convert sample rate

[Enzo-4]

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Hi

I have an mp3 44100 Hz, stereo

I'd like to create a wav mono 11025Hz, so i do:
ecasound -i:9645.mp3 -f:16,1,11025 -o:out.wav
(or ecasound -i:9645.mp3 -f:s16_le,1,11025 -o:out.wav)

ecasound creates a mono wav file, but doesn't change the sample rate, how can i do to convert it also ?

Here is the dump of ecasound command:

- [ Session created ] ----------------------------------------------------------
- [ Chainsetup created (cmdline) ] ---------------------------------------------
- [ Connecting chainsetup ] ----------------------------------------------------
(eca-chainsetup) 'nonrt' buffering mode selected.
(eca-chainsetup) Audio object "9645.mp3", mode "read".
(audio-io) Format: s16_le, channels 2, srate 44100, interleaved.
(eca-chainsetup) Audio object "out.wav", mode "read/write".
(audio-io) Format: s16_le, channels 1, srate 44100, interleaved.
- [ Chainsetup connected ] -----------------------------------------------------
(eca-control-objects) Connected chainsetup: "command-line-setup".
- [ Controller/Starting batch processing ] -------------------------------------
- [ Engine init - Driver start ] -----------------------------------------------
- [ Controller/Batch processing finished (0) ] ---------------------------------
(eca-control-objects) Disconnecting chainsetup: "command-line-setup".
- [ Chainsetup disconnected ] --------------------------------------------------

Thanks
Enzo

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[Rémi R]

Hi,

I hope this answer from a biginer is not stupid...

"-f:s16_le,1,11025" don't resample your signal, that's specify a format
for a part of your chain setup. (even if it can magicly change from 24
to 16 bits sometimes).

You should better use something like :
-f:s16_le,1,11025 -i resample,auto,9645.mp3 -o out.wav

Your chain setup is in format s16_le,1,11025, the input file is 9645.mp3
resampled, the output file is out.wav.

Pick up "resample" stuff in man and examples pages.

Good luck,
Rémi+


Enzo a écrit :

> Hi
>
> I have an mp3 44100 Hz, stereo
>
> I'd like to create a wav mono 11025Hz, so i do:
> ecasound -i:9645.mp3 -f:16,1,11025 -o:out.wav
> (or ecasound -i:9645.mp3 -f:s16_le,1,11025 -o:out.wav)
>
> ecasound creates a mono wav file, but doesn't change the sample rate, how can i do to convert it also ?
>
> Here is the dump of ecasound command:
>
> - [ Session created ] ----------------------------------------------------------
> - [ Chainsetup created (cmdline) ] ---------------------------------------------
> - [ Connecting chainsetup ] ----------------------------------------------------
> (eca-chainsetup) 'nonrt' buffering mode selected.
> (eca-chainsetup) Audio object "9645.mp3", mode "read".
> (audio-io) Format: s16_le, channels 2, srate 44100, interleaved.
> (eca-chainsetup) Audio object "out.wav", mode "read/write".
> (audio-io) Format: s16_le, channels 1, srate 44100, interleaved.
> - [ Chainsetup connected ] -----------------------------------------------------
> (eca-control-objects) Connected chainsetup: "command-line-setup".
> - [ Controller/Starting batch processing ] -------------------------------------
> - [ Engine init - Driver start ] -----------------------------------------------
> - [ Controller/Batch processing finished (0) ] ---------------------------------
> (eca-control-objects) Disconnecting chainsetup: "command-line-setup".
> - [ Chainsetup disconnected ] --------------------------------------------------
>
> Thanks
> Enzo
>
>
>
>
>
>
>
>
>
>       _____________________________________________________________________________
> Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail http://mail.yahoo.fr
>
>
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>
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>
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[Julien Claassen]

Hi Enzo!
  There is another way of doing it, which I found to be of better quality.
ecasound -i in.mp3 -o tmp.wav
sox -r 44100 -c 2 tmp.wav -r 11025 -c 1 out.wav
rm tmp.wav
  Of course you could do the channel conversion with ecasound, but if
youalready use sox, why not do two steps in one.
  I found that still ecasound's conversion has its qualitative drawbacks.
  Kindest regards
      Julien

--------
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======== FIND MY WEB-PROJECT AT: ========
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the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
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[Sergei Steshenko-2]

--- Julien Claassen <julien@...> wrote:

> Hi Enzo!
>   There is another way of doing it, which I found to be of better quality.
> ecasound -i in.mp3 -o tmp.wav
> sox -r 44100 -c 2 tmp.wav -r 11025 -c 1 out.wav
> rm tmp.wav
>   Of course you could do the channel conversion with ecasound, but if
> youalready use sox, why not do two steps in one.
>   I found that still ecasound's conversion has its qualitative drawbacks.
>   Kindest regards
>       Julien
>
> --------
> Music was my first love and it will be my last (John Miles)
>
> ======== FIND MY WEB-PROJECT AT: ========
> http://ltsb.sourceforge.net
> the Linux TextBased Studio guide
> ======= AND MY PERSONAL PAGES AT: =======
> http://www.juliencoder.de
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2008.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Ecasound-list mailing list
> Ecasound-list@...
> https://lists.sourceforge.net/lists/listinfo/ecasound-list
>

IIRC, 'ecasound' has high quality  resampler.

http://ccrma.stanford.edu/planetccrma/man/man1/ecasound.1.html :

"
If ecasound was compiled with support for libsamplerate, you can
              use  ´resample-hq´  to  use the highest quality resampling algo-
              rithm available.
".

Anyway, since output sampling frequency is supposed to be 11025Hz, which is
pretty low, one can't expect high sound quality.

Regards,
  Sergei.

Applications From Scratch: http://appsfromscratch.berlios.de/


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[Julien Claassen]

Hi Sergei!
  I know about the high-quality resampling and I have it. But alas, it still
gives a kind of low-frequency crackling. Don't ask me where that comes from.
  sox also has high quality resampling and it has always worked very fine. You
can adjust some smoothing and anti-aliasing stuff, which can generate nice
quality even with low frequencies.
  Kindest regards
        Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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[Kai Vehmanen]

Hi,

On Fri, 25 Jan 2008, Julien Claassen wrote:

>  I know about the high-quality resampling and I have it. But alas, it still
> gives a kind of low-frequency crackling. Don't ask me where that comes from.

ugh, oh -- that sounds bad! :( That sounds like the one-off buffering
error we had a long time ago with libsamplerate (fixed in 2.4.0).

With which version of libsamplerate you get the low-freq crackle?

Br, Kai

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[Julien Claassen]

Hello Kai!
   I'm running with libsamplerate 0.1.2.
   Thanks!
   Kindest regards
         Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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[Kai Vehmanen]

Hi,

On Fri, 25 Jan 2008, Julien Claassen wrote:

>  I know about the high-quality resampling and I have it. But alas, it still
> gives a kind of low-frequency crackling. Don't ask me where that comes from.

I've been trying to reproduce this but with no success so far (with the
same version of libsamplerate). Could you provide me with an example of
the low-freq crackling? And if possible, by using a generated source (so
that I can reproduce all the steps):

# with git version (5secs of sine as input)
ecasound -f:16,1,96000 -i tone,sine,440,5 -o foo96000.wav
ecasound -f:16,1,44100 -i resample,auto,foo96000.wav -o foo44100.wav

# or with older versions (with LADSPA)
ecasound -f:16,1,96000 -i null -o foo96000.wav -el:sine_fcac,440,1 -t:5
ecasound -f:16,1,44100 -i resample,auto,foo96000.wav -o foo44100.wav

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