ao_sun.c

自己移植的linux下的流媒体播放器原代码,支持mms协议,支持ftp和http协议.

#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	AUDIO_SWFEATURE_MIXER	/* solaris8 or newer? */
# define HAVE_SYS_MIXER_H 1
#endif
#if	HAVE_SYS_MIXER_H
# include <sys/mixer.h>
#endif
#ifdef	__svr4__
#include <stropts.h>
#endif

#include "config.h"
#include "mixer.h"

#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
#include "help_mp.h"

static ao_info_t info = 
{
    "Sun audio output",
    "sun",
    "Juergen Keil",
    ""
};

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


static char *sun_mixer_device = NULL;
static char *audio_dev = NULL;
static int queued_bursts = 0;
static int queued_samples = 0;
static int bytes_per_sample = 0;
static int byte_per_sec = 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 af2sunfmt(int format)
{
    switch (format){
    case AF_FORMAT_MU_LAW:
	return AUDIO_ENCODING_ULAW;
    case AF_FORMAT_A_LAW:
	return AUDIO_ENCODING_ALAW;
    case AF_FORMAT_S16_NE:
	return AUDIO_ENCODING_LINEAR;
#ifdef	AUDIO_ENCODING_LINEAR8	// Missing on SunOS 5.5.1...
    case AF_FORMAT_U8:
	return AUDIO_ENCODING_LINEAR8;
#endif
    case AF_FORMAT_S8:
	return AUDIO_ENCODING_LINEAR;
#ifdef	AUDIO_ENCODING_DVI	// Missing on NetBSD...
    case AF_FORMAT_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.  44kbyte can be sent to all supported
			     * sun audio devices without blocking in the
			     * "write" below.
			     */
    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>0)
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
	goto error;
    }
    
    if (write(fd, silence, len) != len) {
	if (verbose>0)
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
	goto error;
    }

    if (ioctl(fd, AUDIO_GETINFO, &info)) {
	if (verbose>0)
	    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>0)
		perror("rtsc: GETINFO2 failed");
	    goto error;
	}
	if (info.play.samples < last_samplecnt) {
	    if (verbose>0)
		printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples);
	    goto error;
	}

	if ((increment = info.play.samples - last_samplecnt) > 0) {
	    if (verbose>0)
		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;
    }

    /*
     * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes
     * chunks (== 4096 samples) to the audio device.  If we see a minimum
     * sample counter increment from the soundcard driver of less than
     * 2000 samples,  we assume that the driver provides a useable realtime
     * sample counter in the AUDIO_INFO play.samples field.  Timing based
     * on sample counts should be much more accurate than counting whole 
     * 16kbyte chunks.
     */
    if (min_increment < 2000)
	rtsc_ok = RTSC_ENABLED;

    if (verbose>0)
	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;
}


// match the requested sample rate |sample_rate| against the
// sample rates supported by the audio device |dev|.  Return
// a supported sample rate,  if that sample rate is close to
// (< 1% difference) the requested rate; return 0 otherwise.

#define	MAX_RATE_ERR	1

static unsigned
find_close_samplerate_match(int dev, unsigned sample_rate)
{
#if	HAVE_SYS_MIXER_H
    am_sample_rates_t *sr;
    unsigned i, num, err, best_err, best_rate;

    for (num = 16; num < 1024; num *= 2) {
	sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num));
	if (!sr)
	    return 0;
	sr->type = AUDIO_PLAY;
	sr->flags = 0;
	sr->num_samp_rates = num;
	if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) {
	    free(sr);
	    return 0;
	}
	if (sr->num_samp_rates <= num)
	    break;
	free(sr);
    }

    if (sr->flags & MIXER_SR_LIMITS) {
	/*
	 * HW can playback any rate between 
	 * sr->samp_rates[0] .. sr->samp_rates[1]
	 */
	free(sr);
	return 0;
    } else {
	/* HW supports fixed sample rates only */

	best_err = 65535;
	best_rate = 0;

	for (i = 0; i < sr->num_samp_rates; i++) {
	    err = abs(sr->samp_rates[i] - sample_rate);
	    if (err == 0) {
		/*
		 * exact supported sample rate match, no need to
		 * retry something else
		 */
		best_rate = 0;
		break;
	    }
	    if (err < best_err) {
		best_err = err;
		best_rate = sr->samp_rates[i];
	    }
	}

	free(sr);

	if (best_rate > 0 && (100/MAX_RATE_ERR)*best_err < sample_rate) {
	    /* found a supported sample rate with <1% error? */
	    return best_rate;
	}
	return 0;
    }
#else	/* old audioio driver, cannot return list of supported rates */
    /* XXX: hardcoded sample rates */
    unsigned i, err;
    unsigned audiocs_rates[] = {
	5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050,
	27420, 32000, 33075, 37800, 44100, 48000, 0
    };

    for (i = 0; audiocs_rates[i]; i++) {
	err = abs(audiocs_rates[i] - sample_rate);
	if (err == 0) {
	    /* 
	     * exact supported sample rate match, no need to
	     * retry something elise
	     */
	    return 0;
	}
	if ((100/MAX_RATE_ERR)*err < audiocs_rates[i]) {
	    /* <1% error? */
	    return audiocs_rates[i];
	}
    }

    return 0;
#endif
}


// return the highest sample rate supported by audio device |dev|.
static unsigned
find_highest_samplerate(int dev)
{
#if	HAVE_SYS_MIXER_H
    am_sample_rates_t *sr;
    unsigned i, num, max_rate;

    for (num = 16; num < 1024; num *= 2) {
	sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num));
	if (!sr)
	    return 0;
	sr->type = AUDIO_PLAY;
	sr->flags = 0;
	sr->num_samp_rates = num;
	if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) {
	    free(sr);
	    return 0;
	}
	if (sr->num_samp_rates <= num)
	    break;
	free(sr);
    }

    if (sr->flags & MIXER_SR_LIMITS) {
	/*
	 * HW can playback any rate between 
	 * sr->samp_rates[0] .. sr->samp_rates[1]
	 */
	max_rate = sr->samp_rates[1];
    } else {
	/* HW supports fixed sample rates only */
	max_rate = 0;
	for (i = 0; i < sr->num_samp_rates; i++) {
	    if (sr->samp_rates[i] > max_rate)
		max_rate = sr->samp_rates[i];
	}
    }
    free(sr);
    return max_rate;

#else	/* old audioio driver, cannot return list of supported rates */
    return 44100;	/* should be supported even on old ISA SB cards */
#endif
}


static void setup_device_paths()
{
    if (audio_dev == NULL) {
	if ((audio_dev = getenv("AUDIODEV")) == NULL)
	    audio_dev = "/dev/audio";
    }

    if (sun_mixer_device == NULL) {