* Slashdot interview re: vorbis @ Geoff Shang ` Charles Hallenbeck ` Brent Harding 0 siblings, 2 replies; 25+ messages in thread From: Geoff Shang @ UTC (permalink / raw) To: speakup Hi all: About 4 or 5 weeks ago, Kirk got me interested in the new audio format called OGG vorbis. Yeah, strange name I know, but this looks really exciting. This is an opensource, GPL, patent-free codec that, in my opinion, already beats MP3. Anyway, the guy who's running the show was interviewed by slashdot, and it's an interesting read. If you can stick with it through the initial technical details, it gets more readable. You can find it at: http://slashdot.org/article.pl?sid=00/08/14/1034209 I've been chasing this band-wagon for a month already, so feel free to fire any questions you might have my way. Geoff. -- Geoff Shang <gshang10@scu.edu.au> ICQ number 43634701 ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis Slashdot interview re: vorbis Geoff Shang @ ` Charles Hallenbeck ` Brent Harding 1 sibling, 0 replies; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: speakup Geoff - Many thanks for this reference. I have also been playhing with audio formats a bit lately and am very much interested to check out this vorbis thing. Chuck My web site is http://www.mhonline.net/~chuckh Keep grandma off the streets -- legalize bingo. ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis Slashdot interview re: vorbis Geoff Shang ` Charles Hallenbeck @ ` Brent Harding ` Geoff Shang 1 sibling, 1 reply; 25+ messages in thread From: Brent Harding @ UTC (permalink / raw) To: speakup Is there a debian package yet out for vorbis? Can it record directly from the audio device, so I never need the wave file to encode it, that it encodes on the way? At 07:13 PM 8/17/00 +1000, you wrote: >Hi all: > >About 4 or 5 weeks ago, Kirk got me interested in the new audio format >called OGG vorbis. Yeah, strange name I know, but this looks really >exciting. This is an opensource, GPL, patent-free codec that, in my >opinion, already beats MP3. Anyway, the guy who's running the show was >interviewed by slashdot, and it's an interesting read. If you can stick >with it through the initial technical details, it gets more readable. You >can find it at: > >http://slashdot.org/article.pl?sid=00/08/14/1034209 > >I've been chasing this band-wagon for a month already, so feel free to fire >any questions you might have my way. > >Geoff. > >-- >Geoff Shang <gshang10@scu.edu.au> >ICQ number 43634701 > > >_______________________________________________ >Speakup mailing list >Speakup@braille.uwo.ca >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding @ ` Geoff Shang ` Charles Hallenbeck 0 siblings, 1 reply; 25+ messages in thread From: Geoff Shang @ UTC (permalink / raw) To: speakup Hi Brent: hahaha. No, no debian packages for vorbis yet, it's way way too new yet. However, some tools like oggenc can be downloaded as precompiled binaries. You know, you could solve your recording problem pretty easily by using arecord (if you run alsa which I think you do). Arecord is neat because it can record to standard output. This is cool because you can then pass it to something else that can use standard input, such as an MP3 or vorbis encoder. A command example for lame to encode MP3 for example might be: arecord |lame - recording.mp3 Another example using the oggenc program now available for vorbis: arecord |oggenc - -o output.ogg Note that I've seen instances of arecord dropping out when there's other HD activity. Whether piping to standard output would suffer similarly, I do not know. I would be interested if there are other wav recorders that will record to standard output. I really dig this ability to pipe audio. I like to encode stuff in formats like MP3, but I'm impatient. I want to encode at the lowest bitrate I can get away with that still sounds good. I don't want to encode the entire file before I decide I don't like that particular rate, so I pipe it to mpg123 and listen to some of it to see what I think. Note that this breaks up over a certain bitrate, depending on your processor power. Here's an example: lame -hb 160 input.wav - |mpg123 - One other comment on piping. The ultimate application for saving space as Brent alluded to is ripping tracks from CD. Cdparanoia will, you guessed it, rip to standard output. So you can use a command like this: cdparanoia -qz 1 - |lame -hb 160 - track1.mp3 This will encode track 1 on the CD to track1.mp3 using high quality mode and a bitrate of 160kbps. Geoff. -- Geoff Shang <gshang10@scu.edu.au> ICQ number 43634701 ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Geoff Shang @ ` Charles Hallenbeck ` Geoff Shang ` Kerry Hoath 0 siblings, 2 replies; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: Geoff Shang; +Cc: speakup Hi Geoff - After seeing your earlier note about piping audio streams with lame, I decided to give that program a whirl. I have been using bladeenc until now. I noticed after compiling and installing the August 6 beta release that it now supports encoding and decoding the ogg vorbis format as well as mp3. I am curious - is that a new feature? Have you used it? How does it work? I have not been too happy with bladeenc as an mp3 encoder because it requires a minimum of 32 kilobits sampling rate inputs, and the resulting mp3 file loses something in the translation. That is, if you only have 8k or 11.025 or 16k to start with, it seems wasteful to first artificially inflate the sampling rate to 32k in order to compress it with bladeenc, and lots of noise creeps into the signal along the way - probably with sox resampling. I am hopeful that lame can do a better job, and am excited to see ogg vorbis support too. Chuck. My web site is http://www.mhonline.net/~chuckh Keep grandma off the streets -- legalize bingo. ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Charles Hallenbeck @ ` Geoff Shang ` Buddy Brannan ` (2 more replies) ` Kerry Hoath 1 sibling, 3 replies; 25+ messages in thread From: Geoff Shang @ UTC (permalink / raw) To: speakup Hi Chuck: As the bladeenc site says, they're not about low bit-rate encoding. In my personal opinion, lame shits all over it. Bladeenc reckon they're better at rates over 128K, but I'm yet to hear it. Lame is certainly better at 128K and below. I don't have the August 6 beta, didn't know there was one actually. Lame does do an OK job at encoding vorbis streams, but only to a point. The vorbis code up until this week from vorbis.org only really supported encoding at higher bitrates like 160. Lame will do lower, but I think it uses it's MP3 engine and alters the output to encode vorbis instead. Don't get me wrong, it still sounds better than MP3 even produced by the same encoder, but I don't think it sounds as good as vorbis will sound at the same rates. I'd look out for another version of lame to hit the streets soon as Tuesday saw the release of vorbis 1.0beta2. As you may recall from the interview, this does 128K encoding and might even do 96, haven't looked. The program listed on vorbis.com for encoding has changed from an ogg-ised version of lame 3.84alpha to a program called oggenc. I haven't tried it yet but I do intend to. I might also say while I'm here that if you want a straight player, then I'd probably recommend freeamp. But, and it's a big but, you need the latest available code. There is a version labeled freeamp 2.1beta5vorbis, but the vorbis support doesn't work with any other than the X gdk interface. freeamp 2.1beta6 will fix this, but last I checked, it hadn't been released yet. Until it is, you will have to checkout the CVS code. This does work, believe me. I finally after 4 weeks of chasing it and E-mails to freeamp developers, have a freeamp that will play ogg vorbis in console mode. If you want some content to listen to, check out www.vorbis.com. There is a They Might Be Giants song at 96K that sounds quite impressive, and it's only 1.4 meg. Geoff. -- Geoff Shang <gshang10@scu.edu.au> ICQ number 43634701 ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Geoff Shang @ ` Buddy Brannan ` Charles Hallenbeck ` Brent Harding 2 siblings, 0 replies; 25+ messages in thread From: Buddy Brannan @ UTC (permalink / raw) To: speakup Yeah, and lame is quite a bit faster than bladeenc, I think. Certainly seems to be, far as I can tell, but I just installed lame today. Like it quite a lot though. Should be interesting to see what comes of this new thing though. ... -- Buddy Brannan, KB5ELV | And if the ground yawned, Email: davros@ycardz.com | I'd step to the side and say, Phone: (972) 276-6360 | "Hey ground! I'm nobody's lunch!" Voice mail: (877) 791-5298 | --Eddie From Ohio ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Geoff Shang ` Buddy Brannan @ ` Charles Hallenbeck ` Charles Hallenbeck ` Tommy Moore ` Brent Harding 2 siblings, 2 replies; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: speakup Geoff - Thanks for all that info! I agree about lame, after playing with it for a day or so it really does beat out bladeenc with everything I have tried. I am able to create voice files at 16k sampling rate that sound fantastic. I am also more interested in low bit rates. Chuck. My web site is http://www.mhonline.net/~chuckh Keep grandma off the streets -- legalize bingo. ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Charles Hallenbeck @ ` Charles Hallenbeck ` Tommy Moore 1 sibling, 0 replies; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: speakup Sorry for my loose language earlier - I really do know the difference between sampling rate and bandwidth! I should have said 16k bps instead of sampling rate - that yields files of size 2k per second of playback with very acceptable voice quality. My web site is http://www.mhonline.net/~chuckh Keep grandma off the streets -- legalize bingo. ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Charles Hallenbeck ` Charles Hallenbeck @ ` Tommy Moore ` Charles Hallenbeck 1 sibling, 1 reply; 25+ messages in thread From: Tommy Moore @ UTC (permalink / raw) To: speakup Wo, htey can go to 16 KBPS? Hmm, I gotta see that. Where od I get this program? Also is there a way to get a debian package of it some how? ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Tommy Moore @ ` Charles Hallenbeck 0 siblings, 0 replies; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: Tommy Moore; +Cc: speakup Hi Tommy, Hell yes, it can go down to 8 kbps if you want, but it sounds pretty grim that tight. You can specify bandwidth in 8k increments from 8k to 320k. It also does vbr where you can specify the minimum and maximum bit rates you will permit it to do. The tarball is a source package and seems to be distribution independent as far as I can tell. Here is the URL: http://www.sulaco.org/mp3/ Good luck! Chuck. My web site is http://www.mhonline.net/~chuckh Keep grandma off the streets -- legalize bingo. ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Geoff Shang ` Buddy Brannan ` Charles Hallenbeck @ ` Brent Harding ` Kerry Hoath ` (3 more replies) 2 siblings, 4 replies; 25+ messages in thread From: Brent Harding @ UTC (permalink / raw) To: speakup It won't take output from arecord, arecord records by default in 8 bit sampling rate, which lame won't work with. At 02:18 PM 8/20/00 +1000, you wrote: >Hi Chuck: > >As the bladeenc site says, they're not about low bit-rate encoding. In my >personal opinion, lame shits all over it. Bladeenc reckon they're better >at rates over 128K, but I'm yet to hear it. Lame is certainly better at >128K and below. > >I don't have the August 6 beta, didn't know there was one actually. Lame >does do an OK job at encoding vorbis streams, but only to a point. The >vorbis code up until this week from vorbis.org only really supported >encoding at higher bitrates like 160. Lame will do lower, but I think it >uses it's MP3 engine and alters the output to encode vorbis instead. Don't >get me wrong, it still sounds better than MP3 even produced by the same >encoder, but I don't think it sounds as good as vorbis will sound at the >same rates. > >I'd look out for another version of lame to hit the streets soon as Tuesday >saw the release of vorbis 1.0beta2. As you may recall from the interview, >this does 128K encoding and might even do 96, haven't looked. The program >listed on vorbis.com for encoding has changed from an ogg-ised version of >lame 3.84alpha to a program called oggenc. I haven't tried it yet but I do >intend to. > >I might also say while I'm here that if you want a straight player, then >I'd probably recommend freeamp. But, and it's a big but, you need the >latest available code. There is a version labeled freeamp 2.1beta5vorbis, >but the vorbis support doesn't work with any other than the X gdk >interface. freeamp 2.1beta6 will fix this, but last I checked, it hadn't >been released yet. Until it is, you will have to checkout the CVS >code. This does work, believe me. I finally after 4 weeks of chasing it >and E-mails to freeamp developers, have a freeamp that will play ogg vorbis >in console mode. > >If you want some content to listen to, check out www.vorbis.com. There is >a They Might Be Giants song at 96K that sounds quite impressive, and it's >only 1.4 meg. > >Geoff. > > >-- >Geoff Shang <gshang10@scu.edu.au> >ICQ number 43634701 > > > >_______________________________________________ >Speakup mailing list >Speakup@braille.uwo.ca >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding @ ` Kerry Hoath ` Charles Hallenbeck ` (2 subsequent siblings) 3 siblings, 0 replies; 25+ messages in thread From: Kerry Hoath @ UTC (permalink / raw) To: speakup Then try arecord -b16 that gives you 16-bit output. arecord -h is your friend. On Sun, Aug 20, 2000 at 10:15:57PM -0500, Brent Harding wrote: > It won't take output from arecord, arecord records by default in 8 bit > sampling rate, which lame won't work with. > At 02:18 PM 8/20/00 +1000, you wrote: > >Hi Chuck: > > > >As the bladeenc site says, they're not about low bit-rate encoding. In my > >personal opinion, lame shits all over it. Bladeenc reckon they're better > >at rates over 128K, but I'm yet to hear it. Lame is certainly better at > >128K and below. > > > >I don't have the August 6 beta, didn't know there was one actually. Lame > >does do an OK job at encoding vorbis streams, but only to a point. The > >vorbis code up until this week from vorbis.org only really supported > >encoding at higher bitrates like 160. Lame will do lower, but I think it > >uses it's MP3 engine and alters the output to encode vorbis instead. Don't > >get me wrong, it still sounds better than MP3 even produced by the same > >encoder, but I don't think it sounds as good as vorbis will sound at the > >same rates. > > > >I'd look out for another version of lame to hit the streets soon as Tuesday > >saw the release of vorbis 1.0beta2. As you may recall from the interview, > >this does 128K encoding and might even do 96, haven't looked. The program > >listed on vorbis.com for encoding has changed from an ogg-ised version of > >lame 3.84alpha to a program called oggenc. I haven't tried it yet but I do > >intend to. > > > >I might also say while I'm here that if you want a straight player, then > >I'd probably recommend freeamp. But, and it's a big but, you need the > >latest available code. There is a version labeled freeamp 2.1beta5vorbis, > >but the vorbis support doesn't work with any other than the X gdk > >interface. freeamp 2.1beta6 will fix this, but last I checked, it hadn't > >been released yet. Until it is, you will have to checkout the CVS > >code. This does work, believe me. I finally after 4 weeks of chasing it > >and E-mails to freeamp developers, have a freeamp that will play ogg vorbis > >in console mode. > > > >If you want some content to listen to, check out www.vorbis.com. There is > >a They Might Be Giants song at 96K that sounds quite impressive, and it's > >only 1.4 meg. > > > >Geoff. > > > > > >-- > >Geoff Shang <gshang10@scu.edu.au> > >ICQ number 43634701 > > > > > > > >_______________________________________________ > >Speakup mailing list > >Speakup@braille.uwo.ca > >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > > > > > > > _______________________________________________ > Speakup mailing list > Speakup@braille.uwo.ca > http://speech.braille.uwo.ca/mailman/listinfo/speakup > -- -- Kerry Hoath: kerry@gotss.eu.org Alternates: kerry@emusys.com.au kerry@gotss.spice.net.au or khoath@lis.net.au ICQ UIN: 8226547 ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding ` Kerry Hoath @ ` Charles Hallenbeck [not found] ` <Pine.LNX.4.21.0008210640560.164-100000@hudson.mhonline.net > ` Geoff Shang 3 siblings, 0 replies; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: speakup On Sun, 20 Aug 2000, Brent Harding wrote: > It won't take output from arecord, arecord records by default in 8 bit > sampling rate, which lame won't work with. I am using arecord to output 8 bits unsigned at 11025 sampling rate, and lame 3.86 works beautifully to create an mp3 file at 16 kbps bandwidth. Chuck. My web site is http://www.mhonline.net/~chuckh Keep grandma off the streets -- legalize bingo. ^ permalink raw reply [flat|nested] 25+ messages in thread
[parent not found: <Pine.LNX.4.21.0008210640560.164-100000@hudson.mhonline.net >]
* Re: Slashdot interview re: vorbis [not found] ` <Pine.LNX.4.21.0008210640560.164-100000@hudson.mhonline.net > @ ` Brent Harding ` Charles Hallenbeck 0 siblings, 1 reply; 25+ messages in thread From: Brent Harding @ UTC (permalink / raw) To: speakup I took which ever lame I found on freshmeat.net, probably the latest, as only one was found. At 06:43 AM 8/21/00 -0400, you wrote: >On Sun, 20 Aug 2000, Brent Harding wrote: > >> It won't take output from arecord, arecord records by default in 8 bit >> sampling rate, which lame won't work with. > >I am using arecord to output 8 bits unsigned at 11025 sampling rate, and >lame 3.86 works beautifully to create an mp3 file at 16 kbps bandwidth. > >Chuck. > > >My web site is http://www.mhonline.net/~chuckh >Keep grandma off the streets -- legalize bingo. > > >_______________________________________________ >Speakup mailing list >Speakup@braille.uwo.ca >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding @ ` Charles Hallenbeck ` Geoff Shang 0 siblings, 1 reply; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: speakup [-- Attachment #1: Type: TEXT/PLAIN, Size: 1499 bytes --] Hi Brent, My lame is dated August 6, version 3.86. What I have found is that it falls apart with sampling rates below 11025 samples per second - it squeaks and groans when given an 8k wav file, whether it is 8 bit or 16 bit samples, but just raise the sampling rate to 11025 or even 16000 and 8 bit samples work just great. I am attaching a small recording using the parameters I described below. See what y0ou think. Chuck. On Mon, 21 Aug 2000, Brent Harding wrote: > I took which ever lame I found on freshmeat.net, probably the latest, as > only one was found. > At 06:43 AM 8/21/00 -0400, you wrote: > >On Sun, 20 Aug 2000, Brent Harding wrote: > > > >> It won't take output from arecord, arecord records by default in 8 bit > >> sampling rate, which lame won't work with. > > > >I am using arecord to output 8 bits unsigned at 11025 sampling rate, and > >lame 3.86 works beautifully to create an mp3 file at 16 kbps bandwidth. > > > >Chuck. > > > > > >My web site is http://www.mhonline.net/~chuckh > >Keep grandma off the streets -- legalize bingo. > > > > > >_______________________________________________ > >Speakup mailing list > >Speakup@braille.uwo.ca > >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > > > > > > > _______________________________________________ > Speakup mailing list > Speakup@braille.uwo.ca > http://speech.braille.uwo.ca/mailman/listinfo/speakup > My web site is http://www.mhonline.net/~chuckh Keep grandma off the streets -- legalize bingo. [-- Attachment #2: Type: AUDIO/mpeg, Size: 46289 bytes --] ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Charles Hallenbeck @ ` Geoff Shang 0 siblings, 0 replies; 25+ messages in thread From: Geoff Shang @ UTC (permalink / raw) To: speakup Hi Chuck: Hey, that doesn't sound bad at all. I'm impressed. Geoff. -- Geoff Shang <gshang10@scu.edu.au> ICQ number 43634701 ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding ` (2 preceding siblings ...) [not found] ` <Pine.LNX.4.21.0008210640560.164-100000@hudson.mhonline.net > @ ` Geoff Shang ` Brent Harding 3 siblings, 1 reply; 25+ messages in thread From: Geoff Shang @ UTC (permalink / raw) To: speakup On Sun, 20 Aug 2000, Brent Harding wrote: > It won't take output from arecord, arecord records by default in 8 bit > sampling rate, which lame won't work with. Oh yeah. Silly me. Try: arecord -m |lame - file.mp3 For 44.1khz 16 bit stereo. If you want something less, use the s and b switches and ummm... the other one for specifying channel numbers to get what you want. You may also have to use command options in the lame command to get it working properly, but it's definitely doable. Oh by the way, apparently there is a vorbis-tools (not necessarily spelled correctly) debian package available, but I'd guess that it's in unstable. Geoff. ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Geoff Shang @ ` Brent Harding ` Geoff Shang 0 siblings, 1 reply; 25+ messages in thread From: Brent Harding @ UTC (permalink / raw) To: speakup I don't know if I really should upgrade to unstable. Does the capital m for dat quality give better than lower m for CD quality? I tried it this morning, and it worked. What default bandwidth does lame make the files? At 09:04 AM 8/22/00 +1000, you wrote: >On Sun, 20 Aug 2000, Brent Harding wrote: > >> It won't take output from arecord, arecord records by default in 8 bit >> sampling rate, which lame won't work with. > >Oh yeah. Silly me. Try: > >arecord -m |lame - file.mp3 > >For 44.1khz 16 bit stereo. If you want something less, use the s and b >switches and ummm... the other one for specifying channel numbers to get >what you want. You may also have to use command options in the lame >command to get it working properly, but it's definitely doable. > >Oh by the way, apparently there is a vorbis-tools (not necessarily spelled >correctly) debian package available, but I'd guess that it's in unstable. > >Geoff. > > > >_______________________________________________ >Speakup mailing list >Speakup@braille.uwo.ca >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding @ ` Geoff Shang ` Brent Harding 0 siblings, 1 reply; 25+ messages in thread From: Geoff Shang @ UTC (permalink / raw) To: speakup Hi: OK, capital M is 48khz, cd quality is 44.1khz. Depending on your soundcard, you may not be able to use 48khz. Most probably, any 48khz files you can infact use will probably be downsampled to 44.1khz upon playback, unless you've got a funky soundcard. Lame by default encodes at 128kbps stereo. If you use the -a switch to downmix to mono, the default is 64kbps. Finally, you don't need to do a distupgrade necessarily to use a package from unstable. In theory, it should let you know if other packages need to be upgraded in order for it to work. Having said that, however, compiling from source is pretty straight-forward, and you might be able to grab binaries for stuff anyway. What do you want to do with vorbis anyway? There are binaries for oggenc (an encoder) and freeamp 2.1beta6 is apparently out now (according to one of the developers who wrote to me personally to tell me) and that should do for playing them. Geoff. -- Geoff Shang <gshang10@scu.edu.au> ICQ number 43634701 ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Geoff Shang @ ` Brent Harding ` Charles Hallenbeck ` Geoff Shang 0 siblings, 2 replies; 25+ messages in thread From: Brent Harding @ UTC (permalink / raw) To: speakup Just to see if it's better. It seems I can't reverse the process take an mp3 and make a wave out of it. I've tried mpg123 myfile.mp3 | arecord -M myfile.wav, and it plays through the speaker, not to the program. >Hi: > >OK, capital M is 48khz, cd quality is 44.1khz. Depending on your >soundcard, you may not be able to use 48khz. Most probably, any 48khz >files you can infact use will probably be downsampled to 44.1khz upon >playback, unless you've got a funky soundcard. > >Lame by default encodes at 128kbps stereo. If you use the -a switch to >downmix to mono, the default is 64kbps. > >Finally, you don't need to do a distupgrade necessarily to use a package >from unstable. In theory, it should let you know if other packages need to >be upgraded in order for it to work. Having said that, however, compiling >from source is pretty straight-forward, and you might be able to grab >binaries for stuff anyway. What do you want to do with vorbis >anyway? There are binaries for oggenc (an encoder) and freeamp 2.1beta6 is >apparently out now (according to one of the developers who wrote to me >personally to tell me) and that should do for playing them. > >Geoff. > > >-- >Geoff Shang <gshang10@scu.edu.au> >ICQ number 43634701 > > >_______________________________________________ >Speakup mailing list >Speakup@braille.uwo.ca >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding @ ` Charles Hallenbeck ` Geoff Shang 1 sibling, 0 replies; 25+ messages in thread From: Charles Hallenbeck @ UTC (permalink / raw) To: Brent Harding; +Cc: speakup Brent - To make a wav file from an mp3 file, use the -w switch in mpg123. The -w switch is not mentioned in the man pages, but the program's built in help screen mentions it. The format is: mpg123 -w output.wav input.mp3 Hope that helps. Chuck. My web site is http://www.mhonline.net/~chuckh Winter will ask us what we did all summer. ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Brent Harding ` Charles Hallenbeck @ ` Geoff Shang ` Brent Harding 1 sibling, 1 reply; 25+ messages in thread From: Geoff Shang @ UTC (permalink / raw) To: speakup Hi Brent: I see that you've already been given the command to write a WAV file from MPG123. I might just point out that (A) Not all programs will pipe to standard output without being asked, and (B) there's vertually no need to ever pipe to arecord, as you could redirect outjput to a file. For example: mpg123 -s filename.mp3 |arecord -m filename.wav would have done what you wanted, since the -s tells it to write to standard output (stdout). But it's far more efficient to do this: mpg123 -s filename.mp3 >filename.wav This uses the shell to redirect standard output to a file. You could use 2 greaterthans to append rather than to rewrite. Of course, this is a bad example since MPG123 will do it itself with -w which I recommend above both of the above. Geoff. -- Geoff Shang <gshang10@scu.edu.au> ICQ number 43634701 ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Geoff Shang @ ` Brent Harding 0 siblings, 0 replies; 25+ messages in thread From: Brent Harding @ UTC (permalink / raw) To: speakup Why are files I make in linux not playable in windows, shouldn't they be, it can often say not a valid wave file. Or was it a blank .wav that had nothing in it, don't think so. At 05:56 PM 8/24/00 +1000, you wrote: >Hi Brent: > >I see that you've already been given the command to write a WAV file from >MPG123. I might just point out that (A) Not all programs will pipe to >standard output without being asked, and (B) there's vertually no need to >ever pipe to arecord, as you could redirect outjput to a file. For >example: > >mpg123 -s filename.mp3 |arecord -m filename.wav > >would have done what you wanted, since the -s tells it to write to standard >output (stdout). But it's far more efficient to do this: > >mpg123 -s filename.mp3 >filename.wav > >This uses the shell to redirect standard output to a file. You could use >2 greaterthans to append rather than to rewrite. Of course, this is a bad >example since MPG123 will do it itself with -w which I recommend above both >of the above. > >Geoff. > > >-- >Geoff Shang <gshang10@scu.edu.au> >ICQ number 43634701 > > >_______________________________________________ >Speakup mailing list >Speakup@braille.uwo.ca >http://speech.braille.uwo.ca/mailman/listinfo/speakup > > > ^ permalink raw reply [flat|nested] 25+ messages in thread
* Re: Slashdot interview re: vorbis ` Charles Hallenbeck ` Geoff Shang @ ` Kerry Hoath 1 sibling, 0 replies; 25+ messages in thread From: Kerry Hoath @ UTC (permalink / raw) To: speakup Ok the sad news is as follows: The best mp3 encoder for LInux at Low bitrates in my experience is l3enc by fraun hofer (spelling). It manages 32kbps rather well although it does introduce some noise and stuff. The sox resample code is broken unless you are running sox 12.17 I believe and you use the resample option not the rate loop. When I say low bitrates that is bitrates <128kbps. Bladenc seems to shine at rates over 160kbps as do most of the iso-based encoders. There are encoders like LAME and GOGO, but they have nasty artifacts at rates <64kbps mono and insert schreechy sounds like 2 birds mating into your audio stream. Gogo has the advantage that it is blazingly fast with huge chunks in raw assembler. The disadvantage is the documentation is a bit so so and the source comments are in Japanese. If anyone wants a copy of l3enc I have binaries for libc5 and an enabling alternative. Regards, Kerry. On Sat, Aug 19, 2000 at 02:38:25PM -0400, Charles Hallenbeck wrote: > Hi Geoff - > > After seeing your earlier note about piping audio streams with lame, I > decided to give that program a whirl. I have been using bladeenc until > now. I noticed after compiling and installing the August 6 beta release > that it now supports encoding and decoding the ogg vorbis format as well > as mp3. I am curious - is that a new feature? Have you used it? How does > it work? I have not been too happy with bladeenc as an mp3 encoder because > it requires a minimum of 32 kilobits sampling rate inputs, and the > resulting mp3 file loses something in the translation. That is, if you > only have 8k or 11.025 or 16k to start with, it seems wasteful to first > artificially inflate the sampling rate to 32k in order to compress it with > bladeenc, and lots of noise creeps into the signal along the way - > probably with sox resampling. > > I am hopeful that lame can do a better job, and am excited to see ogg > vorbis support too. > > Chuck. > > > My web site is http://www.mhonline.net/~chuckh > Keep grandma off the streets -- legalize bingo. > > > _______________________________________________ > Speakup mailing list > Speakup@braille.uwo.ca > http://speech.braille.uwo.ca/mailman/listinfo/speakup > -- -- Kerry Hoath: kerry@gotss.eu.org Alternates: kerry@emusys.com.au kerry@gotss.spice.net.au or khoath@lis.net.au ICQ UIN: 8226547 ^ permalink raw reply [flat|nested] 25+ messages in thread
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