

14 years, 11 months

compiling hdsploader

by fons＠kokkinizita.net

Hello all, Anyone understands what is the problem here: fons@zita2:~/build/alsa-tools-1.0.23/hdsploader> make if gcc -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE=\"hdsploader\" -DVERSION=\"1.2\" -DSTDC_HEADERS=1 -DHAVE_LIBASOUND=1 -I. -I. -DDATAPATH=\"/usr/local/share/alsa/firmware/hdsploader\" -g -O2 -MT hdsploader.o -MD -MP -MF ".deps/hdsploader.Tpo" -c -o hdsploader.o hdsploader.c; \ then mv -f ".deps/hdsploader.Tpo" ".deps/hdsploader.Po"; else rm -f ".deps/hdsploader.Tpo"; exit 1; fi In file included from hdsploader.c:26:0: /usr/include/alsa/sound/hdsp.h:76:14: error: expected ‘:’, ‘,’, ‘;’, ‘}’ or ‘__attribute__’ before ‘*’ token hdsploader.c: In function ‘upload_firmware’: hdsploader.c:73:13: error: ‘hdsp_firmware_t’ has no member named ‘firmware_data’ make: *** [hdsploader.o] Error 1 TIA, -- FA There are three of them, and Alleline.

14 years, 11 months

Re: [LAD] twice as loud

by lieven moors

On 07/21/2010 07:24 PM, Fons Adriaensen-2 wrote: > On Thu, Jul 22, 2010 at 01:05:01AM +0200, Philipp Überbacher wrote: > > > I think the word loudness is a problem here. Afaik it usually refers to > > how it is perceived, and twice the amplitude doesn't mean twice the > > perceived loudness. It may mean twice the sound pressure level, energy, > > or intensity (if we ignore analogue anomalies, as you wrote in some > other > > answer). > > Subjective loudness is a very complex thing, depending on the > spectrum, duration, and other aspects of the sound, and also > on circumstances not related to the sound itself. > > For mid frequencies and a duraion of one second, the average > subjective impression of 'twice as loud' seems to correspond > to an SPL difference of around +10 dB. > > I often wondered what criterion we use to determine which > objective SPL difference sounds as 'twice as loud'. We don't > have any conscious numerical value (there may be unconscious > ones such as the amount of auditory nerve pulses, or the amount > of neural activity), so what it this impression based on ? Sorry to post this twice, but I think the mail got truncated. Hi Fons, I'm a fool to even try to answer this question. But I couldn't resist... Let's suppose we have two sounds A and B, and sound B has been measured as being twice as loud as A, by somebody. In order to be able to say that, that person needs some kind of reference measurement unit, the equivalent of a measurement stick. That unit has to satisfy two requirements. It has to be big enough, so that people can agree some difference is being measured, and it has to be small enough, so that a multiples of that unit fit into a realistic range. There is a requirement of maximum precision (the smallest value we can measure), and a requirement of minimum precision. The question is, what kind of measurement stick is being used by that person. First of all, we can assume that the length of that stick will be depend on the range of possible input values that we observe, and that we want to measure. If we want to measure the size of a road, we will probably use kilometers, instead of meters. In the same way, when our ears want to measure the amplitude of a sound, our ears will use smaller or bigger units, depending on the ranges observed. What are the ranges we observe? Let's assume that humans are perfect, and observe everything that we can observe with SPL meters. We could do a statistical investigation on a number of people, and make charts of everything they hear. In these charts we would see what frequencies they are exposed to, and what the minimum and maximum SPL's are for that frequencies. After more analyses, we would have one chart that could be representative for most people. >From that chart we could get an estimate of the size of the measurement unit. Frequencies with with bigger SPL variations would be measured with bigger units, and visa versa. And from this we could deduce what the minimum precision is for a certain frequency, when we say it is twice as loud. To satisfy the requirement of maximum precision, we should take into account the smallest observable differences for every frequency in the spectrum. now you can kill me :-) Greetings, Lieven

14 years, 11 months

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Linux-audio-dev