Linux-audio-dev June 2005

linux-audio-dev@lists.linuxaudio.org

120 participants
101 discussions

[linux-audio-dev] [ANN] GNUsound 0.7.2 released

by Pascal Haakmat

Hi, GNUsound 0.7.2 was released. This release fixes a few naggling issues and incorporates support for the gmerlin avdecoder, which provides the ability to load .AAC and Musepack formats (among many others). Changes from 0.7.1: Fixed undo issues with Mix Tool, Move Tool, Pencil Tool. Fixed ILLEGAL_MID_SIDE_FORCE error when saving a FLAC file at compression level 4. Fixed a few make/autoconf snags (Jens Koerber) Added support for gmerlin avdecoder (Burkhard Plaum) Updated config.sub/config.guess. GNUsound 0.7.2 is available here: ftp://ftp.gnu.org/gnu/gnusound/gnusound-0.7.2.tar.bz2 The GNUsound homepage: http://www.gnu.org/software/gnusound Thanks, Pascal.

18 years, 10 months

[linux-audio-dev] Two / Three HDSPM Cards?

by Audio Developer

Howdy, I am currently trying to get two HDSPM cards working with ALSA and JACK. I can use each card individually from each other using -Dhw:0 or -Dhw:1. However I need to get both cards to look like one card for a program that we are developing in house. From what I have read I need either a .asoundrc or a /etc/alsa.conf file. For the time being I have chosen to work with .asoundrc. Here is the problem. I build the .asoundrc file according to the help page in the ALSA documentation, however when I execute any program that calls that virtual card, I get the following error. (This output is from the PCM test program in Alsa-Lib) opteron:/usr/src/alsa-lib-1.0.9/test # ./pcm -Dplug:hdspmbig -c112 Playback device is plug:hdspmbig Stream parameters are 44100Hz, S16_LE, 112 channels Sine wave rate is 440.0000Hz Using transfer method: write pcm: pcm_multi.c:811: snd_pcm_multi_open: Assertion `!slave_map[sidxs[i]][schannels[i]]' failed. Aborted If I reduce the number of channels contained in the virtual card named hdspm_112 to 64 channels, the test will work fine. I have a suspicion that the problem lies in the fact that the virtual card holds more channels than are physically available to the first sound card (hw:0). I am sure that there is something painfully obvious that I am missing but I haven't been able to locate it in the last four days. I am including my .asoundrc file for reference. Any help would me much appreciated. Thanks, Matthew Simmons --------------begin .asoundrc ---------------------- pcm_slave.hdspm_s { pcm hdspm_0 } pcm.hdspm_1 { type hw card 1 } ctl.hdspm_1 { type hw card 1 } pcm.hdspm_0 { type hw card 0 } ctl.hdspm_0 { type hw card 0 } ctl.hdspm_112 { type hw card 0 } pcm.hdspm_112 { type multi; slaves.a.pcm hdspm_0; slaves.a.channels 56; slaves.b.pcm hdspm_1; slaves.b.channels 56; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave a; bindings.2.channel 2; bindings.3.slave a; bindings.3.channel 3; bindings.4.slave a; bindings.4.channel 4; bindings.5.slave a; bindings.5.channel 5; bindings.6.slave a; bindings.6.channel 6; bindings.7.slave a; bindings.7.channel 7; bindings.8.slave a; bindings.8.channel 8; bindings.9.slave a; bindings.9.channel 9; bindings.10.slave a; bindings.10.channel 10; bindings.11.slave a; bindings.11.channel 11; bindings.12.slave a; bindings.12.channel 12; bindings.13.slave a; bindings.13.channel 13; bindings.14.slave a; bindings.14.channel 14; bindings.15.slave a; bindings.15.channel 15; bindings.16.slave a; bindings.16.channel 16; bindings.17.slave a; bindings.17.channel 17; bindings.18.slave a; bindings.18.channel 18; bindings.19.slave a; bindings.19.channel 19; bindings.20.slave a; bindings.20.channel 20; bindings.21.slave a; bindings.21.channel 21; bindings.22.slave a; bindings.22.channel 22; bindings.23.slave a; bindings.23.channel 23; bindings.24.slave a; bindings.24.channel 24; bindings.25.slave a; bindings.25.channel 25; bindings.26.slave a; bindings.26.channel 26; bindings.27.slave a; bindings.27.channel 27; bindings.28.slave a; bindings.28.channel 28; bindings.29.slave a; bindings.29.channel 29; bindings.30.slave a; bindings.30.channel 30; bindings.31.slave a; bindings.31.channel 31; bindings.32.slave a; bindings.32.channel 32; bindings.33.slave a; bindings.33.channel 33; bindings.34.slave a; bindings.34.channel 34; bindings.35.slave a; bindings.35.channel 35; bindings.36.slave a; bindings.36.channel 36; bindings.37.slave a; bindings.37.channel 37; bindings.38.slave a; bindings.38.channel 38; bindings.39.slave a; bindings.39.channel 39; bindings.40.slave a; bindings.40.channel 40; bindings.41.slave a; bindings.41.channel 41; bindings.42.slave a; bindings.42.channel 42; bindings.43.slave a; bindings.43.channel 43; bindings.44.slave a; bindings.44.channel 44; bindings.45.slave a; bindings.45.channel 45; bindings.46.slave a; bindings.46.channel 46; bindings.47.slave a; bindings.47.channel 47; bindings.48.slave a; bindings.48.channel 48; bindings.49.slave a; bindings.49.channel 49; bindings.50.slave a; bindings.50.channel 50; bindings.51.slave a; bindings.51.channel 51; bindings.52.slave a; bindings.52.channel 52; bindings.53.slave a; bindings.53.channel 53; bindings.54.slave a; bindings.54.channel 54; bindings.55.slave a; bindings.55.channel 55; # Use hdspm_1 bindings.56.slave b; bindings.56.channel 0; bindings.57.slave b; bindings.57.channel 1; bindings.58.slave b; bindings.58.channel 2; bindings.59.slave b; bindings.59.channel 3; bindings.60.slave b; bindings.60.channel 4; bindings.61.slave b; bindings.61.channel 5; bindings.62.slave b; bindings.62.channel 6; bindings.63.slave b; bindings.63.channel 7; bindings.64.slave b; bindings.64.channel 8; bindings.65.slave b; bindings.65.channel 9; bindings.66.slave b; bindings.66.channel 10; bindings.67.slave b; bindings.67.channel 11; bindings.68.slave b; bindings.68.channel 12; bindings.69.slave b; bindings.69.channel 13; bindings.70.slave b; bindings.70.channel 14; bindings.71.slave b; bindings.71.channel 15; bindings.72.slave b; bindings.72.channel 16; bindings.73.slave b; bindings.73.channel 17; bindings.74.slave b; bindings.74.channel 18; bindings.75.slave b; bindings.75.channel 19; bindings.76.slave b; bindings.76.channel 20; bindings.77.slave b; bindings.77.channel 21; bindings.78.slave b; bindings.78.channel 22; bindings.79.slave b; bindings.79.channel 23; bindings.80.slave b; bindings.80.channel 24; bindings.81.slave b; bindings.81.channel 25; bindings.82.slave b; bindings.82.channel 26; bindings.83.slave b; bindings.83.channel 27; bindings.84.slave b; bindings.84.channel 28; bindings.85.slave b; bindings.85.channel 29; bindings.86.slave b; bindings.86.channel 30; bindings.87.slave b; bindings.87.channel 31; bindings.88.slave b; bindings.88.channel 32; bindings.89.slave b; bindings.89.channel 33; bindings.90.slave b; bindings.90.channel 34; bindings.91.slave b; bindings.91.channel 35; bindings.92.slave b; bindings.92.channel 36; bindings.93.slave b; bindings.93.channel 37; bindings.94.slave b; bindings.94.channel 38; bindings.95.slave b; bindings.95.channel 39; bindings.96.slave b; bindings.96.channel 40; bindings.97.slave b; bindings.97.channel 41; bindings.98.slave b; bindings.98.channel 42; bindings.99.slave b; bindings.99.channel 43; bindings.100.slave b; bindings.100.channel 44; bindings.101.slave b; bindings.101.channel 45; bindings.102.slave b; bindings.102.channel 46; bindings.103.slave b; bindings.103.channel 47; bindings.104.slave b; bindings.104.channel 48; bindings.105.slave b; bindings.105.channel 49; bindings.106.slave b; bindings.106.channel 50; bindings.107.slave b; bindings.107.channel 51; bindings.108.slave b; bindings.108.channel 52; bindings.109.slave b; bindings.109.channel 53; bindings.110.slave b; bindings.110.channel 54; bindings.111.slave b; bindings.111.channel 55; bindings.112.slave b; bindings.112.channel 56; } pcm.hdspmbig { type route slave.pcm "hdspm_112" ttable.0.0 1 ttable.1.1 1 ttable.2.2 1 ttable.3.3 1 ttable.4.4 1 ttable.5.5 1 ttable.6.6 1 ttable.6.6 1 ttable.7.7 1 ttable.8.8 1 ttable.9.9 1 ttable.10.10 1 ttable.11.11 1 ttable.12.12 1 ttable.13.13 1 ttable.14.14 1 ttable.15.15 1 ttable.16.16 1 ttable.17.17 1 ttable.18.18 1 ttable.19.19 1 ttable.20.20 1 ttable.21.21 1 ttable.22.22 1 ttable.23.23 1 ttable.24.24 1 ttable.25.25 1 ttable.26.26 1 ttable.27.27 1 ttable.28.28 1 ttable.29.29 1 ttable.30.30 1 ttable.31.31 1 ttable.32.32 1 ttable.33.33 1 ttable.34.34 1 ttable.35.35 1 ttable.36.36 1 ttable.37.37 1 ttable.38.38 1 ttable.39.39 1 ttable.40.40 1 ttable.41.41 1 ttable.42.42 1 ttable.43.43 1 ttable.44.44 1 ttable.45.45 1 ttable.46.46 1 ttable.47.47 1 ttable.48.48 1 ttable.49.49 1 ttable.50.50 1 ttable.51.51 1 ttable.52.52 1 ttable.53.53 1 ttable.54.54 1 ttable.55.55 1 ttable.56.56 1 ttable.57.57 1 ttable.58.58 1 ttable.59.59 1 ttable.60.60 1 ttable.61.62 1 ttable.63.63 1 ttable.64.64 1 ttable.65.65 1 ttable.66.66 1 ttable.67.67 1 ttable.68.68 1 ttable.69.69 1 ttable.70.70 1 ttable.71.71 1 ttable.72.72 1 ttable.73.73 1 ttable.74.74 1 ttable.75.75 1 ttable.76.76 1 ttable.77.77 1 ttable.78.78 1 ttable.79.79 1 ttable.80.80 1 ttable.81.81 1 ttable.82.82 1 ttable.83.83 1 ttable.84.84 1 ttable.85.85 1 ttable.86.86 1 ttable.87.87 1 ttable.88.88 1 ttable.89.89 1 ttable.90.90 1 ttable.91.91 1 ttable.92.92 1 ttable.93.93 1 ttable.94.94 1 ttable.95.95 1 ttable.96.96 1 ttable.97.97 1 ttable.98.98 1 ttable.99.99 1 ttable.100.100 1 ttable.101.101 1 ttable.102.102 1 ttable.103.103 1 ttable.104.104 1 ttable.105.105 1 ttable.106.106 1 ttable.107.107 1 ttable.108.108 1 ttable.109.109 1 ttable.110.110 1 ttable.111.111 1 } ctl.hdspmbig { type hw; card 0; }

18 years, 10 months

[linux-audio-dev] [ANN] QjackCtl 0.2.17 released!

by Rui Nuno Capela

Hi, QjackCtl 0.2.17 update release is out. Directly from the change log: - Systemic I/O Latency settings are now featured for the alsa, oss and coreaudio backends, letting you specify the known latency of external hardware for client aware compensation purposes (thanks to Wolfgang Woehl, for the reminder). - Update on last backstage changes to the coreaudio backend options (due to Stéphane Letz. Thanks). As always, you can grab the source tarball from the usual place: http://qjackctl.sourceforge.net/ http://sourceforge.net/projects/qjackctl Enjoy! -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

18 years, 10 months

Re: [linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

by Mickael Vardo

Hi there. >Actually a correct explanation isn't that simple. Yours is much >_too_ simple. Theoretically a 20 kHz bandlimited signal can be >represented _exactly_ as a 40 kHz PCM stream. In order to not >have to use a very steep lowpass filter in the DAC it is better >to use a somewhat higher sampling frequency. 48 kHz should be >enough most of the time. No. Nyquist frequency allow original _sampled_ signal to be fully reconstructed using a IDFT. It does not mean that the reconstructed signal will have anything more than the original samples. If you sample an incoming analog signal at 2 samples per second (sps), you'll definitely not be able to reconstruct ALL the phases and ALL the frequencies between 0 and 1 Hz. You'll get, however, a correct reconstruction of a periodic signal if it lasts long enough. This is due to the inherent impossibility, using a DFT, to get both accurate space and time resolutions at the same time. Just try this "simple" experience: sample a 1 Hz pulse that is triggered after a random non-quantized delay that is less than four seconds. Sample it at a rate of 2 sps and then, try to get the original signal back with all its components (phase and frequency). Good luck. This is just as impossible as saying that you can compress the information stored in any analog signal in a finite number of samples. Mickael Vardo

18 years, 10 months

[linux-audio-dev] New toys for Mx44

by Jens M Andreasen

Modulating "Magic" Numbers. ---------------------------- Given two instances of the rude pseudo sine fuction S: #define S(n)(((n)*((n)^0x8000))>>13)^(((short)((n)^0x8000))>>15) .. and offset them some 60 degrees with the "magic number" 0x14D4, then they will partially cancel each other out to get you something that *looks* like a sine wawe (but actually still have lots of harmonics.) Now, what then if one was to change the offset, I thought. That should result in different blends of harmonics, right? So I made this new tiny button between the waweform selector and the detune slider. Activating this button will do the following: Frequency offset (detune) will detune the two instances against each other. Phase parameters will change, not only the phase relative to the other oscillators, but also the offset between the two instances of S(). I've set up oscillator one in patch H3 ready for experimentation. Forgot to change the name though, so the title "op 1 on quack" is a reference to the sound in H2, which is actually pretty amusing ... but only If you are five years old! :D Old and deaf R&R farts will probably prefer the "fat boy" in E8. However, the implication is that I from this week can offer you two oscillators for the price of one only. Get'em from: http://mx44.linux.dk mvh // Jens M Andreasen

18 years, 10 months

[linux-audio-dev] Callback functions

by Allann Jones

Hi, Anyone knows how can I use callback builtin functions to operations like capture, playback and events using ALSA API? For a while I'd implemented calls to callback functions in each return of 'writei' and 'readi'. Thank you.

18 years, 10 months

Re: What Parts of Linux Audio Simply Work Great? (was Re: [linux-audio-dev] Best-performing Linux-friendly MIDI interfaces?)

by Peter Zubaj

Hi, Why not solve this kinds of problems. Soulutions: 1) Remove OSS drivers from kernel - then all problems will be alsa problems. 2) Remove OSS emulation from alsa - then all problems will be OSS problems. Otherways this will be newer end. Peter Zubaj ___________________________________________________________________________ Podte na navstevu k Wande - k najlepsej priatelke kazdej zeny na internete. http://www.wanda.sk/

18 years, 10 months

Re: [linux-audio-dev] Re: Looking for fast integer resampling code

by Kjetil Svalastog Matheussen <k.s.matheussen＠notam02.no>

Erik de Castro Lopo: > > Kjetil Svalastog Matheussen wrote: > >> I was primarily testing speed, but both libraries where using the sinc >> routine, so... > > Yes, but you can design a sinc filter with 10 coefficients and one > with 10000. The one with 10000 coefficients will have a steeper > transition band and a better signal to noise ratio. > Yes. But back to the point, I think there is an extremely good chance that mus_src (even by setting the width-argument to 5 (default value is 10 which is also faster than SRC_SINC_FASTEST)) will sound better than whats being produced by a linear resampler, and might be exactly be what the person asking the question is looking for.

18 years, 10 months

Re: [linux-audio-dev] How to access "real-time" capabilities

by Jonathan Woithe

Lee Revell wrote: > On Fri, 2005-06-10 at 11:12 -0700, Ian wrote: > > Can anyone tell me *very briefly* what is the current > > state of play regarding realtime-lsm in the 2.6.* > > kernels and whether this feature (or something like it > > ) is likely to be introduced into the kernel anytime > > soon. > > realtime-lsm should be considered deprecated. The solution accepted by > Linus is rlimit based. I think this will be in 2.6.12 (seems to be in > the release candidates). Check the archives for "rlimit". > > This will require patching PAM and bash. In the interim you could try set_rtlimit: http://www.physics.adelaide.edu.au/~jwoithe/set_rtlimits-1.1.0.tgz It allows you to access the enhanced rlimits (ie: RT scheduling and nice levels) under 2.6.12 kernel or later without needing to patch anything. In fact, even once PAM/etc are patched this might be preferred by some since it allows you to control the limits based on the program being run; so for example you can allow /usr/local/bin/ardour to run at realtime priority 50 while /usr/bin/yes (for example) can't. Regards jonathan

18 years, 10 months

Re: [linux-audio-dev] Re: Looking for fast integer resampling code

by Kjetil Svalastog Matheussen

Erik de Castro Lopo: > > > > I recently did a lot of benchmarking between libsamplerate and mus_src > > in clm/sndlib. My result was quite interesting, the fastest mus_src sinc > > resampler where a lot faster than the fastest libsamplerate resampler. > > I think most people would agree that speed is not the most important > aspect when measuring the quality of a sample rate converter. > I was primarily testing speed, but both libraries where using the sinc routine, so... > With libsamplerate, I can state that the three sinc based converters > have the following characteristics: > > SNR Bandwidth > SRC_SINC_FASTEST 102.42 dB 80.23 % > SRC_SINC_MEDIUM_QUALITY 98.99 dB 90.68 % > SRC_SINC_BEST_QUALITY 97.43 dB 96.96% > > where SNR is signal to noise ratio and Bandwidth its a percentage of > the theoretical best bandwidth (ie half of the minimum of the source > and desination sample rate). > > > A third program is the original sinc-resampler from Julius Smith: > > http://ccrma-www.stanford.edu/~jos/resample/ > > I don't know how this one performs compaired to the other two though. > > libsamplerate the same algorithm as this one. I think Julius O. Smith > developed this algorithm. > He did, and Bil Schottstaedt used this algorithm as well for mus_src. When I compaired mus_src with width=5, and libsamplerate with SRC_SINC_FASTEST, the first one is much faster, I don't know what kind of SNR/Bandwidth you get with your way of measuring when using mus_src with width=5, but the source is easy to read so I guess you can figure it out. --

18 years, 10 months

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Linux-audio-dev June 2005