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Added format conversion and resampling through pl_format and pl_resample. Someone please check my implementation for bugs. 

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3661 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
mswitch 2001-12-22 16:19:00 +00:00
parent c4ef688a08
commit a6d1e0ce2f
1 changed files with 52 additions and 20 deletions
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72
libao2/ao_dxr3.c
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@ -15,6 +15,7 @@
#include "audio_out.h"
#include "audio_out_internal.h"
#include "audio_plugin.h"
void perror( const char *s );
#include <errno.h>
@ -33,6 +34,7 @@ LIBAO_EXTERN(dxr3)
static audio_buf_info dxr3_buf_info;
static int fd_control = 0, fd_audio = 0;
int need_conversion = 0;
// to set/get/query special features/parameters
static int control(int cmd,int arg)
@ -54,6 +56,15 @@ static int control(int cmd,int arg)
static int init(int rate,int channels,int format,int flags)
{
int ioval;
ao_plugin_data.rate = rate;
ao_plugin_data.channels = channels;
ao_plugin_data.format = format;
ao_plugin_data.sz_mult = 1;
ao_plugin_data.sz_fix = 0;
ao_plugin_data.delay_mult = 1;
ao_plugin_data.delay_fix = 0;
ao_plugin_cfg.pl_format_type = format;
ao_plugin_cfg.pl_resample_fout = rate;
fd_audio = open( "/dev/em8300_ma", O_WRONLY );
if( fd_audio < 0 )
{
@ -69,18 +80,18 @@ static int init(int rate,int channels,int format,int flags)
}
ioctl(fd_audio, SNDCTL_DSP_RESET, NULL);
ao_data.format = format;
if( ioctl (fd_audio, SNDCTL_DSP_SETFMT, &ao_data.format) < 0 )
printf( "AO: [dxr3] Unable to set audio format\n" );
if(format == AFMT_AC3 && ao_data.format != AFMT_AC3)
printf( "AO: [dxr3] Unable to set audio format\n" );
if(format != ao_data.format)
{
printf("AO: [dxr3] Can't set audio device /dev/em8300_ma to AC3 output\n");
return 0;
need_conversion |= 0x1;
ao_data.format = AFMT_S16_LE;
ao_plugin_data.format = format;
ao_plugin_cfg.pl_format_type = ao_data.format;
}
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;
if(format != AFMT_AC3)
if(channels>2)
@ -102,19 +113,12 @@ static int init(int rate,int channels,int format,int flags)
printf( "AO: [dxr3] Unable to set samplerate\n" );
return 0;
}
if( rate < ao_data.samplerate )
if( rate != ao_data.samplerate )
{
ao_data.samplerate = 44100;
ioctl(fd_audio, SNDCTL_DSP_SPEED, &ao_data.samplerate);
if( ao_data.samplerate != 44100 )
{
printf( "AO: [dxr3] Unable to set samplerate\n" );
return 0;
}
printf("AO: [dxr3] Using %d Hz samplerate (requested: %d) (Upsampling)\n",ao_data.samplerate,rate);
ao_data.samplerate = rate;
need_conversion |= 0x2;
ao_plugin_data.rate = rate;
ao_plugin_cfg.pl_resample_fout = ao_data.samplerate;
}
else printf("AO: [dxr3] Using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
if( ioctl(fd_audio, SNDCTL_DSP_GETOSPACE, &dxr3_buf_info)==-1 )
{
@ -138,6 +142,24 @@ static int init(int rate,int channels,int format,int flags)
ao_data.outburst=dxr3_buf_info.fragsize;
}
if(need_conversion)
{
if(need_conversion & 0x1)
{
if(!audio_plugin_format.init())
return 0;
ao_plugin_data.len = ao_data.buffersize*2;
audio_plugin_format.control(AOCONTROL_PLUGIN_SET_LEN,0);
}
if(need_conversion & 0x2)
{
if(!audio_plugin_resample.init())
return 0;
ao_plugin_data.len = ao_data.buffersize*2;
audio_plugin_resample.control(AOCONTROL_PLUGIN_SET_LEN,0);
}
}
ioval = EM8300_PLAYMODE_PLAY;
if( ioctl( fd_control, EM8300_IOCTL_SET_PLAYMODE, &ioval ) < 0 )
printf( "AO: [dxr3] Unable to set playmode\n" );
@ -152,6 +174,8 @@ static void uninit()
printf( "AO: [dxr3] Uninitializing\n" );
if( ioctl(fd_audio, SNDCTL_DSP_RESET, NULL) < 0 )
printf( "AO: [dxr3] Unable to reset device\n" );
if(need_conversion & 0x1) audio_plugin_format.uninit();
if(need_conversion & 0x2) audio_plugin_resample.uninit();
close( fd_audio );
close( fd_control ); /* Just in case */
}
@ -161,6 +185,8 @@ static void reset()
{
if( ioctl(fd_audio, SNDCTL_DSP_RESET, NULL) < 0 )
printf( "AO: [dxr3] Unable to reset device\n" );
if(need_conversion & 0x1) audio_plugin_format.reset();
if(need_conversion & 0x2) audio_plugin_resample.reset();
}
// stop playing, keep buffers (for pause)
@ -216,8 +242,14 @@ static int get_space()
// return: number of bytes played
static int play(void* data,int len,int flags)
{
len /= ao_data.outburst;
return write(fd_audio,data,len*ao_data.outburst);
int tmp = get_space();
int size = (tmp<len)?tmp:len;
ao_plugin_data.data = data;
ao_plugin_data.len = size;
if(need_conversion & 0x1) audio_plugin_format.play();
if(need_conversion & 0x2) audio_plugin_resample.play();
write(fd_audio,ao_plugin_data.data,ao_plugin_data.len);
return size;
}
// return: delay in seconds between first and last sample in buffer