Linux-audio-user

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20713 discussions

[linux-audio-user] Assistance with splitting up stereo channels in ALSA is appreciated.

by Malcolm Baldridge

Greetings Esteemed Gentlemen: I've been hacking around with ALSA (1.0.3b), Jack [CVS as of two weeks ago], and other audio miscellanea and have found it all to work wonderfully. I've managed to add more tasks to my already overworked home server (which runs my entire networking/firewalling/routing and service operations). A summary of just the audio tasks of possible interest and background: I have a 1.6Ghz Pentium 4 (pre-Northwood) with the PC133 chipset (i840), which has 82801BA-ICH2 audio hardware with the usual AC97 codec. Kernel is 2.4.25-lck1 [Con's low-latency patches, as well as HZ=1000 timebase changes.] The audio output jack is connected to my MERLIN phone system's music-on-hold port, which is fed mono/downmixed output via "mpg123 -z -q -2 -m". I have each (L + R) channels of my LINE-IN jack connected to two Phone Patch boxes to monitor phonecalls "for Quality purposes" automatically. I had to write that recording software, which does configurable SQUELCH noise gating, and the like. The phone patch compensates for the differential levels of the send/receive sides of phone conversations, so the resulting mix has each side of the phonecall sounding about the same amplitude. It's the only way to do this. I found the Jack audio interface to be interesting and friendly, so I wrote the software to the Jack API, starting with one of the example client applications included with Jack. It's multi-threaded and event-driven, with callbacks from Jack when audio arrives. It works well. I then spawn off a child process to "lame -S --silent --nohist -q 2 -h -v -b 8 -B 32" with some ID3 tags representing the call's origin time, and any decoded DTMF tones, etc. This too works better than I imagined. Since it's spawned from the main process of my monitoring program, I don't tie up the audio monitoring/capturing threads during the encoding/processing operation. The jack and audio capturing threads run at RT priority, the rest run at normal priority. I run the encode jobs at nice +18. None of this skips ever, even during large file uploads/downloads from my 100BaseT connected SAMBA users, or during updatedb. This brings me to my question. It turns out that I should be sending a "beep" down the lines I am monitoring to notify the parties that they are being recorded. I have a spare output channel [the R channel of the main PCM device]. The L channel is providing music-on-hold, remember, via mpg123 which is using the ALSA driver. How can I configure ALSA (via asoundrc?) to split the PCM device into two independently accessible components for the L and R channels? I'm very pleased with mpg123 and do not feel I need to reinvent any wheels on that side. I don't feel it necessary to give Jack more to do, so I'd like to stay out of Jack if possible. (I run jackd with "jackd -R -d alsa -S -C -p 4096 -M -n 2 -r 11025"). The documentation for ALSA's asoundrc aspires for more than it can attain, alas. I suspect I need to configure some bindings, but it's not at all clear how I specify which channel in the device specification syntax. hw:0,0 refers the first device of the first soundcard, for example. But how do I specify the first (L) channel of the first device of the first soundcard? Ultimately, all that matters is Quality, =MB= -- A focus on Quality.

21 years, 10 months

[linux-audio-user] well well

by Aaron Trumm

Well guys, I have been up all night and you'd think I'd've been trying to recover my crucial data on my crashed harddrive on my main linux audio box. BUT NOOOO. I've been obsessing over ripping mp3s on my secondary box. So I spent some time failing at compiling jack cuz I hate grip on my machines cuz it's so slow and unreliable at least on my machines, and then said screw this and learned some bash and made my first usuable script tee hee. i'm so proud of my newbie self that I thought I'd share it. don't blame me if it blows you up. it shouldn't, it's a pretty simple inane cdparanoia/lame using script. fun. did I do anything horribly stupid? here: --- #!/bin/bash # Aaron Trumm's cdparanoia/lame mp3 ripping script # It simply takes all the tracks on a cd, rips, converts to mp3, and names them a standard file name # It asks you for the artist and album. # Very simple but useful for me. ARTIST= ALBUM= REMOVE= X= echo "Artist: "; read ARTIST echo "Album: "; read ALBUM echo "Delete wav files afterwards y/n? "; read REMOVE cdparanoia -B for X in 1 2 3 4 5 6 7 8 9 do if [ -e track0"$X".cdda.wav ] then lame track0"$X".cdda.wav "$ARTIST"_"$ALBUM"_"$X".mp3 fi done for X in 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 do if [ -e track"$X".cdda.wav ] then lame track"$X".cdda.wav "$ARTIST"_"$ALBUM"_"$X".mp3 fi done for X in 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 do if [ -e track"$X".cdda.wav ] then lame track"$X".cdda.wav "$ARTIST"_"$ALBUM"_"$X".mp3 fi done for X in 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 do if [ -e track"$X".cdda.wav ] then lame track"$X".cdda.wav "$ARTIST"_"$ALBUM"_"$X".mp3 fi done if [ "$REMOVE" == "Y" ] || [ "$REMOVE" == "y" ] then rm -rf *.wav fi

21 years, 10 months

[linux-audio-user] ANNOUNCE JACK-0.98.1 release

by Taybin Rutkin

These are just minor bug fixes to the configure script and header files. ***CHANGES*** Properly detect libreadline. Fatal error messages for missing get_opt_long(), dl_open(), and pthread_create(). No longer try to optimize for sse/mmx/altivec. Removed stale header from jack.h (jack_port_connect) JACK is available at http://jackit.sf.net Taybin

21 years, 10 months

Re: LiveCDs as Rescue Disks (WAS: [linux-audio-user] Ardour Crash+ can't boot)

by Matthew Allen

> -----Original Message----- > From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > user-bounces(a)music.columbia.edu] On Behalf Of Mark Knecht > > I recently ran DSL, Dynebolic, Gentoo LiveCD AND Knoppix in Frys on all > of the laptops I was considering until I found models that ran all 4 > CDs. This gave me a lot of confidence that once I got the laptop home > I'd be able to put Linux on it. It worked. I know watch DVD using Xine > and Alsa just fine. Aww man what an awesome idea! This sounds like 3 hours of weekend fun. And just in time for my laptop buying adventures. Thanks .matt _________________________________________________ Scanned on 26 Apr 2004 17:10:17 Scanning by http://erado.com

21 years, 10 months

[linux-audio-user] Re: linux-audio-user Digest, Vol 7, Issue 35

by philicorda

>Date: Fri, 16 Apr 2004 09:43:00 -0700 (PDT) >From: R Parker <rtp405(a)yahoo.com> >Subject: [linux-audio-user] sndfile-info and lots of questions >To: linux-audio-user(a)music.columbia.edu >Message-ID: <20040416164300.90068.qmail(a)web20916.mail.yahoo.com> >Content-Type: text/plain; charset=us-ascii > >Hi, > >As I've gone through my most challanging mastering >project, I've developed more questions than answers. > > Hope it all turned out right in the end! Only just caught up with the last email. I reackon Jamin could easily sort out the occasional uncontrollable kick drum. Perhaps in a two pass process, once to sort out that, then another pass for more general mastering. >There's been reference, on this list, to documentation >that explains file formats-- I don't recall the >document title or where to find it. Ultimately, I need >a resouce that explains things like; the number of >available samples for the different bit depths (16bit >range from -ABC to +XYZ, 24bit...), DC offset is, peak >amplitude is, RMS is, etc. > >I need to know if samples are syncronous with decibel >level, is maximum samples equal to 0db? > > Yes. Though 0dbfs is perhaps a better way to put it. Samples in 16bit= 2^16 Samples = -96dbfs range. Samples in 24bit= 2^24 Samples = -144dbfs range. Samples in 32bit float = I'm not sure if you count the Mantissa or not = Shockingly massive dynamic range. (Almost exactly :) ). >Of course another challange is tools like sndfile-info >assume that a file exists. This is not always the case >and in my situation it's almost never true. I return >JAMin output to an Ardour return bus and don't produce >a file until the return bus is exported. Printing a >track to the file system and then analyzing it is no >way to save time. > > I guess the only way some of those figures can be found is for the file to be printed. Ie, you can only tell what the peak sample is if you have compared it to every other sample in the file. To do it in real time, you would have to have 'Peak Sample so far', 'RMS so far' etc, and reset the figures manually when you started playback again. Not to say such a facility would not be useful. It would be nice if Jamin had a statistics page that showed you how many overs there had been, how many of them were consecutive etc. There is a new program called'Jackbitscope' http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/bitscope/ this has displays for highest and lowest sample so far. (And lots of other useful looking things I don't really understand. ) >Anyway, I appreciate all the responses to my past >questions and am hopeful that someone can look at the >current mumbo jumbo and prescribe some effective >medications; coffee, sleep, black bear gallbladders, >urls to useful documents, etc. > > Ground lion's teeth are effective in 90% of these cases in my experience. If not..... "0dbfs+ levels in digital mastering" http://www.studio-systems.com/audiofeatures/SepOct2002/Audio%20Article/0DBF… It's a couple of years old but still interesting, in that it covers what can happen when very hot digital levels hit the analog world. Also, the rec.audio.pro faq is chock full of useful stuff... http://www.faqs.org/faqs/AudioFAQ/pro-audio-faq/

21 years, 10 months

LiveCDs as Rescue Disks (WAS: [linux-audio-user] Ardour Crash + can't boot)

by Matthew Allen

> -----Original Message----- > From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > user-bounces(a)music.columbia.edu] On Behalf Of Mark Knecht > Sent: Sunday, April 25, 2004 8:50 AM > To: A list for linux audio users > Subject: Re: [linux-audio-user] Ardour Crash + can't boot > > And again, I think that having a good LiveCD for looking at these things > helps a lot. I don't know how many times I have used Damn Small Linux to save a bad Lilo or GRUB install. Last weekend I even used it to get data off a Disk that windows refused to boot with. I am seriously thinking about burning DSL to a credit card mini CD and keeping it in my wallet. .matt _________________________________________________ Scanned on 26 Apr 2004 16:37:33 Scanning by http://erado.com

21 years, 10 months

[linux-audio-user] eliminating track gaps

by Jack Bowling

I would like some confirmation that the two methods of eliminating the 2-second pre-gap when burning CDs is to: 1) Burn in DAO mode 2) cat all files into one humungous track. Any info on other options much appreciated. -- Jack Bowling mailto: jbinpg(a)shaw.ca

21 years, 10 months

[linux-audio-user] just for fun

by Dave Phillips

Greetings: I only recently realized that there's a JACK output plugin for XMMS, so I applied it and used qjackctl to wire XMMS to SuperCollider3 (running the BBCut ugen) for some good clean LAU fun. JACK is simply the bee's knees... :) Btw, I agree with Frank N. that Jesse C. should be properly chastized for introducing a new FreqTweak right before the LA conference. Jesse, I have work to do and I can't stop playing around with the new FT !! ;) Such cool stuff... Best, dp

21 years, 10 months

[linux-audio-user] RE: eliminating track gaps

by daniel.vincent＠blueyonder.co.uk

Jack Bowling: > I would like some confirmation that the two methods of eliminating the > 2-second pre-gap when burning CDs is to: > 1) Burn in DAO mode > 2) cat all files into one humungous track. > Any info on other options much appreciated. I would recommend "gcdmaster", which can take loads of individual files and stick them together and DAO them onto a CD, and can also take one file and break it into tracks for you. Recently I have concatenated a load of tracks using Audacity, sorted out the levels and EQ and then exported as a WAV, which I then loaded into gcdmaster and split back into tracks. YMMV obviously... daniel www.onionjack.com <http://www.onionjack.com>

21 years, 10 months

[linux-audio-user] RE: Suddeny, snd-usb-audio has loads of undefined symbols! (SOLVED?)

by David Baron

Simply, did not happen again. So... question: why? There are other sporadic and even consistent bootup errors which seem related to timing. Certain initrd-releated items busy and cannot be removed, etc. (Unfortunately, it goes by too quickly to write them down.) <<Did not upgrade anything touching this, too my knowledge. Doing a <<cat /dev/midi1 from the device connected to USB shows normal operation. <<What gives?

21 years, 10 months

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