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Lots of fixes for digital audio output submitted by Steven Brookes <stevenjb@mda.co.uk> 

Hardware sync-code activated


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3828 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
mswitch 2001-12-28 10:20:16 +00:00
parent 296399a5bb
commit 74fc43e91a
1 changed files with 26 additions and 18 deletions
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44
libao2/ao_dxr3.c
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@ -79,6 +79,11 @@ static int init(int rate,int channels,int format,int flags)
return 0;
}
ioval = (format==AFMT_AC3)?EM8300_AUDIOMODE_DIGITALAC3:
EM8300_AUDIOMODE_ANALOG;
if( ioctl( fd_control, EM8300_IOCTL_SET_AUDIOMODE, &ioval ) < 0 )
printf( "AO: [dxr3] Unable to set audiomode\n" );
ioctl(fd_audio, SNDCTL_DSP_RESET, NULL);
ao_data.format = format;
@ -94,19 +99,25 @@ static int init(int rate,int channels,int format,int flags)
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;
}
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*rate;
if(format != AFMT_U8 && format != AFMT_S8)
ao_data.bps*=2;
if(format == AFMT_AC3)
ao_data.bps*=2;
ao_data.samplerate=rate;
if( ioctl (fd_audio, SNDCTL_DSP_SPEED, &ao_data.samplerate) < 0 )
{
@ -137,8 +148,8 @@ static int init(int rate,int channels,int format,int flags)
else
{
printf("AO: [dxr3] frags: %3d/%d (%d bytes/frag) free: %6d\n",
dxr3_buf_info.fragments+1, dxr3_buf_info.fragstotal, dxr3_buf_info.fragsize, dxr3_buf_info.bytes);
ao_data.buffersize=(dxr3_buf_info.bytes/2);
dxr3_buf_info.fragments, dxr3_buf_info.fragstotal, dxr3_buf_info.fragsize, dxr3_buf_info.bytes);
ao_data.buffersize=dxr3_buf_info.bytes;
ao_data.outburst=dxr3_buf_info.fragsize;
}
@ -160,6 +171,8 @@ static int init(int rate,int channels,int format,int flags)
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" );
@ -226,14 +239,13 @@ static void audio_resume()
static int get_space()
{
int space = 0;
if( ioctl(fd_audio, SNDCTL_DSP_GETODELAY, &space) < 0 )
if( ioctl(fd_audio, SNDCTL_DSP_GETOSPACE, &dxr3_buf_info)==-1 )
{
printf( "AO: [dxr3] Unable to get unplayed bytes in buffer\n" );
return ao_data.outburst;
printf( "AO: [dxr3] Unable to get unplayed bytes in buffer\n" );
return ao_data.outburst;
}
space = ao_data.buffersize - space;
space /= ao_data.outburst; /* This is a smart way of doing a fast modulo reduction */
space *= ao_data.outburst; /* fetched from ao_mpegpes.c */
space=dxr3_buf_info.fragments*dxr3_buf_info.fragsize;
return space;
}
@ -248,6 +260,8 @@ static int play(void* data,int len,int flags)
ao_plugin_data.len = size;
if(need_conversion & 0x1) audio_plugin_format.play();
if(need_conversion & 0x2) audio_plugin_resample.play();
if( ioctl(fd_audio, EM8300_IOCTL_AUDIO_SETPTS, &ao_data.pts) < 0 )
printf( "AO: [dxr3] Unable to set pts\n" );
write(fd_audio,ao_plugin_data.data,ao_plugin_data.len);
return size;
}
@ -255,12 +269,6 @@ static int play(void* data,int len,int flags)
// return: delay in seconds between first and last sample in buffer
static float get_delay()
{
int r=0;
if( ioctl(fd_audio, SNDCTL_DSP_GETODELAY, &r) < 0 )
{
printf( "AO: [dxr3] Unable to get unplayed bytes in buffer\n" );
return ((float)ao_data.buffersize)/(float)ao_data.bps;
}
return (((float)r)/(float)ao_data.bps);
return 0.0;
}