view libao2/ao_sgi.c @ 16429:84174804804b

Updates to NUT spec: 1. remove average_bitrate 2. add other_stream_header, for subtitles and metadata 3. add max_pts to index 4. index_ptr - a 64 bit integer to say the total length of all index packets 5. specify how to write "multiple" indexes 6. change forward_ptr behavior, starts right after forward_ptr, ends after checksum 7. remove stream_id <-> stream_class limitation. 8. time_base_nom must also be non zero. 9. rename time_base_nom and time_base_denom, now timebase means the length of a tick, not amounts of ticks 10. remove (old?) sample_rate_mul stuff. 11. specify what exactly the checksum covers. 12. specify that stream classes which have multiple streams must have an info packet.. (in new Semantic requirements section) 13. Rename 'timestamp' to pts. 14. Change date of draft... 15. Add myself to authors...

/*
  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"
#include "mp_msg.h"
#include "help_mp.h"

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

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){
  
  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_INFO);
  
  return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {

  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(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) {
      mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice);
      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) {
      mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror()));
    }
    
    if (x[0].sizeOut < 0) {
      mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate);
    }

    if (alGetParams(rv,x, 1)<0) {
      mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror()));
    }
    
    if (frate != alFixedToDouble(x[0].value.ll)) {
      mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, 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) {
    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, 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) {
    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
    return 0;
  }
  
  ao_port = alOpenPort("mplayer", "w", ao_config);
  
  if (!ao_port) {
    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, 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! */

  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit);

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

// stop playing and empty buffers (for seeking/pause)
static void reset() {
  
  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Reset);
  
}

// stop playing, keep buffers (for pause)
static void audio_pause() {
    
  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_PauseInfo);
    
}

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

  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_ResumeInfo);

}

// 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);
}