view libao2/ao_sgi.c @ 12334:eeddcef4fb08

"hard" frame duplication for mencoder. this finally makes it possible to generate valid mpeg output from avi's that have duplicate frames in them, or when using inverse telecine filters. to use it, put the "harddup" filter at the end of your filter chain.

/*
  ao_sgi - sgi/irix output plugin for MPlayer

  22oct2001 [e-mail]
  
*/

#include <stdio.h>
#include <stdlib.h>
#include <dmedia/audio.h>

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

static ao_info_t info = 
{
	"sgi audio output",
	"sgi",
	"kopflos",
	""
};

LIBAO_EXTERN(sgi)


static ALconfig	ao_config;
static ALport	ao_port;
static int sample_rate;
static int queue_size;

// to set/get/query special features/parameters
static int control(int cmd, void *arg){
  
  printf("ao_sgi, control\n");
  
  return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {
  
  printf("ao_sgi, init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
  
  { /* from /usr/share/src/dmedia/audio/setrate.c */
  
    int fd;
    int rv;
    double frate;
    ALpv x[2];

    rv = alGetResourceByName(AL_SYSTEM, "out.analog", AL_DEVICE_TYPE);
    if (!rv) {
      printf("ao_sgi, play: invalid device\n");
      return 0;
    }
    
    frate = rate;
   
    x[0].param = AL_RATE;
    x[0].value.ll = alDoubleToFixed(rate);
    x[1].param = AL_MASTER_CLOCK;
    x[1].value.i = AL_CRYSTAL_MCLK_TYPE;

    if (alSetParams(rv,x, 2)<0) {
      printf("ao_sgi, init: setparams failed: %s\n", alGetErrorString(oserror()));
      printf("ao_sgi, init: could not set desired samplerate\n"); 
    }
    
    if (x[0].sizeOut < 0) {
      printf("ao_sgi, init: AL_RATE was not accepted on the given resource\n");
    }

    if (alGetParams(rv,x, 1)<0) {
      printf("ao_sgi, init: getparams failed: %s\n", alGetErrorString(oserror()));
    }
    
    if (frate != alFixedToDouble(x[0].value.ll)) {
      printf("ao_sgi, init: samplerate is now %lf (desired rate is %lf)\n", alFixedToDouble(x[0].value.ll), frate);
    } 
    sample_rate = (int)frate;
  }
  
  ao_data.buffersize=131072;
  ao_data.outburst = ao_data.buffersize/16;
  ao_data.channels = channels;
  
  ao_config = alNewConfig();
  
  if (!ao_config) {
    printf("ao_sgi, init: %s\n", alGetErrorString(oserror()));
    return 0;
  }
  
  if(channels == 2) alSetChannels(ao_config, AL_STEREO);
  else alSetChannels(ao_config, AL_MONO);
  
  alSetWidth(ao_config, AL_SAMPLE_16);
  alSetSampFmt(ao_config, AL_SAMPFMT_TWOSCOMP);
  alSetQueueSize(ao_config, 48000);
  
  if (alSetDevice(ao_config, AL_DEFAULT_OUTPUT) < 0) {
    printf("ao_sgi, init: %s\n", alGetErrorString(oserror()));
    return 0;
  }
  
  ao_port = alOpenPort("mplayer", "w", ao_config);
  
  if (!ao_port) {
    printf("ao_sgi, init: Unable to open audio channel: %s\n", alGetErrorString(oserror()));
    return 0;
  }
  
  // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
  queue_size = alGetQueueSize(ao_config);
  return 1;  

}

// close audio device
static void uninit(int immed) {

  /* TODO: samplerate should be set back to the value before mplayer was started! */

  printf("ao_sgi, uninit: ...\n");

  if (ao_port) {
    while(alGetFilled(ao_port) > 0) sginap(1);  
    alClosePort(ao_port);
    alFreeConfig(ao_config);
  }
	
}

// stop playing and empty buffers (for seeking/pause)
static void reset() {
  
  printf("ao_sgi, reset: ...\n");
  
}

// stop playing, keep buffers (for pause)
static void audio_pause() {
    
  printf("ao_sgi, audio_pause: ...\n");
    
}

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

  printf("ao_sgi, audio_resume: ...\n");

}

// return: how many bytes can be played without blocking
static int get_space() {
  
  // printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_outburst);
  // printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
  
  return alGetFillable(ao_port)*(2*ao_data.channels);
    
}


// 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) {
    
  // printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config);
  // printf("channels %d\n", ao_channels);

  alWriteFrames(ao_port, data, len/(2*ao_data.channels));
  
  return len;
  
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(){
  
  // printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);
  
  //return 0;
  return  (float)queue_size/((float)sample_rate);
}