mpv/libao2/ao_dxr2.c

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/*
* DXR2 audio output driver
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <sys/ioctl.h>
#include <inttypes.h>
#include <dxr2ioctl.h>
#include "config.h"
#include "mp_msg.h"
#include "libavutil/common.h"
#include "mpbswap.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "libmpdemux/mpeg_packetizer.h"
static const ao_info_t info =
{
"DXR2 audio output",
"dxr2",
"Tobias Diedrich <ranma+mplayer@tdiedrich.de>",
""
};
LIBAO_EXTERN(dxr2)
static int volume=19;
static int last_freq_id = -1;
extern int dxr2_fd;
// to set/get/query special features/parameters
static int control(int cmd,void *arg){
switch(cmd){
case AOCONTROL_GET_VOLUME:
if(dxr2_fd > 0) {
ao_control_vol_t* vol = (ao_control_vol_t*)arg;
vol->left = vol->right = volume * 19.0 / 100.0;
return CONTROL_OK;
}
return CONTROL_ERROR;
case AOCONTROL_SET_VOLUME:
if(dxr2_fd > 0) {
dxr2_oneArg_t v;
float diff;
ao_control_vol_t* vol = (ao_control_vol_t*)arg;
// We need this trick because the volume stepping is often too small
diff = ((vol->left+vol->right) / 2 - (volume*19.0/100.0)) * 19.0 / 100.0;
v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff));
if(v.arg > 19) v.arg = 19;
if(v.arg < 0) v.arg = 0;
if(v.arg != volume) {
volume = v.arg;
if( ioctl(dxr2_fd,DXR2_IOC_SET_AUDIO_VOLUME,&v) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO DXR2] Setting volume to %d failed.\n",volume);
return CONTROL_ERROR;
}
}
return CONTROL_OK;
}
return CONTROL_ERROR;
}
return CONTROL_UNKNOWN;
}
static int freq=0;
static int freq_id=0;
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
if(dxr2_fd <= 0)
return 0;
last_freq_id = -1;
ao_data.outburst=2048;
ao_data.samplerate=rate;
ao_data.channels=channels;
ao_data.buffersize=2048;
ao_data.bps=rate*4;
ao_data.format=format;
freq=rate;
switch(rate){
case 48000:
freq_id=DXR2_AUDIO_FREQ_48;
break;
case 96000:
freq_id=DXR2_AUDIO_FREQ_96;
break;
case 44100:
freq_id=DXR2_AUDIO_FREQ_441;
break;
case 32000:
freq_id=DXR2_AUDIO_FREQ_32;
break;
case 22050:
freq_id=DXR2_AUDIO_FREQ_2205;
break;
#ifdef DXR2_AUDIO_FREQ_24
// This is not yet in the dxr2 driver CVS
// you can get the patch at
// http://www.tdiedrich.de/~ranma/patches/dxr2.pcm1723.20020513
case 24000:
freq_id=DXR2_AUDIO_FREQ_24;
break;
case 64000:
freq_id=DXR2_AUDIO_FREQ_64;
break;
case 88200:
freq_id=DXR2_AUDIO_FREQ_882;
break;
#endif
default:
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO DXR2] %d Hz not supported, try to resample.\n",rate);
return 0;
}
return 1;
}
// close audio device
static void uninit(int immed){
}
// stop playing and empty buffers (for seeking/pause)
static void reset(void){
}
// stop playing, keep buffers (for pause)
static void audio_pause(void)
{
// for now, just call reset();
reset();
}
// resume playing, after audio_pause()
static void audio_resume(void)
{
}
extern int vo_pts;
// return: how many bytes can be played without blocking
static int get_space(void){
float x=(float)(vo_pts-ao_data.pts)/90000.0;
int y;
if(x<=0) return 0;
y=freq*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
if(y>32768) y=32768;
return y;
}
static void dxr2_send_lpcm_packet(unsigned char* data,int len,int id,unsigned int timestamp,int freq_id)
{
int write_dxr2(const unsigned char *data, int len);
if(dxr2_fd < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"DXR2 fd is not valid\n");
return;
}
if(last_freq_id != freq_id) {
ioctl(dxr2_fd, DXR2_IOC_SET_AUDIO_SAMPLE_FREQUENCY, &freq_id);
last_freq_id = freq_id;
}
send_mpeg_lpcm_packet (data, len, id, timestamp, freq_id, write_dxr2);
}
// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags){
int write_dxr2(const unsigned char *data, int len);
// MPEG and AC3 don't work :-(
if(ao_data.format==AF_FORMAT_MPEG2)
send_mpeg_ps_packet (data, len, 0xC0, ao_data.pts, 2, write_dxr2);
else if(AF_FORMAT_IS_AC3(ao_data.format))
send_mpeg_ps_packet (data, len, 0x80, ao_data.pts, 2, write_dxr2);
else {
int i;
//unsigned short *s=data;
uint16_t *s=data;
#if !HAVE_BIGENDIAN
for(i=0;i<len/2;i++) s[i] = bswap_16(s[i]);
#endif
dxr2_send_lpcm_packet(data,len,0xA0,ao_data.pts-10000,freq_id);
}
return len;
}
// return: delay in seconds between first and last sample in buffer
static float get_delay(void){
return 0.0;
}