annotate libao2/ao_oss.c @ 13249:a6642a4330fa

ensure that avi files have a valid header as soon as possible. without this, the header says 0x0 video size, which works with mplayer when the video size is stored in the codec data, but it does NOT work with other players or with codecs that don't store size (e.g. snow). actually i don't like having seeks in the muxer module, but i don't know any other way to implement this fix without major changes to mencoder. if you have a better fix, please reverse this and commit yours.

#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 "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 <[e-mail]> */

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,"audio_setup: Can't open mixer device %s: %s\n",
        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,"audio_setup: Audio card mixer does not have channel '%s' using default\n",
              mixer_channel);
            i = SOUND_MIXER_NRDEVICES+1;
            break;
          }
          oss_mixer_channel = i;
          break;
        }
      }
      if(i==SOUND_MIXER_NRDEVICES){
        mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't find mixer channel '%s' using default\n",
          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,"audio_setup: Can't open audio device %s: %s\n", 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,"audio_setup: Can't make filedescriptor blocking: %s\n", 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,"Can't set audio device %s to AC3 output, trying S16...\n", 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,"audio_setup: Failed to set audio device to %d channels\n", 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,"audio_setup: Failed to set audio device to %d channels\n", 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,"audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
      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,"\n   ***  Your audio driver DOES NOT support select()  ***\n"
          "Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
        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,"\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", 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;
}