mirror of https://github.com/mpv-player/mpv
339 lines
9.2 KiB
C
339 lines
9.2 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <sys/ioctl.h>
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#include <unistd.h>
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#include <sys/time.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <string.h>
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//#include <sys/soundcard.h>
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#include "../config.h"
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#include "../mp_msg.h"
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#include "../mixer.h"
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#include "afmt.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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static ao_info_t info =
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{
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"OSS/ioctl audio output",
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"oss",
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"A'rpi",
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""
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};
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/* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */
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LIBAO_EXTERN(oss)
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static char *dsp=PATH_DEV_DSP;
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static audio_buf_info zz;
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static int audio_fd=-1;
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char *oss_mixer_device = PATH_DEV_MIXER;
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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_SET_DEVICE:
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dsp=(char*)arg;
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return CONTROL_OK;
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case AOCONTROL_GET_DEVICE:
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(char*)arg=dsp;
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return CONTROL_OK;
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case AOCONTROL_QUERY_FORMAT:
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return CONTROL_TRUE;
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case AOCONTROL_GET_VOLUME:
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case AOCONTROL_SET_VOLUME:
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{
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ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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int fd, v, devs;
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if(ao_data.format == AFMT_AC3)
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return CONTROL_TRUE;
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if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
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{
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ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
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if (devs & SOUND_MASK_PCM)
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{
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if (cmd == AOCONTROL_GET_VOLUME)
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{
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ioctl(fd, SOUND_MIXER_READ_PCM, &v);
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vol->right = (v & 0xFF00) >> 8;
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vol->left = v & 0x00FF;
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}
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else
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{
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v = ((int)vol->right << 8) | (int)vol->left;
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ioctl(fd, SOUND_MIXER_WRITE_PCM, &v);
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}
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}
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else
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{
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close(fd);
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return CONTROL_ERROR;
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}
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close(fd);
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return CONTROL_OK;
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}
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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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// 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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mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
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audio_out_format_name(format));
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if (ao_subdevice)
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dsp = ao_subdevice;
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if(mixer_device)
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oss_mixer_device=mixer_device;
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp);
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#ifdef __linux__
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audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
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#else
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audio_fd=open(dsp, O_WRONLY);
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#endif
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if(audio_fd<0){
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mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't open audio device %s: %s\n", dsp, strerror(errno));
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return 0;
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}
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#ifdef __linux__
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/* Remove the non-blocking flag */
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if(fcntl(audio_fd, F_SETFL, 0) < 0) {
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mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't make filedescriptor blocking: %s\n", strerror(errno));
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return 0;
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}
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#endif
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#if defined(FD_CLOEXEC) && defined(F_SETFD)
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fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
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#endif
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if(format == AFMT_AC3) {
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ao_data.samplerate=rate;
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ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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}
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ac3_retry:
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ao_data.format=format;
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if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format)<0 ||
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ao_data.format != format) if(format == AFMT_AC3){
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mp_msg(MSGT_AO,MSGL_WARN,"Can't set audio device %s to AC3 output, trying S16...\n", dsp);
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#ifdef WORDS_BIGENDIAN
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format=AFMT_S16_BE;
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#else
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format=AFMT_S16_LE;
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#endif
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goto ac3_retry;
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}
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
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audio_out_format_name(ao_data.format), audio_out_format_name(format));
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#if 0
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if(ao_data.format!=format)
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mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-aop list=format'\n",audio_out_format_name(format));
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#endif
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ao_data.channels = channels;
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if(format != AFMT_AC3) {
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// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
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if (ao_data.channels > 2) {
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if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
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ao_data.channels != channels ) {
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mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Failed to set audio device to %d channels\n", channels);
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return 0;
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}
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}
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else {
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int c = ao_data.channels-1;
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if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
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mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
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return 0;
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}
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ao_data.channels=c+1;
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}
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
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// set rate
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ao_data.samplerate=rate;
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ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
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#if 0
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if(ao_data.samplerate!=rate)
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mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %d Hz samplerate! A-V sync problems or wrong speed are possible! Try with '-aop list=resample:fout=%d'\n",rate,ao_data.samplerate);
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#endif
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}
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if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
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int r=0;
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mp_msg(MSGT_AO,MSGL_WARN,"audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
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if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
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} else {
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ao_data.outburst=r;
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
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}
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} else {
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mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
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zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
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if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
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ao_data.outburst=zz.fragsize;
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}
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if(ao_data.buffersize==-1){
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// Measuring buffer size:
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void* data;
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ao_data.buffersize=0;
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#ifdef HAVE_AUDIO_SELECT
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data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
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while(ao_data.buffersize<0x40000){
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fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
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tv.tv_sec=0; tv.tv_usec = 0;
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if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
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write(audio_fd,data,ao_data.outburst);
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ao_data.buffersize+=ao_data.outburst;
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}
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free(data);
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if(ao_data.buffersize==0){
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mp_msg(MSGT_AO,MSGL_ERR,"\n *** Your audio driver DOES NOT support select() ***\n"
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"Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
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return 0;
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}
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#endif
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}
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ao_data.bps=ao_data.channels;
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if(ao_data.format != AFMT_U8 && ao_data.format != AFMT_S8)
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ao_data.bps*=2;
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ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
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ao_data.bps*=ao_data.samplerate;
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return 1;
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}
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// close audio device
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static void uninit(){
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if(audio_fd == -1) return;
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#ifdef SNDCTL_DSP_RESET
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ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
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#endif
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close(audio_fd);
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audio_fd = -1;
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset(){
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uninit();
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audio_fd=open(dsp, O_WRONLY);
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if(audio_fd < 0){
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mp_msg(MSGT_AO,MSGL_ERR,"\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno));
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return;
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}
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#if defined(FD_CLOEXEC) && defined(F_SETFD)
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fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
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#endif
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ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
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if(ao_data.format != AFMT_AC3) {
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if (ao_data.channels > 2)
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ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
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else {
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int c = ao_data.channels-1;
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ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
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}
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ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
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}
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause()
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{
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uninit();
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}
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// resume playing, after audio_pause()
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static void audio_resume()
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{
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reset();
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}
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// return: how many bytes can be played without blocking
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static int get_space(){
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int playsize=ao_data.outburst;
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#ifdef SNDCTL_DSP_GETOSPACE
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if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
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// calculate exact buffer space:
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playsize = zz.fragments*zz.fragsize;
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if (playsize > MAX_OUTBURST)
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playsize = (MAX_OUTBURST / zz.fragsize) * zz.fragsize;
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return playsize;
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}
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#endif
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// check buffer
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#ifdef HAVE_AUDIO_SELECT
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{ fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds);
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FD_SET(audio_fd, &rfds);
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tv.tv_sec = 0;
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tv.tv_usec = 0;
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if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
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}
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#endif
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return ao_data.outburst;
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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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len/=ao_data.outburst;
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len=write(audio_fd,data,len*ao_data.outburst);
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return len;
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}
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static int audio_delay_method=2;
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// return: delay in seconds between first and last sample in buffer
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static float get_delay(){
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/* Calculate how many bytes/second is sent out */
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if(audio_delay_method==2){
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#ifdef SNDCTL_DSP_GETODELAY
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int r=0;
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if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
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return ((float)r)/(float)ao_data.bps;
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#endif
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audio_delay_method=1; // fallback if not supported
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}
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if(audio_delay_method==1){
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// SNDCTL_DSP_GETOSPACE
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if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
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return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
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audio_delay_method=0; // fallback if not supported
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}
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return ((float)ao_data.buffersize)/(float)ao_data.bps;
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}
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