view libao2/ao_sun.c @ 4218:3931c41f740a

Added new syncengine thanks to a new previously undocumented feature of the em8300, this might fix playback on both slow and fast machines (more testing needed). This also requires users to get the em8300 driver from cvs until the next version is released (will probably happen this weekend) Added lots of comments, should be pretty easy to understand most of the internals now Added lots of brackets to if's for's while's etc, this is not a cosmetical thing but rather due to the fact I got some very odd bugs with else's since I didn't properly use brackets (and it's the K&R standard to have brackets everywhere) Fixed some bugs that would occur when disabling libmp1e Switched to default to the new naming scheme of device nodes, the driver will slowly switch over to this state, if it can't find devices under the new name it will try the old naming scheme I stopped opening devices in non-blocking mode, it would break the new syncengine which tries to burst data to the device (alot of times meaning it will fill the fifo pretty fast which would previously result in jerkyness on fast machines) The device now sets the initial state of the pts and speed (probably not needed, but assumption is the mother of all fuckups =) Keep the control interface open during the entire duration of the libvo device, we might need this to flush video buffers on seeking (currently not implemented, therefore seeking is broken) This is beta stuff to the driver, I will get some users to test it for me and do my best to fix seeking as soon as possible...

#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


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 convert_u8_s8;
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_BE:
    case AFMT_S16_LE:
	return AUDIO_ENCODING_LINEAR;
#ifdef	AUDIO_ENCODING_LINEAR8	// Missing on SunOS 5.5.1...
    case AFMT_U8:
	return AUDIO_ENCODING_LINEAR8;
#endif
#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.  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)
	    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;
    }

    /*
     * 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)
	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 ok;

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

    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_data.format = format);
    info.play.precision =
	(format==AFMT_S16_LE || format==AFMT_S16_BE
	 ? 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;
    ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
    if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) {
	/* sun audiocs hardware does not support U8 format, try S8... */
	info.play.encoding = AUDIO_ENCODING_LINEAR;
	ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
	if (ok) {
	    /* we must perform software U8 -> S8 conversion */
	    convert_u8_s8 = 1;
	}
    }
    if (!ok) {
	printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",
	       channels, audio_out_format_name(format), rate);
	return 0;
    }

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

#ifdef	__not_used__
    /*
     * hmm, ao_data.buffersize is currently not used in this driver, do there's
     * no need to measure it
     */
    if(ao_data.buffersize==-1){
	// Measuring buffer size:
	void* data;
	ao_data.buffersize=0;
#ifdef HAVE_AUDIO_SELECT
	data = malloc(ao_data.outburst);
	memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst);
	while(ao_data.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_data.outburst);
	    ao_data.buffersize+=ao_data.outburst;
	}
	free(data);
	if(ao_data.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_data.buffersize measurement from the
	// audio driver's STREAMS queue
	ioctl(audio_fd, I_FLUSH, FLUSHW);
#endif
	ioctl(audio_fd, AUDIO_DRAIN, 0);
#endif
    }
#endif	/* __not_used__ */

    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_data.format);
    info.play.precision =
	(ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE 
	 ? 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()
{
    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_data.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_data.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	WORDS_BIGENDIAN
    int native_endian = AFMT_S16_BE;
#else
    int native_endian = AFMT_S16_LE;
#endif

    if (len < ao_data.outburst) return 0;
    len /= ao_data.outburst;
    len *= ao_data.outburst;

    /* 16-bit format using the 'wrong' byteorder?  swap words */
    if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE)
	&& ao_data.format != native_endian) {
	static void *swab_buf;
	static int swab_len;
	if (len > swab_len) {
	    if (swab_buf)
		swab_buf = realloc(swab_buf, len);
	    else
		swab_buf = malloc(len);
	    swab_len = len;
	    if (swab_buf == NULL) return 0;
	}
	swab(data, swab_buf, len);
	data = swab_buf;
    } else if (ao_data.format == AFMT_U8 && convert_u8_s8) {
	int i;
	unsigned char *p = data;

	for (i = 0, p = data; i < len; i++, p++)
	    *p ^= 0x80;
    }

    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(){
    audio_info_t info;
    ioctl(audio_fd, AUDIO_GETINFO, &info);
    if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
	return (float)(queued_samples - info.play.samples) / (float)byte_per_sec;
    else
	return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
}