view libao2/ao_sun.c @ 1285:202d9e2dc202

-vcodec option (maybe some other name would be better though) to select between driver types without editing codecs.conf. mplayer will default to normal codec search loop if it does not find codec for the specified driver type. config range checking for the parameter (an integer) should be cleaned, IMHO

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/audioio.h>
#ifdef	__svr4__
#include <stropts.h>
#endif

#include "../config.h"

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

static ao_info_t info = 
{
    "Sun audio output",
    "sun",
    "[e-mail]",
    ""
};

LIBAO_EXTERN(sun)


/* These defines are missing on NetBSD */
#ifndef	AUDIO_PRECISION_8
#define AUDIO_PRECISION_8	8
#define AUDIO_PRECISION_16	16
#endif
#ifndef	AUDIO_CHANNELS_MONO
#define	AUDIO_CHANNELS_MONO	1
#define	AUDIO_CHANNELS_STEREO	2
#endif


// there are some globals:
// ao_samplerate
// ao_channels
// ao_format
// ao_bps
// ao_outburst
// ao_buffersize

static char *audio_dev = "/dev/audio";
static int queued_bursts = 0;
static int queued_samples = 0;
static int bytes_per_sample = 0;
static int audio_fd = -1;
static enum {
    RTSC_UNKNOWN = 0,
    RTSC_ENABLED,
    RTSC_DISABLED
} enable_sample_timing;

extern int verbose;


// convert an OSS audio format specification into a sun audio encoding
static int oss2sunfmt(int oss_format)
{
  switch (oss_format){
  case AFMT_MU_LAW:
    return AUDIO_ENCODING_ULAW;
  case AFMT_A_LAW:
    return AUDIO_ENCODING_ALAW;
  case AFMT_S16_LE:
    return AUDIO_ENCODING_LINEAR;
  case AFMT_U8:
    return AUDIO_ENCODING_LINEAR8;
#ifdef	AUDIO_ENCODING_DVI	// Missing on NetBSD...
  case AFMT_IMA_ADPCM:
    return AUDIO_ENCODING_DVI;
#endif
  default:
    return AUDIO_ENCODING_NONE;
  }
}

// try to figure out, if the soundcard driver provides usable (precise)
// sample counter information
static int realtime_samplecounter_available(char *dev)
{
    int fd = -1;
    audio_info_t info;
    int rtsc_ok = RTSC_DISABLED;
    int len;
    void *silence = NULL;
    struct timeval start, end;
    struct timespec delay;
    int usec_delay;
    unsigned last_samplecnt;
    unsigned increment;
    unsigned min_increment;

    len = 44100 * 4 / 4;    // amount of data for 0.25sec of 44.1khz, stereo, 16bit
    silence = calloc(1, len);
    if (silence == NULL)
	goto error;
    
    if ((fd = open(dev, O_WRONLY)) < 0)
	goto error;

    AUDIO_INITINFO(&info);
    info.play.sample_rate = 44100;
    info.play.channels = AUDIO_CHANNELS_STEREO;
    info.play.precision = AUDIO_PRECISION_16;
    info.play.encoding = AUDIO_ENCODING_LINEAR;
    info.play.samples = 0;
    if (ioctl(fd, AUDIO_SETINFO, &info)) {
	if (verbose)
	    printf("rtsc: SETINFO failed\n");
	goto error;
    }
    
    if (write(fd, silence, len) != len) {
	if (verbose)
	    printf("rtsc: write failed");
	goto error;
    }

    if (ioctl(fd, AUDIO_GETINFO, &info)) {
	if (verbose)
	    perror("rtsc: GETINFO1");
	goto error;
    }

    last_samplecnt = info.play.samples;
    min_increment = ~0;

    gettimeofday(&start, NULL);
    for (;;) {
	delay.tv_sec = 0;
	delay.tv_nsec = 10000000;
	nanosleep(&delay, NULL);
	gettimeofday(&end, NULL);
	usec_delay = (end.tv_sec - start.tv_sec) * 1000000
	    + end.tv_usec - start.tv_usec;

	// stop monitoring sample counter after 0.2 seconds
	if (usec_delay > 200000)
	    break;

	if (ioctl(fd, AUDIO_GETINFO, &info)) {
	    if (verbose)
		perror("rtsc: GETINFO2 failed");
	    goto error;
	}
	if (info.play.samples < last_samplecnt) {
	    if (verbose)
		printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples);
	    goto error;
	}

	if ((increment = info.play.samples - last_samplecnt) > 0) {
	    if (verbose)
		printf("ao_sun: sample counter increment: %d\n", increment);
	    if (increment < min_increment) {
		min_increment = increment;
		if (min_increment < 2000)
		    break;	// looks good
	    }
	}
	last_samplecnt = info.play.samples;
    }

    if (min_increment < 2000)
	rtsc_ok = RTSC_ENABLED;

    if (verbose)
	printf("ao_sun: minimum sample counter increment per 10msec interval: %d\n"
	       "\t%susing sample counter based timing code\n",
	       min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not ");
    

error:
    if (silence != NULL) free(silence);
    if (fd >= 0) {
#ifdef	__svr4__
	// remove the 0 bytes from the above measurement from the
	// audio driver's STREAMS queue
	ioctl(fd, I_FLUSH, FLUSHW);
#endif
	//ioctl(fd, AUDIO_DRAIN, 0);
	close(fd);
    }

    return rtsc_ok;
}

// to set/get/query special features/parameters
static int control(int cmd,int arg){
  switch(cmd){
    case AOCONTROL_SET_DEVICE:
      audio_dev=(char*)arg;
      return CONTROL_OK;
    case AOCONTROL_QUERY_FORMAT:
      return CONTROL_TRUE;
  }
  return CONTROL_UNKNOWN;
}

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

  audio_info_t info;
  int byte_per_sec;

  if (ao_subdevice) audio_dev = ao_subdevice;

  if (enable_sample_timing == RTSC_UNKNOWN
      && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
      enable_sample_timing = realtime_samplecounter_available(audio_dev);
  }

  printf("ao2: %d Hz  %d chans  %s [0x%X]\n",
	 rate,channels,audio_out_format_name(format),format);

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

  ioctl(audio_fd, AUDIO_DRAIN, 0);

  AUDIO_INITINFO(&info);
  info.play.encoding = oss2sunfmt(ao_format = format);
  info.play.precision = (format==AFMT_S16_LE? AUDIO_PRECISION_16:AUDIO_PRECISION_8);
  info.play.channels = ao_channels = channels;
  info.play.sample_rate = ao_samplerate = rate;
  if(ioctl (audio_fd, AUDIO_SETINFO, &info)<0)
    printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",channels,audio_out_format_name(format),rate);
  bytes_per_sample = channels * info.play.precision / 8;
  byte_per_sec = bytes_per_sample * rate;
  ao_outburst = byte_per_sec > 100000 ? 16384 : 8192;

  if(ao_buffersize==-1){
    // Measuring buffer size:
    void* data;
    ao_buffersize=0;
#ifdef HAVE_AUDIO_SELECT
    data=malloc(ao_outburst); memset(data,0,ao_outburst);
    while(ao_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_outburst);
      ao_buffersize+=ao_outburst;
    }
    free(data);
    if(ao_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;
    }
#ifdef	__svr4__
    // remove the 0 bytes from the above ao_buffersize measurement from the
    // audio driver's STREAMS queue
    ioctl(audio_fd, I_FLUSH, FLUSHW);
#endif
    ioctl(audio_fd, AUDIO_DRAIN, 0);
#endif
  }

  AUDIO_INITINFO(&info);
  info.play.samples = 0;
  info.play.eof = 0;
  info.play.error = 0;
  ioctl (audio_fd, AUDIO_SETINFO, &info);

  queued_bursts = 0;
  queued_samples = 0;

  return 1;
}

// close audio device
static void uninit(){
#ifdef	__svr4__
    // throw away buffered data in the audio driver's STREAMS queue
    ioctl(audio_fd, I_FLUSH, FLUSHW);
#endif
    close(audio_fd);
}

// stop playing and empty buffers (for seeking/pause)
static void reset(){
    audio_info_t info;

    uninit();
    audio_fd=open(audio_dev, O_WRONLY);
    if(audio_fd<0){
	printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno));
	return;
    }

    ioctl(audio_fd, AUDIO_DRAIN, 0);

    AUDIO_INITINFO(&info);
    info.play.encoding = oss2sunfmt(ao_format);
    info.play.precision = (ao_format==AFMT_S16_LE? AUDIO_PRECISION_16:AUDIO_PRECISION_8);
    info.play.channels = ao_channels;
    info.play.sample_rate = ao_samplerate;
    info.play.samples = 0;
    info.play.eof = 0;
    info.play.error = 0;
    ioctl (audio_fd, AUDIO_SETINFO, &info);
    queued_bursts = 0;
    queued_samples = 0;
}

// stop playing, keep buffers (for pause)
static void audio_pause()
{
    struct audio_info info;
    AUDIO_INITINFO(&info);
    info.play.pause = 1;
    ioctl(audio_fd, AUDIO_SETINFO, &info);
}

// resume playing, after audio_pause()
static void audio_resume()
{
    struct audio_info info;
    AUDIO_INITINFO(&info);
    info.play.pause = 0;
    ioctl(audio_fd, AUDIO_SETINFO, &info);
}


// return: how many bytes can be played without blocking
static int get_space(){
    int playsize = ao_outburst;
    audio_info_t info;

    // 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

    ioctl(audio_fd, AUDIO_GETINFO, &info);
    if (queued_bursts - info.play.eof > 2)
	return 0;

    return ao_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 < ao_outburst) return 0;
    len /= ao_outburst;
    len = write(audio_fd, data, len*ao_outburst);
    if(len > 0) {
      queued_samples += len / bytes_per_sample;
      if (write(audio_fd,data,0) < 0)
	  perror("ao_sun: send EOF audio record");
      else
	  queued_bursts ++;
    }
    return len;
}


// return: how many unplayed bytes are in the buffer
static int get_delay(){
    audio_info_t info;
    ioctl(audio_fd, AUDIO_GETINFO, &info);
    if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
	return (queued_samples - info.play.samples) * bytes_per_sample;
    else
	return (queued_bursts - info.play.eof) * ao_outburst;
}