Slashdot interview re: vorbis

Slashdot interview re: vorbis

[Geoff Shang]

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

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

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.

Re: Slashdot interview re: vorbis

[Brent Harding]

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
>
>
>

Re: Slashdot interview re: vorbis

[Geoff Shang]

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

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

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.

Re: Slashdot interview re: vorbis

[Geoff Shang]

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

Re: Slashdot interview re: vorbis

[Buddy Brannan]

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

Re: Slashdot interview re: vorbis

[Kerry Hoath]

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

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

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.

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

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.

Re: Slashdot interview re: vorbis

[Tommy Moore]

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?

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

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.

Re: Slashdot interview re: vorbis

[Brent Harding]

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
>
>
>

Re: Slashdot interview re: vorbis

[Kerry Hoath]

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

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

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.

Re: Slashdot interview re: vorbis

[Brent Harding]

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
>
>
>

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

[-- 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 --]

Re: Slashdot interview re: vorbis

[Geoff Shang]

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.

Re: Slashdot interview re: vorbis

[Brent Harding]

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
>
>
>

Re: Slashdot interview re: vorbis

[Geoff Shang]

Hi Chuck:

Hey, that doesn't sound bad at all.  I'm impressed.

Geoff.


-- 
Geoff Shang <gshang10@scu.edu.au>
ICQ number 43634701

Re: Slashdot interview re: vorbis

[Geoff Shang]

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

Re: Slashdot interview re: vorbis

[Brent Harding]

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
>
>
>

Re: Slashdot interview re: vorbis

[Charles Hallenbeck]

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.

Re: Slashdot interview re: vorbis

[Geoff Shang]

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

Re: Slashdot interview re: vorbis

[Brent Harding]

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
>
>
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