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20661 discussions

Re: [linux-audio-user] New Kernel; No Sound. Linux-2.6.5 ALSA

by Martin McCormick

My thanks to a list member who wrote me off the list and told me about the aabuild and aadebug scripts on the alsa site. I ran the aadebug script and found that the biggest problem seems to be that ALSA couldn't find my sound card after plug and play registered it. Putting in the sound card driver as a built-in to the kernel rather than a module seems to have cured that problem and I may actually have it working. Now, for the next part: Since ALSA is part of the kernel, I need some command-line tools to take the place of aplayer and amixer which were in the old distribution. As a computer user who is blind, I do not use X windows because there isn't a truly working interface to X that makes it talk to us yet. It is in the works, but it has been there for about a decade although it really is getting a lot closer. I presently have aumix on the system and it did adjust some of the controls on the sound card under the 2.4.19 kernel and the OSS modules, but I gather that it doesn't talk to /dev/mixer, etc. The old ALSA utilities I had on here had Amixer and aplayer and dated back to December of 2001. They don't even try to work with the present setup. So, my question for now is, are there any command-line-based applications that replace aplayer, amixer and something I saw called alsamixer? This is all a bit confusing since there is a lot of software that is based on the old ALSA which doesn't seem to do anything at all in the new setup. Thanks to all. Martin McCormick

21 years, 7 months

[linux-audio-user] isolate the bass line

by millward

Could someone advise me on what software to use to isolate the bass line in a .wav file? I'm running Linux Red Hat 9.0 but I have a friend with an XP machine, so anything that works on either would be OK. That is, if it's not too expensive.

21 years, 7 months

[linux-audio-user] M-Audio Quattro and 2.6

by metafor＠gmx.net

Hi Is there any posibility to get this card working with 2.6 kernel? If not, is there some resource on the net to get it working under 2.4.xx with jack and pd, because as far i was not able to get it running without a lot of noisy clicks. any ideas? thanks marc

21 years, 7 months

Re: [linux-audio-user] New Kernel; No Sound. Linux-2.6.5 ALSA

by Martin McCormick

tim hall writes: >Now you need to configure it >I have it in /etc/alsa/modultils/ >then do 'update-modules'. I am still not there yet, but I keep finding new reasons why. update-modules does nothing but exercise my fingers and return a new shell prompt without so much as a single squawk about anything. Unfortunately, there is no alsa directory in /lib/modules/2.6.5 although one can sure see all the modules in their proper directory. The whole problem stems from the fact that the module-management environment was and still probably is broken. I installed Debian Linux in January of 2002 and things were different then. Last night, I discovered that depmod which gets its replacement if you install module-init-tools, was a symlink to the old depmod from then. Also, modprobe is another symlink between /sbin/modprobe and something called insmod which documentation says is for inserting modules in to a running kernel. Two years ago, you just inserted modules but never removed them. The code to remove them was there, but experimental. I took the liberty of re-doing both those links so they point to the new versions of depmod and modprobe. I then built a new 2.6.5 kernel, made the modules and installed them. There was no complaining from the process at all. I then did another update-modules and, as is the hallmark of this quest, absolutely nothing new happened. References to /dev/[anything audio] produce "no such device" errors. Another blast from the past that has been haunting me all this time was that I had unsuccessfully tried to get alsa going in 2002 when it was a separate application. It was something like /etc/init.d/alsa and was started and stopped at boot and shutdown by a script in /etc/rc2.d and another in /etc/rc1.d called S20alsa. Those were still trying to work. They are gone now. At this time, that is where I stand. I did try another MAKEDEV.snd script run from the scripts directory in the 2.6.5 source, but that just repeated the same futile action as before. The audio devices all exist, but return "no such device" errors when used. I think it is really funny that usually, it is the hardware end of the setup that is problematic, but log messages show PNP finding the right card and the correct driver being registered. The only thing I see on boot that even looks pathological is a depmod message complaining about something called AF_PACKET not being found. I don't think that relates to sound, so one thing at a time for now.:-) What else could it be? I've sure learned a lot about how modules do or don't get installed, but I bet there is something really stupid behind this whole standoff. I'd say someone stupid, but that's getting close to home. In my straw-grasping mode, I wonder if I should rebuild module-init-tools after discovering the depmod and modprobe links weren't right, but somehow, that doesn't sound really necessary. Any more ideas besides starting from scratch? Many thanks. Martin McCormick

21 years, 7 months

[linux-audio-user] USB Midi Interface - which one to choose?

by Adrian Gschwend

Hi all, I think this is one of the questions which rises on a regular base here but I couldn't find something that maches my requirements so far so I ask as well :-) (is there a FAQ for this list somewhere?) I am finaly rebuilding my studio based on Linux-only apps so I need some compatible hardware as well for my Synths. I still have quite a lot of MIDI devices so I am looking for HW with as many MIDI I/O's as possible. Actually I don't care how I have to connect them to my PC but I would prefer an USB device or a PCI card. What options do I have? Thanks Adrian

21 years, 7 months

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

by philicorda

"Hi, Maybe some folks on this list have insites into the following conversation between Steve and myself. I need all the help I can get. The last three paragraphs are a topic for which I am woefully ignorant." Hiya. I do the occasional master for releases, and "make it louder" is an all too common request. Whether it suits the track, the place in the album or the recording does not matter, they just want it LOUD. It's sad, but he who pays the piper.... Yes, you have to mash the audio (as you have found), and it kills me to hear carefully recorded tracks being brutally handled, so the idea is to do it as kindly as possible. :) Here's some rambling about how I go about it. First, where is all the energy in the track? What is taking up all your headroom? Subsonics can really take up headroom, so going back to the mix and high passing any tracks that don't need bass end can give you more space to work with. Starting at 20hz on most tracks and work up from there. Then, put a fast limiter across your main outputs, set it so it's fairly hammering the track, and listen for the moments when it really ducks. Look what's going on in the track at those points and automate/eq those parts. Much of getting a 'loud' track is in the mixing. Pretty much any limiter will work here. You *want* it to be offensive. :) My chain for mastering is normally - EQ/Exciter -> Comp -> Multiband Comp-> Limiter -> Dither. The first eq gets the general shape. Don't move on till you are happy. Try a steep highpass at 10hz or whatever here too. Creep it up till you can hear it then back off. Getting this right really depends on your monitoring. If your monitors can't do anything below 40hz and you are mastering electronic/dance be careful as there can be a lot going on down there. Listening from outside the control room with the door open seems to help here. I have no idea why. The next compressor is optional. If the tracks fairly dynamic I'll use one. It will subtle though. It's a good way to make the multiband more predictable when going for volume too. Perversely, setting a very low threshold works here. You'd think that would mash the track, as the compressor is working nearly all the time, but as the ratio is so low (1.2:1 or whatever) combined with a long release (1 sec or greater) it just smooths out the dynamics before it hits the multiband. Soft knee is essential here. You should not hear this 'working' at all ideally. An opto comp hear will let some peaks through but still bring up the general ambience in a pleasent way. You *want* to be using a multiband comp for volume. Try finding the range where the vocals are and fitting your middle multibands around that. You need to be able to solo bands to do this ideally. This means that the bass end interfere won't with the vocals, so they don't duck when there is heavy kick or bass. If the vocals remain up front you are half way there. If you have more bands, try finding the space between the kick and bass. Don't go too heavy on the top end bands, they should still have punch. As a rule of thumb, I play the track through the multiband, and play with the thresholds so each band is doing no more than 3-4db reduction at their respective loudest points, perhaps less on the upper bands. Then, play with the input level to the multiband to see what you can get away with. You can get away with more compression in the low end than the high end, and the more you control that low end the less the limiter has to try and control your kick/bass etc. Also, using very fast attack/release in the upper bands can work well, but don't kill the snare. Try soloing the low band and reducing the attack release until you begin to hear distortion, then back off a long way. Now the final limiter. Ideally, the multiband is doing most of the work and the final limiter is not working hard. If you try and get all your level control out of that limiter it's going to be working too hard and everything will mash up. I can not get the same amount of clean limiting out of any apps on Linux as I can out of a Finalizer or Waves L2. That's just Life. :) If you are doing anything more than 6db reduction here, something is wrong earlier in the chain. Overloads on your final master. Take care. If there are too many flattened peaks you can end up with a CD that sounds OK on your player, but starts getting unpleasent on older/crappier ones. I have had this happen, even with stuff that is not actually digitally clipped, just heavily limited. You should not need to do hard digital clipping at all, ever. You will end up with high frequency hash that will make radio station's limiters do odd things and some CD players cry. If you are not getting the volume you need in other ways, buy or rent a finalizer and stick it through that. It's got automatic wizards to get you most of the way there. It's what everyone else does. If you do want to drive it a bit, try mastering to a decent 2-track tape and pushing that. Depending on the machine it'll get rid of peaks wonderfully. Just mastering on to tape at a normal level and re-recording+normalising the result can get you a few extra dbs without any obvious change to the sound. A little clipping on an analog desk is not a problem. If it sounds good, go with it. :) Just remember to keep an un-mashed master for when the current passion for square-waved CDs dies away. Your clients will thank you for it someday. But really, a Finalizer does this kind of thing really well. They don't sound great but they do the job for volume.

21 years, 7 months

[linux-audio-user] (no subject)

by Martin McCormick

Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group Subject: Re: [linux-audio-user] Fwd: [Jamin] Re: soft clip: Achieving Gain, inconsequential overloads -------- Steve Harris writes: >Actually thats not quite right - AD converters generally work in 1's or >2's complement, depending on the brand/variety, so the middle point is at >0, and the range is slightly larger in the -ve voltage range. Correct. We may be talking about the difference between signed and unsigned number representation. If you have an unsigned value, the only possibilities are from 0 to the maximum value represented by all ones. There are no negative numbers. When you have a signed value, then an 8-bit number, for example can be represented by +1 to +127 or -1 to -128. Now, here is something I am a bit unsure of, myself. I know, from actual observations that a straight PCM output from your basic A/D converter if read as unsigned numbers moves in steps from 0 to all 1's on. I honestly have not tried to interpret those data as signed numbers because it wasn't convenient at the time. If one wants to have a valid representation of what the wave form is doing for graphical or calculation purposes, then the mid-point would have to be what one would call 0 level with -1 being one below and 1 being 1 above, etc. I am certainly not arguing with anyone, but am a wee bit confused as to the correct way to represent the numbers. This is probably off-topic, but any graphical software that shows you your music wave forms or does DSP functions in an arithmetic manner that is based upon an AC model has to behave as if the mid-point value was 0. Maybe someone can set me/us all straight in this manner. G

21 years, 7 months

Re: [linux-audio-user] New Kernel; No Sound. Linux-2.6.5 ALSA

by Martin McCormick

tim hall writes: >It sounds like you've installed ALSA ok. >Now you need to configure it >I have it in /etc/alsa/modultils/ >then do 'update-modules'. I am logging in to the system in question from my work place, but I see that the path described in /etc/modutils/alsa points to a directory that doesn't yet exist. Somehow, I missed the update-modules step. It is elegant how that actually works since by using the uname command, one's module path changes when the kernel does so you automatically get the correct set of modules attached to your kernel no matter what it is. anyway, I can try it all out when I go home tonight. I bet you have solved the puzzle. A thousand thanks. Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group

21 years, 7 months

[linux-audio-user] New Kernel; No Sound. Linux-2.6.5 ALSA

by Martin McCormick

I have spent this weekend attempting to get ALSA running on a Dell Dimension that uses the CS4236+ sound driver. Surprisingly enough, the hardware appears to be working in that PNP finds the sound card. I built the 2.6.5 kernel with modularized support for ALSA and OSS compatibility under ALSA. When I got to the make modules_install stage, I got a torrent of error messages from depmod and a link to the document post-halloween-2.6.txt. that and a trip through the kernel documentation got me to install module-init-tools-3.0 which appears to be the newest version. That, when run, got all the dependency problems taken care of and I see no errors at all now when installing modules. It looks for all the world like it could work, but it doesn't. Another trip through the ALSA section of the 2.6.5 kernel tells me to use MAKEDEV.snd in the scripts directory. I did that and it successfully re-made all the audio devices as symlinks to similar names appended by a number. /dev/dsp or /dev/mixer got linked to /dev/dsp0 and /dev/mixer0. Makes perfect sense to me. I thought, "Now it will surely work!" Nope. Here is what I can tell so far. Nothing complains upon boot. I do have alsa-base installed as a Debian package and it did install without problem. One trouble-shooting suggestion I read said to look in /proc/asound/oss/sndstat to see a list of the cards. There is no such list so that is significant, but I am not sure what causes that. I play around a lot with /dev/dsp to record low-fidelity audio and even wrote an application in C that simulates a voice-activated recorder to catch amateur radio and police scanner traffic. You can normally just cat /dev/dsp >some_file and it will dump 8,000 8-bit samples per second of PCM audio to your file. It at least used to work under the 2.4.19 kernel I ran for a couple of years so I know the sound card can work fine if it sees the right setup. Now, if I try to do anything with the audio devices, the complaint is "no such device." The only complaint I see from the system except for when I try to use sound occurs on shutdown. Broadcast message from root (ttyS1) (Sun Apr 11 07:54:31 2004): The system is going down for reboot NOW! INIT: INIT: Sending processes the TERM signal ALSA detected, but alsactl not usable, carrying on with aumix. Saving mixer settings: failed. ALSA driver isn't running. To be a little more clear, if I look for /proc/asound/oss/sndstat, there is no asound directory under /proc although there are several other directories. ALSA just isn't getting that far yet. Thanks for any constructive ideas. I did a check of the archives, but didn't see anything that sounded exactly like this problem although that could just be that I picked the wrong search terms. to connect the dots, I Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group

21 years, 7 months

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

by Martin McCormick

I am new to this list and got in in the middle of this discussion, but Here are some thoughts on 0 VU and what it means in the digital domain. You probably already know that an A/D converter when fed silence normally produces an output that is exactly halfway between the number 0 and 0xFFFF for a 16-bit converter. For a converter with more or less bits, the maximum reading is all bits on so it is 2^N power. The halfway point makes the D/A converter output a voltage that is halfway between 0 and the maximum peak voltage that particular A/D converter can put out. All that being said, the A/D converter can't go so much as one bit more than maximum or one bit less than minimum so there is absolutely no head room. If you send out a sine wave to the A/D converter, any part of that wave that goes above the maximum input threshold or below the minimum level will cause the output to flatten out. If you graph the levels, you would see a sine wave with a flat spot on top and another on bottom. A little flat spot probably doesn't hurt anything if it happens infrequently, but the more it happens, the worse the distortion. In other words, if you stay below the hard limits, the distortion is next to nothing, but it hits big time when you reach the limits of the sample value the A/D converter was designed to read. As to how you can tell when you are flat-topping, one could run a program designed to read the data for each channel and look for extreme limits such as all 0's or all 1's for one or two samples in succession and register this. In the amateur experimenting I have done with writing level detectors for 8-bit audio, I look for low values between 0 and some arbitrary value such as 1 to 3 and also values between 0xFF and maybe 0xFc to 0xFE. If I see any of those, the alarm goes off because that is darn close to the limit and may actually be the limit. I haven't messed much with stereo 16-bit audio such as a .wav file, but the bits of the left-channel sample and the right-channel sample are interleaved so you'll have to reassemble the 16-bit word for each channel to test it. Cheers. Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group

21 years, 7 months

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Linux-audio-user