Re: [mp3encoder] filtering bad recordings
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 ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
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 ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
The version of postfish I found is a pre-release that's pretty old from here: http://svn.xiph.org/trunk/postfish/ It won't compile in Slackware either.where are you getting postfish? I tried the -h switch and thought the speaker sounded bad at times. 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. Marty ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
On 3/17/07, Marty Huntzberry <[EMAIL PROTECTED]> wrote:
> Would you post the commands that you used with postfish, sox, and lame?
Postfish is UI-based; but you can skip the Postfish step as the
declipping was the only interesting bit.
> I cannot get postfish to compile on my Athlon 64 running Fedora Core Linux 64
> bit edition. I'll try compiling in Slackware 11 (32 bit i86 edition)
> and see if it works there.
I'll have a look at that. Last time I tried building on FC x86_64 was
rawhide between FC5 and FC6 (and it did build then). I haven't put
any time into it lately.
> What I'm trying to do is reduce it to a 22 kHz mono mp3 file like you did. I
> loaded the original file into Audacity so I
> could see it...it seems to have been recorded too loud and the peak sounds go
> off the chart.
Yeah. Even worse it looks like it ran through a 'soft clipper'
(whoever produced this did it on purpose, whether or not he realized
it) so the declipper on Postfish is of limited usefulness. Clipping
would have been reversable, but a 'soft clipper' thoroughly screws the
signal... it makes clipping look like not clipping, but tends not to
actually improve the sound quality at all.
The declipper might still make an improvement, but only at the expense
of real processor time... I'm talking about cranking the trigger down
to -6dB and letting it crunch for days :-(
> In lame I've been reducing the sound, encoding in mono
> at 22 kHz and 24 kbps with this:
>
> lame -m m -b 24 --resample 22.05 -f --scale-r -2 --mp3input original.mp3
> compressed .mp3
Thanks for this.
--scale-r -2 is increasing the volume, not decreasing it. You want
--scale-r .5 or the like.
-f is telling the encoder 'quality doesn't matter; be as fast as
possible'. This could be the source of your problems. You want -h
for low bitrate ('quality matters more than speed').
With those changes, eg:
lame -h -m m -b 24 --resample 22.05 --scale-r .5 --mp3input
SH20070313\ Nehem4_5b.mp3 nehem-h.mp3
The output sounds pretty good. Still clipped, but the lowpass takes
the edge off it.
Monty
>
> Marty
>
> On Fri, 16 Mar 2
> On Fri, 16 Mar 2007 18:48:00 -0400
> [EMAIL PROTECTED] wrote:
>
> 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
> ___
> mp3encoder mailing list
> [email protected]
> https://minnie.tuhs.org/mailman/listinfo/mp3encoder
>
>
> ___
> mp3encoder mailing list
> [email protected]
> https://minnie.tuhs.org/mailman/listinfo/mp3encoder
>
___
mp3encoder mailing list
[email protected]
https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
Would you post the commands that you used with postfish, sox, and lame? I cannot get postfish to compile on my Athlon 64 running Fedora Core Linux 64 bit edition. I'll try compiling in Slackware 11 (32 bit i86 edition) and see if it works there. What I'm trying to do is reduce it to a 22 kHz mono mp3 file like you did. I loaded the original file into Audacity so I could see it...it seems to have been recorded too loud and the peak sounds go off the chart. In lame I've been reducing the sound, encoding in mono at 22 kHz and 24 kbps with this: lame -m m -b 24 --resample 22.05 -f --scale-r -2 --mp3input original.mp3 compressed .mp3 Marty On Fri, 16 Mar 2 On Fri, 16 Mar 2007 18:48:00 -0400 [EMAIL PROTECTED] wrote: 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 ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
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 ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
I will look into these codecs. Are these lame codecs or audacity? I'm running Fedora Core and Slackware Linux. Some are wondering why I want such a low bit rate. Here's why-I run an mp3 server that streams and sounds best at low bitrates (it's not choppy on low bandwith connections). I have ~ 5000 mp3s and most are speeches. Some were pre-compressed at 24 kbps and 22 khz and sound great. The ones I'm trying to compress with lame are recorded at 192 kbps and 44 khz. So, it's been done and it should work for me. Here are links to 2 speeches: A 24 kpbs 22 khz speech download that sounds fine: http://media.ccphilly.org:81/Teaching/Audio/B02_Exodus/WED54924.mp3 The 192 kpbs 44 khz speech I want to compress: http://www.ccbellmawr.com/radioprogram/Week20070312/SH20070313%20Nehem4_5b.mp3 Marty On Thu, 15 Mar 2007 23:03:47 -0400 "Ishaan Dalal" <[EMAIL PROTECTED]> wrote: Marty, Unfortunately, Lame is known to be not really optimized for low-bitrate encoding. The closest option would be to the Fraunhofer codec at 22 KHz/32 kbps (the ACM codec bundled with Windows can encode at that rate). If it is voice, you should try speech codecs - they will get you much better quality at the kind of bitrates you want. I'd suggest Speex. Are you transcoding? (i.e. you don't have the original 22 KHz/16-bit PCM (WAV) audio, but are using the 192 kbps MP3 as the "source")? Cheers, -Ishaan 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. > > 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 > > > ___ > mp3encoder mailing list > [email protected] > https://minnie.tuhs.org/mailman/listinfo/mp3encoder > ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
Marty, Unfortunately, Lame is known to be not really optimized for low-bitrate encoding. The closest option would be to the Fraunhofer codec at 22 KHz/32 kbps (the ACM codec bundled with Windows can encode at that rate). If it is voice, you should try speech codecs - they will get you much better quality at the kind of bitrates you want. I'd suggest Speex. Are you transcoding? (i.e. you don't have the original 22 KHz/16-bit PCM (WAV) audio, but are using the 192 kbps MP3 as the "source")? Cheers, -Ishaan 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. > > 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 > > > ___ > mp3encoder mailing list > [email protected] > https://minnie.tuhs.org/mailman/listinfo/mp3encoder > ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
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 ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
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 > ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
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 ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
Re: [mp3encoder] filtering bad recordings
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 > > ___ > mp3encoder mailing list > [email protected] > https://minnie.tuhs.org/mailman/listinfo/mp3encoder > ___ mp3encoder mailing list [email protected] https://minnie.tuhs.org/mailman/listinfo/mp3encoder
