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diff --git a/cinelerra-5.1/libzmpeg3/a52dec-0.7.3/libao/audio_out_oss.c b/cinelerra-5.1/libzmpeg3/a52dec-0.7.3/libao/audio_out_oss.c
new file mode 100644
index 00000000..3247a2df
--- /dev/null
+++ b/cinelerra-5.1/libzmpeg3/a52dec-0.7.3/libao/audio_out_oss.c
@@ -0,0 +1,194 @@
+/*
+ * audio_out_oss.c
+ * Copyright (C) 2000-2002 Michel Lespinasse <walken@zoy.org>
+ * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
+ *
+ * This file is part of a52dec, a free ATSC A-52 stream decoder.
+ * See http://liba52.sourceforge.net/ for updates.
+ *
+ * a52dec 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.
+ *
+ * a52dec 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 this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include "config.h"
+
+#ifdef LIBAO_OSS
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <inttypes.h>
+
+#if defined(__OpenBSD__)
+#include <soundcard.h>
+#elif defined(__FreeBSD__)
+#include <machine/soundcard.h>
+#ifndef AFMT_S16_NE
+#include <machine/endian.h>
+#if BYTE_ORDER == LITTLE_ENDIAN
+#define AFMT_S16_NE AFMT_S16_LE
+#else
+#define AFMT_S16_NE AFMT_S16_BE
+#endif
+#endif
+#else
+#include <sys/soundcard.h>
+#endif
+
+#include "a52.h"
+#include "audio_out.h"
+#include "audio_out_internal.h"
+
+typedef struct oss_instance_s {
+    ao_instance_t ao;
+    int fd;
+    int sample_rate;
+    int set_params;
+    int flags;
+} oss_instance_t;
+
+int oss_setup (ao_instance_t * _instance, int sample_rate, int * flags,
+	       sample_t * level, sample_t * bias)
+{
+    oss_instance_t * instance = (oss_instance_t *) _instance;
+
+    if ((instance->set_params == 0) && (instance->sample_rate != sample_rate))
+	return 1;
+    instance->sample_rate = sample_rate;
+
+    *flags = instance->flags;
+    *level = 1;
+    *bias = 384;
+
+    return 0;
+}
+
+int oss_play (ao_instance_t * _instance, int flags, sample_t * _samples)
+{
+    oss_instance_t * instance = (oss_instance_t *) _instance;
+    int16_t int16_samples[256*6];
+    int chans = -1;
+
+#ifdef LIBA52_DOUBLE
+    float samples[256 * 6];
+    int i;
+
+    for (i = 0; i < 256 * 6; i++)
+	samples[i] = _samples[i];
+#else
+    float * samples = _samples;
+#endif
+
+    chans = channels_multi (flags);
+    flags &= A52_CHANNEL_MASK | A52_LFE;
+
+    if (instance->set_params) {
+	int tmp;
+
+	tmp = chans;
+	if ((ioctl (instance->fd, SNDCTL_DSP_CHANNELS, &tmp) < 0) ||
+	    (tmp != chans)) {
+	    fprintf (stderr, "Can not set number of channels\n");
+	    return 1;
+	}
+
+	tmp = instance->sample_rate;
+	if ((ioctl (instance->fd, SNDCTL_DSP_SPEED, &tmp) < 0) ||
+	    (tmp != instance->sample_rate)) {
+	    fprintf (stderr, "Can not set sample rate\n");
+	    return 1;
+	}
+
+	instance->flags = flags;
+	instance->set_params = 0;
+    } else if ((flags == A52_DOLBY) && (instance->flags == A52_STEREO)) {
+	fprintf (stderr, "Switching from stereo to dolby surround\n");
+	instance->flags = A52_DOLBY;
+    } else if ((flags == A52_STEREO) && (instance->flags == A52_DOLBY)) {
+	fprintf (stderr, "Switching from dolby surround to stereo\n");
+	instance->flags = A52_STEREO;
+    } else if (flags != instance->flags)
+	return 1;
+
+    float2s16_multi (samples, int16_samples, flags);
+    write (instance->fd, int16_samples, 256 * sizeof (int16_t) * chans);
+
+    return 0;
+}
+
+void oss_close (ao_instance_t * _instance)
+{
+    oss_instance_t * instance = (oss_instance_t *) _instance;
+
+    close (instance->fd);
+}
+
+ao_instance_t * oss_open (int flags)
+{
+    oss_instance_t * instance;
+    int format;
+
+    instance = malloc (sizeof (oss_instance_t));
+    if (instance == NULL)
+	return NULL;
+
+    instance->ao.setup = oss_setup;
+    instance->ao.play = oss_play;
+    instance->ao.close = oss_close;
+
+    instance->sample_rate = 0;
+    instance->set_params = 1;
+    instance->flags = flags;
+
+    instance->fd = open ("/dev/dsp", O_WRONLY);
+    if (instance->fd < 0) {
+	fprintf (stderr, "Can not open /dev/dsp\n");
+	free (instance);
+	return NULL;
+    }
+
+    format = AFMT_S16_NE;
+    if ((ioctl (instance->fd, SNDCTL_DSP_SETFMT, &format) < 0) ||
+	(format != AFMT_S16_NE)) {
+	fprintf (stderr, "Can not set sample format\n");
+	free (instance);
+	return NULL;
+    }
+
+    return (ao_instance_t *) instance;
+}
+
+ao_instance_t * ao_oss_open (void)
+{
+    return oss_open (A52_STEREO);
+}
+
+ao_instance_t * ao_ossdolby_open (void)
+{
+    return oss_open (A52_DOLBY);
+}
+
+ao_instance_t * ao_oss4_open (void)
+{
+    return oss_open (A52_2F2R);
+}
+
+ao_instance_t * ao_oss6_open (void)
+{
+    return oss_open (A52_3F2R | A52_LFE);
+}
+
+#endif