osalp-dev

[Osalp-dev] OSALP -> OS X -> convert MP3 to WAV

[J. T. S. <mai...@ja...>]

Hi All,

I am interested in this project from a usage standpoint and also a  
contribution standpoint.

Can OSALP run on OS X?

With OSALP, can I take a .mp3, clip out 10 seconds (specifying a start  
time and a duration or end time) and write just that 10 second piece  
as a wav so I can use LAME to reencode as a new .mp3?

Can OSALP replace what I would use LAME for?

Thoughts are appreciated.

-Jason

Re: [Osalp-dev] Mix two wav files

[<da...@dc...>]

Yes.  In the source example directory there is a command line program
which shows how to do this.

Take a look at the mix function in audioChain.cc

The latest cvs has a configure script to help compile this example.

Deborah Venturelli wrote:
> Hello, 
> 
>  
> 
> I've just seen the pages about the OSALP software.
> 
> I've a question:
> 
> With this library is it possible to mix two wav files? 
> 
> I explain: I have two stereo wav files, and I need to create a single stereo
> wav file which has the left channel of the first wav file, and the right
> channel of the second wav file. Or I could have two mono wav files, and I
> have to create a stereo wav file in which in the left channel I have the
> first file, and in the right channel the second file.
> 
> Sorry for my English, I hope you have understood!
> 
>  
> 
>> Deborah Venturelli
> 
>> Leonardo Multimedia S.r.l.
> 
>> e-mail:  <mailto:d.v...@le...>
> d.v...@le...
> 
>  
> 
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[Osalp-dev] Mix two wav files

[Deborah V. <d.v...@le...>]

Hello, 

 

I've just seen the pages about the OSALP software.

I've a question:

With this library is it possible to mix two wav files? 

I explain: I have two stereo wav files, and I need to create a single stereo
wav file which has the left channel of the first wav file, and the right
channel of the second wav file. Or I could have two mono wav files, and I
have to create a stereo wav file in which in the left channel I have the
first file, and in the right channel the second file.

Sorry for my English, I hope you have understood!

 

>Deborah Venturelli

>Leonardo Multimedia S.r.l.

>e-mail:  <mailto:d.v...@le...>
d.v...@le...

 

Re: [Osalp-dev] Current OSALP status

[Anthony Z. <zw...@us...>]

osa...@li... wrote on 01/17/2008 01:26:54 AM:

> Hi Folks
> 
> I still occasionally get feedback regarding this project.  I looked at 
> the logs today and the releases are still holding steady at about 500 
> downloads a month.  I also still get a few emails about it a month.
> 
> Obviously, I'm not actively developing it and haven't been since 2002. 
> But I do still like the code especially because it's chaining is very 
> simple to use.
> 
> So, I'm curious about who is on this list and why?  Are you using OSALP 
> or did you just sign up and forget about during your search for a c++ 
> audio library?
> 
> Thanks
> -- Darrick
> 

At the time, I needed something like OSALP.  We were reading from an http 
stream and writing out an rtp stream and I had hoped OSALP would help with 
that.  At the time, it was not feasible, but I too liked how it worked, 
and didn't unsubscribe from the mailing list when we found it unsuitable 
for our usage.  I had thought eventually that it might evolve into 
something I could use in my own projects.

[Osalp-dev] Current OSALP status

[Darrick S. <da...@dc...>]

Hi Folks

I still occasionally get feedback regarding this project.  I looked at 
the logs today and the releases are still holding steady at about 500 
downloads a month.  I also still get a few emails about it a month.

Obviously, I'm not actively developing it and haven't been since 2002. 
But I do still like the code especially because it's chaining is very 
simple to use.

So, I'm curious about who is on this list and why?  Are you using OSALP 
or did you just sign up and forget about during your search for a c++ 
audio library?

Thanks
-- Darrick

Re: [Osalp-dev] converting audio formats

[darrick <da...@dc...>]

Hi Chris

Osalp is quite unmaintained at the momemnt.  Even then it only handled 
PCM data in wave files and quality settings weren't handled very well.

Have you looked into SOX?  Or mpg321?  Or Lame?

With all three you can easily do what you want and they are very well 
supported.

-- Darrick

Christian Seiringer wrote:
> Hello,
> 
> I want to make an application which can convert wav to mp3 or mp3 to wav or only change the quality of one of these formats. So I searched for libraries, with which I can do this work. I found OSALP and I thought it could be a good library for this. But now I have the problem, that I get an error, when I am trying converting a wav into a mp3. I also have problems, when i am changing the quality of a wav file. The next problem I have is that I could only change wav files with PCM format. So, is it possible to change wav files in an other format?
> So my question is, is it possible to make all the things I want with the library.
> 
> thanks chris
> ______________________________________________________________
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[Osalp-dev] converting audio formats

[Christian S. <chr...@we...>]

Hello,

I want to make an application which can convert wav to mp3 or mp3 to wav or only change the quality of one of these formats. So I searched for libraries, with which I can do this work. I found OSALP and I thought it could be a good library for this. But now I have the problem, that I get an error, when I am trying converting a wav into a mp3. I also have problems, when i am changing the quality of a wav file. The next problem I have is that I could only change wav files with PCM format. So, is it possible to change wav files in an other format?
So my question is, is it possible to make all the things I want with the library.

thanks chris
______________________________________________________________
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[Osalp-dev] Spectrum analyzer (fft) w/ graphics display

[Exflux-6 <dia...@gm...>]

Greetings,=20
  I found the xm.h in lessmotif package (i think thats the one), still
having problems building the editor.

However, I am essentially only looking to do an fft on the audio data
(in whichever format), and display the values from the stereo channels
on screen. I am using GL. Any simple programming examples you may have
available would be greatly usefull on how to use the library. Digging
through the library itself looking for answers is providing to be
increasingly difficult. The documentation which comes with the
package, (appears) to have no programming examples itself. Help!

-Brian.
--=20
di=B7a=B7lec=B7tic: (n)
The art or practice of arriving at the truth by the exchange of
logical arguments.

[Osalp-dev] Building examples error (editor, etc)

[Exflux-6 <dia...@gm...>]

y0,
 just joined the list. continuously attempting to develop some cool
sound visualizer software, after checking out just about every
resource and open source program on the net. I've stumbled upon osalp,
it seems hopefull that this could have what I need.

Problem is: its not building the editor example, its looking for
Xm/Xm.h I don't have that, not sure where to get it, any suggestions?
or even simpler examples available I can build to make sure this all
works?

Thanks,
-Brian

--=20
di=B7a=B7lec=B7tic: (n)
The art or practice of arriving at the truth by the exchange of
logical arguments.

[Osalp-dev] AC_CHECK_LIB question

[Paul S. <pau...@uc...>]

ayo,
I am developing an open source linux music player called TagFarm 
(tagfarm.sf.net) and would like to use the osalp library to handle 
playing music files. However, I cant seem to get autoconfig to recognize 
aflib. When i put the following line in configure.in I get an error 
saying it cant find aflibConfig in -laflib.

AC_CHECK_LIB( aflib, aflibConfig , , AC_MSG_ERROR( [**** no aflib go get 
it http://osalp.sourceforge.net/ ****]),)

i have tried several other symbols besides aflibConfig such as enable, 
process, output_config, etc but none seem to work. Does anyone know what 
im doing wrong....?

Also, I was admiring the beauty and easy of use this library brings. 
However, i believe that the aflibMpgFile implementation can be improved 
using mad source rather than Mpegtoraw. I was looking through the 
AlsaPlayer source and discovered that with a little time and tweaking, 
the aflibMpgFile could be implemented with mad. The benefits would be 
that it could play mp3 encoded with VBR and could seek mid-song without 
those sqweeking sounds. If you guys want I could start messing around 
with this.

-Dinocore-

Re: [Osalp-dev] Saving files...

[darrick <da...@dc...>]

Rua Haszard Morris wrote:

> Hi list, Darrick...
> 
> The last release was in 2002, is this project very
> active?  

It's very inactive.  Coding wise I've been more active with other projects.

> 
> Does anybody use this lib under windows - if so, what
> development environment?
> 
> Ogg seems to be supported, but I don't see an
> aflibOggFile on the web documentation - is the web doc
> out of date?

Yes.  But all the formats work essentially the same.  A big problem with the 
loadable modules was being able to pass format specific to them (i.e. bitrate). 
    I spent a lot of time trying to work this out but never came up with 
anything I liked all that much.  That's why things have been rather inactive 
because I was spending way to much time and going nowhere (except in circles.)

Note that aflibOggFile only reads ogg but doesn't write them.

-- Darrick

Re: [Osalp-dev] Saving files...

[darrick <da...@dc...>]

Hugh Macdonald wrote:
> 
> When loading, AUTO works fine for me - it's when creating a new file, AUTO doesn't pick up the format from the extension (which I'd expect it to)
The AUTO detection works only for opening files not creating them, as it's based 
on the content of the file (i.e. the header) and not the extension.
> 
> And no, I don't think I am deleting sFile... good point - I take it some critical code is in the destructor...

When the destructor is called the aiff header is updated with the number of 
samples written to the file.  So it is rather critical.

-- Darrick

Re: [Osalp-dev] Saving files...

[Hugh M. <hu...@mo...>]

On Thu, 04 Nov 2004 18:04:13 -0800
darrick <da...@dc...> wrote:

> Well regarding this code.  I have to ask.  You are deleting sFile
> right?

Thanks - it appears that this was exactly the problem I was having!

-- 
Hugh Macdonald
The Moving Picture Company

Re: [Osalp-dev] Saving files...

[Hugh M. <hu...@mo...>]

On Thu, 04 Nov 2004 18:04:13 -0800
darrick <da...@dc...> wrote:

> Well regarding this code.  I have to ask.  You are deleting sFile
> right?  Also, could you send me off list a file you are that is not
> detected with the "AUTO" option.

When loading, AUTO works fine for me - it's when creating a new file, AUTO doesn't pick up the format from the extension (which I'd expect it to)

And no, I don't think I am deleting sFile... good point - I take it some critical code is in the destructor...

will try that one - cheers

-- 
Hugh Macdonald
The Moving Picture Company

Re: [Osalp-dev] Saving files...

[darrick <da...@dc...>]

Well regarding this code.  I have to ask.  You are deleting sFile right?  Also, 
could you send me off list a file you are that is not detected with the "AUTO" 
option.

Thanks
-- Darrick

Hugh Macdonald wrote:
  > Here's a chunk of my code:
> (audioData is a 2D array of size audioChans*audioSamples)
> 
> aflibConfig sInfo;
> aflibStatus sStatus;
> 
> sInfo.setChannels(audioChans);
> sInfo.setSampleSize(AFLIB_DATA_16S);
> sInfo.setSamplesPerSecond(audioSampleRate);
> sInfo.setTotalSamples(audioSamples);
> 
> aflibFile *sFile = aflibFile::create("AIFF", audioFile, sInfo, &sStatus);
> if(sStatus != AFLIB_SUCCESS)
> {
>   // Check what the error is, print error message and return
> }
> aflibData audData(sInfo, audioSamples);
> for(chan = 0; chan < audioChans; chan++)
> {
>   for(sample = 0; sample < audioSamples; sample++)
>   {
>     audData.setSample(audioData[chan][sample], sample, chan);
>   }
> }
> sFile->afwrite(audData);

Re: [Osalp-dev] Saving files...

[Rua H. M. <ru...@ya...>]

Hi list, Darrick...

I'm thinking about jumping on and starting to use
Osalp.  I have some questions - I'd appreciate any
answers/info anyone can give...

The last release was in 2002, is this project very
active?  

Does anybody use this lib under windows - if so, what
development environment?

Ogg seems to be supported, but I don't see an
aflibOggFile on the web documentation - is the web doc
out of date?

cheers,
Rua HM.

 --- darrick <da...@dc...> wrote: 
> Hi Hugh
> 
> What type of system are you running osalp on?  Are
> you running a straight 
> compile of osalp-0.7.3?  I'm not having any trouble
> on my slackware-10 x86 box 
> auto detecting aiff files nor with writing out the
> length fields.
> 
> Thanks
> -- Darrick
> 
> Hugh Macdonald wrote:
> >>I've been using osalp for a short while now for
> opening files (which I
> >>then store in a custom format before passing to
> libquicktime), but I'm
> >>now trying to go the other way, and am having
> slight problems....
> 
> 
>
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Re: [Osalp-dev] Saving files...

[darrick <da...@dc...>]

Hi Hugh

What type of system are you running osalp on?  Are you running a straight 
compile of osalp-0.7.3?  I'm not having any trouble on my slackware-10 x86 box 
auto detecting aiff files nor with writing out the length fields.

Thanks
-- Darrick

Hugh Macdonald wrote:
>>I've been using osalp for a short while now for opening files (which I
>>then store in a custom format before passing to libquicktime), but I'm
>>now trying to go the other way, and am having slight problems....

Re: [Osalp-dev] Saving files...

[Hugh M. <hu...@mo...>]

I think I've found part of the problem....

Again, maybe I'm doing something wrong, or maybe it's a dodgy file writer...

But when I look at a dump of my created file, the 'length' field (bytes 4-7) is set to 0. Am I missing a function to write this later, or is the code at fault?

I know this is a pretty quiet list, but I'd love any info I can get on this...

Cheers

Hugh Macdonald (at a slightly different e-mail address)

On Thu, 4 Nov 2004 09:08:32 +0000
Hugh Macdonald <hug...@gm...> wrote:

> I've been using osalp for a short while now for opening files (which I
> then store in a custom format before passing to libquicktime), but I'm
> now trying to go the other way, and am having slight problems....
> 
> Here's a chunk of my code:
> (audioData is a 2D array of size audioChans*audioSamples)
> 
> aflibConfig sInfo;
> aflibStatus sStatus;
> 
> sInfo.setChannels(audioChans);
> sInfo.setSampleSize(AFLIB_DATA_16S);
> sInfo.setSamplesPerSecond(audioSampleRate);
> sInfo.setTotalSamples(audioSamples);
> 
> aflibFile *sFile = aflibFile::create("AIFF", audioFile, sInfo, &sStatus);
> if(sStatus != AFLIB_SUCCESS)
> {
>   // Check what the error is, print error message and return
> }
> aflibData audData(sInfo, audioSamples);
> for(chan = 0; chan < audioChans; chan++)
> {
>   for(sample = 0; sample < audioSamples; sample++)
>   {
>     audData.setSample(audioData[chan][sample], sample, chan);
>   }
> }
> sFile->afwrite(audData);
> 
> 
> At the end of this, I have my file, but the header information seems
> to be corrupt... I've got a tool for printing out the header
> information of a .aif file - it uses libsndfile (this was what I used
> before I discovered osalp) - this gives correct header information
> except the number of frames (samples) which it reports as 0.
> 
> The other thing that's bugging me (which I suspect I can't do anything
> about) is that I don't seem to be able to set the format as "AUTO"
> when creating a file - If I do, I get
> AFLIB_ERROR_INITIALIZATION_FAILURE - it's not going to be too hard for
> me to write code that checks the needed extension and gives the right
> format, but I was surprised that this wasn't what "AUTO" did....
> 
> anyway, cheers for any help...
> 
> -- 
> Hugh Macdonald
> 
> 
> -------------------------------------------------------
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-- 
Hugh Macdonald
The Moving Picture Company

[Osalp-dev] Saving files...

[Hugh M. <hug...@gm...>]

I've been using osalp for a short while now for opening files (which I
then store in a custom format before passing to libquicktime), but I'm
now trying to go the other way, and am having slight problems....

Here's a chunk of my code:
(audioData is a 2D array of size audioChans*audioSamples)

aflibConfig sInfo;
aflibStatus sStatus;

sInfo.setChannels(audioChans);
sInfo.setSampleSize(AFLIB_DATA_16S);
sInfo.setSamplesPerSecond(audioSampleRate);
sInfo.setTotalSamples(audioSamples);

aflibFile *sFile = aflibFile::create("AIFF", audioFile, sInfo, &sStatus);
if(sStatus != AFLIB_SUCCESS)
{
  // Check what the error is, print error message and return
}
aflibData audData(sInfo, audioSamples);
for(chan = 0; chan < audioChans; chan++)
{
  for(sample = 0; sample < audioSamples; sample++)
  {
    audData.setSample(audioData[chan][sample], sample, chan);
  }
}
sFile->afwrite(audData);


At the end of this, I have my file, but the header information seems
to be corrupt... I've got a tool for printing out the header
information of a .aif file - it uses libsndfile (this was what I used
before I discovered osalp) - this gives correct header information
except the number of frames (samples) which it reports as 0.

The other thing that's bugging me (which I suspect I can't do anything
about) is that I don't seem to be able to set the format as "AUTO"
when creating a file - If I do, I get
AFLIB_ERROR_INITIALIZATION_FAILURE - it's not going to be too hard for
me to write code that checks the needed extension and gives the right
format, but I was surprised that this wasn't what "AUTO" did....

anyway, cheers for any help...

-- 
Hugh Macdonald

[Osalp-dev] Mac OS X Support ?

[<kar...@ya...>]

Hello,

I'am trying to compile osalp on MacOs X (using Fink)

I've got this error:
bash-2.05a$ make
make  all-recursive
Making all in aflib
source='aflibAudioPitch.cc'
object='aflibAudioPitch.lo' libtool=yes \
depfile='.deps/aflibAudioPitch.Plo'
tmpdepfile='.deps/aflibAudioPitch.TPlo' \
depmode=gcc3 /bin/sh ../config/depcomp \
/bin/sh ../libtool --mode=compile g++ -DHAVE_CONFIG_H
-I. -I. -I..     -g -O2 -Wall 
-DMODULE_INSTALL_DIR="\"/usr/local/lib/aflib\"" -c -o
aflibAudioPitch.lo `test -f 'aflibAudioPitch.cc' ||
echo './'`aflibAudioPitch.cc
g++ -DHAVE_CONFIG_H -I. -I. -I.. -g -O2 -Wall
-DMODULE_INSTALL_DIR=\"/usr/local/lib/aflib\" -c
aflibAudioPitch.cc -MT aflibAudioPitch.lo -MD -MP -MF
.deps/aflibAudioPitch.TPlo  -fno-common -DPIC
In file included from aflibAudioSampleRateCvt.h:32,
                 from aflibAudioPitch.h:30,
                 from aflibAudioPitch.cc:28:
aflibConverter.h:225: error: parse error before
numeric constant
make[2]: *** [aflibAudioPitch.lo] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

It appear that some type are not defined...
Does the library work on Mac Os ?
What about little/big endian ?

Regard



	

	
		
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[Osalp-dev] questions about audio mixer libs

[Kun W. <we...@ca...>]

Hi;
I read the code and want to what ares the input formats supported by aflibAudioMixer? linear audio sample, wav, u-law?

Thanks
Kun

[Osalp-dev] adding classes

[<ad...@pr...>]

hi,

i'm new to this list, and new to programming audio really, but I was just
wondering about where to add all the source files of this library=2E I'm
running VC++=2Enet and I wasn't sure=2E Or if this is too general a questi=
on,
could someone point me to a good site where there is more information I ca=
n
read about programming audio? Thanks in advance =2E =2E=20

reason lahalla

--------------------------------------------------------------------
mail2web - Check your email from the web at
http://mail2web=2Ecom/ =2E

[Osalp-dev] MP3 mixing

[<mat...@pi...>]

Hi everybody,

Nice project!

Sorry if I ask a question previously answerd, but the archives of the
mailing list does'nt work...

Is it possible to mix multiple MP3 files into one MP3 file using the
libraries as is?

I have browsed trough the API reference and I've noticed aflibAudioMixer =
and
the compute_segment method. I took a look at the code but I'm not figurin=
g
everything out. And before going further, I would like to know if it can =
do
the job...

If not, is there any other open source code doing that?

Thanks,
G=E9rard Materna.

Re: [Osalp-dev] To play wav file

[Senthil K. <sen...@ho...>]

Dear Sir,

   If possible and time permits please give me some feedback on the sample 
program which i have written to play the wav file.

Attachment : WavPlayer.c
                  prompt.wav       ( properties of the wav file : 8 bit , 
mono, 11025 Hz)

When i play through XMMS (player), i am able to hear good quality of sound.
But with my program some noise comes along with the sound.

In the programm i am setting the Master volume of Left speaker as 100 %. 
Will it have any impact on the sound quality( because of amplification).

with thanks and regards

Senthil








>From: Darrick Servis <da...@dc...>
>To: "Senthil Kumar" <sen...@ho...>,   
>osa...@li...
>Subject: Re: [Osalp-dev] To play wav file
>Date: Mon, 19 May 2003 10:47:31 -0700
>
>To convert 8 bit unsigned mono to 16 bit signed stereo ...
>
>int data_length = 4096;
>int channels = 2;
>char* orig_ptr = malloc(data_length*sizeof(unsigned char));
>short* dest_ptr = malloc(data_length*channels*sizeof(signed int));
>
>data_length = fread(orig_ptr,sizeof(unsigned char),data_length,FILE_IN);
>
>for (i = 0; i < data_length; i++){
>      /* convert to signed 16 */
>      *dest_ptr++ = (*orig_ptr++ << 8)^0x8000 ;
>      /* copy left channel to right */
>      *dest_ptr++ = *dest_ptr;
>}
>
>fwrite(dest_ptr,sizeof(signed int), data_length*channels,DSP_OUT);
>
>Stereo data is interleaved...
>
>You're getting dropouts because the timer is not synced to the soundcard.
>That's not the way to do it anyway.  Make a seperate thread from your
>interface thread and do...
>
>while( play_file() )
>{
>       play_chunk();
>}
>
>Writing to the soundcard blocks by default.
>
>Good luck,
>Darrick
>
>On Monday 19 May 2003 03:44 am, Senthil Kumar wrote:
> > Dear All,
> >
> >    I am working in C with Linux as OS.
> > I am having some problem in playing the raw pcm data extracted from wav
> > file.
> > My wav file is 8bit, Mono, 11025Hz.
> >
> > My audio driver supports 16bit, Stereo (2 channels), Sampling rate (can 
>be
> > varied using ioctl system call).
> >
> > Since the Audio Driver properties cannot be changed for the Bits/sample 
>and
> > channel, i made a convertion from 8 to 16 bits by multipying 8_bit_data
> > *255 and assinged to 16_bit_array.
> > But sound comes with some noise.
> >
> > Since the audio driver expects two 16 bit data at a time (because of 2
> > channel), i defined an int32 bit array of size 4096 . Read 4096 sample 
>from
> > the wav file and filled in the array.
> >
> > I placed the 8 bit PCM data (extracted from wav file )in the first byte
> > int32 bit array and wrote it to the dsp device.
> >
> > for(i=0;i<BUFFERSIZE;i++)
> >      {
> >     fread(&data,sizeof(unsigned char),1,ptr);
> > //   fread(&buffer1,sizeof(unsigned char),1,ptr1);
> >     audio_buffer1[i]=((((unsigned  int) data)<<24)|0x00000000);
> > }
> >
> >
> >
> > With the volume level as 100 % for the left speaker (since i am playing 
>the
> > sound in the left speaker) i am able to get the sound, but not better
> > quality.
> >
> > As i want to control play and stop immideately, i wrote the sound data 
>in
> > sizes of 4096 samples.
> >
> > I kept one timer process, so that it will send message to my process at
> > every 371ms.
> > I calculated the time as
> >
> > 1 bit=( 1/11025*32) =2.834 micro seconds
> > 4096*32 bits = 2micro seconds * 4096 *32 =371milli second.
> >
> > After every reading of 4096 samples from the wav file till the end of 
>the
> > file, i wrote the data into the dsp device using the write function.
> >
> > But  the sound played was not continuous and  some jerk in the sound.
> > Please help me to solve this problem.
> >
> >
> >
> > with thanks and regards
> > Senthil
> >
> > _________________________________________________________________
> > Technical writer? Earn more now.
> > http://server1.msn.co.in/msnleads/tis/index.asp Click here.
> >
> >
> >
> > -------------------------------------------------------
> > This SF.net email is sponsored by: If flattening out C++ or Java
> > code to make your application fit in a relational database is painful,
> > don't do it! Check out ObjectStore. Now part of Progress Software.
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> > _______________________________________________
> > Osalp-dev mailing list
> > Osa...@li...
> > https://lists.sourceforge.net/lists/listinfo/osalp-dev
>

_________________________________________________________________
Narain Karthikeyan. The fastest Indian. 
http://server1.msn.co.in/msnspecials/narain/index.asp Know more about him.

Re: [Osalp-dev] To play wav file

[Darrick S. <da...@dc...>]

To convert 8 bit unsigned mono to 16 bit signed stereo ...

int data_length = 4096;
int channels = 2;
char* orig_ptr = malloc(data_length*sizeof(unsigned char));  
short* dest_ptr = malloc(data_length*channels*sizeof(signed int));

data_length = fread(orig_ptr,sizeof(unsigned char),data_length,FILE_IN);

for (i = 0; i < data_length; i++){
     /* convert to signed 16 */
     *dest_ptr++ = (*orig_ptr++ << 8)^0x8000 ;
     /* copy left channel to right */
     *dest_ptr++ = *dest_ptr;
}

fwrite(dest_ptr,sizeof(signed int), data_length*channels,DSP_OUT);

Stereo data is interleaved...

You're getting dropouts because the timer is not synced to the soundcard.  
That's not the way to do it anyway.  Make a seperate thread from your 
interface thread and do...

while( play_file() )
{
      play_chunk();
}

Writing to the soundcard blocks by default.

Good luck,
Darrick

On Monday 19 May 2003 03:44 am, Senthil Kumar wrote:
> Dear All,
>
>    I am working in C with Linux as OS.
> I am having some problem in playing the raw pcm data extracted from wav
> file.
> My wav file is 8bit, Mono, 11025Hz.
>
> My audio driver supports 16bit, Stereo (2 channels), Sampling rate (can be
> varied using ioctl system call).
>
> Since the Audio Driver properties cannot be changed for the Bits/sample and
> channel, i made a convertion from 8 to 16 bits by multipying 8_bit_data
> *255 and assinged to 16_bit_array.
> But sound comes with some noise.
>
> Since the audio driver expects two 16 bit data at a time (because of 2
> channel), i defined an int32 bit array of size 4096 . Read 4096 sample from
> the wav file and filled in the array.
>
> I placed the 8 bit PCM data (extracted from wav file )in the first byte
> int32 bit array and wrote it to the dsp device.
>
> for(i=0;i<BUFFERSIZE;i++)
>      {
>     fread(&data,sizeof(unsigned char),1,ptr);
> //   fread(&buffer1,sizeof(unsigned char),1,ptr1);
>     audio_buffer1[i]=((((unsigned  int) data)<<24)|0x00000000);
> }
>
>
>
> With the volume level as 100 % for the left speaker (since i am playing the
> sound in the left speaker) i am able to get the sound, but not better
> quality.
>
> As i want to control play and stop immideately, i wrote the sound data in
> sizes of 4096 samples.
>
> I kept one timer process, so that it will send message to my process at
> every 371ms.
> I calculated the time as
>
> 1 bit=( 1/11025*32) =2.834 micro seconds
> 4096*32 bits = 2micro seconds * 4096 *32 =371milli second.
>
> After every reading of 4096 samples from the wav file till the end of the
> file, i wrote the data into the dsp device using the write function.
>
> But  the sound played was not continuous and  some jerk in the sound.
> Please help me to solve this problem.
>
>
>
> with thanks and regards
> Senthil
>
> _________________________________________________________________
> Technical writer? Earn more now.
> http://server1.msn.co.in/msnleads/tis/index.asp Click here.
>
>
>
> -------------------------------------------------------
> This SF.net email is sponsored by: If flattening out C++ or Java
> code to make your application fit in a relational database is painful,
> don't do it! Check out ObjectStore. Now part of Progress Software.
> http://www.objectstore.net/sourceforge
> _______________________________________________
> Osalp-dev mailing list
> Osa...@li...
> https://lists.sourceforge.net/lists/listinfo/osalp-dev