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mpv/libao2/ao_oss.c

378 lines
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C
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#include <stdio.h>
#include <stdlib.h>
#include <sys/ioctl.h>
#include <unistd.h>
#include <sys/time.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <errno.h>
#include <string.h>
//#include <sys/soundcard.h>
#include "config.h"
#include "mp_msg.h"
#include "mixer.h"
#include "help_mp.h"
#include "afmt.h"
#include "audio_out.h"
#include "audio_out_internal.h"
static ao_info_t info =
{
"OSS/ioctl audio output",
"oss",
"A'rpi",
""
};
/* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */
LIBAO_EXTERN(oss)
static char *dsp=PATH_DEV_DSP;
static audio_buf_info zz;
static int audio_fd=-1;
char *oss_mixer_device = PATH_DEV_MIXER;
int oss_mixer_channel = SOUND_MIXER_PCM;
// to set/get/query special features/parameters
static int control(int cmd,void *arg){
switch(cmd){
case AOCONTROL_SET_DEVICE:
dsp=(char*)arg;
return CONTROL_OK;
case AOCONTROL_GET_DEVICE:
*(char**)arg=dsp;
return CONTROL_OK;
case AOCONTROL_QUERY_FORMAT:
return CONTROL_TRUE;
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
{
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int fd, v, devs;
if(ao_data.format == AFMT_AC3)
return CONTROL_TRUE;
if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
{
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
if (devs & (1 << oss_mixer_channel))
{
if (cmd == AOCONTROL_GET_VOLUME)
{
ioctl(fd, MIXER_READ(oss_mixer_channel), &v);
vol->right = (v & 0xFF00) >> 8;
vol->left = v & 0x00FF;
}
else
{
v = ((int)vol->right << 8) | (int)vol->left;
ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v);
}
}
else
{
close(fd);
return CONTROL_ERROR;
}
close(fd);
return CONTROL_OK;
}
}
return CONTROL_ERROR;
}
return CONTROL_UNKNOWN;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
audio_out_format_name(format));
if (ao_subdevice)
dsp = ao_subdevice;
if(mixer_device)
oss_mixer_device=mixer_device;
if(mixer_channel){
int fd, devs, i;
if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer,
oss_mixer_device, strerror(errno));
}else{
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
close(fd);
for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
if(!strcasecmp(mixer_channels[i], mixer_channel)){
if(!(devs & (1 << i))){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,
mixer_channel);
i = SOUND_MIXER_NRDEVICES+1;
break;
}
oss_mixer_channel = i;
break;
}
}
if(i==SOUND_MIXER_NRDEVICES){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,
mixer_channel);
}
}
}
mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp);
mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device);
mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]);
#ifdef __linux__
audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
#else
audio_fd=open(dsp, O_WRONLY);
#endif
if(audio_fd<0){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenDev, dsp, strerror(errno));
return 0;
}
#ifdef __linux__
/* Remove the non-blocking flag */
if(fcntl(audio_fd, F_SETFL, 0) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantMakeFd, strerror(errno));
return 0;
}
#endif
#if defined(FD_CLOEXEC) && defined(F_SETFD)
fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
#endif
if(format == AFMT_AC3) {
ao_data.samplerate=rate;
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
}
ac3_retry:
ao_data.format=format;
if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format)<0 ||
ao_data.format != format) if(format == AFMT_AC3){
mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSetAC3, dsp);
#ifdef WORDS_BIGENDIAN
format=AFMT_S16_BE;
#else
format=AFMT_S16_LE;
#endif
goto ac3_retry;
}
mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
audio_out_format_name(ao_data.format), audio_out_format_name(format));
#if 0
if(ao_data.format!=format)
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));
#endif
ao_data.channels = channels;
if(format != AFMT_AC3) {
// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
if (ao_data.channels > 2) {
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
ao_data.channels != channels ) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, channels);
return 0;
}
}
else {
int c = ao_data.channels-1;
if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, ao_data.channels);
return 0;
}
ao_data.channels=c+1;
}
mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
// set rate
ao_data.samplerate=rate;
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
#if 0
if(ao_data.samplerate!=rate)
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);
#endif
}
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
int r=0;
mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_OSS_CantUseGetospace);
if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
} else {
ao_data.outburst=r;
mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
}
} else {
mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
ao_data.outburst=zz.fragsize;
}
if(ao_data.buffersize==-1){
// Measuring buffer size:
void* data;
ao_data.buffersize=0;
#ifdef HAVE_AUDIO_SELECT
data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
while(ao_data.buffersize<0x40000){
fd_set rfds;
struct timeval tv;
FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
tv.tv_sec=0; tv.tv_usec = 0;
if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
write(audio_fd,data,ao_data.outburst);
ao_data.buffersize+=ao_data.outburst;
}
free(data);
if(ao_data.buffersize==0){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantUseSelect);
return 0;
}
#endif
}
ao_data.bps=ao_data.channels;
if(ao_data.format != AFMT_U8 && ao_data.format != AFMT_S8)
ao_data.bps*=2;
ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
ao_data.bps*=ao_data.samplerate;
return 1;
}
// close audio device
static void uninit(int immed){
if(audio_fd == -1) return;
#ifdef SNDCTL_DSP_SYNC
// to get the buffer played
if (!immed)
ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL);
#endif
#ifdef SNDCTL_DSP_RESET
if (immed)
ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
#endif
close(audio_fd);
audio_fd = -1;
}
// stop playing and empty buffers (for seeking/pause)
static void reset(){
uninit(1);
audio_fd=open(dsp, O_WRONLY);
if(audio_fd < 0){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantReopen, strerror(errno));
return;
}
#if defined(FD_CLOEXEC) && defined(F_SETFD)
fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
#endif
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
if(ao_data.format != AFMT_AC3) {
if (ao_data.channels > 2)
ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
else {
int c = ao_data.channels-1;
ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
}
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
}
}
// stop playing, keep buffers (for pause)
static void audio_pause()
{
uninit(1);
}
// resume playing, after audio_pause()
static void audio_resume()
{
reset();
}
// return: how many bytes can be played without blocking
static int get_space(){
int playsize=ao_data.outburst;
#ifdef SNDCTL_DSP_GETOSPACE
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
// calculate exact buffer space:
playsize = zz.fragments*zz.fragsize;
if (playsize > MAX_OUTBURST)
playsize = (MAX_OUTBURST / zz.fragsize) * zz.fragsize;
return playsize;
}
#endif
// check buffer
#ifdef HAVE_AUDIO_SELECT
{ fd_set rfds;
struct timeval tv;
FD_ZERO(&rfds);
FD_SET(audio_fd, &rfds);
tv.tv_sec = 0;
tv.tv_usec = 0;
if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
}
#endif
return ao_data.outburst;
}
// 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){
len/=ao_data.outburst;
len=write(audio_fd,data,len*ao_data.outburst);
return len;
}
static int audio_delay_method=2;
// return: delay in seconds between first and last sample in buffer
static float get_delay(){
/* Calculate how many bytes/second is sent out */
if(audio_delay_method==2){
#ifdef SNDCTL_DSP_GETODELAY
int r=0;
if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
return ((float)r)/(float)ao_data.bps;
#endif
audio_delay_method=1; // fallback if not supported
}
if(audio_delay_method==1){
// SNDCTL_DSP_GETOSPACE
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
audio_delay_method=0; // fallback if not supported
}
return ((float)ao_data.buffersize)/(float)ao_data.bps;
}
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