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
> >
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