annotate libao2/ao_esd.c @ 36004:b81edd9d28ab

Move PIC detection to the front of PIE added. It seems that clang turns off PIC when -fpie is set, which causes part of ffmpeg failed to be compiled on OS X. In fact, in the configure of ffmpeg, PIC is unconditionally enabled, which could be introduced as ffmpeg seems to be an essential part of mplayer.

/*
 * EsounD audio output driver for MPlayer
 *
 * copyright (c) 2002 Juergen Keil <[e-mail]>
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

 /*
 * TODO / known problems:
 * - does not work well when the esd daemon has autostandby disabled
 *   (workaround: run esd with option "-as 2" - fortunatelly this is
 *    the default)
 * - plays noise on a linux 2.4.4 kernel with a SB16PCI card, when using
 *   a local tcp connection to the esd daemon; there is no noise when using
 *   a unix domain socket connection.
 *   (there are EIO errors reported by the sound card driver, so this is
 *   most likely a linux sound card driver problem)
 */

#include <sys/types.h>
#include <sys/time.h>
#include <sys/socket.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include <fcntl.h>
#include <time.h>
#ifdef	__svr4__
#include <stropts.h>
#endif
#include <esd.h>

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


#define ESD_RESAMPLES 0
#define ESD_DEBUG 0

#if	ESD_DEBUG
#define	dprintf(...)	printf(__VA_ARGS__)
#else
#define	dprintf(...)	/**/
#endif


#define	ESD_CLIENT_NAME	"MPlayer"
#define	ESD_MAX_DELAY	(1.0f)	/* max amount of data buffered in esd (#sec) */

static const ao_info_t info =
{
    "EsounD audio output",
    "esd",
    "Juergen Keil <[e-mail]>",
    ""
};

LIBAO_EXTERN(esd)

static int esd_fd = -1;
static int esd_play_fd = -1;
static esd_server_info_t *esd_svinfo;
static int esd_latency;
static int esd_bytes_per_sample;
static unsigned long esd_samples_written;
static struct timeval esd_play_start;

/*
 * to set/get/query special features/parameters
 */
static int control(int cmd, void *arg)
{
    esd_player_info_t *esd_pi;
    esd_info_t        *esd_i;
    time_t	       now;
    static time_t      vol_cache_time;
    static ao_control_vol_t vol_cache;

    switch (cmd) {
    case AOCONTROL_GET_VOLUME:
	time(&now);
	if (now == vol_cache_time) {
	    *(ao_control_vol_t *)arg = vol_cache;
	    return CONTROL_OK;
	}

	dprintf("esd: get vol\n");
	if ((esd_i = esd_get_all_info(esd_fd)) == NULL)
	    return CONTROL_ERROR;

	for (esd_pi = esd_i->player_list; esd_pi != NULL; esd_pi = esd_pi->next)
	    if (strcmp(esd_pi->name, ESD_CLIENT_NAME) == 0)
		break;

	if (esd_pi != NULL) {
	    ao_control_vol_t *vol = (ao_control_vol_t *)arg;
	    vol->left =  esd_pi->left_vol_scale  * 100 / ESD_VOLUME_BASE;
	    vol->right = esd_pi->right_vol_scale * 100 / ESD_VOLUME_BASE;

	    vol_cache = *vol;
	    vol_cache_time = now;
	}
	esd_free_all_info(esd_i);

	return CONTROL_OK;

    case AOCONTROL_SET_VOLUME:
	dprintf("esd: set vol\n");
	if ((esd_i = esd_get_all_info(esd_fd)) == NULL)
	    return CONTROL_ERROR;

	for (esd_pi = esd_i->player_list; esd_pi != NULL; esd_pi = esd_pi->next)
	    if (strcmp(esd_pi->name, ESD_CLIENT_NAME) == 0)
		break;

	if (esd_pi != NULL) {
	    ao_control_vol_t *vol = (ao_control_vol_t *)arg;
	    esd_set_stream_pan(esd_fd, esd_pi->source_id,
			       vol->left  * ESD_VOLUME_BASE / 100,
			       vol->right * ESD_VOLUME_BASE / 100);

	    vol_cache = *vol;
	    time(&vol_cache_time);
	}
	esd_free_all_info(esd_i);
	return CONTROL_OK;

    default:
	return CONTROL_UNKNOWN;
    }
}


/*
 * open & setup audio device
 * return: 1=success 0=fail
 */
static int init(int rate_hz, int channels, int format, int flags)
{
    esd_format_t esd_fmt;
    int bytes_per_sample;
    int fl;
    char *server = ao_subdevice;  /* NULL for localhost */
    float lag_seconds, lag_net, lag_serv;
    struct timeval proto_start, proto_end;

    if (esd_fd < 0) {
	esd_fd = esd_open_sound(server);
	if (esd_fd < 0) {
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenSound,
		   strerror(errno));
	    return 0;
	}

	/* get server info, and measure network latency */
	gettimeofday(&proto_start, NULL);
	esd_svinfo = esd_get_server_info(esd_fd);
	if(server) {
	    gettimeofday(&proto_end, NULL);
	    lag_net  = (proto_end.tv_sec  - proto_start.tv_sec) +
		(proto_end.tv_usec - proto_start.tv_usec) / 1000000.0;
	    lag_net /= 2.0; /* round trip -> one way */
	} else
	    lag_net = 0.0;  /* no network lag */

	/*
	if (esd_svinfo) {
	    mp_msg(MSGT_AO, MSGL_INFO, "AO: [esd] server info:\n");
	    esd_print_server_info(esd_svinfo);
	}
	*/
    }

    esd_fmt = ESD_STREAM | ESD_PLAY;

#if	ESD_RESAMPLES
    /* let the esd daemon convert sample rate */
#else
    /* let mplayer's audio filter convert the sample rate */
    if (esd_svinfo != NULL)
	rate_hz = esd_svinfo->rate;
#endif
    ao_data.samplerate = rate_hz;

    /* EsounD can play mono or stereo */
    switch (channels) {
    case 1:
	esd_fmt |= ESD_MONO;
	ao_data.channels = bytes_per_sample = 1;
	break;
    default:
	esd_fmt |= ESD_STEREO;
	ao_data.channels = bytes_per_sample = 2;
	break;
    }

    /* EsounD can play 8bit unsigned and 16bit signed native */
    switch (format) {
    case AF_FORMAT_S8:
    case AF_FORMAT_U8:
	esd_fmt |= ESD_BITS8;
	ao_data.format = AF_FORMAT_U8;
	break;
    default:
	esd_fmt |= ESD_BITS16;
	ao_data.format = AF_FORMAT_S16_NE;
	bytes_per_sample *= 2;
	break;
    }

    /* modify audio_delay depending on esd_latency
     * latency is number of samples @ 44.1khz stereo 16 bit
     * adjust according to rate_hz & bytes_per_sample
     */
#ifdef CONFIG_ESD_LATENCY
    esd_latency = esd_get_latency(esd_fd);
#else
    esd_latency = ((channels == 1 ? 2 : 1) * ESD_DEFAULT_RATE *
		   (ESD_BUF_SIZE + 64 * (4.0f / bytes_per_sample))
		   ) / rate_hz;
    esd_latency += ESD_BUF_SIZE * 2;
#endif
    if(esd_latency > 0) {
	lag_serv = (esd_latency * 4.0f) / (bytes_per_sample * rate_hz);
	lag_seconds = lag_net + lag_serv;
	audio_delay += lag_seconds;
	mp_msg(MSGT_AO, MSGL_INFO,MSGTR_AO_ESD_LatencyInfo,
	       lag_serv, lag_net, lag_seconds);
    }

    esd_play_fd = esd_play_stream_fallback(esd_fmt, rate_hz,
					   server, ESD_CLIENT_NAME);
    if (esd_play_fd < 0) {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenPBStream, strerror(errno));
	return 0;
    }

    /* enable non-blocking i/o on the socket connection to the esd server */
    if ((fl = fcntl(esd_play_fd, F_GETFL)) >= 0)
	fcntl(esd_play_fd, F_SETFL, O_NDELAY|fl);

#if ESD_DEBUG
    {
	int sbuf, rbuf, len;
	len = sizeof(sbuf);
	getsockopt(esd_play_fd, SOL_SOCKET, SO_SNDBUF, &sbuf, &len);
	len = sizeof(rbuf);
	getsockopt(esd_play_fd, SOL_SOCKET, SO_RCVBUF, &rbuf, &len);
	dprintf("esd: send/receive socket buffer space %d/%d bytes\n",
		sbuf, rbuf);
    }
#endif

    ao_data.bps = bytes_per_sample * rate_hz;
    ao_data.outburst = ao_data.bps > 100000 ? 4*ESD_BUF_SIZE : 2*ESD_BUF_SIZE;

    esd_play_start.tv_sec = 0;
    esd_samples_written = 0;
    esd_bytes_per_sample = bytes_per_sample;

    return 1;
}


/*
 * close audio device
 */
static void uninit(int immed)
{
    if (esd_play_fd >= 0) {
	esd_close(esd_play_fd);
	esd_play_fd = -1;
    }

    if (esd_svinfo) {
	esd_free_server_info(esd_svinfo);
	esd_svinfo = NULL;
    }

    if (esd_fd >= 0) {
	esd_close(esd_fd);
	esd_fd = -1;
    }
}


/*
 * 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)
{
    int offs;
    int nwritten;
    int nsamples;
    int n;

    /* round down buffersize to a multiple of ESD_BUF_SIZE bytes */
    len = len / ESD_BUF_SIZE * ESD_BUF_SIZE;
    if (len <= 0)
	return 0;

#define	SINGLE_WRITE 0
#if	SINGLE_WRITE
    nwritten = write(esd_play_fd, data, len);
#else
    for (offs = 0, nwritten=0; offs + ESD_BUF_SIZE <= len; offs += ESD_BUF_SIZE) {
	/*
	 * note: we're writing to a non-blocking socket here.
	 * A partial write means, that the socket buffer is full.
	 */
	n = write(esd_play_fd, (char*)data + offs, ESD_BUF_SIZE);
	if ( n < 0 ) {
	    if ( errno != EAGAIN )
		dprintf("esd play: write failed: %s\n", strerror(errno));
	    break;
	} else if ( n != ESD_BUF_SIZE ) {
	    nwritten += n;
	    break;
	} else
	    nwritten += n;
    }
#endif

    if (nwritten > 0) {
	if (!esd_play_start.tv_sec)
	    gettimeofday(&esd_play_start, NULL);
	nsamples = nwritten / esd_bytes_per_sample;
	esd_samples_written += nsamples;

	dprintf("esd play: %d %lu\n", nsamples, esd_samples_written);
    } else {
	dprintf("esd play: blocked / %lu\n", esd_samples_written);
    }

    return nwritten;
}


/*
 * stop playing, keep buffers (for pause)
 */
static void audio_pause(void)
{
    /*
     * not possible with esd.  the esd daemom will continue playing
     * buffered data (not more than ESD_MAX_DELAY seconds of samples)
     */
}


/*
 * resume playing, after audio_pause()
 */
static void audio_resume(void)
{
    /*
     * not possible with esd.
     *
     * Let's hope the pause was long enough that the esd ran out of
     * buffered data;  we restart our time based delay computation
     * for an audio resume.
     */
    esd_play_start.tv_sec = 0;
    esd_samples_written = 0;
}


/*
 * stop playing and empty buffers (for seeking/pause)
 */
static void reset(void)
{
#ifdef	__svr4__
    /* throw away data buffered in the esd connection */
    if (ioctl(esd_play_fd, I_FLUSH, FLUSHW))
	perror("I_FLUSH");
#endif
}


/*
 * return: how many bytes can be played without blocking
 */
static int get_space(void)
{
    struct timeval tmout;
    fd_set wfds;
    float current_delay;
    int space;

    /*
     * Don't buffer too much data in the esd daemon.
     *
     * If we send too much, esd will block in write()s to the sound
     * device, and the consequence is a huge slow down for things like
     * esd_get_all_info().
     */
    if ((current_delay = get_delay()) >= ESD_MAX_DELAY) {
	dprintf("esd get_space: too much data buffered\n");
	return 0;
    }

    FD_ZERO(&wfds);
    FD_SET(esd_play_fd, &wfds);
    tmout.tv_sec = 0;
    tmout.tv_usec = 0;

    if (select(esd_play_fd + 1, NULL, &wfds, NULL, &tmout) != 1)
	return 0;

    if (!FD_ISSET(esd_play_fd, &wfds))
	return 0;

    /* try to fill 50% of the remaining "free" buffer space */
    space = (ESD_MAX_DELAY - current_delay) * ao_data.bps * 0.5f;

    /* round up to next multiple of ESD_BUF_SIZE */
    space = (space + ESD_BUF_SIZE-1) / ESD_BUF_SIZE * ESD_BUF_SIZE;

    dprintf("esd get_space: %d\n", space);
    return space;
}


/*
 * return: delay in seconds between first and last sample in buffer
 */
static float get_delay(void)
{
    struct timeval now;
    double buffered_samples_time;
    double play_time;

    if (!esd_play_start.tv_sec)
	return 0;

    buffered_samples_time = (float)esd_samples_written / ao_data.samplerate;
    gettimeofday(&now, NULL);
    play_time  =  now.tv_sec  - esd_play_start.tv_sec;
    play_time += (now.tv_usec - esd_play_start.tv_usec) / 1000000.;

    /* dprintf("esd delay: %f %f\n", play_time, buffered_samples_time); */

    if (play_time > buffered_samples_time) {
	dprintf("esd: underflow\n");
	esd_play_start.tv_sec = 0;
	esd_samples_written = 0;
	return 0;
    }

    dprintf("esd: get_delay %f\n", buffered_samples_time - play_time);
    return buffered_samples_time - play_time;
}