[MP3 ENCODER] LAME encoder

2003-03-13 Thread Pusillus
Hi,
I need to play a 128bit stereo mp3 file. Im tryng to
use lame library to do, but I obtain a noised pcm
frames.
what's wrong?

#include stdio.h
#includefcntl.h
#includestdlib.h
#includeunistd.h
#include sys/ioctl.h
#include linux/soundcard.h
#include lame.h

FILE *rec;
FILE *raw;

mp3data_struct mp3str;
mp3data_struct *mp3data;
short int pcm_l[1152],pcm_r[1152];

unsigned char buf[200];
int ret,len,ARG;
int audio;

int main()
{
mp3data=mp3str;
rec=fopen(test.mp3,r);
raw=fopen(raw.raw,w);
audio=open(/dev/dsp,O_RDWR);
if (audio  0) {
perror(open of /dev/dsp failed);
exit(1);
  }


lame_decode_init();

while(mp3data-header_parsed!=1)
{
len=fread(buf,1,sizeof(buf),rec);
ret=lame_decode1_headers(buf, len,
pcm_l, pcm_r, mp3data);
}
printf
(stereo=%i\nsamlpr=%i\nbitrate=%i\nmode%i\nmode_ext=%i\nframesize%i\nnsamp=%d\ntotalfr=%i,mp3data-stereo,mp3data-samplerate,mp3data-bitrate,mp3data-mode,mp3data-mode_ext,mp3data-framesize,mp3data-nsamp,mp3data-totalframes,mp3data-framenum);
mp3data-header_parsed=0;
ARG=mp3data-samplerate/2;
ioctl(audio,SOUND_PCM_WRITE_RATE,ARG);
ioctl(audio,SOUND_PCM_READ_RATE,ARG);

printf(\nrate = %i\n,ARG);

ARG=mp3data-stereo;
ioctl(audio, SOUND_PCM_WRITE_CHANNELS, ARG);

ARG=16;
ioctl(audio, SOUND_PCM_WRITE_BITS,ARG);
write(audio,pcm_l,ret);
 write(audio,pcm_r,ret);
 fwrite(pcm_l,sizeof(short),ret,raw);
 fwrite(pcm_r,sizeof(short),ret,raw);

   do{

while((len0)(ret1))
{ 
len=fread(buf,1,sizeof(buf),rec);
  ret=lame_decode(buf,
len, pcm_l, pcm_r);
}
printf(ret=%i,ret);

mp3data-header_parsed=0;
   write(audio,pcm_l,ret);
   write(audio,pcm_r,ret);
fwrite(pcm_l,sizeof(short),ret,raw);
fwrite(pcm_r,sizeof(short),ret,raw);
ret=0;


   }while (len0);





lame_decode_exit();
fclose(rec);
fclose(raw);
return 0;
}

thank's

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Re: [MP3 ENCODER] lame encoder

2003-03-07 Thread bob singer


--- Damian Pitcon [EMAIL PROTECTED] wrote:
Hi all, I have a question that I'd love if someone
could help me with. I'm using cool edit pro 2 on
windows XP and have installed the lame cooledit plugin
into the main CE directory. But when I try to encode
it says invalid file or format'. Any ideas whats up?
Please e-mail me at: [EMAIL PROTECTED] 
thanks!
Damian

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Damien

I believe the Cool Edit plugins are on a seperate directory for plugins and data 
libraries only. I'll double check that when I get back to work in a couple days, but I 
think it's the same as sound forge or vegas, with a plugins directory, possibly in 
another options or accessories directory off the main. Check your windows registry 
too, to make sure it shows up in the local_machine,software area. Sometimes .dll files 
can end up attached to other software and cause conflicts. Sorry Mister Gates... 
Microsoft is not an exact science.

B$


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[MP3 ENCODER] lame encoder

2003-03-06 Thread Damian Pitcon
Hi all, I have a question that I'd love if someone
could help me with. I'm using cool edit pro 2 on
windows XP and have installed the lame cooledit plugin
into the main CE directory. But when I try to encode
it says invalid file or format'. Any ideas whats up?
Please e-mail me at: [EMAIL PROTECTED] 
thanks!
Damian

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Re: [MP3 ENCODER] lame encoder

2003-03-06 Thread PhillLop
sorry mp3encoder



Re: [MP3 ENCODER] Lame Encoder!

2002-12-10 Thread Gabriel Bouvigne
 I have found that lame encoder (lame.exe) can't encode wav file that have
 spaces in its' filename/folder. For examples:

 C:\My Music\Wav 01.wav (both have spaces in folder  filename)
 C:\My Music\Wav01.wav (have space in folder name)
 C:\MyMusic\Wav 01.wav (have space in filename)


It is the common behaviour of win32 console applications. You have either to
use the short name, or to put the filename between double quotes.

Regards,


Gabriel Bouvigne
www.mp3-tech.org
personal page: http://gabriel.mp3-tech.org

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Antwort: [MP3 ENCODER] Lame Encoder!

2002-12-10 Thread Detlef BIERITZ

hi there

put it in 

C:\My Music\Wav 01.wav

regards
d. bieritz
[EMAIL PROTECTED]







Jutawan Muda [EMAIL PROTECTED]
Gesendet von: [EMAIL PROTECTED]
09.12.2002 13:19
Bitte antworten an mp3encoder

An:
   [EMAIL PROTECTED]
Kopie:
   
Thema:
   [MP3 ENCODER] Lame Encoder!


Hi  Hello,

I have found that lame encoder (lame.exe) can't encode wav file that have

spaces in its' filename/folder. For examples:

C:\My Music\Wav 01.wav (both have spaces in folder  filename)
C:\My Music\Wav01.wav (have space in folder name)
C:\MyMusic\Wav 01.wav (have space in filename)

Please fix these in the next version.

Thanks
Mr.Bean




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[MP3 ENCODER] Lame Encoder!

2002-12-09 Thread Jutawan Muda
Hi  Hello,

I have found that lame encoder (lame.exe) can't encode wav file that have 
spaces in its' filename/folder. For examples:

C:\My Music\Wav 01.wav (both have spaces in folder  filename)
C:\My Music\Wav01.wav (have space in folder name)
C:\MyMusic\Wav 01.wav (have space in filename)

Please fix these in the next version.

Thanks
Mr.Bean




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Re: [MP3 ENCODER] Lame encoder input- Help needed

2001-12-04 Thread Stefan Ohlsson

On Mon, 3 Dec 2001, Mark Taylor wrote:
2. I encoded a file in two ways. One - using the lame encoder command line
options and the other by writing a small program to read data from a file
and then using libmp3lame functions to encode. In both cases I used the ABR
(average bitrate mode) at 12kbps. The only difference being that in the
second case the encoder doesnt know the filesize( and hence doesnt know the
total number of samples to be encoded either).
 When playing back on WinAmp the command line encoded mp3 file bitrate keeps
switching between 8 and 16kbps (as displayed in the player window). But the
player shows a bitrate of 13kbps for the second file.
Am I doing something wrong or does the encoder need to know the filesize
beforehand?
play the files with lame --decode and see what it does.

One difference might be that the libmp3lame version will not
have the Xing VBR tag unless you specifically called the routine
from the LAME API to add the tag.  Maybe someone who uses WinAmp will
know if it treats VBR files with and without the VBR tag differently?

Yes it does. For files with VBR tag it shows the /current/ bitrate
of the stream as it changes, but for files without the tag it
shows the /average/ bitrate so far.

/Stefan
-- 
[ Stefan Ohlsson ] · http://www.cixit.se/ · [ 020-4702019 / +46-709-561314 ]

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Re: [MP3 ENCODER] Lame encoder input- Help needed

2001-12-04 Thread Alexander Leidinger

On  3 Dez, Manoj Palki wrote:

 2. I encoded a file in two ways. One - using the lame encoder command line
 options and the other by writing a small program to read data from a file
 and then using libmp3lame functions to encode. In both cases I used the ABR
 (average bitrate mode) at 12kbps. The only difference being that in the
 second case the encoder doesnt know the filesize( and hence doesnt know the
 total number of samples to be encoded either).

Have a look at slame (= simple lame) in
http://www.leidinger.net/lame/py-lame.tar.bz2 (python bindings for
libmp3lame, still a lot to do (no ID3 yet, only output to a file and
only encoding of 16bit interleaved stereo at the moment)), it does the
right thing in this case (set the total number of samples of the input).

Bye,
Alexander.

-- 
  Loose bits sink chips.

http://www.Leidinger.net   Alexander @ Leidinger.net
  GPG fingerprint = C518 BC70 E67F 143F BE91  3365 79E2 9C60 B006 3FE7

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RE: [MP3 ENCODER] Lame encoder input- Help needed

2001-12-03 Thread Manoj Palki

Thanks a lot Mark. I have some more questions and would be very grateful if
someone can answer them.

1. When lame combines standard bitrates to get the required average, over
what range is this average achieved? ( For ex : does lame achieve the
average over a specified number of frames or is it over a specified time or
is it the entire file)

2. I encoded a file in two ways. One - using the lame encoder command line
options and the other by writing a small program to read data from a file
and then using libmp3lame functions to encode. In both cases I used the ABR
(average bitrate mode) at 12kbps. The only difference being that in the
second case the encoder doesnt know the filesize( and hence doesnt know the
total number of samples to be encoded either).
 When playing back on WinAmp the command line encoded mp3 file bitrate keeps
switching between 8 and 16kbps (as displayed in the player window). But the
player shows a bitrate of 13kbps for the second file. 
Am I doing something wrong or does the encoder need to know the filesize
beforehand?

Thanks


 -Original Message-
 From: Mark Taylor [mailto:[EMAIL PROTECTED]]
 Sent: Friday, November 30, 2001 2:07 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [MP3 ENCODER] Lame encoder input- Help needed
 
 
  
  Hi,
 I want to encode mp3's at 12kbps (constant bit rate). 
 Lame presently only
  supports 8, 16, 32kbps and upwards. Does anyone know of the 
 amount of effort
  which would be involved in modifying the code to support 
 12kbps? Any help on
  how to go about it would be greatly appreciated.
  
  Thanks
  Manoj
 
 You have two choices:
 
 lame --freeformat 12
 
 which will make all frames 12kbps, but only a few fully complient
 decoders will be able to handle it, or:
 
 lame --abr 12 
 
 which will use a combination of standard bitrate frames to get
 an average of 12kbps.
 
 Mark
 
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Re: [MP3 ENCODER] Lame encoder input- Help needed

2001-12-03 Thread Mark Taylor


 
 Thanks a lot Mark. I have some more questions and would be very grateful if
 someone can answer them.
 
 1. When lame combines standard bitrates to get the required average, over
 what range is this average achieved? ( For ex : does lame achieve the
 average over a specified number of frames or is it over a specified time or
 is it the entire file)
 
the average is not guaranteed.  I think it usually comes within 95%,
and almost always on the low side.  It's an easy fix to 
tweak the algorithm to get this to come out exact, but right now it
is close enough no one has bothered to fix it.


 2. I encoded a file in two ways. One - using the lame encoder command line
 options and the other by writing a small program to read data from a file
 and then using libmp3lame functions to encode. In both cases I used the ABR
 (average bitrate mode) at 12kbps. The only difference being that in the
 second case the encoder doesnt know the filesize( and hence doesnt know the
 total number of samples to be encoded either).
  When playing back on WinAmp the command line encoded mp3 file bitrate keeps
 switching between 8 and 16kbps (as displayed in the player window). But the
 player shows a bitrate of 13kbps for the second file. 
 Am I doing something wrong or does the encoder need to know the filesize
 beforehand?
 
 Thanks
 

play the files with lame --decode and see what it does.  

One difference might be that the libmp3lame version will not
have the Xing VBR tag unless you specifically called the routine
from the LAME API to add the tag.  Maybe someone who uses WinAmp will
know if it treats VBR files with and without the VBR tag differently?

Mark
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Re: [MP3 ENCODER] Lame encoder input- Help needed

2001-11-30 Thread Mark Taylor

 
 Hi,
I am trying to encode raw PCM files into mp3 using the libmp3lame
 library.
 The PCM files are sampled at 8000khz, 16bit mono.
 My rather basic questions are
 
 1) Should I read from the PCM file into an unsigned or a signed buffer?
 2) When I call the function 'lame_encode_buffer_int' should I input 576
 samples or 1152 samples?
 

#2: 
The input buffer can be any size you are willing to allocate memory
for.  I would use whatever buffer you already have to input the data.
Just make sure the mp3 output buffer is large enough to
handle the output.  see the file API for details. 

#1:  
There are 3 possible routines, all of which take signed data
but the scaling differs depending on the input type.  This
is tersely documented in lame.h.  


   input   data range

lame_encode_buffer() short int  +/- 32768
lame_encode_buffer_float() float+/- 32768
lame_encode_buffer_long()  long +/- 32768
lame_encode_buffer_int()int +/- 2^(8*size(int)-1)


The scaling used by lame_encode_buffer_long() is a mistake,
since I believe long is usually 4 bytes, so we dont allow
for the full range.  This should probably be fixed.

Internally, the data is only used as 'float', so if you are
doing any type of DSP and have converted to float, dont 
quantize back to an integer before calling lame.


Mark

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Re: [MP3 ENCODER] Lame encoder input- Help needed

2001-11-30 Thread Mark Taylor

 
 Hi,
I want to encode mp3's at 12kbps (constant bit rate). Lame presently only
 supports 8, 16, 32kbps and upwards. Does anyone know of the amount of effort
 which would be involved in modifying the code to support 12kbps? Any help on
 how to go about it would be greatly appreciated.
 
 Thanks
 Manoj

You have two choices:

lame --freeformat 12

which will make all frames 12kbps, but only a few fully complient
decoders will be able to handle it, or:

lame --abr 12 

which will use a combination of standard bitrate frames to get
an average of 12kbps.

Mark

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