view libao2/ao_oss.c @ 27518:e54c9b7eb0d8

Revert bad changes to SSA/ASS subtitle packet format The following commits are reverted partially or completely: "a valid ASS line contains 9 ',' before actual text" "demux_mkv: output correctly formated ASS packets" "libass: add a new ass_process_data() to process demuxed subtitle packets" These commits converted the internal representation of SSA/ASS subtitle packets from the format used by Matroska to a custom format where each packet has contents exactly matching one line in complete SSA script files. AFAIK no files natively use such a format for muxed subtitles. The stated reason for this change was to use a format that could in principle be muxed into a maximal number of containers. SSA subtitles do not have an implicit duration so both start time and duration or end time need to be specified explicitly; the new format moved timing information inside the codec packet data so it could be muxed without modification into containers that can represent only start time at the container level. However such a change is wrong from the viewpoint of program architecture. Timing information belongs to the demuxer level, but these commits moved not only the duration but also the authoritative value of the start time to inside the codec data. Additionally the new format lost the value of the Matroska ReadOrder field which is used by MPlayer. This commit changes the internal packet format back to that used by Matroska and makes the internal Matroska demuxer output that format again. Libavformat still outputs the "new" format; it could be converted back to the Matroska format in demux_lavf.c, but I'm not adding that code at least yet. The current lavf code has similar problems as the reverted code in MPlayer, and it also currently fails to provide any way to access the value of the ReadOrder field. I hope that the lavf side will be improved; if it isn't conversion can be added later. For now I'll make MPlayer default to the internal Matroska demuxer instead of the lavf one in a separate commit.

#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 <[e-mail]> */

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 ((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) {
#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;
}