view libao2/ao_oss.c @ 34234:4ec96d5d2e4c

build: drop releaseclean target The target is supposed to remove files that are created during the XML build process without removing the generated documentation. Unfortunately, it does not work as expected and is not worth the extra complication.

/*
 * 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 <[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_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_NE: 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_NE;
#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(AF_FORMAT_IS_AC3(ao_data.format))
		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(AF_FORMAT_IS_AC3(format)) {
    ao_data.samplerate=rate;
    ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
  }

ac3_retry:
  if (AF_FORMAT_IS_AC3(format))
    format = AF_FORMAT_AC3_NE;
  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(!AF_FORMAT_IS_AC3(format)) {
    // 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;
  switch (ao_data.format & AF_FORMAT_BITS_MASK) {
  case AF_FORMAT_8BIT:
    break;
  case AF_FORMAT_16BIT:
    ao_data.bps*=2;
    break;
  case AF_FORMAT_24BIT:
    ao_data.bps*=3;
    break;
  case AF_FORMAT_32BIT:
    ao_data.bps*=4;
    break;
  }

  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(AF_FORMAT_IS_AC3(ao_data.format))
    ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
  if(!AF_FORMAT_IS_AC3(ao_data.format)) {
    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;
}