er Frank Neumann or me. Please use the
keywords "ZKM registration" in the subject. We will post the list of
talks/presentations and further information on the meeting from time
to time, so that you can decide about joining this meeting.
Matthias
--
Martin Wolters
Creative ATC
1500
ge to make it more
representative are appreciated. I want the database to be accesible to
anyone who can provide professional support in any way shape or form
for Linux Audio.
http://www.djcj.org
Best regards.
--
Martin Wolters
Creative ATC
1500 Green Hills Road
Scotts Valley, CA 95066
Oh well. So now the "official" code looks like this:
#include
unsigned char * in; //point to buffer returned by cd_read_audio
#if __BYTE_ORDER == __LITTLE_ENDIAN
short input = (short)(in[0]|(in[1] << 8));
#else
short input = (short)(in[1]|(in[0] << 8));
// ?? didn'
Thanks everybody for the help. To bring the topic to an end, here is the code
that works:
#include
unsigned char * in; //point to buffer returned by cd_read_audio
#ifdef __LITTLE_ENDIAN
short input = (short)(in[0]|(in[1] << 8)); // PC
#else
short input = (short)(in[1]|(in[0]
correct? I now use the following lines:
#include
#define s16 __s16
#define u16 __u16
> On Fri, Aug 09, 2002 at 06:18:44PM -0700, Martin Wolters wrote:
> > I am trying to read out data from a CDDA and convert it into float (on
> > Linux x86). The data from the CD is ret
I am trying to read out data from a CDDA and convert it into float (on
Linux x86). The data from the CD is returned in an unsigned char buffer.
That code looks something like this:
int cd_read_audio(int cdrom_fd, int lba, int num, unsigned char *buf)
{
struct cdrom_read_audio ra;
ra.a
d probably be very Linux minded :) Anybody knows one? And
> otherwise, are people interested in starting one?
>
> greetz,
> Kasper
>
--
Martin Wolters
Creative ATC
1500 Green Hills Road
Scotts Valley, CA 95066
Phone: ++1 (831) 440-2848
Fax: ++1 (831) 440-2882
submitted to either Dave or myself and we will give advice or make
> changes to ensure our publishing standard is met. If you are not able
> to send html don't worry because I am willing to format any braindumps
> and submit to Dave on your behalf.
>
> Regards.
>
--
Martin
The additional data at the beginning and end of the mp3 files is caused
by the need of ovxerlapping frames for the transformation between the
time and frequency domain. The length of added samples should be mostly
stable: There is only a very finite number of possibilities and it
should be eas
an "put it all
together" view. That's what I would like to see being addressed in the
new HowTo.
I am also interested in learning (and maybe conducting a poll) who is
using which libraries/systems.
Your comments and suggestions are welcome!
-M
--
Martin Wolters
Creative AT
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