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mpv/libao2/ao_oss.c

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/*
* OSS 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 <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 "config.h"
#include "mp_msg.h"
#include "mixer.h"
#include "help_mp.h"
#ifdef HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#else
#ifdef HAVE_SOUNDCARD_H
#include <soundcard.h>
#endif
#endif
#include "libaf/af_format.h"
#include "audio_out.h"
#include "audio_out_internal.h"
static const 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 int format2oss(int format)
{
switch(format)
{
case AF_FORMAT_U8: return AFMT_U8;
case AF_FORMAT_S8: return AFMT_S8;
case AF_FORMAT_U16_LE: return AFMT_U16_LE;
case AF_FORMAT_U16_BE: return AFMT_U16_BE;
case AF_FORMAT_S16_LE: return AFMT_S16_LE;
case AF_FORMAT_S16_BE: return AFMT_S16_BE;
#ifdef AFMT_S24_PACKED
case AF_FORMAT_S24_LE: return AFMT_S24_PACKED;
#endif
#ifdef AFMT_U32_LE
case AF_FORMAT_U32_LE: return AFMT_U32_LE;
#endif
#ifdef AFMT_U32_BE
case AF_FORMAT_U32_BE: return AFMT_U32_BE;
#endif
#ifdef AFMT_S32_LE
case AF_FORMAT_S32_LE: return AFMT_S32_LE;
#endif
#ifdef AFMT_S32_BE
case AF_FORMAT_S32_BE: return AFMT_S32_BE;
#endif
#ifdef AFMT_FLOAT
case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
#endif
// SPECIALS
case AF_FORMAT_MU_LAW: return AFMT_MU_LAW;
case AF_FORMAT_A_LAW: return AFMT_A_LAW;
case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
#ifdef AFMT_MPEG
case AF_FORMAT_MPEG2: return AFMT_MPEG;
#endif
#ifdef AFMT_AC3
case AF_FORMAT_AC3: return AFMT_AC3;
#endif
}
mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
return -1;
}
static int oss2format(int format)
{
switch(format)
{
case AFMT_U8: return AF_FORMAT_U8;
case AFMT_S8: return AF_FORMAT_S8;
case AFMT_U16_LE: return AF_FORMAT_U16_LE;
case AFMT_U16_BE: return AF_FORMAT_U16_BE;
case AFMT_S16_LE: return AF_FORMAT_S16_LE;
case AFMT_S16_BE: return AF_FORMAT_S16_BE;
#ifdef AFMT_S24_PACKED
case AFMT_S24_PACKED: return AF_FORMAT_S24_LE;
#endif
#ifdef AFMT_U32_LE
case AFMT_U32_LE: return AF_FORMAT_U32_LE;
#endif
#ifdef AFMT_U32_BE
case AFMT_U32_BE: return AF_FORMAT_U32_BE;
#endif
#ifdef AFMT_S32_LE
case AFMT_S32_LE: return AF_FORMAT_S32_LE;
#endif
#ifdef AFMT_S32_BE
case AFMT_S32_BE: return AF_FORMAT_S32_BE;
#endif
#ifdef AFMT_FLOAT
case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
#endif
// SPECIALS
case AFMT_MU_LAW: return AF_FORMAT_MU_LAW;
case AFMT_A_LAW: return AF_FORMAT_A_LAW;
case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM;
#ifdef AFMT_MPEG
case AFMT_MPEG: return AF_FORMAT_MPEG2;
#endif
#ifdef AFMT_AC3
case AFMT_AC3: return AF_FORMAT_AC3;
#endif
}
mp_msg(MSGT_GLOBAL,MSGL_ERR,MSGTR_AO_OSS_UnknownUnsupportedFormat, format);
return -1;
}
static char *dsp=PATH_DEV_DSP;
static audio_buf_info zz;
static int audio_fd=-1;
static int prepause_space;
static const char *oss_mixer_device = PATH_DEV_MIXER;
static 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;
#ifdef SNDCTL_DSP_GETFMTS
case AOCONTROL_QUERY_FORMAT:
{
int format;
if (!ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format))
if ((unsigned int)format & (unsigned long)arg)
return CONTROL_TRUE;
return CONTROL_FALSE;
}
#endif
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 == AF_FORMAT_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;
int oss_format;
char *mdev = mixer_device, *mchan = mixer_channel;
mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
af_fmt2str_short(format));
if (ao_subdevice) {
char *m,*c;
m = strchr(ao_subdevice,':');
if(m) {
c = strchr(m+1,':');
if(c) {
mchan = c+1;
c[0] = '\0';
}
mdev = m+1;
m[0] = '\0';
}
dsp = ao_subdevice;
}
if(mdev)
oss_mixer_device=mdev;
else
oss_mixer_device=PATH_DEV_MIXER;
if(mchan){
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], mchan)){
if(!(devs & (1 << i))){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,mchan);
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,mchan);
}
}
} else
oss_mixer_channel = SOUND_MIXER_PCM;
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 == AF_FORMAT_AC3) {
ao_data.samplerate=rate;
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
}
ac3_retry:
ao_data.format=format;
oss_format=format2oss(format);
if (oss_format == -1) {
#if HAVE_BIGENDIAN
oss_format=AFMT_S16_BE;
#else
oss_format=AFMT_S16_LE;
#endif
format=AF_FORMAT_S16_NE;
}
if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
oss_format != format2oss(format)) {
mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSet, dsp,
af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) );
format=AF_FORMAT_S16_NE;
goto ac3_retry;
}
#if 0
if(oss_format!=format2oss(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 '-af format'\n",audio_out_format_name(format));
#endif
ao_data.format = oss2format(oss_format);
if (ao_data.format == -1) return 0;
mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
ao_data.channels = channels;
if(format != AF_FORMAT_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(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 != AF_FORMAT_U8 && ao_data.format != AF_FORMAT_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(void){
int oss_format;
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
oss_format = format2oss(ao_data.format);
if(ao_data.format == AF_FORMAT_AC3)
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
if(ao_data.format != AF_FORMAT_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(void)
{
prepause_space = get_space();
uninit(1);
}
// resume playing, after audio_pause()
static void audio_resume(void)
{
int fillcnt;
reset();
fillcnt = get_space() - prepause_space;
if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) {
void *silence = calloc(fillcnt, 1);
play(silence, fillcnt, 0);
free(silence);
}
}
// return: how many bytes can be played without blocking
static int get_space(void){
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){
if(len==0)
return len;
if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
len/=ao_data.outburst;
len*=ao_data.outburst;
}
len=write(audio_fd,data,len);
return len;
}
static int audio_delay_method=2;
// return: delay in seconds between first and last sample in buffer
static float get_delay(void){
/* 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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