mirror of
https://github.com/mpv-player/mpv
synced 2024-12-24 15:52:25 +00:00
18004d2d98
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@21523 b3059339-0415-0410-9bf9-f77b7e298cf2
204 lines
4.5 KiB
C
204 lines
4.5 KiB
C
#include <math.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <sys/ioctl.h>
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#include <inttypes.h>
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#include <dxr2ioctl.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "help_mp.h"
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#include "libavutil/common.h"
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#include "mpbswap.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "libaf/af_format.h"
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#include "libmpdemux/mpeg_packetizer.h"
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static ao_info_t info =
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{
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"DXR2 audio output",
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"dxr2",
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"Tobias Diedrich <ranma+mplayer@tdiedrich.de>",
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""
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};
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LIBAO_EXTERN(dxr2)
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static int volume=19;
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static int last_freq_id = -1;
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extern int dxr2_fd;
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// to set/get/query special features/parameters
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static int control(int cmd,void *arg){
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switch(cmd){
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case AOCONTROL_GET_VOLUME:
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if(dxr2_fd > 0) {
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ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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vol->left = vol->right = volume * 19.0 / 100.0;
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return CONTROL_OK;
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}
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return CONTROL_ERROR;
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case AOCONTROL_SET_VOLUME:
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if(dxr2_fd > 0) {
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dxr2_oneArg_t v;
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float diff;
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ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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// We need this trick because the volume stepping is often too small
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diff = ((vol->left+vol->right) / 2 - (volume*19.0/100.0)) * 19.0 / 100.0;
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v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff));
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if(v.arg > 19) v.arg = 19;
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if(v.arg < 0) v.arg = 0;
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if(v.arg != volume) {
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volume = v.arg;
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if( ioctl(dxr2_fd,DXR2_IOC_SET_AUDIO_VOLUME,&v) < 0) {
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mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_SetVolFailed,volume);
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return CONTROL_ERROR;
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}
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}
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return CONTROL_OK;
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}
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return CONTROL_ERROR;
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}
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return CONTROL_UNKNOWN;
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}
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static int freq=0;
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static int freq_id=0;
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(int rate,int channels,int format,int flags){
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if(dxr2_fd <= 0)
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return 0;
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last_freq_id = -1;
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ao_data.outburst=2048;
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ao_data.samplerate=rate;
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ao_data.channels=channels;
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ao_data.buffersize=2048;
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ao_data.bps=rate*4;
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ao_data.format=format;
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freq=rate;
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switch(rate){
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case 48000:
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freq_id=DXR2_AUDIO_FREQ_48;
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break;
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case 96000:
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freq_id=DXR2_AUDIO_FREQ_96;
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break;
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case 44100:
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freq_id=DXR2_AUDIO_FREQ_441;
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break;
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case 32000:
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freq_id=DXR2_AUDIO_FREQ_32;
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break;
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case 22050:
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freq_id=DXR2_AUDIO_FREQ_2205;
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break;
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#ifdef DXR2_AUDIO_FREQ_24
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// This is not yet in the dxr2 driver CVS
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// you can get the patch at
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// http://www.tdiedrich.de/~ranma/patches/dxr2.pcm1723.20020513
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case 24000:
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freq_id=DXR2_AUDIO_FREQ_24;
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break;
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case 64000:
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freq_id=DXR2_AUDIO_FREQ_64;
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break;
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case 88200:
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freq_id=DXR2_AUDIO_FREQ_882;
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break;
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#endif
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default:
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mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_UnsupSamplerate,rate);
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return 0;
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}
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return 1;
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}
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// close audio device
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static void uninit(int immed){
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset(void){
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause(void)
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{
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// for now, just call reset();
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reset();
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}
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// resume playing, after audio_pause()
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static void audio_resume(void)
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{
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}
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extern int vo_pts;
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// return: how many bytes can be played without blocking
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static int get_space(void){
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float x=(float)(vo_pts-ao_data.pts)/90000.0;
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int y;
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if(x<=0) return 0;
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y=freq*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
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if(y>32768) y=32768;
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return y;
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}
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static void dxr2_send_lpcm_packet(unsigned char* data,int len,int id,unsigned int timestamp,int freq_id)
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{
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extern int write_dxr2(unsigned char *data, int len);
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if(dxr2_fd < 0) {
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mp_msg(MSGT_AO,MSGL_ERR,"DXR2 fd is not valid\n");
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return;
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}
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if(last_freq_id != freq_id) {
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ioctl(dxr2_fd, DXR2_IOC_SET_AUDIO_SAMPLE_FREQUENCY, &freq_id);
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last_freq_id = freq_id;
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}
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send_mpeg_lpcm_packet (data, len, id, timestamp, freq_id, write_dxr2);
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}
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// plays 'len' bytes of 'data'
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// it should round it down to outburst*n
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// return: number of bytes played
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static int play(void* data,int len,int flags){
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extern int write_dxr2(unsigned char *data, int len);
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// MPEG and AC3 don't work :-(
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if(ao_data.format==AF_FORMAT_MPEG2)
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send_mpeg_ps_packet (data, len, 0xC0, ao_data.pts, 2, write_dxr2);
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else if(ao_data.format==AF_FORMAT_AC3)
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send_mpeg_ps_packet (data, len, 0x80, ao_data.pts, 2, write_dxr2);
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else {
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int i;
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//unsigned short *s=data;
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uint16_t *s=data;
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#ifndef WORDS_BIGENDIAN
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for(i=0;i<len/2;i++) s[i] = bswap_16(s[i]);
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#endif
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dxr2_send_lpcm_packet(data,len,0xA0,ao_data.pts-10000,freq_id);
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}
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return len;
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}
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// return: delay in seconds between first and last sample in buffer
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static float get_delay(void){
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return 0.0;
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}
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