annotate libao2/ao_sun.c @ 24787:02535b3216c5

Avoid text deformation and subtitles moving outside the screen in pan-and-scan mode. For this, crop amounts are passed from vo_gl as negative margins sizes. They are used to calculate aspect ratio. They are ignored when calculating subtitle positions, so subtitles will stay on screen most of the time. Based on a patch by Jindrich Makovicka [makovick gmail com].

#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;


static void flush_audio(int fd) {
#ifdef AUDIO_FLUSH
  ioctl(fd, AUDIO_FLUSH, 0);
#elif defined(__svr4__)
  ioctl(fd, I_FLUSH, FLUSHW);
#endif
}

// 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 ( mp_msg_test(MSGT_AO,MSGL_V) )
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
	goto error;
    }
    
    if (write(fd, silence, len) != len) {
	if ( mp_msg_test(MSGT_AO,MSGL_V) )
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
	goto error;
    }

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

	if ((increment = info.play.samples - last_samplecnt) > 0) {
	    if ( mp_msg_test(MSGT_AO,MSGL_V) )
	        mp_msg(MSGT_AO,MSGL_V,"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 ( mp_msg_test(MSGT_AO,MSGL_V) )
	mp_msg(MSGT_AO,MSGL_V,"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) {
	// remove the 0 bytes from the above measurement from the
	// audio driver's STREAMS queue
        flush_audio(fd);
	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(void)
{
    if (audio_dev == NULL) {
	if ((audio_dev = getenv("AUDIODEV")) == NULL)
	    audio_dev = "/dev/audio";
    }

    if (sun_mixer_device == NULL) {
	if ((sun_mixer_device = mixer_device) == NULL || !sun_mixer_device[0]) {
	    sun_mixer_device = malloc(strlen(audio_dev) + 4);
	    strcpy(sun_mixer_device, audio_dev);
	    strcat(sun_mixer_device, "ctl");
	}
    }

    if (ao_subdevice) audio_dev = ao_subdevice;
}

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

	if ( !sun_mixer_device )    /* control function is used before init? */
	    setup_device_paths();

	fd=open( sun_mixer_device,O_RDONLY );
	if ( fd != -1 )
	{
	    ao_control_vol_t *vol = (ao_control_vol_t *)arg;
	    float volume;
	    struct audio_info info;
	    ioctl( fd,AUDIO_GETINFO,&info);
	    volume = info.play.gain * 100. / AUDIO_MAX_GAIN;
	    if ( info.play.balance == AUDIO_MID_BALANCE ) {
		vol->right = vol->left = volume;
	    } else if ( info.play.balance < AUDIO_MID_BALANCE ) {
		vol->left  = volume;
		vol->right = volume * info.play.balance / AUDIO_MID_BALANCE;
	    } else {
		vol->left  = volume * (AUDIO_RIGHT_BALANCE-info.play.balance)
							/ AUDIO_MID_BALANCE;
		vol->right = volume;
	    }
	    close( fd );
	    return CONTROL_OK;
	}	
	return CONTROL_ERROR;
    }
    case AOCONTROL_SET_VOLUME:
    {
	ao_control_vol_t *vol = (ao_control_vol_t *)arg;
        int fd;

	if ( !sun_mixer_device )    /* control function is used before init? */
	    setup_device_paths();

	fd=open( sun_mixer_device,O_RDONLY );
	if ( fd != -1 )
	{
	    struct audio_info info;
	    float volume;
	    AUDIO_INITINFO(&info);
	    volume = vol->right > vol->left ? vol->right : vol->left;
	    if ( volume != 0 ) {
		info.play.gain = volume * AUDIO_MAX_GAIN / 100;
		if ( vol->right == vol->left )
		    info.play.balance = AUDIO_MID_BALANCE;
		else
		    info.play.balance = (vol->right - vol->left + volume) * AUDIO_RIGHT_BALANCE / (2*volume);
	    }
#if !defined (__OpenBSD__) && !defined (__NetBSD__)
	    info.output_muted = (volume == 0);
#endif
	    ioctl( fd,AUDIO_SETINFO,&info );
	    close( fd );
	    return CONTROL_OK;
	}	
	return CONTROL_ERROR;
    }
    }
    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 pass;
    int ok;
    int convert_u8_s8;

    setup_device_paths();

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

    mp_msg(MSGT_AO,MSGL_STATUS,"ao2: %d Hz  %d chans  %s [0x%X]\n",
	   rate,channels,af_fmt2str_short(format),format);

    audio_fd=open(audio_dev, O_WRONLY);
    if(audio_fd<0){
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantOpenAudioDev, audio_dev, strerror(errno));
	return 0;
    }

    ioctl(audio_fd, AUDIO_DRAIN, 0);

    if (af2sunfmt(format) == AUDIO_ENCODING_NONE)
      format = AF_FORMAT_S16_NE;

    for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */

	AUDIO_INITINFO(&info);
	info.play.encoding = af2sunfmt(ao_data.format = format);
	info.play.precision =
	    (format==AF_FORMAT_S16_NE
	     ? AUDIO_PRECISION_16
	     : AUDIO_PRECISION_8);
	info.play.channels = ao_data.channels = channels;
	info.play.sample_rate = ao_data.samplerate = rate;

	convert_u8_s8 = 0;

	if (pass & 1) {
	    /*
	     * on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is 
	     * not supported, but 8-bit signed encoding is.
	     *
	     * Try S8, and if it works, use our own U8->S8 conversion before
	     * sending the samples to the sound driver.
	     */
#ifdef AUDIO_ENCODING_LINEAR8
	    if (info.play.encoding != AUDIO_ENCODING_LINEAR8)
#endif
		continue;
	    info.play.encoding = AUDIO_ENCODING_LINEAR;
	    convert_u8_s8 = 1;
	}

	if (pass & 2) {
	    /*
	     * on some sun audio drivers, only certain fixed sample rates are
	     * supported.
	     *
	     * In case the requested sample rate is very close to one of the
	     * supported rates,  use the fixed supported rate instead.
	     */
	    if (!(info.play.sample_rate =
		  find_close_samplerate_match(audio_fd, rate))) 
	      continue;

	    /*
	     * I'm not returning the correct sample rate in
	     * |ao_data.samplerate|, to avoid software resampling.
	     *
	     * ao_data.samplerate = info.play.sample_rate;
	     */
	}

	if (pass & 4) {
	    /* like "pass & 2", but use the highest supported sample rate */
	    if (!(info.play.sample_rate
		  = ao_data.samplerate
		  = find_highest_samplerate(audio_fd)))
		continue;
	}

	ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
	if (ok) {
	    /* audio format accepted by audio driver */
	    break;
	}

	/*
	 * format not supported?
	 * retry with different encoding and/or sample rate
	 */
    }

    if (!ok) {
	char buf[128];
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
	       channels, af_fmt2str(format, buf, 128), rate);
	return 0;
    }

    if (convert_u8_s8)
      ao_data.format = AF_FORMAT_S8;

    bytes_per_sample = channels * info.play.precision / 8;
    ao_data.bps = byte_per_sec = bytes_per_sample * ao_data.samplerate;
    ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192;

    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(int immed){
    // throw away buffered data in the audio driver's STREAMS queue
    if (immed)
	flush_audio(audio_fd);
    close(audio_fd);
}

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

    uninit(1);
    audio_fd=open(audio_dev, O_WRONLY);
    if(audio_fd<0){
	mp_msg(MSGT_AO, MSGL_FATAL, MSGTR_AO_SUN_CantReopenReset, strerror(errno));
	return;
    }

    ioctl(audio_fd, AUDIO_DRAIN, 0);

    AUDIO_INITINFO(&info);
    info.play.encoding = af2sunfmt(ao_data.format);
    info.play.precision =
	(ao_data.format==AF_FORMAT_S16_NE 
	 ? AUDIO_PRECISION_16
	 : AUDIO_PRECISION_8);
    info.play.channels = ao_data.channels;
    info.play.sample_rate = ao_data.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(void)
{
    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(void)
{
    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(void){
    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 !defined (__OpenBSD__) && !defined(__NetBSD__)
    if (queued_bursts - info.play.eof > 2)
	return 0;
    return ao_data.outburst;
#else
    return info.hiwat * info.blocksize - info.play.seek;
#endif

}

// 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_data.outburst) return 0;
    len /= ao_data.outburst;
    len *= ao_data.outburst;

    len = write(audio_fd, data, len);
    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: delay in seconds between first and last sample in buffer
static float get_delay(void){
    audio_info_t info;
    ioctl(audio_fd, AUDIO_GETINFO, &info);
#if defined (__OpenBSD__) || defined(__NetBSD__)
    return (float) info.play.seek/ (float)byte_per_sec ;
#else
    if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
	return (float)(queued_samples - info.play.samples) / (float)ao_data.samplerate;
    else
	return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
#endif
}