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Remove lame upsampler, cleaned out stuff relating to setting channels. 

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3616 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
mswitch 2001-12-19 14:31:31 +00:00
parent 343b941007
commit 7a6d1acd7c
1 changed files with 16 additions and 49 deletions
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65
libao2/ao_dxr3.c
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@ -81,11 +81,21 @@ static int init(int rate,int channels,int format,int flags)
printf("AO: [dxr3] Sample format: %s (requested: %s)\n",
audio_out_format_name(ao_data.format), audio_out_format_name(format));
ao_data.channels=channels-1;
if( ioctl (fd_audio, SNDCTL_DSP_STEREO, &ao_data.channels) < 0 )
printf( "AO: [dxr3] Unable to set number of channels\n" );
ao_data.channels=channels;
if(format != AFMT_AC3)
if(channels>2)
if( ioctl (fd_audio, SNDCTL_DSP_CHANNELS, &ao_data.channels) < 0 )
printf( "AO: [dxr3] Unable to set number of channels\n" );
else
{
int c = channels-1;
if( ioctl(fd_audio,SNDCTL_DSP_STEREO,&c) < 0)
printf( "AO: [dxr3] Unable to set number of channels for AC3\n" );
}
ao_data.bps = (channels+1)*rate;
ao_data.bps = channels*rate;
if(format != AFMT_U8 && format != AFMT_S8)
ao_data.bps*=2;
ao_data.samplerate=rate;
if( ioctl (fd_audio, SNDCTL_DSP_SPEED, &ao_data.samplerate) < 0 )
{
@ -105,7 +115,6 @@ static int init(int rate,int channels,int format,int flags)
ao_data.samplerate = rate;
}
else printf("AO: [dxr3] Using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
if(format == AFMT_AC3 ) ao_data.bps *= 2;
if( ioctl(fd_audio, SNDCTL_DSP_GETOSPACE, &dxr3_buf_info)==-1 )
{
@ -207,50 +216,8 @@ static int get_space()
// return: number of bytes played
static int play(void* data,int len,int flags)
{
if( ao_data.samplerate < 44100 )
{
int i,j,ratio,len2;
unsigned char *data2,*s,*d;
ratio = 44100/ao_data.samplerate;ratio/=2;ratio*=2;
len2 = len * ratio;
data2 = malloc(len2);
s = data;
d = data2;
//Upsampler
if( ao_data.format == AFMT_U8 )
{
for(i=0;i<ratio/2;i++)
for(j=0;j<len;j++)
{
*d = *s;
d++;
*d = *s;
d++;s++;
}
}
else
{
for(i=0;i<ratio/2;i++)
for(j=0;j<len/2;j++)
{
*d = *s;
d++;s++;
*d = *s;
d++;s--;
*d = *s;
d++;s++;
*d = *s;
d++;s++;
}
}
if( len2 < 0 ) return 0;
write(fd_audio,data2,len2);
return len;
}
return write(fd_audio,data,len);
len /= ao_data.outburst;
return write(fd_audio,data,len*ao_data.outburst);
}
// return: delay in seconds between first and last sample in buffer