annotate libao2/ao_sun.c @ 1196:fc51929ec8ea

Applied patch by J¸«ärgen Keil ([e-mail]), improves smoothness of video playback.

#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 (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;
}