Re: [mp3encoder] filtering bad recordings

2007-03-19 Thread Marty Huntzberry
wow-that worked!  I tried that in the past with another speech and didn't like 
16 kHz sampling but with this mp3 it works fine.  I dropped the
re-sampling spec and the r switch and kept it mono:

lame -m m -b 24 --mp3input original.mp3 compressed.mp3


Thanks for all the help!

Marty

Visit my media server: http://linuxhippy.servemp3.com:8001

On Mon, 19 Mar 2007 20:10:47 +0100
Gabriel Bouvigne [EMAIL PROTECTED] wrote:

Marty Huntzberry a écrit :

 That r -2 switch makes it sound softer.  Other values did increase (r -1, 
 r-3)maybe it's a glitch in lame.  Looking at the output in audacity you
 can see it decreaes.

To be honest we are expecting --scale to be used with a float number 
(ex: 0.5 or 1.1), but not with a negative number, so there might be a 
parsing problem regarding this in Lame (I'll try to check it).

Regarding your situation, I'd suggest you to try first an encoding 
without specifying the output sampling rate. This will let Lame choose 
it based on the target bitrate and its own knowledge of its compression 
abilities.
As an example, in mono and targetting 24kbps, current Lame version would 
choose to use a sampling rate of 16kHz, which will probably sound better 
than 22kHz.

Regards,

-- 
Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org
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Re: [mp3encoder] filtering bad recordings

2007-03-16 Thread xiphmont
Having listened to your sample, aside from *alot* of clipping, that
audio should reduce to 22kHz mono very nicely.  It's a nice clear
recording.

I did a quick declip/channel merge with postfish, downsample with sox
and encoded it with lame and it came out sounding fine-- at least as
good as the 'good' sample you referenced.

So... what are you actually doing?  Don't spare the details.  Are you
encoding as a stereo 24kbpbs file by accident?

Monty
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Re: [mp3encoder] filtering bad recordings

2007-03-15 Thread xiphmont
On 3/15/07, tech list [EMAIL PROTECTED] wrote:
 Marty, unfortunately, there's not much you can do since the recording itself
 was low quality. 22KHz should have been OK for voice, but looks like the mic
 was placed badly
 or the acoustics of your room/hall was not the best.
 Your best bet would be to forget about lame for the moment and try out with
 some audio
 tools like audacity. Keep a copy of the original recording and play around
 with some noise
 cancellation, filtering etc. to see what sounds best.

The Postfish tool (linux only) has a filter called 'deverb' that can
remove some amount of excess reverberation, so long as there's a
reasonably strong direct signal in the midst of the echo/reverb.

Monty
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Re: [mp3encoder] filtering bad recordings

2007-03-15 Thread Marty Huntzberry
The only effect in audacity and lame that seems to work is reducing the audio.  
Why does it sound good in it's original mp3 format of 192 kbps and 44
khz but sounds bad at 24 kbps and 22 khz?  It's mainly a voice lecture.

On Thu, 15 Mar 2007 16:12:53 +0530
tech list [EMAIL PROTECTED] wrote:

Marty, unfortunately, there's not much you can do since the recording itself
was low quality. 22KHz should have been OK for voice, but looks like the mic
was placed badly
or the acoustics of your room/hall was not the best.
Your best bet would be to forget about lame for the moment and try out with
some audio
tools like audacity. Keep a copy of the original recording and play around
with some noise
cancellation, filtering etc. to see what sounds best.


On 1/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

 I have a lecture recording (no music, just voice) that was recorded with a
 digital recorder about 1 foot from the speaker at 22 kHz and 16 bit
 mono.  It sounds like the speaker is in a well.  Can I clean the recording
 up a bit with a filter in lame?

 Marty

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Re: [mp3encoder] filtering bad recordings

2007-03-15 Thread xiphmont
On 3/15/07, Marty Huntzberry [EMAIL PROTECTED] wrote:
 The only effect in audacity and lame that seems to work is reducing the 
 audio.  Why does it sound good in it's original mp3 format of 192 kbps and 44
 khz but sounds bad at 24 kbps and 22 khz?  It's mainly a voice lecture.

...maybe becasue you've eliminated 87% of the data?  Just a hunch.

(That would sound good in a voice-specific codec at 8kHz.  I don't
know why you think mp3 at 24kHz would sound good).

Oh, also, if you're going from stereo-mono, that's going to make an
echoey environment way less intelligible.  You've collapsed a 360
degree circular soundfield right into the center along with the voice.
 You don't get to use any of the brain's spiffy localization hardware
to pick out voice from the ambient anymore, now the voice and all the
noise localize to the same place.

Monty
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