Re: (no subject)

Re: (no subject)

[David Graff]

kevinc@doink.com said:
> Is there a program which will convert a sound file into a text (.txt)
> or comma separated value (.csv) file?  I'm thinking that would be a
> good route to go in order to import the data from a sound file into a
> math manipulation program, e.g. octave, maxima, Matlab, Mathcad,
> Gnumeric, etc.

Someone else will probably know of existing tools for importing audio 
sample data into one or another math package, but if I had to convert 
from binary to ascii numerics, I'd probably use sox to convert the 
input file to a raw single-channel output of appropriate endian-ness, 
and run a perl one-liner to reformat the binary numbers into strings; 
e.g.:

 sox 1chn_pcm.wav -t raw - | 
  perl -e 'while(read(STDIN,$_,2)==2){printf("%d\n",unpack("v",$_))}' > 1ch_pcm.txt


(You probably want to elaborate the perl script to produce a different
output format, depending on what your math package wants as input.)

	Dave Graff



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Re: (no subject)

[Vijay Gaur]

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Hi,
  Did anyone used sox to convert mp3 to wav.
  I would greatly appreciate if someone can post me the instructions.
  I need to convert mp3 file to wav(16KHz, 16bit, mono).

Thanks a bunch in advance.
Vijay


On 7/25/07, David Graff <graff@ldc.upenn.edu> wrote:
>
>
> kevinc@doink.com said:
> > Is there a program which will convert a sound file into a text (.txt)
> > or comma separated value (.csv) file?  I'm thinking that would be a
> > good route to go in order to import the data from a sound file into a
> > math manipulation program, e.g. octave, maxima, Matlab, Mathcad,
> > Gnumeric, etc.
>
> Someone else will probably know of existing tools for importing audio
> sample data into one or another math package, but if I had to convert
> from binary to ascii numerics, I'd probably use sox to convert the
> input file to a raw single-channel output of appropriate endian-ness,
> and run a perl one-liner to reformat the binary numbers into strings;
> e.g.:
>
> sox 1chn_pcm.wav -t raw - |
> perl -e 'while(read(STDIN,$_,2)==2){printf("%d\n",unpack("v",$_))}' >
> 1ch_pcm.txt
>
>
> (You probably want to elaborate the perl script to produce a different
> output format, depending on what your math package wants as input.)
>
>        Dave Graff
>
>
>
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-- 
Regards,
Vijay Gaur
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www.LimeAll.com

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Re: (no subject)

[Rafal Maszkowski]

On Wed, Jul 25, 2007 at 10:02:21AM -0700, Kevin Cosgrove wrote:
> Is there a program which will convert a sound file into a text
> (.txt) or comma separated value (.csv) file?  I'm thinking that

.dat format.

R.
-- 
Jeżeli upadnie cywilizacja zachodnia, to będzie szansa na odbudowę cywilizacji
      chrześcijańskiej. (dr Stanisław Krajski, Radio Maryja, 10 X 2001)

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Re: (no subject)

[David Graff]

Sorry -- there was a mistake in that perl script -- it should have been:

  perl -e 'while(read(STDIN,$_,2)==2){printf("%d\n",unpack("s",$_))}'

Note the "s" (for signed short int in native byte order), instead "v"
(unsigned short int in "VAX" (little-endian) byte order) -- the "s"
assumes that you're using a little-endian machine.  If you aren't, use
the "-x" (xinu) option on sox to invert the byte order of ms-wav pcm
files.

	Dave Graff




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(no subject)

[Daniel Payne]

I have a requriement to provide a Java based application service to trim and
segment audio files of various formats. SoX seems to have the capabilities I
need, but I have a few questions:

1. Is there a known Java JNI interface to SoX? If not, do you consider it
feasible to create one?

2. If JNI isn't available and I make calls to the program itself, will I run
into any problems by making multiple, concurrent requests?

3. And for the real kicker... What about integrating third-party codecs not
already listed as being supported (such as, say, RealAudio)?

Thanks,
Dan

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(no subject)

[JJENGR]

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SoX Users,
 
I have a serial digital stereo audio bit stream coming in to my PC. The  
digital audio samples are 12 bits/sample at a sample rate of  44.1KSamples/sec.  I 
want to play the audio through using a standard audio  player that would be 
present in most windows PC's (for example the standard  Mirosoft Audio Player). 
To use that player I need to get the audio into some  sort of standard audio 
format.  I will use SoX to do that so my question  is:
 
Can I stream the raw incoming data samples into the SoX converter while  
playing the output of the SoX converter through the audio player? I dont mind a  
short dealy of a few seconds.
 
Thanks in adance for your help

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(no subject)

[maanu]

Hello there,

I noticed the following .au support discrepancy while trying 3 sox
versions I've compiled myself on my Ubuntu 8.10.

a) The following command line works well with this given ring.au [1]
(notice: not warning):
 me@machine:/tmp$ ./sox-14.0.1/src/sox /tmp/ring.au /tmp/ring-14.0.1.wav
 me@machine:/tmp$ ./sox-14.1.0/src/sox /tmp/ring.au /tmp/ring-14.1.0.wav
 me@machine:/tmp$ ./sox-14.2.0/src/sox /tmp/ring.au /tmp/ring-14.2.0.wav


b) Same test with another .au file [2]: (notice the warnings with 14.10
and 14.2.0):
 me@machine:/tmp$ ./sox-14.0.1/src/sox /tmp/ringbad.au
 /tmp/ringbad-14.0.1.wav

 me@machine:/tmp$ ./sox-14.1.0/src/sox /tmp/ringbad.au
 /tmp/ringbad-14.1.0.wav
 /tmp/sox-14.1.0/src/.libs/lt-sox au: header size 24 is too small

 me@machine:/tmp$ ./sox-14.2.0/src/sox /tmp/ringbad.au
 /tmp/ringbad-14.2.0.wav
 /tmp/sox-14.2.0/src/.libs/lt-sox au: header size 24 is too small


Despite the warning, the conversion is correctly done, but why this
warning message from 14.1.0 ?
Could the file be wrong (but the header looks good, it is playable, and
no windows or linux audio editors I tried complained) ? 

Note the 3 sox versions have been compiled the same way ("./configure
--without-sndfile --without-flac --without-ogg --without-wavpack
--without-lame --without-mad && make")

Emmanuel.

[1] is located at:
http://src.opensolaris.org/source/xref/bluetooth/bluetooth-gate/usr/src/cmd/audio/samples/au/ring.au
Its header looks good:
          0011 2233 4455 6677 8899 aabb ccdd eeff
00000000: 2e73 6e64 0000 0030 0000 98c7 0000 0001  .snd...0........
00000010: 0000 1f40 0000 0001 6f6e 6520 7269 6e67  ...@....one ring
00000020: 2028 6c6f 6f70 2074 6869 7329 0000 0000   (loop this)....
00000030: f775 645a 5555 5964 f2d7 cac2 bebe c3cc  .udZUUYd........
00000040: e558 443b 3634 363b 476d cab9 b0ad acaf  .XD;646;Gm......
[...]

[2] is not on internet (but I can provide it if required), its header
looks good as well:
          0011 2233 4455 6677 8899 aabb ccdd eeff
00000000: 2e73 6e64 0000 0018 0006 aeb2 0000 0001  .snd............
00000010: 0000 1f40 0000 0001 5756 534f 5054 5755  ...@....WVSOPTWU
00000020: 4f4e 4f56 5953 4c4b 515e 5e4d 4248 c9a6  ONOVYSLKQ^^MBH..
00000030: 968d 8a8a 8d94 a278 2112 0c09 080b 0f1c  .......x!.......
00000040: 3caa 968e 8a8a 8c91 9ec5 2615 0d09 080a  <.........&.....
[...]


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(no subject)

[Eli Sanders]

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I am using the compand effect and getting a lot of clipping. Does the -G and norm options apply to the input? 
How do I normalize the output and prevent clipping?


sox -G --norm=-3 alien.wav alienout.wav. compand  0.3,1 -90,-90,-70,-70,-60,-20,0,0 -5 0 0.2
sox WARN compand: compand clipped 46934 samples; decrease volume? 		 	   		  

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(no subject)

[Paul Arnold]

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I am looking for an autotune function that I can control the parameters of
via a CLI/script. Hence a LADSPA plugin for sox would be fine. Autotalent
and VocProc look promising, but are GUI from what I can make out...? Any
ideas?

Paul

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(no subject)

[James Spencer]

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So I am speculating this happens because of using Ogg Vorbis.
I don't know if the effect is intriduced while mixing the
6-channel file (I doubt that), or only when playing it back


AFAICT it is introduced when mixing the 6 channel file (FWIW ffmpeg
exhibits the same behaviour).

In a freshly merged 6 channel file, I've tried remixing the channels into a
different order with the result that both the original LFE channel and the
new one sound "filtered" and have lost data.

Remixing a 7 channel file down to 6 channels also introduces the effect on
the 6th channel.

-- 

-----------------------
James Spencer
p.ota.to

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(no subject)

[paul fellows]

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Re: (no subject)

[Eric Wong]

paul fellows <4-werk@gmx.com> wrote:
>  how do i aviod the <=style crap

Disable HTML in your email client and use plain-text instead.
Plain-text is the most compatible with all mail software.

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(no subject)

[urg]

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(no subject)

[Jeff Learman]

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Thanks, that would be perfect!  Also, a flag to print skipped tags, if
there isn't one already.

Please forgive the late reply.  I'd just signed up and you replied so
fast the exploder didn't know me yet!

Jeff


Måns Rullgård writes:
>
> Jeff Learman <jjlearman@...>  writes:
>
>> I noticed that when converting stereo sample files to mono, the 'smpl'
>> chunks (which encode loop information used by sample players), SoX omits
>> the 'smpl' chunk in the output file.
>>
>> For example,
>>
>> sox input.wav -c 1 output.wav
>>
>> sums a stereo input.wav to a mono output.wav. However, the 'smpl' chunk in
>> input.wav is discarded.
>>
>> I have a workaround; I wrote code to extract the smpl chunk and append it
>> to the output. But it would be nice if 'smpl' chunks were supported by
>> default, at least for the easy cases. (I realize it could be difficult or
>> impossible for many cases.)
>>
>> What's the best way to file a bug report?
>
> The smpl chunk isn't part of the official WAV format, so SoX ignores it.
> It's the only reasonable thing to do with unknown chunks. Copying the macross
> to the output is risky since we can't know that the processing didn't
> invalidate them. For example, a chunk referencing specific samples is (likely)
> still relevant after downmixing stereo to mono, whereas resampling to a higher
> or lower rate will obviously make it useless.
>
> The best we can reasonably do
> is add a flag to specify extra chunks topreserve if found in the input. Would
> that be helpful in your case?
>
> -- Måns Rullgård

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