mirror of
https://github.com/mpv-player/mpv
synced 2024-12-15 11:25:10 +00:00
7be3e8694b
send 0-samples according to the amount of data lost during pause. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@23829 b3059339-0415-0410-9bf9-f77b7e298cf2
534 lines
14 KiB
C
534 lines
14 KiB
C
#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 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_U24_LE
|
|
case AF_FORMAT_U24_LE: return AFMT_U24_LE;
|
|
#endif
|
|
#ifdef AFMT_U24_BE
|
|
case AF_FORMAT_U24_BE: return AFMT_U24_BE;
|
|
#endif
|
|
#ifdef AFMT_S24_LE
|
|
case AF_FORMAT_S24_LE: return AFMT_S24_LE;
|
|
#endif
|
|
#ifdef AFMT_S24_BE
|
|
case AF_FORMAT_S24_BE: return AFMT_S24_BE;
|
|
#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_U24_LE
|
|
case AFMT_U24_LE: return AF_FORMAT_U24_LE;
|
|
#endif
|
|
#ifdef AFMT_U24_BE
|
|
case AFMT_U24_BE: return AF_FORMAT_U24_BE;
|
|
#endif
|
|
#ifdef AFMT_S24_LE
|
|
case AFMT_S24_LE: return AF_FORMAT_S24_LE;
|
|
#endif
|
|
#ifdef AFMT_S24_BE
|
|
case AFMT_S24_BE: return AF_FORMAT_S24_BE;
|
|
#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 (format & (int)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) {
|
|
#ifdef WORDS_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);
|
|
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) {
|
|
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;
|
|
}
|