Linux-audio-dev

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7501 discussions

[linux-audio-dev] Re: Re: What Parts of Linux Audio Simply Work Great?

by Lachlan Davison

> > 3. 'amateur' users vs. 'professionals'. > > Since a took a clear stand on this matter, let me define what > I mean by a 'professional' in this context: someone who makes > a living by providing a service to customers who pay for it. > I still maintain that someone in this position will not mind > logging in as different users for his personal and pro work. > He / she will probably use different machines anyway. It's > just a matter of 'best practice' and professional conduct. > > > > > As was already pointed out, prosumer and professional users > > > will in all probably have two different audio cards anyway. > > > At least for the hobbyist (and I can speak for them): > > disagreed :) . Just to join in... from a dance music making perspective i'm not sure if i understand this customer concept. Does your fav rock band use "best practice" and professional conduct? are they not professional? I hope at least some of the people on here are interested in making music too, and i hope we can focus on that. The key thing i've learned from my limited experience with music, more time making music and less time getting your computer to work means better music. So i think the goal really needs to be as Jay said, it just works. Though i'm coming from the just doing it for fun camp so take that into account.

20 years

Subject: Re: [linux-audio-dev] Re: What Parts of Linux Audio Simply Work Great?

by lazzaro

> On Sat, Jun 18, 2005 at 02:34:46AM +0200, Christoph Eckert wrote: >>> I think we should (and can) keep the desktop and 'pro' >>> worlds separate. >>> >> >> I do not agree :) . We're in the free software world, so >> there's no need to tell the non-pro-audio-users "use anything >> else". How OS X solves this problem may be instructive. [most facts below are probably right, I'm sure I got a few wrong ...] In the System Preferences->Sound menu, a user can choose the default audio input and audio output device. In the CoreAudio API, this choice becomes the current value of kAudioHardwarePropertyDefaultInputDevice and kAudioHardwarePropertyDefaultOutputDevice. Consumer-oriented apps use those Defaults, as a rule; content-creation apps usually have their own Preferences that let you select the audio devices for the app. CoreMIDI works this way too. So, the underlying system is "pro", but there are provisions in the API to handle the mindset of both pro and consumer worlds. Finally, note that Apple solved the legacy problem by emulating Sound Manager (OS 9 audio API). Until Tiger, QuickTime actually used Sound Manager. Many other apps still do. If Linux followed this model, one would write emulators over jack for all of the consumer audio APIs being used in jack, to accommodate the installed base, and have the emulations work well enough to convince distros to just ship jack as the bottom layer, and use its emulated APIs. Then, you can start evangelizing direct jack calls to all application developers, consumer and pro alike. Yes, this takes a lot of work, and is not in the direct path of solving pro audio problems. Just like I've spent 5 years on pushing RTP MIDI through the standards process because I felt someone had to do it, I think the Linux audio API problem only gets solved if someone decides to dedicate 5 years of their life to doing it. That's how long a systems project takes to have an impact on the world. Good luck. --- John Lazzaro http://www.cs.berkeley.edu/~lazzaro lazzaro [at] cs [dot] berkeley [dot] edu ---

20 years

[linux-audio-dev] alsa ethernet streaming?

by Garett Shulman

Hello, I would like push audio streams over ethernet and was wondering what avenues people have tried. I intend to have two linux boxes (or more). Box1 has the user interface and all of the audio data. Box2 has no user interface and is plugged into an amplifier and speakers. Eventually I would like to have many Box2's. I envision an application with a server running on box2 and a client running on box1. Ideally the client would create an ALSA input and the server would create an ALSA output. Then, the client connects to the server, any ALSA enabled app connects to the input created by the client, and audio is heard from the speakers connected to the server. I have no expectations that such an application actually exists. I'm curious as to what options do exists to push audio over etherenet. -Garett

20 years

[linux-audio-dev] Re: What Parts of Linux Audio Simply Work Great?

by Peter Zubaj

>What I mean is if you have your soundserver, you are still left with >rewritting every single app you ever want to use to use the soundserver, >otherwise the problem still remains. There are a hell of a lot of tools >around that use either plain OSS or direct ALSA. Why not simply use alsa api. alsa-lib is not dependand on alsa-driver. You can write(for example)esd alsa-lib plugin. All applications which uses alsa api can be redirected to esd using this plugin withou user notify this. If needed, it can go directly to soundcard or dmix. What makes alsa unique is alsa-lib (I personally thing this is best alsa part) not alsa-drivers. Peter Zubaj ___________________________________________________________________________ Podte na navstevu k Wande - k najlepsej priatelke kazdej zeny na internete. http://www.wanda.sk/

20 years

Re: [linux-audio-dev] Re: What Parts of Linux Audio Simply Work Great?

by Peter Zubaj

>this woud IMHO be the best solution, but it will not happen or >at least last very long until > >* all distros will have JACK running per default >* all developers of any audio program have rewritten their >code. Why not use only ALSA API. There is alsa to jack plugin. If someone wants use jack it can be used to route audio to jack. Also there are alsa <-> OSS driver plugins. This means you can output sound from alsa app to oss driver or capture from alsa driver without any source change. Peter Zubaj ___________________________________________________________________________ Podte na navstevu k Wande - k najlepsej priatelke kazdej zeny na internete. http://www.wanda.sk/

20 years

[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.

20 years

[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; }

20 years

[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

20 years

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

20 years

[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

20 years

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Linux-audio-dev