view libao2/ao_oss.c @ 4559:5dc383bb1c82

added mga_top_reserved module parameter to skip a configurable amount of space at the top of video memory. this is needed to prevent corruption of the kernel's console font when using the "fastfont" option with matroxfb.
author rfelker
date Thu, 07 Feb 2002 02:07:29 +0000
parents f648f699eda6
children d678ce495a75
line wrap: on
line source

#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 <sys/soundcard.h>

#include "../config.h"

#include "afmt.h"

#include "audio_out.h"
#include "audio_out_internal.h"

extern int verbose;

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 char *dsp="/dev/dsp";
static audio_buf_info zz;
static int audio_fd=-1;

char *oss_mixer_device = "/dev/mixer";
int oss_mixer_usemaster = 0;

// to set/get/query special features/parameters
static int control(int cmd,int arg){
    switch(cmd){
	case AOCONTROL_SET_DEVICE:
	    dsp=(char*)arg;
	    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, mcmd, devs;

	    if(ao_data.format == AFMT_AC3)
		return CONTROL_TRUE;
    
	    if ((fd = open("/dev/mixer", O_RDONLY)) > 0)
	    {
		ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
		if ((devs & SOUND_MASK_PCM) && (oss_mixer_usemaster == 0))
		    if (cmd == AOCONTROL_GET_VOLUME)
			mcmd = SOUND_MIXER_READ_PCM;
		    else
			mcmd = SOUND_MIXER_WRITE_PCM;
		else if ((devs & SOUND_MASK_VOLUME) && (oss_mixer_usemaster == 1))
		    if (cmd == AOCONTROL_GET_VOLUME)
			mcmd = SOUND_MIXER_READ_VOLUME;
		    else
			mcmd = SOUND_MIXER_WRITE_VOLUME;
		else
		{
		    close(fd);
		    return CONTROL_ERROR;
		}

		if (cmd == AOCONTROL_GET_VOLUME)
		{
		    ioctl(fd, cmd, &v);
		    vol->right = (v & 0xFF00) >> 8;
		    vol->left = v & 0x00FF;
		}
		else
		{
		    v = ((int)vol->right << 8) | (int)vol->left;
		    ioctl(fd, cmd, &v);
		}
		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){

  printf("ao2: %d Hz  %d chans  %s\n",rate,channels,
    audio_out_format_name(format));

  if (ao_subdevice)
    dsp = ao_subdevice;

  if (verbose)
    printf("audio_setup: using '%s' dsp device\n", dsp);

  audio_fd=open(dsp, O_WRONLY);
  if(audio_fd<0){
    printf("Can't open audio device %s  -> nosound\n",dsp);
    return 0;
  }

  ao_data.bps=channels*rate;
  if(format != AFMT_U8 && format != AFMT_S8)
    ao_data.bps*=2;

  if(format == AFMT_AC3) {
    ao_data.samplerate=rate;
    ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
  }
  
  ao_data.format=format;
  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
  if(format == AFMT_AC3 && ao_data.format != AFMT_AC3) {
      printf("Can't set audio device %s to AC3 output\n", dsp);
      return 0;
  }
  printf("audio_setup: sample format: %s (requested: %s)\n",
    audio_out_format_name(ao_data.format), audio_out_format_name(format));
  
  if(format != AFMT_AC3) {
    // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
    ao_data.channels = channels;
    if (ao_data.channels > 2) {
      if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
	   ao_data.channels != channels ) {
	printf("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) {
	printf("audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
	return 0;
      }
    }
    printf("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);
    printf("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;
      printf("audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
      if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
          printf("audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
      } else {
          ao_data.outburst=r;
          printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
      }
  } else {
      printf("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){
        printf("\n   ***  Your audio driver DOES NOT support select()  ***\n");
          printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
        return 0;
    }
#endif
  }

    return 1;
}

// close audio device
static void uninit(){
#ifdef SNDCTL_DSP_RESET
    ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
#endif
    close(audio_fd);
}

// stop playing and empty buffers (for seeking/pause)
static void reset(){
    uninit();
    audio_fd=open(dsp, O_WRONLY);
    if(audio_fd<0){
	printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE ***\n");
	return;
    }

  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()
{
    // for now, just call reset();
    reset();
}

// resume playing, after audio_pause()
static void audio_resume()
{
}


// 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){
      int r=0;
      if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
         return ((float)r)/(float)ao_data.bps;
      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;
}