Linux-audio-user May 2004

linux-audio-user@lists.linuxaudio.org

154 participants
165 discussions

Re: [linux-audio-user] Indexing the Read Pointer in /dev/dsp

by Martin McCormick

Paul Winkler writes: >> That sounds impressive. Someone correct me if I am wrong, but >> Jack requires X windows for just about everything. > >Not at all. Many of the clients do, but the server certainly doesn't. >Ecasound is a featureful client that doesn't either. Thank you. I am glad to be wrong about things like that. I started reading some about Jack and thought it was totally X. I guess you can say I don't know Jack. Martin McCormick

21 years, 2 months

Re: [linux-audio-user] Indexing the Read Pointer in /dev/dsp

by Martin McCormick

Malcolm Baldridge writes: >Funny, I've written a simple program (derived from the Jack "simple_client") >recent to do something similar. That sounds impressive. Someone correct me if I am wrong, but Jack requires X windows for just about everything. As a computer user who is blind, X is just not quite ready to productively use yet. My program uses the raw /dev/dsp PCM audio device. Without any IOCTL directives, this device reads and writes a 8 thousand sample-per-second stream of 8-bit audio. It is great for communications audio and no good for anything else.:-) My program idles by reading the samples and looking for any that are a given value above or below 0X80 which is the value one gets from the A/D converter under silence. If there is a swing through 0 to the other side of the cycle, a flag gets set. I make sure that several 0 crossings go by to avoid transient trips and then I set the VOX delay timer which tells the rest of the program to store samples in the output file. There is a buffer between the input and output so as to preserve the wave form and not chop it off. I burn roughly 25 megs for each hour of captured audio. I don't see anything wrong with making a new file for each recording, but I think there is a limit to the number of files one can have in a directory. That is the only pitfall I can see. There is one more bell to this program. If the sample read is 0 or 1 at one extreme or 0Xfe or 0Xff at the other, I send the bell character to warn that the audio is probably going in to clipping and to reduce the input level. The warning functions like a peak indicator. >What it does (for now, I'm sure I'll be adding more to it) is: > >1) Monitor the command-line specified input port for a sound above the >squelch level. > >2) Apply a configurable DC Bias adjustment [squelch comparison is performed >after this] > >3) Keep track of the peak samples processed to pass a "scaling value" to >normalise the sound data. [This is also post-DC Bias adjustment.] > >3a) I'm also looking at various dynamic compression [gain reduction] strategie

21 years, 2 months

Re: [linux-audio-user] Fwd: [Jamin] Re: soft clip: Achieving Gain, inconsequential overloads

by davidrclark＠earthlink.net

Hi Mark, > No, in my experience it doesn't favor louder always. I'm talking about historical trends, not individual experience, individual instruments, every single genre, etc. > had pictures of how volumes have grown over the years That's it! > I do not for a second think that this has all happened because the best > producers were doing a bad job 15-20 years ago. I agree. (Doesn't everybody?) > I'm just saying that I've not heard this argued in any other forum. That's why I brought it up. > Where do these come from, and why? See my earlier posting here (nonlinear processes in inner ear). I'm not claiming that everybody prefers this under all circumstances, just that there MAY be a general preference for denser spectra that drives this loudness increase. I also suggest that a preference for better dynamics occasionally reverses this trend. Other explanations (for the gradual increase in loudness) I've seen are not very satisfactory, esp. "Oh, it's just fashionable." I think it's related to the general trend towards richer orchestration which occurs in many genres.

21 years, 2 months

[linux-audio-user] Audacity/Windows/Plugins

by Mark Knecht

Hi, Sorry, but a very non-linux question. I wonder if anyone here can say anyting about how the Windows version of Audacity works? Is it stable? Does the VST enabler work? Is it stable? Does it only provide for audio suite/non-realtime operation on Audacity Audio, or does it allow Audacity to become a VSti platform of any type? Thanks in advance, Mark

21 years, 2 months

[linux-audio-user] [ANN] Linux Audio Conference #2 live streams, cams and chat available tomorrow

by Joern Nettingsmeier

hi everyone! for those who have not heard it yet, the second international Linux Audio Conference is taking place at the ZKM Karlsruhe/Germany from 29.4. to 2.5.2004. see http://www.zkm.de/lad/ for details. we have a number of very interesting presentations, all of which will be streamed out live, for the unlucky folks who can't be here in person. additionally, you will be able to download the presentation slides in advance should you wish to follow a lecture. there will be feedback channels on IRC, operated by folks who are in the lecture rooms. they will relay questions from you to the live audience. if all goes well, webcams will upload still images every 30 seconds to give you an idea of the ambience and of which slide is currently up. all important information on streaming relays, downloadable material, irc channels etc. will be dumped to http://linuxaudiodev.org/eventszkm2004.php3 . this page will be updated very frequently during the next days. the streams won't be up until tomorrow morning, but the chat rooms are already there. please forward this mail to any interested people. and no, we do not fear the slashdot effect :) enjoy, joern

21 years, 2 months

[linux-audio-user] Indexing the Read Pointer in /dev/dsp

by Martin McCormick

I have the beginnings of a program I have written that listens to the digital stream from /dev/dsp and records sound to a file when there is any sound. Communications types call this a VOX or Voice Operated relay with X being the abbreviation for relay. I want to also make a log of the time each recording started which is relatively easy to do especially if one uses the extensive time and date functions in UNIX. I think I can even store the stamps in the binary file with the audio. Retrieving those data in a meaningful manner, however, seems to be a problem. I want the playback program to display the time of the start of each recording as it begins to play. The /dev/dsp device is buffered so what really happens when you play a file is that the file pointer runs ahead of the actual byte being pushed through the sound card at any given time. The operating system blocks any more data being stuffed in to /dev/dsp, but the byte being pushed in may be 64 Kbytes ahead of what is playing right now. That's the problem. If I wrote a program to display the time a recording started, you would see that stamp some variable amount of time from instantly to maybe 8 seconds ahead of when the corresponding sound came out the speakers. I have been experimenting with methods of editing binary files and one has the same problem there in that when you reach the place you wanted to cut the virtual tape, the data which are flowing in to /dev/dsp are about 8 seconds past that point and it is hard to tell exactly where to edit. Is there any function that returns some offset or index value as to how much buffer is left? The trick to putting time stamps in with the audio should be to not ever allow a value of 0 to be stored from the A/D converter. When a new recording starts, store a NULL or 0 in the file followed by the 32-bit time and date value that UNIX systems store. Audio can follow after that. When playing back the file, look for the 0 and then use the next 4 bytes to recover the time and date. This would be a piece of cake if not for that variable buffer. The buffer is essential, however, for proper sound. I think that this buffer is actually the DMA channel at work so an indicator that it had run out or would soon run out would also be useful for those who want to synchronize sound with other activities. I have been playing with the VOX program for a couple of years and it works pretty well for recording radio transmissions but two transmissions could be anywhere from a quarter of a second apart to days apart. The program neatly cuts out the silence, but one has no idea when each recording actually started. In amateur radio, such information is useful when unusual skip or propagation conditions are present or in cases of malicious interference or malfunctions when one is trying to document when a particular event happened. For those of you used to recording high fidelity, the 8 seconds I refer to is with mono audio at 8,000 samples per second. Other sample rates and sample widths will certainly de pleat the buffer much faster. Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group

21 years, 2 months

[linux-audio-user] All-Soft audio question

by Aaron Trumm

Hello all I would like to know if anyone knows how I would set up a fluxbox/gentoo system to stream 96khz wave files to multiple websites, remotely, without installing too much audio software or hardware - there isn't room for an actual machine, so I'm wondering if I can set up a computerless Linux system, ie: just software no, hardware, just gentoo, fluxbox, and the wave files, that I could hopefully control remotely from my laptop via wireless ethernet - but again, only wireless ethernet protocol software, not hardware - can't afford that hardware and Linux is open source so I figured if I did a totally soft system, had all my waves, and fluxbox and gentoo, and controlled it via my laptop, then I wouldn't have to buy more hardware and I wouldn't have to find space for the server, because the server would be hardwareless - this would make the applications machine independant too I guess, so I was thinking I might develope using my totally soft linux system, as well. Also I can't install fluxbox or gentoo because I want it to run quickly and quietly in the background, so just a kernel and no x window system, just a kernel EMULATING an x window system, again, remotely, via the All-Soft wireless network connection (All-Soft meaning only software, no hardware) Also, if I could also be pointed toward a way to encode my 96khz wave files at 92, 89, 43.2 and 5000.66 KhZ that would be great... Thanks!

21 years, 2 months

[linux-audio-user] Re: Fwd: [Jamin] Re: soft clip: Achieving Gain, inconsequential overloads

by davidrclark＠earthlink.net

Tim, > I don't think it's psycho-acoustics, distortion creates more harmonics, > that's physics CMIIW. "CMIIW" --- OK: While it is well-known that distortion creates more frequency components, in a classic experiment done in 1924 by Wegel and Lane [1], it was shown that due to the nonlinear processing done by the inner ear, additional tones could be heard provided that the intensity was loud enough. That there are psychophysical effects which result in "louder produces more harmonics" has long been established. If there are also other nonlinear effects beyond this (nonlinear processing by the brain), then there may also be psychoacoustic effects that result in "louder produces more harmonics." This is very likely, not nonexistent. -------------------- [1] R. Wegel and C. Lane. The auditory masking of one pure tone by another and its probable relation to the dynamics of the inner ear. In Physics Review, volume 23, pages 266-285. Cited in Curtis Roads, The Computer Music Tutorial, MIT, 1996.

21 years, 2 months

Re: [linux-audio-user] Fwd: [Jamin] Re: soft clip: Achieving Gain, inconsequential overloads

by davidrclark＠earthlink.net

Hey Mark, I don't think it's any accident that people who like distortion guitars also like them loud. So I think there's more to it that "I'm louder than you." But yes, there certainly seems to be some of what say there, unfortunately. It forces some of us to crunch our stuff more than we want to. Maybe we should "Just say 'No'?" :-)

21 years, 2 months

Re: [linux-audio-user] Fwd: [Jamin] Re: soft clip: Achieving Gain, inconsequential overloads

by davidrclark＠earthlink.net

Some speculation on the preference for loudness: I almost posted something on this previously, but decided not to. I suspect that the "loud is better" actually comes from a desire for denser spectra and that strategies for providing denser spectra have been thwarted by some people simply increasing the volume. Increasing the volume can produce the illusion of a denser spectrum be increasing the amplitude of some frequencies so that they can be heard, or it may simply cause more things in the room to rattle. But once you set the volume for the louder songs, those that actually have denser spectra may sound as though they don't because many frequencies fall below the threshold and/or don't rattle anything. They sound thin again. So people creating dense spectra ALSO need to increase the volume to keep up with the volume settings, sad to say. There may also be a psychoacoustic effect that louder produces more harmonics or an illusion of such. Two trends in the history of pop music that seem to support this denser spectra idea are 1) Electric guitars; 2) Wall of sound. Occasionally people back off from these, but the pop charts are full of songs with dense spectra. I don't know about others, but every time I hear a female solo voice, I am expecting to hear it followed by something with a dense spectrum such as power chords. There are those who, of course, do not like this and prefer purer tones, but even these people often prefer richer tones that have lots of harmonics, such as organs, over simple sinusoids. If you look at the development of any particular genre, you can also see (hear) an increasing spectral density. Often this is accomplished by simply adding more instruments.

21 years, 2 months

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Linux-audio-user May 2004