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
>
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