Add initial support for OSSv4. Mixer still needs a bit more love, but even magic has its limits.

Original commit message from CVS:
* configure.ac:
* sys/Makefile.am:
* sys/oss4/Makefile.am:
* sys/oss4/oss4-audio.c:
* sys/oss4/oss4-audio.h:
* sys/oss4/oss4-mixer-enum.c:
* sys/oss4/oss4-mixer-enum.h:
* sys/oss4/oss4-mixer-slider.c:
* sys/oss4/oss4-mixer-slider.h:
* sys/oss4/oss4-mixer-switch.c:
* sys/oss4/oss4-mixer-switch.h:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-mixer.h:
* sys/oss4/oss4-property-probe.c:
* sys/oss4/oss4-property-probe.h:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-sink.h:
* sys/oss4/oss4-soundcard.h:
* sys/oss4/oss4-source.c:
* sys/oss4/oss4-source.h:
Add initial support for OSSv4. Mixer still needs a bit more love,
but even magic has its limits.

19 files changed, 7866 insertions, 2 deletions

diff --git a/sys/Makefile.am b/sys/Makefile.am
index d7730426..55a01374 100644
--- a/sys/Makefile.am
+++ b/sys/Makefile.am
@@ -34,13 +34,19 @@ else
 DVB_DIR=
 endif
 
+if USE_OSS4
+OSS4_DIR=oss4
+else
+OSS4_DIR=
+endif
+
 if USE_QUICKTIME
 QT_DIR=qtwrapper
 else
 QT_DIR=
 endif
 
-SUBDIRS = $(FBDEV_DIR) $(DVB_DIR) $(VCD_DIR) $(QT_DIR)
+SUBDIRS = $(FBDEV_DIR) $(DVB_DIR) $(VCD_DIR) $(QT_DIR) $(OSS4_DIR)
 
-DIST_SUBDIRS = dvb fbdev vcd qtwrapper dshowdecwrapper dshowsrcwrapper
+DIST_SUBDIRS = dvb fbdev dshowdecwrapper dshowsrcwrapper oss4 qtwrapper vcd
 
diff --git a/sys/oss4/Makefile.am b/sys/oss4/Makefile.am
new file mode 100644
index 00000000..34472c0a
--- /dev/null
+++ b/sys/oss4/Makefile.am
@@ -0,0 +1,32 @@
+plugin_LTLIBRARIES = libgstoss4audio.la
+
+libgstoss4audio_la_SOURCES = \
+	oss4-audio.c          \
+	oss4-mixer.c          \
+	oss4-mixer-enum.c     \
+	oss4-mixer-slider.c   \
+	oss4-mixer-switch.c   \
+	oss4-property-probe.c \
+	oss4-sink.c           \
+	oss4-source.c
+
+libgstoss4audio_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+libgstoss4audio_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) \
+        -lgstinterfaces-$(GST_MAJORMINOR) \
+	-lgstaudio-$(GST_MAJORMINOR) \
+	$(GST_LIBS)
+libgstoss4audio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) 
+
+noinst_HEADERS = \
+	oss4-audio.h          \
+	oss4-mixer.h          \
+	oss4-mixer-enum.h     \
+	oss4-mixer-slider.h   \
+	oss4-mixer-switch.h   \
+	oss4-property-probe.h \
+	oss4-sink.h           \
+	oss4-soundcard.h      \
+	oss4-source.h
+
+
diff --git a/sys/oss4/oss4-audio.c b/sys/oss4/oss4-audio.c
new file mode 100644
index 00000000..a2404cc8
--- /dev/null
+++ b/sys/oss4/oss4-audio.c
@@ -0,0 +1,721 @@
+/* GStreamer OSS4 audio plugin
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <sys/ioctl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "gst/gst-i18n-plugin.h"
+#include <gst/audio/multichannel.h>
+
+#include "oss4-audio.h"
+#include "oss4-mixer.h"
+#include "oss4-property-probe.h"
+#include "oss4-sink.h"
+#include "oss4-source.h"
+#include "oss4-soundcard.h"
+
+GST_DEBUG_CATEGORY (oss4mixer_debug);
+GST_DEBUG_CATEGORY (oss4sink_debug);
+GST_DEBUG_CATEGORY (oss4src_debug);
+GST_DEBUG_CATEGORY (oss4_debug);
+
+#define GST_CAT_DEFAULT oss4_debug
+
+static const struct
+{
+  const GstBufferFormat gst_fmt;
+  const gint oss_fmt;
+  const gchar name[16];
+  const gint depth;
+  const gint width;
+  const gint endianness;
+  const gboolean signedness;
+} fmt_map[] = {
+  /* note: keep sorted by preference, prefered formats first */
+  {
+  GST_MU_LAW, AFMT_MU_LAW, "audio/x-mulaw", 0, 0, 0, FALSE}, {
+  GST_A_LAW, AFMT_A_LAW, "audio/x-alaw", 0, 0, 0, FALSE}, {
+  GST_S32_LE, AFMT_S32_LE, "audio/x-raw-int", 32, 32, G_LITTLE_ENDIAN, TRUE}, {
+  GST_S32_BE, AFMT_S32_BE, "audio/x-raw-int", 32, 32, G_BIG_ENDIAN, TRUE}, {
+  GST_S24_LE, AFMT_S24_LE, "audio/x-raw-int", 24, 32, G_LITTLE_ENDIAN, TRUE}, {
+  GST_S24_BE, AFMT_S24_BE, "audio/x-raw-int", 24, 32, G_BIG_ENDIAN, TRUE}, {
+  GST_S24_3LE, AFMT_S24_PACKED, "audio/x-raw-int", 24, 24, G_LITTLE_ENDIAN,
+        TRUE}, {
+  GST_S16_LE, AFMT_S16_LE, "audio/x-raw-int", 16, 16, G_LITTLE_ENDIAN, TRUE}, {
+  GST_S16_BE, AFMT_S16_BE, "audio/x-raw-int", 16, 16, G_BIG_ENDIAN, TRUE}, {
+  GST_U16_LE, AFMT_U16_LE, "audio/x-raw-int", 16, 16, G_LITTLE_ENDIAN, FALSE}, {
+  GST_U16_BE, AFMT_U16_BE, "audio/x-raw-int", 16, 16, G_BIG_ENDIAN, FALSE}, {
+  GST_S8, AFMT_S8, "audio/x-raw-int", 8, 8, 0, TRUE}, {
+  GST_U8, AFMT_U8, "audio/x-raw-int", 8, 8, 0, FALSE}
+};
+
+static gboolean
+gst_oss4_append_format_to_caps (gint fmt, GstCaps * caps)
+{
+  gint i;
+
+  for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
+    if (fmt_map[i].oss_fmt == fmt) {
+      GstStructure *s;
+
+      s = gst_structure_empty_new (fmt_map[i].name);
+      if (fmt_map[i].width != 0 && fmt_map[i].depth != 0) {
+        gst_structure_set (s, "width", G_TYPE_INT, fmt_map[i].width,
+            "depth", G_TYPE_INT, fmt_map[i].depth, "endianness", G_TYPE_INT,
+            fmt_map[i].endianness, "signed", G_TYPE_BOOLEAN,
+            fmt_map[i].signedness, NULL);
+      }
+      gst_caps_append_structure (caps, s);
+      return TRUE;
+    }
+  }
+  return FALSE;
+}
+
+static gint
+gst_oss4_audio_get_oss_format (GstBufferFormat fmt)
+{
+  guint i;
+
+  for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
+    if (fmt_map[i].gst_fmt == fmt)
+      return fmt_map[i].oss_fmt;
+  }
+  return 0;
+}
+
+/* These are pretty random */
+#define GST_OSS4_MIN_SAMPLE_RATE 1
+#define GST_OSS4_MAX_SAMPLE_RATE 192000
+
+static gboolean
+gst_oss4_audio_detect_rates (GstObject * obj, oss_audioinfo * ai,
+    GstCaps * caps)
+{
+  GValue val = { 0, };
+  int minrate, maxrate, i;
+
+  minrate = ai->min_rate;
+  maxrate = ai->max_rate;
+
+  /* sanity check */
+  if (minrate > maxrate) {
+    GST_WARNING_OBJECT (obj, "min_rate %d > max_rate %d (buggy driver?)",
+        minrate, maxrate);
+    maxrate = ai->min_rate;     /* swap */
+    minrate = ai->max_rate;
+  }
+
+  /* limit to something sensible */
+  if (minrate < GST_OSS4_MIN_SAMPLE_RATE)
+    minrate = GST_OSS4_MIN_SAMPLE_RATE;
+  if (maxrate > GST_OSS4_MAX_SAMPLE_RATE)
+    maxrate = GST_OSS4_MAX_SAMPLE_RATE;
+
+  if (maxrate < GST_OSS4_MIN_SAMPLE_RATE) {
+    GST_WARNING_OBJECT (obj, "max_rate < %d, which makes no sense",
+        GST_OSS4_MIN_SAMPLE_RATE);
+    return FALSE;
+  }
+
+  GST_LOG_OBJECT (obj, "min_rate %d, max_rate %d (originally: %d, %d)",
+      minrate, maxrate, ai->min_rate, ai->max_rate);
+
+  if ((ai->caps & PCM_CAP_FREERATE)) {
+    GST_LOG_OBJECT (obj, "device supports any sample rate between min and max");
+    if (minrate == maxrate) {
+      g_value_init (&val, G_TYPE_INT);
+      g_value_set_int (&val, maxrate);
+    } else {
+      g_value_init (&val, GST_TYPE_INT_RANGE);
+      gst_value_set_int_range (&val, minrate, maxrate);
+    }
+  } else {
+    GST_LOG_OBJECT (obj, "%d sample rates:", ai->nrates);
+    g_value_init (&val, GST_TYPE_LIST);
+    for (i = 0; i < ai->nrates; ++i) {
+      GST_LOG_OBJECT (obj, " rate: %d", ai->rates[i]);
+
+      if (ai->rates[i] >= minrate && ai->rates[i] <= maxrate) {
+        GValue rate_val = { 0, };
+
+        g_value_init (&rate_val, G_TYPE_INT);
+        g_value_set_int (&rate_val, ai->rates[i]);
+        gst_value_list_append_value (&val, &rate_val);
+        g_value_unset (&rate_val);
+      }
+    }
+
+    if (gst_value_list_get_size (&val) == 0) {
+      g_value_unset (&val);
+      return FALSE;
+    }
+  }
+
+  for (i = 0; i < gst_caps_get_size (caps); ++i) {
+    GstStructure *s;
+
+    s = gst_caps_get_structure (caps, i);
+    gst_structure_set_value (s, "rate", &val);
+  }
+
+  g_value_unset (&val);
+
+  return TRUE;
+}
+
+static void
+gst_oss4_audio_add_channel_layout (GstObject * obj, guint64 layout,
+    guint num_channels, GstStructure * s)
+{
+  const GstAudioChannelPosition pos_map[16] = {
+    GST_AUDIO_CHANNEL_POSITION_NONE,    /* 0 = dunno          */
+    GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,      /* 1 = left           */
+    GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,     /* 2 = right          */
+    GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,    /* 3 = center         */
+    GST_AUDIO_CHANNEL_POSITION_LFE,     /* 4 = lfe            */
+    GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,       /* 5 = left surround  */
+    GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,      /* 6 = right surround */
+    GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,       /* 7 = left rear      */
+    GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,      /* 8 = right rear     */
+    GST_AUDIO_CHANNEL_POSITION_NONE,
+    GST_AUDIO_CHANNEL_POSITION_NONE,
+    GST_AUDIO_CHANNEL_POSITION_NONE,
+    GST_AUDIO_CHANNEL_POSITION_NONE,
+    GST_AUDIO_CHANNEL_POSITION_NONE,
+    GST_AUDIO_CHANNEL_POSITION_NONE,
+    GST_AUDIO_CHANNEL_POSITION_NONE
+  };
+  GstAudioChannelPosition ch_layout[8] = { 0, };
+  guint speaker_pos;            /* speaker position as defined by OSS */
+  guint i;
+
+  g_return_if_fail (num_channels <= G_N_ELEMENTS (ch_layout));
+
+  for (i = 0; i < num_channels; ++i) {
+    /* layout contains up to 16 speaker positions, with each taking up 4 bits */
+    speaker_pos = (guint) ((layout >> (i * 4)) & 0x0f);
+
+    /* if it's a channel position that's unknown to us, set all to NONE and
+     * bail out */
+    if (G_UNLIKELY (pos_map[speaker_pos] == GST_AUDIO_CHANNEL_POSITION_NONE))
+      goto no_layout;
+
+    ch_layout[i] = pos_map[speaker_pos];
+  }
+  gst_audio_set_channel_positions (s, ch_layout);
+  return;
+
+no_layout:
+  {
+    /* only warn if it's really unknown, position 0 is ok and represents NONE
+     * (in which case we also just set all others to NONE ignoring the other
+     * positions in the OSS-given layout, because that's what we currently
+     * require in GStreamer) */
+    if (speaker_pos != 0) {
+      GST_WARNING_OBJECT (obj, "unknown OSS channel position %x", ch_layout[i]);
+    }
+    for (i = 0; i < num_channels; ++i) {
+      ch_layout[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+    }
+    gst_audio_set_channel_positions (s, ch_layout);
+    return;
+  }
+}
+
+/* arbitrary max. limit */
+#define GST_OSS4_MIN_CHANNELS 1
+#define GST_OSS4_MAX_CHANNELS 4096
+
+/* takes ownership of the input caps */
+static GstCaps *
+gst_oss4_audio_detect_channels (GstObject * obj, int fd, oss_audioinfo * ai,
+    GstCaps * in_caps)
+{
+  const gchar *forced_layout;
+  GstStructure *s = NULL;
+  guint64 layout = 0;
+  GstCaps *chan_caps = NULL;
+  GstCaps *out_caps = NULL;
+  int minchans, maxchans;
+  int c, i, j;
+
+  /* GST_OSS4_CHANNEL_LAYOUT environment variable: may be used to force a
+   * particular channel layout (if it contains an odd number of channel
+   * positions it will also make us advertise a channel layout for that
+   * channel count, even if we'd usually skip it; this is especially useful
+   * for folks with 2.1 speakers, I guess) */
+  forced_layout = g_getenv ("GST_OSS4_CHANNEL_LAYOUT");
+
+  minchans = ai->min_channels;
+  maxchans = ai->max_channels;
+
+  /* sanity check */
+  if (minchans > maxchans) {
+    GST_WARNING_OBJECT (obj, "min_chans %d > max_chans %d (buggy driver?)",
+        minchans, maxchans);
+    maxchans = ai->min_channels;        /* swap */
+    minchans = ai->max_channels;
+  }
+
+  /* limit to something sensible */
+  if (minchans < GST_OSS4_MIN_CHANNELS)
+    minchans = GST_OSS4_MIN_CHANNELS;
+  if (maxchans > GST_OSS4_MAX_CHANNELS)
+    maxchans = GST_OSS4_MAX_CHANNELS;
+
+  if (maxchans < GST_OSS4_MIN_CHANNELS) {
+    GST_WARNING_OBJECT (obj, "max_chans < %d, which makes no sense",
+        GST_OSS4_MIN_CHANNELS);
+    gst_caps_unref (in_caps);
+    return NULL;
+  }
+
+  GST_LOG_OBJECT (obj, "min_channels %d, max_channels %d (originally: %d, %d)",
+      minchans, maxchans, ai->min_channels, ai->max_channels);
+
+  chan_caps = gst_caps_new_empty ();
+
+  /* first do the simple cases: mono + stereo (channel layout implied) */
+  if (minchans == 1 && maxchans == 1)
+    s = gst_structure_new ("x", "channels", G_TYPE_INT, 1, NULL);
+  else if (minchans == 2 && maxchans >= 2)
+    s = gst_structure_new ("x", "channels", G_TYPE_INT, 2, NULL);
+  else if (minchans == 1 && maxchans >= 2)
+    s = gst_structure_new ("x", "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
+  gst_caps_append_structure (chan_caps, s);
+  s = NULL;
+
+  /* TODO: we assume all drivers use a left/right layout for stereo here */
+  if (maxchans <= 2)
+    goto done;
+
+  if (ioctl (fd, SNDCTL_DSP_GET_CHNORDER, &layout) == -1) {
+    GST_WARNING_OBJECT (obj, "couldn't query channel layout, assuming default");
+    layout = CHNORDER_NORMAL;
+  }
+  GST_DEBUG_OBJECT (obj, "channel layout: %08" G_GINT64_MODIFIER "x", layout);
+
+  /* e.g. forced 2.1 layout would be GST_OSS4_CHANNEL_LAYOUT=421 */
+  if (forced_layout != NULL && *forced_layout != '\0') {
+    guint layout_len;
+
+    layout_len = strlen (forced_layout);
+    if (layout_len >= minchans && layout_len <= maxchans) {
+      layout = g_ascii_strtoull (forced_layout, NULL, 16);
+      maxchans = layout_len;
+      GST_DEBUG_OBJECT (obj, "forced channel layout: %08" G_GINT64_MODIFIER "x"
+          " ('%s'), maxchans now %d", layout, forced_layout, maxchans);
+    } else {
+      GST_WARNING_OBJECT (obj, "ignoring forced channel layout: layout has %d "
+          "channel positions but maxchans is %d", layout_len, maxchans);
+    }
+  }
+
+  /* need to advertise channel layouts for anything >2 and <=8 channels */
+  for (c = MAX (3, minchans); c <= MIN (maxchans, 8); c++) {
+    /* "The min_channels and max_channels fields define the limits for the
+     * number of channels. However some devices don't support all channels
+     * within this range. It's possible that the odd values (3, 5, 7, 9, etc).
+     * are not supported. There is currently no way to check for this other
+     * than checking if SNDCTL_DSP_CHANNELS accepts the requested value.
+     * Another approach is trying to avoid using odd number of channels."
+     *
+     * So, we don't know for sure if these odd values are supported:
+     */
+    if ((c == 3 || c == 5 || c == 7) && (c != maxchans)) {
+      GST_LOG_OBJECT (obj, "not adding layout with %d channels", c);
+      continue;
+    }
+
+    s = gst_structure_new ("x", "channels", G_TYPE_INT, c, NULL);
+    gst_oss4_audio_add_channel_layout (obj, layout, c, s);
+    GST_LOG_OBJECT (obj, "c=%u, appending struct %" GST_PTR_FORMAT, c, s);
+    gst_caps_append_structure (chan_caps, s);
+    s = NULL;
+  }
+
+  if (maxchans <= 8)
+    goto done;
+
+  /* for everything >8 channels, CHANNEL_POSITION_NONE is implied. */
+  if (minchans == maxchans || maxchans == 9) {
+    s = gst_structure_new ("x", "channels", G_TYPE_INT, maxchans, NULL);
+  } else {
+    s = gst_structure_new ("x", "channels", GST_TYPE_INT_RANGE,
+        MAX (9, minchans), maxchans, NULL);
+  }
+  gst_caps_append_structure (chan_caps, s);
+  s = NULL;
+
+done:
+
+  GST_LOG_OBJECT (obj, "channel structures: %" GST_PTR_FORMAT, chan_caps);
+
+  out_caps = gst_caps_new_empty ();
+
+  /* combine each structure in the input caps with each channel caps struct */
+  for (i = 0; i < gst_caps_get_size (in_caps); ++i) {
+    const GstStructure *in_s;
+
+    in_s = gst_caps_get_structure (in_caps, i);
+
+    for (j = 0; j < gst_caps_get_size (chan_caps); ++j) {
+      const GstStructure *chan_s;
+      const GValue *val;
+
+      s = gst_structure_copy (in_s);
+      chan_s = gst_caps_get_structure (chan_caps, j);
+      if ((val = gst_structure_get_value (chan_s, "channels")))
+        gst_structure_set_value (s, "channels", val);
+      if ((val = gst_structure_get_value (chan_s, "channel-positions")))
+        gst_structure_set_value (s, "channel-positions", val);
+
+      gst_caps_append_structure (out_caps, s);
+      s = NULL;
+    }
+  }
+
+  gst_caps_unref (in_caps);
+  gst_caps_unref (chan_caps);
+  return out_caps;
+}
+
+GstCaps *
+gst_oss4_audio_probe_caps (GstObject * obj, int fd)
+{
+  oss_audioinfo ai = { 0, };
+  gboolean output;
+  GstCaps *caps;
+  int formats, i;
+
+  output = GST_IS_OSS4_SINK (obj);
+
+  /* -1 = get info for currently open device (fd). This will fail with
+   * OSS build <= 1013 because of a bug in OSS */
+  ai.dev = -1;
+  if (ioctl (fd, SNDCTL_ENGINEINFO, &ai) == -1)
+    goto engineinfo_failed;
+
+  formats = (output) ? ai.oformats : ai.iformats;
+
+  GST_LOG_OBJECT (obj, "%s formats : 0x%08x", (output) ? "out" : "in", formats);
+
+  caps = gst_caps_new_empty ();
+
+  for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
+    if ((formats & fmt_map[i].oss_fmt)) {
+      gst_oss4_append_format_to_caps (fmt_map[i].oss_fmt, caps);
+    }
+  }
+
+  gst_caps_do_simplify (caps);
+  GST_LOG_OBJECT (obj, "formats: %" GST_PTR_FORMAT, caps);
+
+  if (!gst_oss4_audio_detect_rates (obj, &ai, caps))
+    goto detect_rates_failed;
+
+  caps = gst_oss4_audio_detect_channels (obj, fd, &ai, caps);
+  if (caps == NULL)
+    goto detect_channels_failed;
+
+  GST_LOG_OBJECT (obj, "probed caps: %" GST_PTR_FORMAT, caps);
+
+  return caps;
+
+/* ERRORS */
+engineinfo_failed:
+  {
+    GST_WARNING ("ENGINEINFO supported formats probe failed: %s",
+        g_strerror (errno));
+    return NULL;
+  }
+detect_rates_failed:
+  {
+    GST_WARNING_OBJECT (obj, "failed to detect supported sample rates");
+    gst_caps_unref (caps);
+    return NULL;
+  }
+detect_channels_failed:
+  {
+    GST_WARNING_OBJECT (obj, "failed to detect supported channels");
+    gst_caps_unref (caps);
+    return NULL;
+  }
+}
+
+GstCaps *
+gst_oss4_audio_get_template_caps (void)
+{
+  GstCaps *caps;
+  gint i;
+
+  caps = gst_caps_new_empty ();
+
+  for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
+    gst_oss4_append_format_to_caps (fmt_map[i].oss_fmt, caps);
+  }
+
+  gst_caps_do_simplify (caps);
+
+  for (i = 0; i < gst_caps_get_size (caps); ++i) {
+    GstStructure *s;
+
+    s = gst_caps_get_structure (caps, i);
+    gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, GST_OSS4_MIN_SAMPLE_RATE,
+        GST_OSS4_MAX_SAMPLE_RATE, "channels", GST_TYPE_INT_RANGE,
+        GST_OSS4_MIN_CHANNELS, GST_OSS4_MAX_CHANNELS, NULL);
+  }
+
+  return caps;
+}
+
+static gint
+gst_oss4_audio_ilog2 (gint x)
+{
+  /* well... hacker's delight explains... */
+  x = x | (x >> 1);
+  x = x | (x >> 2);
+  x = x | (x >> 4);
+  x = x | (x >> 8);
+  x = x | (x >> 16);
+  x = x - ((x >> 1) & 0x55555555);
+  x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
+  x = (x + (x >> 4)) & 0x0f0f0f0f;
+  x = x + (x >> 8);
+  x = x + (x >> 16);
+  return (x & 0x0000003f) - 1;
+}
+
+/* called by gst_oss4_sink_prepare() and gst_oss4_source_prepare() */
+gboolean
+gst_oss4_audio_set_format (GstObject * obj, int fd, GstRingBufferSpec * spec)
+{
+  struct audio_buf_info info = { 0, };
+  int fmt, chans, rate, fragsize;
+
+  fmt = gst_oss4_audio_get_oss_format (spec->format);
+  if (fmt == 0)
+    goto wrong_format;
+
+  if (spec->type == GST_BUFTYPE_LINEAR && spec->width != 32 &&
+      spec->width != 24 && spec->width != 16 && spec->width != 8) {
+    goto dodgy_width;
+  }
+
+  /* format */
+  GST_LOG_OBJECT (obj, "setting format: %d", fmt);
+  if (ioctl (fd, SNDCTL_DSP_SETFMT, &fmt) == -1)
+    goto set_format_failed;
+
+  /* channels */
+  GST_LOG_OBJECT (obj, "setting channels: %d", spec->channels);
+  chans = spec->channels;
+  if (ioctl (fd, SNDCTL_DSP_CHANNELS, &chans) == -1)
+    goto set_channels_failed;
+
+  /* rate */
+  GST_LOG_OBJECT (obj, "setting rate: %d", spec->rate);
+  rate = spec->rate;
+  if (ioctl (fd, SNDCTL_DSP_SPEED, &rate) == -1)
+    goto set_rate_failed;
+
+  GST_DEBUG_OBJECT (obj, "effective format   : %d", fmt);
+  GST_DEBUG_OBJECT (obj, "effective channels : %d", chans);
+  GST_DEBUG_OBJECT (obj, "effective rate     : %d", rate);
+
+  /* make sure format, channels, and rate are the ones we requested */
+  if (fmt != gst_oss4_audio_get_oss_format (spec->format) ||
+      chans != spec->channels || rate != spec->rate) {
+    /* This shouldn't happen, but hey */
+    goto format_not_what_was_requested;
+  }
+
+  /* CHECKME: maybe we should just leave the fragsize alone? (tpm) */
+  fragsize = gst_oss4_audio_ilog2 (spec->segsize);
+  fragsize = ((spec->segtotal & 0x7fff) << 16) | fragsize;
+  GST_DEBUG_OBJECT (obj, "setting segsize: %d, segtotal: %d, value: %08x",
+      spec->segsize, spec->segtotal, fragsize);
+
+  /* we could also use the new SNDCTL_DSP_POLICY if there's something in
+   * particular we're trying to achieve here */
+  if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &fragsize) == -1)
+    goto set_fragsize_failed;
+
+  if (GST_IS_OSS4_SOURCE (obj)) {
+    if (ioctl (fd, SNDCTL_DSP_GETISPACE, &info) == -1)
+      goto get_ispace_failed;
+  } else {
+    if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &info) == -1)
+      goto get_ospace_failed;
+  }
+
+  spec->segsize = info.fragsize;
+  spec->segtotal = info.fragstotal;
+
+  spec->bytes_per_sample = (spec->width / 8) * spec->channels;
+
+  GST_DEBUG_OBJECT (obj, "got segsize: %d, segtotal: %d, value: %08x",
+      spec->segsize, spec->segtotal, fragsize);
+
+  return TRUE;
+
+/* ERRORS */
+wrong_format:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("Unable to get format %d", spec->format));
+    return FALSE;
+  }
+dodgy_width:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("unexpected width %d", spec->width));
+    return FALSE;
+  }
+set_format_failed:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("DSP_SETFMT(%d) failed: %s", fmt, g_strerror (errno)));
+    return FALSE;
+  }
+set_channels_failed:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("DSP_CHANNELS(%d) failed: %s", chans, g_strerror (errno)));
+    return FALSE;
+  }
+set_rate_failed:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("DSP_SPEED(%d) failed: %s", rate, g_strerror (errno)));
+    return FALSE;
+  }
+set_fragsize_failed:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("DSP_SETFRAGMENT(%d) failed: %s", fragsize, g_strerror (errno)));
+    return FALSE;
+  }
+get_ospace_failed:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("DSP_GETOSPACE failed: %s", g_strerror (errno)));
+    return FALSE;
+  }
+get_ispace_failed:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("DSP_GETISPACE failed: %s", g_strerror (errno)));
+    return FALSE;
+  }
+format_not_what_was_requested:
+  {
+    GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+        ("Format actually configured wasn't the one we requested. This is "
+            "probably either a bug in the driver or in the format probing code."));
+    return FALSE;
+  }
+}
+
+int
+gst_oss4_audio_get_version (GstObject * obj, int fd)
+{
+  gint ver = 0;
+
+  /* we use the old ioctl here on purpose instead of SNDCTL_SYSINFO */
+  if (ioctl (fd, OSS_GETVERSION, &ver) < 0) {
+    GST_LOG_OBJECT (obj, "OSS_GETVERSION failed: %s", g_strerror (errno));
+    return -1;
+  }
+  GST_LOG_OBJECT (obj, "OSS version: 0x%08x", ver);
+  return ver;
+}
+
+gboolean
+gst_oss4_audio_check_version (GstObject * obj, int fd)
+{
+  return (gst_oss4_audio_get_version (obj, fd) >= GST_MIN_OSS4_VERSION);
+}
+
+gchar *
+gst_oss4_audio_find_device (GstObject * oss)
+{
+  GValueArray *arr;
+  gchar *ret = NULL;
+
+  arr = gst_property_probe_probe_and_get_values_name (GST_PROPERTY_PROBE (oss),
+      "device");
+
+  if (arr != NULL) {
+    if (arr->n_values > 0) {
+      const GValue *val;
+
+      val = g_value_array_get_nth (arr, 0);
+      ret = g_value_dup_string (val);
+    }
+    g_value_array_free (arr);
+  }
+
+  GST_LOG_OBJECT (oss, "first device found: %s", GST_STR_NULL (ret));
+
+  return ret;
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  gint rank;
+
+  GST_DEBUG_CATEGORY_INIT (oss4sink_debug, "oss4sink", 0, "OSS4 audio sink");
+  GST_DEBUG_CATEGORY_INIT (oss4src_debug, "oss4src", 0, "OSS4 audio src");
+  GST_DEBUG_CATEGORY_INIT (oss4mixer_debug, "oss4mixer", 0, "OSS4 mixer");
+  GST_DEBUG_CATEGORY_INIT (oss4_debug, "oss4", 0, "OSS4 plugin");
+
+#ifdef ENABLE_NLS
+  GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
+      LOCALEDIR);
+  bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+#endif
+
+  /* we want a higher rank than the legacy OSS elements have now */
+  rank = GST_RANK_SECONDARY + 1;
+
+  if (!gst_element_register (plugin, "oss4sink", rank, GST_TYPE_OSS4_SINK) ||
+      !gst_element_register (plugin, "oss4src", rank, GST_TYPE_OSS4_SOURCE) ||
+      !gst_element_register (plugin, "oss4mixer", rank, GST_TYPE_OSS4_MIXER)) {
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "oss4",
+    "Open Sound System (OSS) version 4 support for GStreamer",
+    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/sys/oss4/oss4-audio.h b/sys/oss4/oss4-audio.h
new file mode 100644
index 00000000..d2203641
--- /dev/null
+++ b/sys/oss4/oss4-audio.h
@@ -0,0 +1,43 @@
+/* GStreamer OSS4 audio plugin
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef GST_OSS4_AUDIO_H
+#define GST_OSS4_AUDIO_H_
+
+#include <gst/gst.h>
+#include <gst/audio/gstringbuffer.h>
+
+/* This is the minimum version we require */
+#define GST_MIN_OSS4_VERSION  0x040003
+
+int       gst_oss4_audio_get_version (GstObject * obj, int fd);
+
+gboolean  gst_oss4_audio_check_version (GstObject * obj, int fd);
+
+GstCaps * gst_oss4_audio_probe_caps  (GstObject * obj, int fd);
+
+gboolean  gst_oss4_audio_set_format  (GstObject * obj, int fd, GstRingBufferSpec * spec);
+
+GstCaps * gst_oss4_audio_get_template_caps (void);
+
+gchar   * gst_oss4_audio_find_device (GstObject * oss);
+
+#endif /* GST_OSS4_AUDIO_H */
+
+
diff --git a/sys/oss4/oss4-mixer-enum.c b/sys/oss4/oss4-mixer-enum.c
new file mode 100644
index 00000000..edd0d7bb
--- /dev/null
+++ b/sys/oss4/oss4-mixer-enum.c
@@ -0,0 +1,269 @@
+/* GStreamer OSS4 mixer enumeration control
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/* An 'enum' in gnome-volume-control / GstMixer is represented by a
+ * GstMixerOptions object
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-mixer.h"
+#include "oss4-mixer-enum.h"
+#include "oss4-soundcard.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+static GList *gst_oss4_mixer_enum_get_values (GstMixerOptions * options);
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4MixerEnum, gst_oss4_mixer_enum, GST_TYPE_MIXER_OPTIONS);
+
+static void
+gst_oss4_mixer_enum_init (GstOss4MixerEnum * e)
+{
+  e->need_update = TRUE;
+}
+
+static void
+gst_oss4_mixer_enum_dispose (GObject * obj)
+{
+  GstMixerOptions *options = GST_MIXER_OPTIONS (obj);
+
+  /* our list is a flat list with constant strings, but the GstMixerOptions
+   * dispose will try to g_free the contained strings, so clean up the list
+   * before chaining up to GstMixerOptions */
+  g_list_free (options->values);
+  options->values = NULL;
+
+  G_OBJECT_CLASS (gst_oss4_mixer_enum_parent_class)->dispose (obj);
+}
+
+static void
+gst_oss4_mixer_enum_class_init (GstOss4MixerEnumClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstMixerOptionsClass *mixeroptions_class = (GstMixerOptionsClass *) klass;
+
+  gobject_class->dispose = gst_oss4_mixer_enum_dispose;
+  mixeroptions_class->get_values = gst_oss4_mixer_enum_get_values;
+}
+
+static GList *
+gst_oss4_mixer_enum_get_values_locked (GstMixerOptions * options)
+{
+  GstOss4MixerEnum *e = GST_OSS4_MIXER_ENUM_CAST (options);
+  GList *oldlist, *list = NULL;
+  int i;
+
+  /* if current list of values is empty, update/re-check in any case */
+  if (!e->need_update && options->values != NULL)
+    return options->values;
+
+  GST_LOG_OBJECT (e, "updating available values for %s", e->mc->mixext.extname);
+
+  for (i = 0; i < e->mc->mixext.maxvalue; ++i) {
+    if (MIXEXT_ENUM_IS_AVAILABLE (e->mc->mixext, i)) {
+      const gchar *s;
+
+      s = g_quark_to_string (e->mc->enum_vals[i]);
+      GST_LOG_OBJECT (e, "option '%s' is available", s);
+      list = g_list_prepend (list, (gpointer) s);
+    } else {
+      GST_LOG_OBJECT (e, "option '%s' is currently not available");
+    }
+  }
+
+  list = g_list_reverse (list);
+
+  /* this is not thread-safe, but then the entire GstMixer API isn't really,
+   * since we return foo->list and not a copy and don't take any locks, so
+   * not much we can do here but pray; we're usually either called from _new()
+   * or from within _get_values() though, so it should be okay. We could use
+   * atomic ops here, but I'm not sure how much more that really buys us.*/
+  oldlist = options->values;    /* keep window small */
+  options->values = list;
+  g_list_free (oldlist);
+
+  e->need_update = FALSE;
+
+  return options->values;
+}
+
+static GList *
+gst_oss4_mixer_enum_get_values (GstMixerOptions * options)
+{
+  GstOss4MixerEnum *e = GST_OSS4_MIXER_ENUM (options);
+  GList *list;
+
+  /* we take the lock here mostly to serialise ioctls with the watch thread */
+  GST_OBJECT_LOCK (e->mixer);
+
+  list = gst_oss4_mixer_enum_get_values_locked (options);
+
+  GST_OBJECT_UNLOCK (e->mixer);
+
+  return list;
+}
+
+static const gchar *
+gst_oss4_mixer_enum_get_current_value (GstOss4MixerEnum * e)
+{
+  const gchar *cur_val = NULL;
+
+  if (e->mc->enum_vals != NULL && e->mc->last_val < e->mc->mixext.maxvalue) {
+    cur_val = g_quark_to_string (e->mc->enum_vals[e->mc->last_val]);
+  }
+
+  return cur_val;
+}
+
+static gboolean
+gst_oss4_mixer_enum_update_current (GstOss4MixerEnum * e)
+{
+  int cur = -1;
+
+  if (!gst_oss4_mixer_get_control_val (e->mixer, e->mc, &cur))
+    return FALSE;
+
+  if (cur < 0 || cur >= e->mc->mixext.maxvalue) {
+    GST_WARNING_OBJECT (e, "read value %d out of bounds [0-%d]", cur,
+        e->mc->mixext.maxvalue - 1);
+    e->mc->last_val = 0;
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_enum_set_option (GstOss4MixerEnum * e, const gchar * value)
+{
+  GQuark q;
+  int i;
+
+  q = g_quark_try_string (value);
+  if (q == 0) {
+    GST_WARNING_OBJECT (e, "unknown option '%s'", value);
+    return FALSE;
+  }
+
+  for (i = 0; i < e->mc->mixext.maxvalue; ++i) {
+    if (q == e->mc->enum_vals[i])
+      break;
+  }
+
+  if (i >= e->mc->mixext.maxvalue) {
+    GST_WARNING_OBJECT (e, "option '%s' is not valid for this control", value);
+    return FALSE;
+  }
+
+  GST_LOG_OBJECT (e, "option '%s' = %d", value, i);
+
+  if (!MIXEXT_ENUM_IS_AVAILABLE (e->mc->mixext, i)) {
+    GST_WARNING_OBJECT (e, "option '%s' is not selectable currently", value);
+    return FALSE;
+  }
+
+  if (!gst_oss4_mixer_set_control_val (e->mixer, e->mc, i)) {
+    GST_WARNING_OBJECT (e, "could not set option '%s' (%d)", value, i);
+    return FALSE;
+  }
+
+  /* and re-read current value with sanity checks (or could just assign here) */
+  gst_oss4_mixer_enum_update_current (e);
+
+  return TRUE;
+}
+
+const gchar *
+gst_oss4_mixer_enum_get_option (GstOss4MixerEnum * e)
+{
+  const gchar *cur_str = NULL;
+
+  if (!gst_oss4_mixer_enum_update_current (e)) {
+    GST_WARNING_OBJECT (e, "failed to read current value");
+    return NULL;
+  }
+
+  cur_str = gst_oss4_mixer_enum_get_current_value (e);
+  GST_LOG_OBJECT (e, "%s (%d)", GST_STR_NULL (cur_str), e->mc->last_val);
+  return cur_str;
+}
+
+GstMixerTrack *
+gst_oss4_mixer_enum_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc)
+{
+  GstOss4MixerEnum *e;
+  GstMixerTrack *track;
+
+  e = g_object_new (GST_TYPE_OSS4_MIXER_ENUM, "untranslated-label",
+      mc->mixext.extname, NULL);
+  e->mixer = mixer;
+  e->mc = mc;
+
+  track = GST_MIXER_TRACK (e);
+
+  /* caller will set track->label and track->flags */
+
+  track->num_channels = 0;
+  track->min_volume = 0;
+  track->max_volume = 0;
+
+  (void) gst_oss4_mixer_enum_get_values_locked (GST_MIXER_OPTIONS (track));
+
+  if (!gst_oss4_mixer_enum_update_current (e)) {
+    GST_WARNING_OBJECT (track, "failed to read current value, returning NULL");
+    g_object_unref (track);
+    track = NULL;
+  }
+
+  GST_LOG_OBJECT (e, "current value: %d (%s)", e->mc->last_val,
+      gst_oss4_mixer_enum_get_current_value (e));
+
+  return track;
+}
+
+/* This is called from the watch thread */
+void
+gst_oss4_mixer_enum_process_change_unlocked (GstMixerTrack * track)
+{
+  GstOss4MixerEnum *e = GST_OSS4_MIXER_ENUM_CAST (track);
+
+  gchar *cur;
+
+  if (!e->mc->changed && !e->mc->list_changed)
+    return;
+
+  if (e->mc->list_changed) {
+    gst_mixer_options_list_changed (GST_MIXER (e->mixer),
+        GST_MIXER_OPTIONS (e));
+  }
+
+  GST_OBJECT_LOCK (e->mixer);
+  cur = (gchar *) gst_oss4_mixer_enum_get_current_value (e);
+  GST_OBJECT_UNLOCK (e->mixer);
+
+  gst_mixer_option_changed (GST_MIXER (e->mixer), GST_MIXER_OPTIONS (e), cur);
+}
diff --git a/sys/oss4/oss4-mixer-enum.h b/sys/oss4/oss4-mixer-enum.h
new file mode 100644
index 00000000..9fd81669
--- /dev/null
+++ b/sys/oss4/oss4-mixer-enum.h
@@ -0,0 +1,67 @@
+/* GStreamer OSS4 mixer on/off enum control
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef GST_OSS4_MIXER_ENUM_H
+#define GST_OSS4_MIXER_ENUM_H
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "oss4-mixer.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_MIXER_ENUM            (gst_oss4_mixer_enum_get_type())
+#define GST_OSS4_MIXER_ENUM(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER_ENUM,GstOss4MixerEnum))
+#define GST_OSS4_MIXER_ENUM_CAST(obj)       ((GstOss4MixerEnum *)(obj))
+#define GST_OSS4_MIXER_ENUM_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER_ENUM,GstOss4MixerEnumClass))
+#define GST_IS_OSS4_MIXER_ENUM(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER_ENUM))
+#define GST_IS_OSS4_MIXER_ENUM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER_ENUM))
+
+typedef struct _GstOss4MixerEnum GstOss4MixerEnum;
+typedef struct _GstOss4MixerEnumClass GstOss4MixerEnumClass;
+
+struct _GstOss4MixerEnum {
+  GstMixerOptions  mixer_option;
+
+  GstOss4MixerControl  * mc;
+  GstOss4Mixer         * mixer;      /* the mixer we belong to (no ref taken) */
+
+  gboolean               need_update;
+};
+
+struct _GstOss4MixerEnumClass {
+  GstMixerOptionsClass mixer_option_class;
+};
+
+GType           gst_oss4_mixer_enum_get_type (void);
+
+gboolean        gst_oss4_mixer_enum_set_option (GstOss4MixerEnum * e, const gchar * value);
+
+const gchar   * gst_oss4_mixer_enum_get_option (GstOss4MixerEnum * e);
+
+GstMixerTrack * gst_oss4_mixer_enum_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc);
+
+void            gst_oss4_mixer_enum_process_change_unlocked (GstMixerTrack * track);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_MIXER_ENUM_H */
+
+
diff --git a/sys/oss4/oss4-mixer-slider.c b/sys/oss4/oss4-mixer-slider.c
new file mode 100644
index 00000000..a56dcae8
--- /dev/null
+++ b/sys/oss4/oss4-mixer-slider.c
@@ -0,0 +1,296 @@
+/* GStreamer OSS4 mixer slider control
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/* A 'slider' in gnome-volume-control / GstMixer is represented by a
+ * GstMixerTrack with one or more channels.
+ *
+ * A slider should be either flagged as INPUT or OUTPUT (mostly because of
+ * gnome-volume-control being littered with g_asserts for everything it doesn't
+ * expect).
+ *
+ * From mixertrack.h:
+ * "Input tracks can have 'recording' enabled, which means that any input will
+ * be hearable into the speakers that are attached to the output. Mute is
+ * obvious."
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-mixer-slider.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4MixerSlider, gst_oss4_mixer_slider, GST_TYPE_MIXER_TRACK);
+
+static void
+gst_oss4_mixer_slider_class_init (GstOss4MixerSliderClass * klass)
+{
+  /* nothing to do here */
+}
+
+static void
+gst_oss4_mixer_slider_init (GstOss4MixerSlider * s)
+{
+  /* nothing to do here */
+}
+
+static int
+gst_oss4_mixer_slider_pack_volume (GstOss4MixerSlider * s, const gint * volumes)
+{
+  int val = 0;
+
+  switch (s->mc->mixext.type) {
+    case MIXT_MONOSLIDER:
+    case MIXT_MONOSLIDER16:
+    case MIXT_SLIDER:
+      val = volumes[0];
+      break;
+    case MIXT_STEREOSLIDER:
+      val = ((volumes[1] & 0xff) << 8) | (volumes[0] & 0xff);
+      break;
+    case MIXT_STEREOSLIDER16:
+      val = ((volumes[1] & 0xffff) << 16) | (volumes[0] & 0xffff);
+      break;
+    default:
+      g_return_val_if_reached (0);
+  }
+  return val;
+}
+
+static void
+gst_oss4_mixer_slider_unpack_volume (GstOss4MixerSlider * s, int v,
+    gint * volumes)
+{
+  guint32 val;                  /* use uint so bitshifting the highest bit works right */
+
+  val = (guint32) v;
+  switch (s->mc->mixext.type) {
+    case MIXT_MONOSLIDER:
+    case MIXT_MONOSLIDER16:
+    case MIXT_SLIDER:
+      volumes[0] = val;
+      break;
+    case MIXT_STEREOSLIDER:
+      volumes[0] = (val & 0x00ff);
+      volumes[1] = (val & 0xff00) >> 8;
+      break;
+    case MIXT_STEREOSLIDER16:
+      volumes[0] = (val & 0x0000ffff);
+      volumes[1] = (val & 0xffff0000) >> 16;
+      break;
+    default:
+      g_return_if_reached ();
+  }
+}
+
+gboolean
+gst_oss4_mixer_slider_get_volume (GstOss4MixerSlider * s, gint * volumes)
+{
+  GstMixerTrack *track = GST_MIXER_TRACK (s);
+  int v = 0;
+
+  /* if we're supposed to be muted, and don't have an actual mute control
+   * (ie. 'simulate' the mute), then just return the volume as saved, not
+   * the actually set volume which is most likely 0 */
+  if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_MUTE) && !s->mc->mute) {
+    volumes[0] = s->volumes[0];
+    if (track->num_channels == 2)
+      volumes[1] = s->volumes[1];
+    return TRUE;
+  }
+
+  if (!gst_oss4_mixer_get_control_val (s->mixer, s->mc, &v))
+    return FALSE;
+
+  gst_oss4_mixer_slider_unpack_volume (s, v, volumes);
+
+  if (track->num_channels > 1) {
+    GST_LOG_OBJECT (s, "volume: left=%d, right=%d", volumes[0], volumes[1]);
+  } else {
+    GST_LOG_OBJECT (s, "volume: mono=%d", volumes[0]);
+  }
+
+  return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_slider_set_volume (GstOss4MixerSlider * s, const gint * volumes)
+{
+  GstMixerTrack *track = GST_MIXER_TRACK (s);
+  int val = 0;
+
+  /* if we're supposed to be muted, and are 'simulating' the mute because
+   * we don't have a mute control, don't actually change the volume, just
+   * save it as the new desired volume for later when we get unmuted again */
+  if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_MUTE) && !s->mc->mute)
+    goto done;
+
+  val = gst_oss4_mixer_slider_pack_volume (s, volumes);
+
+  if (track->num_channels > 1) {
+    GST_LOG_OBJECT (s, "left=%d, right=%d", volumes[0], volumes[1]);
+  } else {
+    GST_LOG_OBJECT (s, "mono=%d", volumes[0]);
+  }
+
+  if (!gst_oss4_mixer_set_control_val (s->mixer, s->mc, val))
+    return FALSE;
+
+done:
+
+  s->volumes[0] = volumes[0];
+  if (track->num_channels == 2)
+    s->volumes[1] = volumes[1];
+
+  return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_slider_set_record (GstOss4MixerSlider * s, gboolean record)
+{
+  /* There doesn't seem to be a way to do this using the OSS4 mixer API, so
+   * just do nothing here for now. */
+  return FALSE;
+}
+
+gboolean
+gst_oss4_mixer_slider_set_mute (GstOss4MixerSlider * s, gboolean mute)
+{
+  GstMixerTrack *track = GST_MIXER_TRACK (s);
+  gboolean ret;
+
+  /* if we don't have a mute control, simulate mute (which is a bit broken,
+   * since we can't differentiate between capture/playback volume etc., so
+   * we just assume that setting the volume to 0 would be the same as muting
+   * this control) */
+  if (s->mc->mute == NULL) {
+    int volume;
+
+    if (mute) {
+      volume = 0;
+    } else {
+      volume = gst_oss4_mixer_slider_pack_volume (s, s->volumes);
+    }
+    ret = gst_oss4_mixer_set_control_val (s->mixer, s->mc, volume);
+  } else {
+    ret = gst_oss4_mixer_set_control_val (s->mixer, s->mc->mute, !!mute);
+  }
+
+  if (mute) {
+    track->flags |= GST_MIXER_TRACK_MUTE;
+  } else {
+    track->flags &= ~GST_MIXER_TRACK_MUTE;
+  }
+
+  return FALSE;
+}
+
+GstMixerTrack *
+gst_oss4_mixer_slider_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc)
+{
+  GstOss4MixerSlider *s;
+  GstMixerTrack *track;
+  gint volumes[2] = { 0, };
+
+  s = g_object_new (GST_TYPE_OSS4_MIXER_SLIDER, "untranslated-label",
+      mc->mixext.extname, NULL);
+
+  track = GST_MIXER_TRACK (s);
+
+  /* caller will set track->label and track->flags */
+
+  s->mc = mc;
+  s->mixer = mixer;
+
+  /* we don't do value scaling but just present a scale of 0-maxvalue */
+  track->min_volume = 0;
+  track->max_volume = mc->mixext.maxvalue;
+
+  switch (mc->mixext.type) {
+    case MIXT_MONOSLIDER:
+    case MIXT_MONOSLIDER16:
+    case MIXT_SLIDER:
+      track->num_channels = 1;
+      break;
+    case MIXT_STEREOSLIDER:
+    case MIXT_STEREOSLIDER16:
+      track->num_channels = 2;
+      break;
+    default:
+      g_return_val_if_reached (NULL);
+  }
+
+  GST_LOG_OBJECT (track, "min=%d, max=%d, channels=%d", track->min_volume,
+      track->max_volume, track->num_channels);
+
+  if (!gst_oss4_mixer_slider_get_volume (s, volumes)) {
+    GST_WARNING_OBJECT (track, "failed to read volume, returning NULL");
+    g_object_unref (track);
+    track = NULL;
+  }
+
+  return track;
+}
+
+/* This is called from the watch thread */
+void
+gst_oss4_mixer_slider_process_change_unlocked (GstMixerTrack * track)
+{
+  GstOss4MixerSlider *s = GST_OSS4_MIXER_SLIDER_CAST (track);
+
+  if (s->mc->mute != NULL && s->mc->mute->changed) {
+    gst_mixer_mute_toggled (GST_MIXER (s->mixer), track,
+        !!s->mc->mute->last_val);
+  } else {
+    /* nothing to do here, since we don't/can't easily implement the record
+     * flag */
+  }
+
+  if (s->mc->changed) {
+    gint volumes[2] = { 0, 0 };
+
+    gst_oss4_mixer_slider_unpack_volume (s, s->mc->last_val, volumes);
+
+    /* if we 'simulate' the mute, update flag when the volume changes */
+    if (s->mc->mute == NULL) {
+      if (volumes[0] == 0 && volumes[1] == 0) {
+        track->flags |= GST_MIXER_TRACK_MUTE;
+      } else {
+        track->flags &= ~GST_MIXER_TRACK_MUTE;
+      }
+    }
+
+    gst_mixer_volume_changed (GST_MIXER (s->mixer), track, volumes);
+  }
+}
diff --git a/sys/oss4/oss4-mixer-slider.h b/sys/oss4/oss4-mixer-slider.h
new file mode 100644
index 00000000..3bee33f8
--- /dev/null
+++ b/sys/oss4/oss4-mixer-slider.h
@@ -0,0 +1,70 @@
+/* GStreamer OSS4 mixer slider control
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef GST_OSS4_MIXER_SLIDER_H
+#define GST_OSS4_MIXER_SLIDER_H
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "oss4-mixer.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_MIXER_SLIDER            (gst_oss4_mixer_slider_get_type())
+#define GST_OSS4_MIXER_SLIDER(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER_SLIDER,GstOss4MixerSlider))
+#define GST_OSS4_MIXER_SLIDER_CAST(obj)       ((GstOss4MixerSlider *)(obj))
+#define GST_OSS4_MIXER_SLIDER_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER_SLIDER,GstOss4MixerSliderClass))
+#define GST_IS_OSS4_MIXER_SLIDER(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER_SLIDER))
+#define GST_IS_OSS4_MIXER_SLIDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER_SLIDER))
+
+typedef struct _GstOss4MixerSlider GstOss4MixerSlider;
+typedef struct _GstOss4MixerSliderClass GstOss4MixerSliderClass;
+
+struct _GstOss4MixerSlider {
+  GstMixerTrack  mixer_track;
+
+  GstOss4MixerControl * mc;
+  GstOss4Mixer        * mixer;      /* the mixer we belong to (no ref taken) */
+  gint                  volumes[2]; /* left/mono, right                      */
+};
+
+struct _GstOss4MixerSliderClass {
+  GstMixerTrackClass mixer_track_class;
+};
+
+GType           gst_oss4_mixer_slider_get_type (void);
+
+GstMixerTrack * gst_oss4_mixer_slider_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc);
+
+gboolean        gst_oss4_mixer_slider_get_volume (GstOss4MixerSlider * s, gint * volumes);
+
+gboolean        gst_oss4_mixer_slider_set_volume (GstOss4MixerSlider * s, const gint * volumes);
+
+gboolean        gst_oss4_mixer_slider_set_record (GstOss4MixerSlider * s, gboolean record);
+
+gboolean        gst_oss4_mixer_slider_set_mute (GstOss4MixerSlider * s, gboolean mute);
+
+void            gst_oss4_mixer_slider_process_change_unlocked (GstMixerTrack * track);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_MIXER_SLIDER_H */
+
+
diff --git a/sys/oss4/oss4-mixer-switch.c b/sys/oss4/oss4-mixer-switch.c
new file mode 100644
index 00000000..403abbc5
--- /dev/null
+++ b/sys/oss4/oss4-mixer-switch.c
@@ -0,0 +1,169 @@
+/* GStreamer OSS4 mixer on/off switch control
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/* A simple ON/OFF 'switch' in gnome-volume-control / GstMixer is represented
+ * by a GstMixerTrack with no channels.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-mixer-switch.h"
+#include "oss4-soundcard.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4MixerSwitch, gst_oss4_mixer_switch, GST_TYPE_MIXER_TRACK);
+
+static void
+gst_oss4_mixer_switch_class_init (GstOss4MixerSwitchClass * klass)
+{
+  /* nothing to do here */
+}
+
+static void
+gst_oss4_mixer_switch_init (GstOss4MixerSwitch * s)
+{
+  /* nothing to do here */
+}
+
+static GstMixerTrackFlags
+gst_oss4_mixer_switch_get_switch_flag (GstMixerTrack * track)
+{
+  if ((track->flags & GST_MIXER_TRACK_INPUT)) {
+    return GST_MIXER_TRACK_RECORD;
+  } else if ((track->flags & GST_MIXER_TRACK_OUTPUT)) {
+    return GST_MIXER_TRACK_MUTE;
+  } else {
+    GST_ERROR_OBJECT (track, "switch neither input nor output track!?");
+  }
+  return 0;
+}
+
+gboolean
+gst_oss4_mixer_switch_set (GstOss4MixerSwitch * s, gboolean enabled)
+{
+  GstMixerTrackFlags switch_flag;
+  GstMixerTrack *track;
+  int newval;
+
+  track = GST_MIXER_TRACK (s);
+  switch_flag = gst_oss4_mixer_switch_get_switch_flag (track);
+
+  newval = (enabled) ? 1 : 0;
+
+  if (!!newval == !!(track->flags & switch_flag)) {
+    GST_LOG_OBJECT (s, "switch is already %d, doing nothing", newval);
+    return TRUE;
+  }
+
+  if (!gst_oss4_mixer_set_control_val (s->mixer, s->mc, newval)) {
+    GST_WARNING_OBJECT (s, "could not set switch to %d", newval);
+    return FALSE;
+  }
+
+  if (newval) {
+    track->flags |= switch_flag;
+  } else {
+    track->flags &= ~switch_flag;
+  }
+
+  GST_LOG_OBJECT (s, "set switch to %d", newval);
+
+  return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_switch_get (GstOss4MixerSwitch * s, gboolean * enabled)
+{
+  GstMixerTrackFlags switch_flag;
+  GstMixerTrack *track;
+  int val = -1;
+
+  track = GST_MIXER_TRACK (s);
+  switch_flag = gst_oss4_mixer_switch_get_switch_flag (track);
+
+  if (!gst_oss4_mixer_get_control_val (s->mixer, s->mc, &val) || val < 0) {
+    GST_WARNING_OBJECT (s, "could not get switch state");
+    return FALSE;
+  }
+
+  *enabled = (val != 0);
+
+  if (!!val != !!(track->flags & switch_flag)) {
+    GST_INFO_OBJECT (s, "updating inconsistent switch state to %d", !!val);
+    if (*enabled) {
+      track->flags |= switch_flag;
+    } else {
+      track->flags &= ~switch_flag;
+    }
+  }
+
+
+  return TRUE;
+}
+
+GstMixerTrack *
+gst_oss4_mixer_switch_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc)
+{
+  GstOss4MixerSwitch *s;
+  GstMixerTrack *track;
+  int cur = -1;
+
+  s = g_object_new (GST_TYPE_OSS4_MIXER_SWITCH, "untranslated-label",
+      mc->mixext.extname, NULL);
+
+  s->mixer = mixer;
+  s->mc = mc;
+
+  track = GST_MIXER_TRACK (s);
+
+  /* caller will set track->label and track->flags */
+
+  track->num_channels = 0;
+  track->min_volume = 0;
+  track->max_volume = 0;
+
+  if (!gst_oss4_mixer_get_control_val (s->mixer, s->mc, &cur) || cur < 0)
+    return NULL;
+
+  return track;
+}
+
+/* This is called from the watch thread */
+void
+gst_oss4_mixer_switch_process_change_unlocked (GstMixerTrack * track)
+{
+  GstOss4MixerSwitch *s = GST_OSS4_MIXER_SWITCH_CAST (track);
+
+  if (!s->mc->changed)
+    return;
+
+  if ((track->flags & GST_MIXER_TRACK_INPUT)) {
+    gst_mixer_record_toggled (GST_MIXER (s->mixer), track, !!s->mc->last_val);
+  } else {
+    gst_mixer_mute_toggled (GST_MIXER (s->mixer), track, !!s->mc->last_val);
+  }
+}
diff --git a/sys/oss4/oss4-mixer-switch.h b/sys/oss4/oss4-mixer-switch.h
new file mode 100644
index 00000000..a8ab83ff
--- /dev/null
+++ b/sys/oss4/oss4-mixer-switch.h
@@ -0,0 +1,65 @@
+/* GStreamer OSS4 mixer on/off switch control
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef GST_OSS4_MIXER_SWITCH_H
+#define GST_OSS4_MIXER_SWITCH_H
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "oss4-mixer.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_MIXER_SWITCH            (gst_oss4_mixer_switch_get_type())
+#define GST_OSS4_MIXER_SWITCH(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER_SWITCH,GstOss4MixerSwitch))
+#define GST_OSS4_MIXER_SWITCH_CAST(obj)       ((GstOss4MixerSwitch *)(obj))
+#define GST_OSS4_MIXER_SWITCH_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER_SWITCH,GstOss4MixerSwitchClass))
+#define GST_IS_OSS4_MIXER_SWITCH(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER_SWITCH))
+#define GST_IS_OSS4_MIXER_SWITCH_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER_SWITCH))
+
+typedef struct _GstOss4MixerSwitch GstOss4MixerSwitch;
+typedef struct _GstOss4MixerSwitchClass GstOss4MixerSwitchClass;
+
+struct _GstOss4MixerSwitch {
+  GstMixerTrack          mixer_track;
+
+  GstOss4MixerControl  * mc;
+  GstOss4Mixer         * mixer;      /* the mixer we belong to (no ref taken) */
+};
+
+struct _GstOss4MixerSwitchClass {
+  GstMixerTrackClass mixer_track_class;
+};
+
+GType            gst_oss4_mixer_switch_get_type (void);
+
+gboolean         gst_oss4_mixer_switch_set (GstOss4MixerSwitch * s, gboolean enabled);
+
+gboolean         gst_oss4_mixer_switch_get (GstOss4MixerSwitch * s, gboolean * enabled);
+
+GstMixerTrack  * gst_oss4_mixer_switch_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc);
+
+void             gst_oss4_mixer_switch_process_change_unlocked (GstMixerTrack * track);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_MIXER_SWITCH_H */
+
+
diff --git a/sys/oss4/oss4-mixer.c b/sys/oss4/oss4-mixer.c
new file mode 100644
index 00000000..62e271e5
--- /dev/null
+++ b/sys/oss4/oss4-mixer.c
@@ -0,0 +1,1774 @@
+/* GStreamer OSS4 mixer implementation
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with mixer library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-oss4mixer
+ * @short_description: element to control sound input and output levels with OSS4
+ *
+ * <refsect2>
+ * <para>
+ * This element lets you adjust sound input and output levels with the
+ * Open Sound System (OSS) version 4. It supports the GstMixer interface, which
+ * can be used to obtain a list of available mixer tracks. Set the mixer
+ * element to READY state before using the GstMixer interface on it.
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * oss4mixer can&apos;t be used in a sensible way in gst-launch.
+ * </para>
+ * </refsect2>
+ *
+ * Since: 0.10.7
+ */
+
+/* Note: ioctl calls on the same open mixer device are serialised via
+ * the object lock to make sure we don't do concurrent ioctls from two
+ * different threads (e.g. app thread and mixer watch thread), since that
+ * will probably confuse OSS. */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+
+#include <gst/interfaces/mixer.h>
+#include <gst/gst-i18n-plugin.h>
+
+#include <glib/gprintf.h>
+
+#define NO_LEGACY_MIXER
+
+#include "oss4-audio.h"
+#include "oss4-mixer.h"
+#include "oss4-mixer-enum.h"
+#include "oss4-mixer-slider.h"
+#include "oss4-mixer-switch.h"
+#include "oss4-property-probe.h"
+#include "oss4-soundcard.h"
+
+#define GST_OSS4_MIXER_WATCH_INTERVAL   500     /* in millisecs, ie. 0.5s */
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+#define DEFAULT_DEVICE       NULL
+#define DEFAULT_DEVICE_NAME  NULL
+
+enum
+{
+  PROP_0,
+  PROP_DEVICE,
+  PROP_DEVICE_NAME
+};
+
+static void gst_oss4_mixer_init_interfaces (GType type);
+
+GST_BOILERPLATE_FULL (GstOss4Mixer, gst_oss4_mixer, GstElement,
+    GST_TYPE_ELEMENT, gst_oss4_mixer_init_interfaces);
+
+static GstStateChangeReturn gst_oss4_mixer_change_state (GstElement *
+    element, GstStateChange transition);
+
+static void gst_oss4_mixer_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_oss4_mixer_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_oss4_mixer_finalize (GObject * object);
+
+static gboolean gst_oss4_mixer_open (GstOss4Mixer * mixer,
+    gboolean silent_errors);
+static void gst_oss4_mixer_close (GstOss4Mixer * mixer);
+
+static gboolean gst_oss4_mixer_enum_control_update_enum_list (GstOss4Mixer * m,
+    GstOss4MixerControl * mc);
+
+#ifndef GST_DISABLE_DEBUG
+static const gchar *mixer_ext_type_get_name (gint type);
+static const gchar *mixer_ext_flags_get_string (gint flags);
+#endif
+
+static void
+gst_oss4_mixer_base_init (gpointer klass)
+{
+  gst_element_class_set_details_simple (GST_ELEMENT_CLASS (klass),
+      "OSS v4 Audio Mixer", "Generic/Audio",
+      "Control sound input and output levels with OSS4",
+      "Tim-Philipp Müller <tim centricular net>");
+}
+
+static void
+gst_oss4_mixer_class_init (GstOss4MixerClass * klass)
+{
+  GstElementClass *element_class;
+  GObjectClass *gobject_class;
+
+  element_class = (GstElementClass *) klass;
+  gobject_class = (GObjectClass *) klass;
+
+  gobject_class->finalize = gst_oss4_mixer_finalize;
+  gobject_class->set_property = gst_oss4_mixer_set_property;
+  gobject_class->get_property = gst_oss4_mixer_get_property;
+
+  /**
+   * GstOss4Mixer:device
+   *
+   * OSS4 mixer device (e.g. /dev/oss/hdaudio0/mix0 or /dev/mixerN)
+   *
+   **/
+  g_object_class_install_property (gobject_class, PROP_DEVICE,
+      g_param_spec_string ("device", "Device",
+          "OSS mixer device (e.g. /dev/oss/hdaudio0/mix0 or /dev/mixerN) "
+          "(NULL = use first mixer device found)", DEFAULT_DEVICE,
+          G_PARAM_READWRITE));
+
+  /**
+   * GstOss4Mixer:device-name
+   *
+   * Human-readable name of the sound device. May be NULL if the device is
+   * not open (ie. when the mixer is in NULL state)
+   *
+   **/
+  g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+      g_param_spec_string ("device-name", "Device name",
+          "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+          G_PARAM_READABLE));
+
+  element_class->change_state = GST_DEBUG_FUNCPTR (gst_oss4_mixer_change_state);
+}
+
+static void
+gst_oss4_mixer_finalize (GObject * obj)
+{
+  GstOss4Mixer *mixer = GST_OSS4_MIXER (obj);
+
+  g_free (mixer->device);
+
+  G_OBJECT_CLASS (parent_class)->finalize (obj);
+}
+
+static void
+gst_oss4_mixer_reset (GstOss4Mixer * mixer)
+{
+  mixer->fd = -1;
+  mixer->need_update = TRUE;
+  memset (&mixer->last_mixext, 0, sizeof (oss_mixext));
+}
+
+static void
+gst_oss4_mixer_init (GstOss4Mixer * mixer, GstOss4MixerClass * g_class)
+{
+  mixer->device = g_strdup (DEFAULT_DEVICE);
+  mixer->device_name = NULL;
+
+  gst_oss4_mixer_reset (mixer);
+}
+
+static void
+gst_oss4_mixer_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOss4Mixer *mixer = GST_OSS4_MIXER (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      GST_OBJECT_LOCK (mixer);
+      if (!GST_OSS4_MIXER_IS_OPEN (mixer)) {
+        g_free (mixer->device);
+        mixer->device = g_value_dup_string (value);
+        /* unset any cached device-name */
+        g_free (mixer->device_name);
+        mixer->device_name = NULL;
+      } else {
+        g_warning ("%s: can't change \"device\" property while mixer is open",
+            GST_OBJECT_NAME (mixer));
+      }
+      GST_OBJECT_UNLOCK (mixer);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_oss4_mixer_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstOss4Mixer *mixer = GST_OSS4_MIXER (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      GST_OBJECT_LOCK (mixer);
+      g_value_set_string (value, mixer->device);
+      GST_OBJECT_UNLOCK (mixer);
+      break;
+    case PROP_DEVICE_NAME:
+      GST_OBJECT_LOCK (mixer);
+      /* If device is set, try to retrieve the name even if we're not open */
+      if (mixer->fd == -1 && mixer->device != NULL) {
+        if (gst_oss4_mixer_open (mixer, TRUE)) {
+          g_value_set_string (value, mixer->device_name);
+          gst_oss4_mixer_close (mixer);
+        } else {
+          g_value_set_string (value, mixer->device_name);
+        }
+      } else {
+        g_value_set_string (value, mixer->device_name);
+      }
+      GST_OBJECT_UNLOCK (mixer);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+gst_oss4_mixer_open (GstOss4Mixer * mixer, gboolean silent_errors)
+{
+  struct oss_mixerinfo mi = { 0, };
+  gchar *device;
+
+  g_return_val_if_fail (!GST_OSS4_MIXER_IS_OPEN (mixer), FALSE);
+
+  if (mixer->device)
+    device = g_strdup (mixer->device);
+  else
+    device = gst_oss4_audio_find_device (GST_OBJECT_CAST (mixer));
+
+  /* desperate times, desperate measures */
+  if (device == NULL)
+    device = g_strdup ("/dev/mixer");
+
+  GST_INFO_OBJECT (mixer, "Trying to open OSS4 mixer device '%s'", device);
+
+  mixer->fd = open (device, O_RDWR, 0);
+  if (mixer->fd < 0)
+    goto open_failed;
+
+  /* Make sure it's OSS4. If it's old OSS, let the old ossmixer handle it */
+  if (!gst_oss4_audio_check_version (GST_OBJECT (mixer), mixer->fd))
+    goto legacy_oss;
+
+  GST_INFO_OBJECT (mixer, "Opened mixer device '%s', which is mixer %d",
+      device, mi.dev);
+
+  /* Get device name for currently open fd */
+  mi.dev = -1;
+  if (ioctl (mixer->fd, SNDCTL_MIXERINFO, &mi) == 0) {
+    mixer->modify_counter = mi.modify_counter;
+    if (mi.name[0] != '\0') {
+      mixer->device_name = g_strdup (mi.name);
+    }
+  } else {
+    mixer->modify_counter = 0;
+  }
+
+  if (mixer->device_name == NULL) {
+    mixer->device_name = g_strdup ("Unknown");
+  }
+  GST_INFO_OBJECT (mixer, "device name = '%s'", mixer->device_name);
+
+  mixer->open_device = device;
+
+  return TRUE;
+
+  /* ERRORS */
+open_failed:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (mixer, RESOURCE, OPEN_READ_WRITE,
+          (_("Could not open audio device for mixer control handling.")),
+          GST_ERROR_SYSTEM);
+    }
+    g_free (device);
+    return FALSE;
+  }
+legacy_oss:
+  {
+    gst_oss4_mixer_close (mixer);
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (mixer, RESOURCE, OPEN_READ_WRITE,
+          (_("Could not open audio device for mixer control handling."
+                  "This version of the Open Sound System is not supported by this "
+                  "element.")), ("Try the 'ossmixer' element instead"));
+    }
+    g_free (device);
+    return FALSE;
+  }
+}
+
+static void
+gst_oss4_mixer_control_free (GstOss4MixerControl * mc)
+{
+  g_list_free (mc->children);
+  g_list_free (mc->mute_group);
+  g_free (mc->enum_vals);
+  memset (mc, 0, sizeof (GstOss4MixerControl));
+  g_free (mc);
+}
+
+static void
+gst_oss4_mixer_free_tracks (GstOss4Mixer * mixer)
+{
+  g_list_foreach (mixer->tracks, (GFunc) g_object_unref, NULL);
+  g_list_free (mixer->tracks);
+  mixer->tracks = NULL;
+
+  g_list_foreach (mixer->controls, (GFunc) gst_oss4_mixer_control_free, NULL);
+  g_list_free (mixer->controls);
+  mixer->controls = NULL;
+}
+
+static void
+gst_oss4_mixer_close (GstOss4Mixer * mixer)
+{
+  g_free (mixer->device_name);
+  mixer->device_name = NULL;
+
+  g_free (mixer->open_device);
+  mixer->open_device = NULL;
+
+  gst_oss4_mixer_free_tracks (mixer);
+
+  if (mixer->fd != -1) {
+    close (mixer->fd);
+    mixer->fd = -1;
+  }
+
+  gst_oss4_mixer_reset (mixer);
+}
+
+static void
+gst_oss4_mixer_watch_process_changes (GstOss4Mixer * mixer)
+{
+  GList *c, *t, *tracks = NULL;
+
+  GST_INFO_OBJECT (mixer, "mixer interface or control changed");
+
+  /* this is all with the mixer object lock held */
+
+  /* we go through the list backwards so we can bail out faster when the entire
+   * interface needs to be rebuilt */
+  for (c = g_list_last (mixer->controls); c != NULL; c = c->prev) {
+    GstOss4MixerControl *mc = c->data;
+    oss_mixer_value ossval = { 0, };
+
+    mc->changed = FALSE;
+    mc->list_changed = FALSE;
+
+    /* not interested in controls we don't expose in the mixer interface */
+    if (!mc->used)
+      continue;
+
+    /* don't try to read a value from controls that don't have one */
+    if (mc->mixext.type == MIXT_DEVROOT || mc->mixext.type == MIXT_GROUP)
+      continue;
+
+    /* is this an enum control whose list may change? */
+    if (mc->mixext.type == MIXT_ENUM && mc->enum_version != 0) {
+      if (gst_oss4_mixer_enum_control_update_enum_list (mixer, mc))
+        mc->list_changed = TRUE;
+    }
+
+    ossval.dev = mc->mixext.dev;
+    ossval.ctrl = mc->mixext.ctrl;
+    ossval.timestamp = mc->mixext.timestamp;
+
+    if (ioctl (mixer->fd, SNDCTL_MIX_READ, &ossval) == -1) {
+      if (errno == EIDRM || errno == EFAULT) {
+        GST_DEBUG ("%s has disappeared", mc->mixext.extname);
+        goto mixer_changed;
+      }
+      GST_WARNING_OBJECT (mixer, "MIX_READ failed: %s", g_strerror (errno));
+      /* just ignore, move on to next one */
+      continue;
+    }
+
+    if (ossval.value == mc->last_val) { /* no change */
+      /* GST_LOG_OBJECT (mixer, "%s hasn't changed", mc->mixext.extname); */
+      continue;
+    }
+
+    mc->last_val = ossval.value;
+    GST_LOG_OBJECT (mixer, "%s changed value to %u 0x%08x",
+        mc->mixext.extname, ossval.value, ossval.value);
+    mc->changed = TRUE;
+  }
+
+  /* copy list and take track refs, so we can safely drop the object lock,
+   * which we need to do to be able to post messages on the bus */
+  tracks = g_list_copy (mixer->tracks);
+  g_list_foreach (tracks, (GFunc) g_object_ref, NULL);
+
+  GST_OBJECT_UNLOCK (mixer);
+
+  /* since we don't know (or want to know exactly) which controls belong to
+   * which track, we just go through the tracks one-by-one now and make them
+   * check themselves if any of their controls have changed and which messages
+   * to post on the bus as a result */
+  for (t = tracks; t != NULL; t = t->next) {
+    GstMixerTrack *track = t->data;
+
+    if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+      gst_oss4_mixer_slider_process_change_unlocked (track);
+    } else if (GST_IS_OSS4_MIXER_SWITCH (track)) {
+      gst_oss4_mixer_switch_process_change_unlocked (track);
+    } else if (GST_IS_OSS4_MIXER_ENUM (track)) {
+      gst_oss4_mixer_enum_process_change_unlocked (track);
+    }
+
+    g_object_unref (track);
+  }
+  g_list_free (tracks);
+
+  GST_OBJECT_LOCK (mixer);
+  return;
+
+mixer_changed:
+  {
+    GST_OBJECT_UNLOCK (mixer);
+    gst_mixer_mixer_changed (GST_MIXER (mixer));
+    GST_OBJECT_LOCK (mixer);
+    return;
+  }
+}
+
+/* This thread watches the mixer for changes in a somewhat inefficient way
+ * (running an ioctl every half second or so). This is still better and
+ * cheaper than apps polling all tracks for changes a few times a second
+ * though. Needs more thought. There's probably (hopefully) a way to get
+ * notifications via the fd directly somehow. */
+static gpointer
+gst_oss4_mixer_watch_thread (gpointer thread_data)
+{
+  GstOss4Mixer *mixer = GST_OSS4_MIXER_CAST (thread_data);
+
+  GST_DEBUG_OBJECT (mixer, "watch thread running");
+
+  GST_OBJECT_LOCK (mixer);
+  while (!mixer->watch_shutdown) {
+    oss_mixerinfo mi = { 0, };
+    GTimeVal tv;
+
+    mi.dev = -1;
+    if (ioctl (mixer->fd, SNDCTL_MIXERINFO, &mi) == 0) {
+      if (mixer->modify_counter != mi.modify_counter) {
+        /* GST_LOG ("processing changes"); */
+        gst_oss4_mixer_watch_process_changes (mixer);
+        mixer->modify_counter = mi.modify_counter;
+      } else {
+        /* GST_LOG ("no changes"); */
+      }
+    } else {
+      GST_WARNING_OBJECT (mixer, "MIXERINFO failed: %s", g_strerror (errno));
+    }
+
+    /* we could move the _get_current_time out of the loop and just do the
+     * add in ever iteration, which would be less exact, but who cares */
+    g_get_current_time (&tv);
+    g_time_val_add (&tv, GST_OSS4_MIXER_WATCH_INTERVAL * 1000);
+    g_cond_timed_wait (mixer->watch_cond, GST_OBJECT_GET_LOCK (mixer), &tv);
+  }
+  GST_OBJECT_UNLOCK (mixer);
+
+  GST_DEBUG_OBJECT (mixer, "watch thread done");
+  gst_object_unref (mixer);
+  return NULL;
+}
+
+/* call with object lock held */
+static void
+gst_oss4_mixer_wake_up_watch_task (GstOss4Mixer * mixer)
+{
+  GST_LOG_OBJECT (mixer, "signalling watch thread to wake up");
+  g_cond_signal (mixer->watch_cond);
+}
+
+static void
+gst_oss4_mixer_stop_watch_task (GstOss4Mixer * mixer)
+{
+  if (mixer->watch_thread) {
+    GST_OBJECT_LOCK (mixer);
+    mixer->watch_shutdown = TRUE;
+    GST_LOG_OBJECT (mixer, "signalling watch thread to stop");
+    g_cond_signal (mixer->watch_cond);
+    GST_OBJECT_UNLOCK (mixer);
+    GST_LOG_OBJECT (mixer, "waiting for watch thread to join");
+    g_thread_join (mixer->watch_thread);
+    GST_DEBUG_OBJECT (mixer, "watch thread stopped");
+    mixer->watch_thread = NULL;
+  }
+
+  if (mixer->watch_cond) {
+    g_cond_free (mixer->watch_cond);
+    mixer->watch_cond = NULL;
+  }
+}
+
+static void
+gst_oss4_mixer_start_watch_task (GstOss4Mixer * mixer)
+{
+  GError *err = NULL;
+
+  mixer->watch_cond = g_cond_new ();
+  mixer->watch_shutdown = FALSE;
+
+  mixer->watch_thread = g_thread_create (gst_oss4_mixer_watch_thread,
+      gst_object_ref (mixer), TRUE, &err);
+
+  if (mixer->watch_thread == NULL) {
+    GST_ERROR_OBJECT (mixer, "Could not create watch thread: %s", err->message);
+    g_cond_free (mixer->watch_cond);
+    mixer->watch_cond = NULL;
+    g_error_free (err);
+  }
+}
+
+static GstStateChangeReturn
+gst_oss4_mixer_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+  GstOss4Mixer *mixer = GST_OSS4_MIXER (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (!gst_oss4_mixer_open (mixer, FALSE))
+        return GST_STATE_CHANGE_FAILURE;
+      gst_oss4_mixer_start_watch_task (mixer);
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+  if (ret == GST_STATE_CHANGE_FAILURE)
+    return ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      gst_oss4_mixer_stop_watch_task (mixer);
+      gst_oss4_mixer_close (mixer);
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+/* === GstMixer interface === */
+
+static inline gboolean
+gst_oss4_mixer_contains_track (GstMixer * mixer, GstMixerTrack * track)
+{
+  return (g_list_find (GST_OSS4_MIXER (mixer)->tracks, track) != NULL);
+}
+
+static inline gboolean
+gst_oss4_mixer_contains_options (GstMixer * mixer, GstMixerOptions * options)
+{
+  return (g_list_find (GST_OSS4_MIXER (mixer)->tracks, options) != NULL);
+}
+
+static void
+gst_oss4_mixer_post_mixer_changed_msg (GstOss4Mixer * mixer)
+{
+  /* only post mixer-changed message once */
+  if (!mixer->need_update) {
+    gst_mixer_mixer_changed (GST_MIXER (mixer));
+    mixer->need_update = TRUE;
+  }
+}
+
+/* call with mixer object lock held to serialise ioctl */
+gboolean
+gst_oss4_mixer_get_control_val (GstOss4Mixer * mixer, GstOss4MixerControl * mc,
+    int *val)
+{
+  oss_mixer_value ossval = { 0, };
+
+  if (GST_OBJECT_TRYLOCK (mixer)) {
+    GST_ERROR ("must be called with mixer lock held");
+    GST_OBJECT_UNLOCK (mixer);
+  }
+
+  ossval.dev = mc->mixext.dev;
+  ossval.ctrl = mc->mixext.ctrl;
+  ossval.timestamp = mc->mixext.timestamp;
+
+  if (ioctl (mixer->fd, SNDCTL_MIX_READ, &ossval) == -1) {
+    if (errno == EIDRM) {
+      GST_DEBUG_OBJECT (mixer, "MIX_READ failed: mixer interface has changed");
+      gst_oss4_mixer_post_mixer_changed_msg (mixer);
+    } else {
+      GST_WARNING_OBJECT (mixer, "MIX_READ failed: %s", g_strerror (errno));
+    }
+    *val = 0;
+    mc->last_val = 0;
+    return FALSE;
+  }
+
+  *val = ossval.value;
+  mc->last_val = ossval.value;
+  GST_LOG_OBJECT (mixer, "got value 0x%08x from %s)", *val, mc->mixext.extname);
+  return TRUE;
+}
+
+/* call with mixer object lock held to serialise ioctl */
+gboolean
+gst_oss4_mixer_set_control_val (GstOss4Mixer * mixer, GstOss4MixerControl * mc,
+    int val)
+{
+  oss_mixer_value ossval = { 0, };
+
+  ossval.dev = mc->mixext.dev;
+  ossval.ctrl = mc->mixext.ctrl;
+  ossval.timestamp = mc->mixext.timestamp;
+  ossval.value = val;
+
+  if (GST_OBJECT_TRYLOCK (mixer)) {
+    GST_ERROR ("must be called with mixer lock held");
+    GST_OBJECT_UNLOCK (mixer);
+  }
+
+  if (ioctl (mixer->fd, SNDCTL_MIX_WRITE, &ossval) == -1) {
+    if (errno == EIDRM) {
+      GST_LOG_OBJECT (mixer, "MIX_WRITE failed: mixer interface has changed");
+      gst_oss4_mixer_post_mixer_changed_msg (mixer);
+    } else {
+      GST_WARNING_OBJECT (mixer, "MIX_WRITE failed: %s", g_strerror (errno));
+    }
+    return FALSE;
+  }
+
+  mc->last_val = val;
+  GST_LOG_OBJECT (mixer, "set value 0x%08x on %s", val, mc->mixext.extname);
+  return TRUE;
+}
+
+static gchar *
+gst_oss4_mixer_control_get_pretty_name (GstOss4MixerControl * mc)
+{
+  gchar *name;
+
+#if 0
+  const gchar *name, *u;
+
+  /* "The id field is the original name given by the driver when it called
+   * mixer_ext_create_control. This name can be used by fully featured GUI
+   * mixers. However this name should be downshifted and cut before the last
+   * underscore ("_") to get the proper name. For example mixer control name
+   * created as "MYDRV_MAINVOL" will become just "mainvol" after this
+   * transformation." */
+  name = mc->mixext.extname;
+  u = MAX (strrchr (name, '_'), strrchr (name, '.'));
+  if (u != NULL)
+    name = u + 1;
+
+  /* maybe capitalize the first letter? */
+  return g_ascii_strdown (name, -1);
+#endif
+  /* the .id thing doesn't really seem to work right, ie. for some sliders
+   * it's just '-' so you have to use the name of the parent control etc.
+   * let's not use it for now, much too painful. */
+  if (g_str_has_prefix (mc->mixext.extname, "misc."))
+    name = g_strdup (mc->mixext.extname + 5);
+  else
+    name = g_strdup (mc->mixext.extname);
+  /* chop off trailing digit (only one for now) */
+  if (strlen (name) > 0 && g_ascii_isdigit (name[strlen (name) - 1]))
+    name[strlen (name) - 1] = '\0';
+  g_strdelimit (name, ".", ' ');
+  return name;
+}
+
+#if 0
+/* FIXME: translations for common option strings */
+static struct
+{
+  const gchar oss_name[32];
+  const gchar *label;
+} option_labels[] = {
+  {
+  "Fast", N_("Fast")}, {
+  "Low", N_("Low")}, {
+  "Medium", N_("Medium")}, {
+  "High", N_("High")}, {
+  "High+", N_("Very high")}, {
+  "Production", N_("Production")}, {
+  "OFF", N_("Off")}, {
+  "ON", N_("On")}, {
+  "Stereo", N_("Stereo")}, {
+  "Multich", N_("Surround sound")}, {
+  "input-mix", N_("Input mix")}, {
+  "front", N_("Front")}, {
+  "rear", N_("Rear")}, {
+  "side", N_("Side")}, {
+  "center/LFE", N_("Center / LFE")}, {
+  "mic", N_("Microphone")}, {
+  "fp-mic", N_("Front panel microphone")}, {
+  "input", N_("Input")}, {
+  "linein", N_("Line-in")}, {
+  "pcm1", N_("PCM 1")}, {
+  "pcm2", N_("PCM 2")}, {
+  "pcm3", N_("PCM 3")}, {
+"pcm4", N_("PCM 4")},};
+#endif
+
+/* these translations are a bit ad-hoc and horribly incomplete; it's not
+ * really going to work this way with all the different chipsets and drivers */
+static struct
+{
+  const gchar oss_name[32];
+  const gchar *label;
+} labels[] = {
+  /* connectors (e.g. hdaudio) */
+  {
+  "jack.green", N_("Green connector")}, {
+  "jack.fp-green", N_("Green front panel connector")}, {
+  "jack.pink", N_("Pink connector")}, {
+  "jack.fp-pink", N_("Pink front panel connector")}, {
+  "jack.blue", N_("Blue connector")}, {
+  "jack.fp-blue", N_("Blue front panel connector")}, {
+  "jack.orange", N_("Orange connector")}, {
+  "jack.fp-orange", N_("Orange front panel connector")}, {
+  "jack.black", N_("Black connector")}, {
+  "jack.fp-black", N_("Black front panel connector")}, {
+  "jack.gray", N_("Gray connector")}, {
+  "jack.fp-gray", N_("Gray front panel connector")}, {
+  "jack.white", N_("White connector")}, {
+  "jack.fp-white", N_("White front panel connector")}, {
+  "jack.red", N_("Red connector")}, {
+  "jack.fp-red", N_("Red front panel connector")}, {
+  "jack.yellow", N_("Yellow connector")}, {
+  "jack.fp-yellow", N_("Yellow front panel connector")},
+      /* connector functions (e.g. hdaudio) */
+  {
+  "jack.green.mode", N_("Green connector function")}, {
+  "jack.fp-green.mode", N_("Green front panel connector function")}, {
+  "jack.pink.mode", N_("Pink connector function")}, {
+  "jack.fp-pink.mode", N_("Pink front panel connector function")}, {
+  "jack.blue.mode", N_("Blue connector function")}, {
+  "jack.fp-blue.mode", N_("Blue front panel connector function")}, {
+  "jack.orange.mode", N_("Orange connector function")}, {
+  "jack.fp-orange.mode", N_("Orange front panel connector function")}, {
+  "jack.black.mode", N_("Black connector function")}, {
+  "jack.fp-black.mode", N_("Black front panel connector function")}, {
+  "jack.gray.mode", N_("Gray connector function")}, {
+  "jack.fp-gray.mode", N_("Gray front panel connector function")}, {
+  "jack.white.mode", N_("White connector function")}, {
+  "jack.fp-white.mode", N_("White front panel connector function")}, {
+  "jack.red.mode", N_("Red connector function")}, {
+  "jack.fp-red.mode", N_("Red front panel connector function")}, {
+  "jack.yellow.mode", N_("Yellow connector function")}, {
+  "jack.fp-yellow.mode", N_("Yellow front panel connector function")},
+      /* other */
+  {
+  "misc.mic", N_("Microphone")}, {
+  "misc.fp-mic", N_("Front panel microphone")}, {
+  "misc.linein", N_("Line-in")}, {
+  "misc.fp-linein", N_("Front panel line-in")}, {
+  "misc.headphone", N_("Headphones")}, {
+  "misc.fp-headphone", N_("Front panel headphones")}, {
+  "misc.front", N_("Front")}, {
+  "misc.rear", N_("Rear")}, {
+  "misc.side", N_("Side")}, {
+  "misc.center/lfe", N_("Center / LFE")}, {
+  "misc.pcm", N_("PCM")}, {
+  "misc.input-mix", N_("Input mix")}
+  /* FIXME translate Audigy NX USB labels) */
+/*
+  { "rec.src", N_("Record Source") },
+  { "output.mute", N_("Mute output") }
+ headph (Controller group)
+   headph.src (Enumeration control)
+   headph.mute (On/Off switch)
+ digital2 (Controller group)
+   digital2.src (Enumeration control)
+   digital2.mute (On/Off switch)
+ digital (Controller group)
+   digital.mute1 (On/Off switch)
+   digital.vol (Controller group)
+     digital.vol.front (Stereo slider (0-255))
+     digital.vol.surr (Stereo slider (0-255))
+     digital.vol.c/l (Stereo slider (0-255))
+     digital.vol.center (Stereo slider (0-255))
+   digital.mute2 (On/Off switch)
+   digital.vol (Stereo slider (0-255))
+ line (Controller group)
+   line.mute (On/Off switch)
+   line.vol (Stereo slider (0-255))
+ play-altset (Enumeration control)
+ rec-altset (Enumeration control)
+*/
+};
+
+/* Decent i18n is pretty much impossible with OSS's way of providing us with
+ * mixer labels (and the fact that they are pretty much random), but that
+ * doesn't mean we shouldn't at least try. */
+const gchar *
+gst_oss4_mixer_control_get_translated_name (GstOss4MixerControl * mc)
+{
+  gchar name[33] = { 0, };
+  char vmix_str[32] = { '\0', };
+  int dummy, i;
+
+  /* main virtual mixer controls (we hide the stream volumes) */
+  if (sscanf (mc->mixext.extname, "vmix%d-%32c", &dummy, vmix_str) == 2) {
+    if (strcmp (vmix_str, "src") == 0)
+      return _("Virtual mixer input");
+    else if (strcmp (vmix_str, "vol") == 0)
+      return _("Virtual mixer output");
+    else if (strcmp (vmix_str, "channels") == 0)
+      return _("Virtual mixer channel configuration");
+  }
+
+  /* munge connector.foo => jack.foo (change from 4.0 -> 4.1) */
+  if (g_str_has_prefix (mc->mixext.extname, "connector.")) {
+    g_snprintf (name, sizeof (name), "jack.%s", mc->mixext.extname + 10);
+  } else {
+    g_strlcpy (name, mc->mixext.extname, sizeof (name));
+  }
+
+  /* munge foo.function => foo.mode (change from 4.0 -> 4.1) */
+  if (g_str_has_suffix (name, ".function"))
+    memcpy (name + strlen (name) - strlen (".function"), ".mode", 5 + 1);
+
+  /* chop off trailing numbers */
+  while (strlen (name) > 0 && g_ascii_isdigit (name[strlen (name) - 1]))
+    name[strlen (name) - 1] = '\0';
+
+  for (i = 0; i < G_N_ELEMENTS (labels); ++i) {
+    if (strcmp (name, labels[i].oss_name) == 0)
+      return _(labels[i].label);
+  }
+
+  g_strdelimit (name, ".", ' ');
+  return g_quark_to_string (g_quark_from_string (name));        /* eek */
+}
+
+#ifndef GST_DISABLE_DEBUG
+static const gchar *
+mixer_ext_type_get_name (gint type)
+{
+  switch (type) {
+    case MIXT_DEVROOT:
+      return "Device root entry";
+    case MIXT_GROUP:
+      return "Controller group";
+    case MIXT_ONOFF:
+      return "On/Off switch";
+    case MIXT_ENUM:
+      return "Enumeration control";
+    case MIXT_MONOSLIDER:
+      return "Mono slider (0-255)";
+    case MIXT_STEREOSLIDER:
+      return "Stereo slider (0-255)";
+    case MIXT_MESSAGE:
+      return "Textual message"; /* whatever this is */
+    case MIXT_MONOVU:
+      return "Mono VU meter value";
+    case MIXT_STEREOVU:
+      return "Stereo VU meter value";
+    case MIXT_MONOPEAK:
+      return "Mono VU meter peak value";
+    case MIXT_STEREOPEAK:
+      return "Stereo VU meter peak value";
+    case MIXT_RADIOGROUP:
+      return "Radio button group";
+    case MIXT_MARKER:          /* Separator between normal and extension entries */
+      return "Separator";
+    case MIXT_VALUE:
+      return "Decimal value entry";
+    case MIXT_HEXVALUE:
+      return "Hex value entry";
+    case MIXT_SLIDER:
+      return "Mono slider (31-bit value range)";
+    case MIXT_3D:
+      return "3D";              /* what's this? */
+    case MIXT_MONOSLIDER16:
+      return "Mono slider (0-32767)";
+    case MIXT_STEREOSLIDER16:
+      return "Stereo slider (0-32767)";
+    case MIXT_MUTE:
+      return "Mute switch";
+    default:
+      break;
+  }
+  return "unknown";
+}
+#endif /* GST_DISABLE_DEBUG */
+
+#ifndef GST_DISABLE_DEBUG
+static const gchar *
+mixer_ext_flags_get_string (gint flags)
+{
+  struct
+  {
+    gint flag;
+    gchar nick[16];
+  } all_flags[] = {
+    /* first the important ones */
+    {
+    MIXF_MAINVOL, "MAINVOL"}, {
+    MIXF_PCMVOL, "PCMVOL"}, {
+    MIXF_RECVOL, "RECVOL"}, {
+    MIXF_MONVOL, "MONVOL"}, {
+    MIXF_DESCR, "DESCR"},
+        /* now the rest in the right order */
+    {
+    MIXF_READABLE, "READABLE"}, {
+    MIXF_WRITEABLE, "WRITABLE"}, {
+    MIXF_POLL, "POLL"}, {
+    MIXF_HZ, "HZ"}, {
+    MIXF_STRING, "STRING"}, {
+    MIXF_DYNAMIC, "DYNAMIC"}, {
+    MIXF_OKFAIL, "OKFAIL"}, {
+    MIXF_FLAT, "FLAT"}, {
+    MIXF_LEGACY, "LEGACY"}, {
+    MIXF_CENTIBEL, "CENTIBEL"}, {
+    MIXF_DECIBEL, "DECIBEL"}, {
+    MIXF_WIDE, "WIDE"}
+  };
+  GString *s;
+  GQuark q;
+  gint i;
+
+  if (flags == 0)
+    return "None";
+
+  s = g_string_new ("");
+  for (i = 0; i < G_N_ELEMENTS (all_flags); ++i) {
+    if ((flags & all_flags[i].flag)) {
+      if (s->len > 0)
+        g_string_append (s, " | ");
+      g_string_append (s, all_flags[i].nick);
+      flags &= ~all_flags[i].flag;      /* unset */
+    }
+  }
+
+  /* unknown flags? */
+  if (flags != 0) {
+    if (s->len > 0)
+      g_string_append (s, " | ");
+    g_string_append (s, "???");
+  }
+
+  /* we'll just put it into the global quark table (cheeky, eh?) */
+  q = g_quark_from_string (s->str);
+  g_string_free (s, TRUE);
+
+  return g_quark_to_string (q);
+}
+#endif /* GST_DISABLE_DEBUG */
+
+#ifndef GST_DISABLE_DEBUG
+static void
+gst_oss4_mixer_control_dump_tree (GstOss4MixerControl * mc, gint depth)
+{
+  GList *c;
+  gchar spaces[64];
+  gint i;
+
+  depth = MIN (sizeof (spaces) - 1, depth);
+  for (i = 0; i < depth; ++i)
+    spaces[i] = ' ';
+  spaces[i] = '\0';
+
+  GST_LOG ("%s%s (%s)", spaces, mc->mixext.extname,
+      mixer_ext_type_get_name (mc->mixext.type));
+  for (c = mc->children; c != NULL; c = c->next) {
+    GstOss4MixerControl *child_mc = (GstOss4MixerControl *) c->data;
+
+    gst_oss4_mixer_control_dump_tree (child_mc, depth + 2);
+  }
+}
+#endif /* GST_DISABLE_DEBUG */
+
+static GList *
+gst_oss4_mixer_get_controls (GstOss4Mixer * mixer)
+{
+  GstOss4MixerControl *root_mc = NULL;
+  oss_mixerinfo mi = { 0, };
+  GList *controls = NULL;
+  GList *l;
+  guint i;
+
+  /* Get info for currently open fd */
+  mi.dev = -1;
+  if (ioctl (mixer->fd, SNDCTL_MIXERINFO, &mi) == -1)
+    goto no_mixerinfo;
+
+  if (mi.nrext <= 0) {
+    GST_DEBUG ("mixer %s has no controls", mi.id);
+    return NULL;
+  }
+
+  GST_INFO ("Reading controls for mixer %s", mi.id);
+
+  for (i = 0; i < mi.nrext; ++i) {
+    GstOss4MixerControl *mc;
+    oss_mixext mix_ext = { 0, };
+
+    mix_ext.dev = mi.dev;
+    mix_ext.ctrl = i;
+
+    if (ioctl (mixer->fd, SNDCTL_MIX_EXTINFO, &mix_ext) == -1) {
+      GST_DEBUG ("SNDCTL_MIX_EXTINFO failed on mixer %s, control %d: %s",
+          mi.id, i, g_strerror (errno));
+      continue;
+    }
+
+    /* if this is the last one, save it for is-interface-up-to-date checking */
+    if (i == mi.nrext)
+      mixer->last_mixext = mix_ext;
+
+    mc = g_new0 (GstOss4MixerControl, 1);
+    mc->mixext = mix_ext;
+
+    /* both control_no and desc fields are pretty useless, ie. not always set,
+     * if ever, so not listed here */
+    GST_INFO ("Control %d", mix_ext.ctrl);
+    GST_INFO ("  name   : %s", mix_ext.extname);
+    GST_INFO ("  type   : %s (%d)", mixer_ext_type_get_name (mix_ext.type),
+        mix_ext.type);
+    GST_INFO ("  flags  : %s (0x%04x)",
+        mixer_ext_flags_get_string (mix_ext.flags), mix_ext.flags);
+    GST_INFO ("  parent : %d", mix_ext.parent);
+
+    /* get tooltip (just for informational purposes for now) */
+    if (MIXEXT_HAS_DESCRIPTION (mix_ext)) {
+      oss_mixer_enuminfo desc = { 0, };
+
+      desc.dev = mix_ext.dev;
+      desc.ctrl = mix_ext.ctrl;
+      if (ioctl (mixer->fd, SNDCTL_MIX_DESCRIPTION, &desc) >= 0) {
+        /* "The string may contain multiple lines. The first line is the
+         *  'tooltip'. Optional subsequent lines may contain more detailed
+         *  help text. Lines are separated by a linefeed character." */
+        g_strdelimit (&desc.strings[desc.strindex[0]], "\n\r", '\0');
+        GST_INFO ("  tooltip: %s", &desc.strings[desc.strindex[0]]);
+      }
+    }
+
+    if (!MIXEXT_IS_ROOT (mix_ext)) {
+      /* find parent (we assume it comes in the list before the child) */
+      for (l = controls; l != NULL; l = l->next) {
+        GstOss4MixerControl *parent_mc = (GstOss4MixerControl *) l->data;
+
+        if (parent_mc->mixext.ctrl == mix_ext.parent) {
+          mc->parent = parent_mc;
+          parent_mc->children = g_list_append (parent_mc->children, mc);
+          break;
+        }
+      }
+      if (mc->parent == NULL) {
+        GST_ERROR_OBJECT (mixer, "couldn't find parent for control %d", i);
+        g_free (mc);
+        continue;
+      }
+    } else if (root_mc == NULL) {
+      root_mc = mc;
+    } else {
+      GST_WARNING_OBJECT (mixer, "two root controls?!");
+    }
+
+    controls = g_list_prepend (controls, mc);
+  }
+
+#ifndef GST_DISABLE_DEBUG
+  gst_oss4_mixer_control_dump_tree (root_mc, 0);
+#endif
+
+  return g_list_reverse (controls);
+
+/* ERRORS */
+no_mixerinfo:
+  {
+    GST_WARNING ("SNDCTL_MIXERINFO failed on mixer device %s: %s", mi.id,
+        g_strerror (errno));
+    return NULL;
+  }
+}
+
+static void
+gst_oss4_mixer_controls_guess_master (GstOss4Mixer * mixer,
+    const GList * controls)
+{
+  GstOss4MixerControl *firstpcm_mc = NULL;
+  GstOss4MixerControl *master_mc = NULL;
+  const GList *l;
+
+  for (l = controls; l != NULL; l = l->next) {
+    GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+    if (((mc->mixext.flags & MIXF_MAINVOL)) && master_mc == NULL) {
+      GST_INFO_OBJECT (mixer, "Master control: %s", mc->mixext.extname);
+      master_mc = mc;
+      break;
+    }
+    /* do we need to check if it's a slider type here? */
+    if (((mc->mixext.flags & MIXF_PCMVOL)) && firstpcm_mc == NULL) {
+      GST_INFO_OBJECT (mixer, "First PCM control: %s", mc->mixext.extname);
+      firstpcm_mc = mc;
+    }
+  }
+
+  /* if no control with MIXF_MAINVOL found, use first one with PCMVOL */
+  if (master_mc == NULL && firstpcm_mc != NULL) {
+    GST_INFO_OBJECT (mixer, "Marking first PCM control as master: %s",
+        firstpcm_mc->mixext.extname);
+    master_mc = firstpcm_mc;
+  }
+
+  if (master_mc != NULL)
+    master_mc->is_master = TRUE;
+}
+
+/* type: -1 for all types, otherwise just return siblings with requested type */
+static GList *
+gst_oss4_mixer_control_get_siblings (GstOss4MixerControl * mc, gint type)
+{
+  GList *s, *siblings = NULL;
+
+  if (mc->parent == NULL)
+    return NULL;
+
+  for (s = mc->parent->children; s != NULL; s = s->next) {
+    GstOss4MixerControl *sibling = (GstOss4MixerControl *) s->data;
+
+    if (mc != sibling && (type < 0 || sibling->mixext.type == type))
+      siblings = g_list_append (siblings, sibling);
+  }
+
+  return siblings;
+}
+
+static void
+gst_oss4_mixer_controls_find_sliders (GstOss4Mixer * mixer,
+    const GList * controls)
+{
+  const GList *l;
+
+  for (l = controls; l != NULL; l = l->next) {
+    GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+    GList *s, *siblings;
+
+    if (!MIXEXT_IS_SLIDER (mc->mixext) || mc->parent == NULL || mc->used)
+      continue;
+
+    mc->is_slider = TRUE;
+    mc->used = TRUE;
+
+    siblings = gst_oss4_mixer_control_get_siblings (mc, -1);
+
+    /* Note: the names can be misleading and may not reflect the actual
+     * hierarchy of the controls, e.g. it's possible that a slider is called
+     * connector.green and the mute control then connector.green.mute, but
+     * both controls are in fact siblings and both children of the group
+     * 'green' instead of mute being a child of connector.green as the naming
+     * would seem to suggest */
+    GST_LOG ("Slider: %s, parent=%s, %d siblings", mc->mixext.extname,
+        mc->parent->mixext.extname, g_list_length (siblings));
+
+    for (s = siblings; s != NULL; s = s->next) {
+      GstOss4MixerControl *sibling = (GstOss4MixerControl *) s->data;
+
+      GST_LOG ("    %s (%s)", sibling->mixext.extname,
+          mixer_ext_type_get_name (sibling->mixext.type));
+
+      if (sibling->mixext.type == MIXT_MUTE ||
+          g_str_has_suffix (sibling->mixext.extname, ".mute")) {
+        /* simple case: slider with single mute sibling. We assume the .mute
+         * suffix in the name won't change - can't really do much else anyway */
+        if (sibling->mixext.type == MIXT_ONOFF ||
+            sibling->mixext.type == MIXT_MUTE) {
+          GST_LOG ("    mute control for %s is %s", mc->mixext.extname,
+              sibling->mixext.extname);
+          mc->mute = sibling;
+          sibling->used = TRUE;
+        }
+        /* a group of .mute controls. We assume they are all switches here */
+        if (sibling->mixext.type == MIXT_GROUP) {
+          GList *m;
+
+          for (m = sibling->children; m != NULL; m = m->next) {
+            GstOss4MixerControl *grouped_sibling = m->data;
+
+            if (grouped_sibling->mixext.type == MIXT_ONOFF ||
+                grouped_sibling->mixext.type == MIXT_MUTE) {
+              GST_LOG ("    %s is grouped mute control for %s",
+                  grouped_sibling->mixext.extname, mc->mixext.extname);
+              mc->mute_group = g_list_append (mc->mute_group, grouped_sibling);
+            }
+          }
+
+          GST_LOG ("    %s has a group of %d mute controls",
+              mc->mixext.extname, g_list_length (mc->mute_group));
+
+          /* we don't mark the individual mute controls as used, only the
+           * group control, as we still want individual switches for the
+           * individual controls */
+          sibling->used = TRUE;
+        }
+      }
+    }
+    g_list_free (siblings);
+  }
+}
+
+/* should be called with the mixer object lock held because of the ioctl;
+ * returns TRUE if the list was read the first time or modified */
+static gboolean
+gst_oss4_mixer_enum_control_update_enum_list (GstOss4Mixer * mixer,
+    GstOss4MixerControl * mc)
+{
+  oss_mixer_enuminfo ei = { 0, };
+  guint num_existing = 0;
+  int i;
+
+  /* count and existing values */
+  while (mc->enum_vals != NULL && mc->enum_vals[num_existing] != 0)
+    ++num_existing;
+
+  ei.dev = mc->mixext.dev;
+  ei.ctrl = mc->mixext.ctrl;
+
+  /* if we have create a generic list with numeric IDs already and the
+   * number of values hasn't changed, then there's not much to do here */
+  if (mc->no_list && mc->enum_vals != NULL &&
+      num_existing == mc->mixext.maxvalue) {
+    return FALSE;
+  }
+
+  /* if we have a list and it doesn't change, there's nothing to do either */
+  if (mc->enum_vals != NULL && mc->enum_version == 0)
+    return FALSE;
+
+  if (ioctl (mixer->fd, SNDCTL_MIX_ENUMINFO, &ei) == -1) {
+    g_free (mc->enum_vals);
+    mc->enum_vals = g_new0 (GQuark, mc->mixext.maxvalue + 1);
+
+    GST_DEBUG ("no enum info available, creating numeric values from 0-%d",
+        mc->mixext.maxvalue - 1);
+
+    /* "It is possible that some enum controls don't have any name list
+     * available. In this case the application should automatically generate
+     * list of numbers (0 to N-1)" */
+    for (i = 0; i < mc->mixext.maxvalue; ++i) {
+      gchar num_str[8];
+
+      g_snprintf (num_str, sizeof (num_str), "%d", i);
+      mc->enum_vals[i] = g_quark_from_string (num_str);
+    }
+    mc->enum_version = 0;       /* the only way to change is via maxvalue */
+  } else {
+    /* old list same as current list? */
+    if (mc->enum_vals != NULL && mc->enum_version == ei.version)
+      return FALSE;
+
+    /* no list yet, or the list has changed */
+    GST_LOG ("%s", (mc->enum_vals) ? "enum list has changed" : "reading list");
+    if (ei.nvalues != mc->mixext.maxvalue) {
+      GST_WARNING_OBJECT (mixer, "Enum: %s, nvalues %d != maxvalue %d",
+          mc->mixext.extname, ei.nvalues, mc->mixext.maxvalue);
+      mc->mixext.maxvalue = MIN (ei.nvalues, mc->mixext.maxvalue);
+    }
+
+    mc->mixext.maxvalue = MIN (mc->mixext.maxvalue, OSS_ENUM_MAXVALUE);
+
+    g_free (mc->enum_vals);
+    mc->enum_vals = g_new0 (GQuark, mc->mixext.maxvalue + 1);
+    for (i = 0; i < mc->mixext.maxvalue; ++i) {
+      GST_LOG ("  %s", ei.strings + ei.strindex[i]);
+      mc->enum_vals[i] = g_quark_from_string (ei.strings + ei.strindex[i]);
+    }
+  }
+
+  return TRUE;
+}
+
+static void
+gst_oss4_mixer_controls_find_enums (GstOss4Mixer * mixer,
+    const GList * controls)
+{
+  const GList *l;
+
+  for (l = controls; l != NULL; l = l->next) {
+    GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+    if (mc->mixext.type != MIXT_ENUM || mc->used)
+      continue;
+
+    mc->is_enum = TRUE;
+    mc->used = TRUE;
+
+    /* Note: enums are special: for most controls, the maxvalue is inclusive,
+     * but for enum controls it's actually exclusive (boggle), so that
+     * mixext.maxvalue = num_values */
+
+    GST_LOG ("Enum: %s, parent=%s, num_enums=%d", mc->mixext.extname,
+        mc->parent->mixext.extname, mc->mixext.maxvalue);
+
+    gst_oss4_mixer_enum_control_update_enum_list (mixer, mc);
+  }
+}
+
+static void
+gst_oss4_mixer_controls_find_switches (GstOss4Mixer * mixer,
+    const GList * controls)
+{
+  const GList *l;
+
+  for (l = controls; l != NULL; l = l->next) {
+    GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+    if (mc->used)
+      continue;
+
+    if (mc->mixext.type != MIXT_ONOFF && mc->mixext.type != MIXT_MUTE)
+      continue;
+
+    mc->is_switch = TRUE;
+    mc->used = TRUE;
+
+    GST_LOG ("Switch: %s, parent=%s", mc->mixext.extname,
+        mc->parent->mixext.extname);
+  }
+}
+
+static void
+gst_oss4_mixer_controls_find_virtual (GstOss4Mixer * mixer,
+    const GList * controls)
+{
+  const GList *l;
+
+  for (l = controls; l != NULL; l = l->next) {
+    GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+    /* or sscanf (mc->mixext.extname, "vmix%d-out.", &n) == 1 ? */
+    /* for now we just flag all virtual controls with managed labels, those
+     * are really more appropriate for a pavucontrol-type control thing than
+     * the (more hardware-oriented) mixer interface */
+    if (mc->mixext.id[0] == '@') {
+      mc->is_virtual = TRUE;
+      GST_LOG ("%s is virtual control with managed label", mc->mixext.extname);
+    }
+  }
+}
+
+static void
+gst_oss4_mixer_controls_dump_unused (GstOss4Mixer * mixer,
+    const GList * controls)
+{
+  const GList *l;
+
+  for (l = controls; l != NULL; l = l->next) {
+    GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+    if (mc->used)
+      continue;
+
+    switch (mc->mixext.type) {
+      case MIXT_DEVROOT:
+      case MIXT_GROUP:
+      case MIXT_MESSAGE:
+      case MIXT_MONOVU:
+      case MIXT_STEREOVU:
+      case MIXT_MONOPEAK:
+      case MIXT_STEREOPEAK:
+      case MIXT_MARKER:
+        continue;               /* not interested in these types of controls */
+      case MIXT_MONODB:
+      case MIXT_STEREODB:
+        GST_DEBUG ("obsolete control type %d", mc->mixext.type);
+        continue;
+      case MIXT_MONOSLIDER:
+      case MIXT_STEREOSLIDER:
+      case MIXT_SLIDER:
+      case MIXT_MONOSLIDER16:
+      case MIXT_STEREOSLIDER16:
+        /* this shouldn't happen */
+        GST_ERROR ("unused slider control?!");
+        continue;
+      case MIXT_VALUE:
+      case MIXT_HEXVALUE:
+        /* value entry, not sure what to do with that, skip for now */
+        continue;
+      case MIXT_ONOFF:
+      case MIXT_MUTE:
+      case MIXT_ENUM:
+      case MIXT_3D:
+      case MIXT_RADIOGROUP:
+        GST_DEBUG ("FIXME: handle %s %s",
+            mixer_ext_type_get_name (mc->mixext.type), mc->mixext.extname);
+        break;
+      default:
+        GST_WARNING ("unknown control type %d", mc->mixext.type);
+        continue;
+    }
+  }
+}
+
+static GList *
+gst_oss4_mixer_create_tracks (GstOss4Mixer * mixer, const GList * controls)
+{
+  const GList *c;
+  GList *tracks = NULL;
+
+  for (c = controls; c != NULL; c = c->next) {
+    GstOss4MixerControl *mc = (GstOss4MixerControl *) c->data;
+    GstMixerTrack *track = NULL;
+
+    if (mc->is_virtual)
+      continue;
+
+    if (mc->is_slider) {
+      track = gst_oss4_mixer_slider_new (mixer, mc);
+    } else if (mc->is_enum) {
+      track = gst_oss4_mixer_enum_new (mixer, mc);
+    } else if (mc->is_switch) {
+      track = gst_oss4_mixer_switch_new (mixer, mc);
+    }
+
+    if (track == NULL)
+      continue;
+
+    track->label = gst_oss4_mixer_control_get_pretty_name (mc);
+    track->flags = 0;
+
+    GST_LOG ("translated label: %s [%s] = %s", track->label, mc->mixext.id,
+        gst_oss4_mixer_control_get_translated_name (mc));
+
+    /* This whole 'a track is either INPUT or OUTPUT' model is just flawed,
+     * esp. if a slider's role can be changed on the fly, like when you change
+     * function of a connector. What should we do in that case? Change the flag
+     * and make the app rebuild the interface? Ignore it? */
+    if (g_str_has_prefix (mc->mixext.extname, "record.")) {
+      mc->is_output = FALSE;
+      mc->is_input = TRUE;
+    }
+
+    /* FIXME: determine is_input and is_output */
+    /* must be either INPUT or OUTPUT (but not both and not neither) for now,
+     * or gnome-volume-control aborts */
+    if (mc->is_input)
+      track->flags |= GST_MIXER_TRACK_INPUT;
+    else if (mc->is_output)
+      track->flags |= GST_MIXER_TRACK_OUTPUT;
+    else
+      track->flags |= GST_MIXER_TRACK_OUTPUT;
+
+    if (mc->is_master)
+      track->flags |= GST_MIXER_TRACK_MASTER;
+
+    tracks = g_list_append (tracks, track);
+  }
+
+  return tracks;
+}
+
+static void
+gst_oss4_mixer_update_tracks (GstOss4Mixer * mixer)
+{
+  GList *controls, *tracks;
+
+  /* read and process controls */
+  controls = gst_oss4_mixer_get_controls (mixer);
+
+  gst_oss4_mixer_controls_guess_master (mixer, controls);
+
+  gst_oss4_mixer_controls_find_sliders (mixer, controls);
+
+  gst_oss4_mixer_controls_find_enums (mixer, controls);
+
+  gst_oss4_mixer_controls_find_switches (mixer, controls);
+
+  gst_oss4_mixer_controls_find_virtual (mixer, controls);
+
+  gst_oss4_mixer_controls_dump_unused (mixer, controls);
+
+  tracks = gst_oss4_mixer_create_tracks (mixer, controls);
+
+  /* free old tracks and controls */
+  gst_oss4_mixer_free_tracks (mixer);
+
+  /* replace old with new */
+  mixer->tracks = tracks;
+  mixer->controls = controls;
+}
+
+static const GList *
+gst_oss4_mixer_list_tracks (GstMixer * mixer_iface)
+{
+  GstOss4Mixer *mixer = GST_OSS4_MIXER (mixer_iface);
+
+  g_return_val_if_fail (mixer != NULL, NULL);
+  g_return_val_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer), NULL);
+
+  GST_OBJECT_LOCK (mixer);
+
+  /* Do a read on the last control to check if the interface has changed */
+  if (!mixer->need_update && mixer->last_mixext.ctrl > 0) {
+    GstOss4MixerControl mc = { {0,}
+    ,
+    };
+    int val;
+
+    mc.mixext = mixer->last_mixext;
+    gst_oss4_mixer_get_control_val (mixer, &mc, &val);
+  }
+
+  if (mixer->need_update || mixer->tracks == NULL) {
+    gst_oss4_mixer_update_tracks (mixer);
+    mixer->need_update = FALSE;
+  }
+
+  GST_OBJECT_UNLOCK (mixer);
+
+  return (const GList *) mixer->tracks;
+}
+
+static void
+gst_oss4_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
+    gint * volumes)
+{
+  GstOss4Mixer *oss;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+  g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+  g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+  g_return_if_fail (volumes != NULL);
+
+  oss = GST_OSS4_MIXER (mixer);
+
+  GST_OBJECT_LOCK (oss);
+
+  if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+    gst_oss4_mixer_slider_set_volume (GST_OSS4_MIXER_SLIDER (track), volumes);
+  }
+
+  GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
+    gint * volumes)
+{
+  GstOss4Mixer *oss;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+  g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+  g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+  g_return_if_fail (volumes != NULL);
+
+  oss = GST_OSS4_MIXER (mixer);
+
+  GST_OBJECT_LOCK (oss);
+
+  memset (volumes, 0, track->num_channels * sizeof (gint));
+
+  if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+    gst_oss4_mixer_slider_get_volume (GST_OSS4_MIXER_SLIDER (track), volumes);
+  }
+
+  GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_set_record (GstMixer * mixer, GstMixerTrack * track,
+    gboolean record)
+{
+  GstOss4Mixer *oss;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+  g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+  g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+
+  oss = GST_OSS4_MIXER (mixer);
+
+  GST_OBJECT_LOCK (oss);
+
+  if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+    gst_oss4_mixer_slider_set_record (GST_OSS4_MIXER_SLIDER (track), record);
+  } else if (GST_IS_OSS4_MIXER_SWITCH (track)) {
+    if ((track->flags & GST_MIXER_TRACK_INPUT)) {
+      gst_oss4_mixer_switch_set (GST_OSS4_MIXER_SWITCH (track), record);
+    } else {
+      GST_WARNING_OBJECT (track, "set_record called on non-INPUT track");
+    }
+  }
+
+  GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
+{
+  GstOss4Mixer *oss;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+  g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+  g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+
+  oss = GST_OSS4_MIXER (mixer);
+
+  GST_OBJECT_LOCK (oss);
+
+  if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+    gst_oss4_mixer_slider_set_mute (GST_OSS4_MIXER_SLIDER (track), mute);
+  } else if (GST_IS_OSS4_MIXER_SWITCH (track)) {
+    if ((track->flags & GST_MIXER_TRACK_OUTPUT)) {
+      gst_oss4_mixer_switch_set (GST_OSS4_MIXER_SWITCH (track), mute);
+    } else {
+      GST_WARNING_OBJECT (track, "set_mute called on non-OUTPUT track");
+    }
+  }
+
+  GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_set_option (GstMixer * mixer, GstMixerOptions * options,
+    gchar * value)
+{
+  GstOss4Mixer *oss;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (value != NULL);
+  g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+  g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+  g_return_if_fail (GST_IS_OSS4_MIXER_ENUM (options));
+  g_return_if_fail (gst_oss4_mixer_contains_options (mixer, options));
+
+  oss = GST_OSS4_MIXER (mixer);
+
+  GST_OBJECT_LOCK (oss);
+
+  if (!gst_oss4_mixer_enum_set_option (GST_OSS4_MIXER_ENUM (options), value)) {
+    /* not much we can do here but wake up the watch thread early, so it
+     * can do its thing and post messages if anything has changed */
+    gst_oss4_mixer_wake_up_watch_task (oss);
+  }
+
+  GST_OBJECT_UNLOCK (oss);
+}
+
+static const gchar *
+gst_oss4_mixer_get_option (GstMixer * mixer, GstMixerOptions * options)
+{
+  GstOss4Mixer *oss;
+  const gchar *current_val;
+
+  g_return_val_if_fail (mixer != NULL, NULL);
+  g_return_val_if_fail (GST_IS_OSS4_MIXER (mixer), NULL);
+  g_return_val_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer), NULL);
+  g_return_val_if_fail (GST_IS_OSS4_MIXER_ENUM (options), NULL);
+  g_return_val_if_fail (gst_oss4_mixer_contains_options (mixer, options), NULL);
+
+  oss = GST_OSS4_MIXER (mixer);
+
+  GST_OBJECT_LOCK (oss);
+
+  current_val = gst_oss4_mixer_enum_get_option (GST_OSS4_MIXER_ENUM (options));
+
+  if (current_val == NULL) {
+    /* not much we can do here but wake up the watch thread early, so it
+     * can do its thing and post messages if anything has changed */
+    gst_oss4_mixer_wake_up_watch_task (oss);
+  }
+
+  GST_OBJECT_UNLOCK (oss);
+
+  return current_val;
+}
+
+static GstMixerFlags
+gst_oss4_mixer_get_mixer_flags (GstMixer * mixer)
+{
+  return GST_MIXER_FLAG_AUTO_NOTIFICATIONS;
+}
+
+static void
+gst_oss4_mixer_interface_init (GstMixerClass * klass)
+{
+  GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE;
+
+  klass->list_tracks = gst_oss4_mixer_list_tracks;
+  klass->set_volume = gst_oss4_mixer_set_volume;
+  klass->get_volume = gst_oss4_mixer_get_volume;
+  klass->set_mute = gst_oss4_mixer_set_mute;
+  klass->set_record = gst_oss4_mixer_set_record;
+  klass->set_option = gst_oss4_mixer_set_option;
+  klass->get_option = gst_oss4_mixer_get_option;
+  klass->get_mixer_flags = gst_oss4_mixer_get_mixer_flags;
+}
+
+/* Implement the horror that is GstImplementsInterface */
+
+static gboolean
+gst_oss4_mixer_supported (GstImplementsInterface * iface, GType iface_type)
+{
+  GstOss4Mixer *mixer;
+
+  g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
+
+  mixer = GST_OSS4_MIXER (iface);
+
+  return GST_OSS4_MIXER_IS_OPEN (mixer);
+}
+
+static void
+gst_oss4_mixer_implements_interface_init (GstImplementsInterfaceClass * klass)
+{
+  klass->supported = gst_oss4_mixer_supported;
+}
+
+static void
+gst_oss4_mixer_init_interfaces (GType type)
+{
+  static const GInterfaceInfo implements_iface_info = {
+    (GInterfaceInitFunc) gst_oss4_mixer_implements_interface_init,
+    NULL,
+    NULL,
+  };
+  static const GInterfaceInfo mixer_iface_info = {
+    (GInterfaceInitFunc) gst_oss4_mixer_interface_init,
+    NULL,
+    NULL,
+  };
+
+  g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
+      &implements_iface_info);
+  g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
+
+  gst_oss4_add_property_probe_interface (type);
+}
diff --git a/sys/oss4/oss4-mixer.h b/sys/oss4/oss4-mixer.h
new file mode 100644
index 00000000..4eaff50c
--- /dev/null
+++ b/sys/oss4/oss4-mixer.h
@@ -0,0 +1,128 @@
+/* GStreamer OSS4 mixer implementation
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef OSS4_MIXER_H
+#define OSS4_MIXER_H
+
+#include <gst/gst.h>
+
+#include "oss4-soundcard.h"
+
+#define GST_OSS4_MIXER(obj)              (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER,GstOss4Mixer))
+#define GST_OSS4_MIXER_CAST(obj)         ((GstOss4Mixer *)(obj))
+#define GST_OSS4_MIXER_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER,GstOss4MixerClass))
+#define GST_IS_OSS4_MIXER(obj)           (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER))
+#define GST_IS_OSS4_MIXER_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER))
+#define GST_TYPE_OSS4_MIXER              (gst_oss4_mixer_get_type())
+
+#define GST_OSS4_MIXER_IS_OPEN(obj)      (GST_OSS4_MIXER(obj)->fd != -1)
+
+typedef struct _GstOss4Mixer GstOss4Mixer;
+typedef struct _GstOss4MixerClass GstOss4MixerClass;
+
+struct _GstOss4Mixer {
+  GstElement            element;
+
+  /*< private >*/
+
+  /* element bits'n'bops */ 
+  gchar               * device;
+
+  /* mixer details */
+  gint                  fd;             /* file descriptor if open, or -1    */
+  gchar               * device_name;    /* device description, or NULL       */
+  gchar               * open_device;    /* the device we opened              */
+
+  GList               * tracks;         /* list of available tracks          */
+  GList               * controls;       /* list of available controls        */
+  gboolean              need_update;    /* re-read list of available tracks? */
+
+  oss_mixext            last_mixext;    /* we keep this around so we can
+                                         * easily check if the mixer
+                                         * interface has changed             */
+
+  GThread             * watch_thread;   /* thread watching for value changes */
+  GCond               * watch_cond;
+  gint                  watch_shutdown;
+  gint                  modify_counter; /* from MIXERINFO */
+
+  /* for property probe interface */
+  GList               * property_probe_list;
+};
+
+struct _GstOss4MixerClass {
+  GstElementClass       element_class;
+};
+
+/* helper struct holding info about one control */
+typedef struct _GstOss4MixerControl GstOss4MixerControl;
+
+struct _GstOss4MixerControl {
+  oss_mixext           mixext;
+  GstOss4MixerControl *parent;         /* NULL if root                         */
+  GstOss4MixerControl *mute;           /* sibling with mute function, or NULL  */
+  GList               *mute_group;     /* group of mute controls, or NULL      */
+  GList               *children;       /* GstOss4MixerControls (no ownership)  */
+
+  GQuark              *enum_vals;      /* 0-terminated array of values or NULL */
+  int                  enum_version;   /* 0 = list won't change                */
+
+  int                  last_val;       /* last value seen                      */
+
+  gboolean             is_virtual : 1; /* is a vmix control with dynamic label */
+  gboolean             is_master  : 1;
+  gboolean             is_slider  : 1; /* represent as slider                  */
+  gboolean             is_switch  : 1; /* represent as switch                  */
+  gboolean             is_enum    : 1; /* represent as combo/enumeration       */
+  gboolean             no_list    : 1; /* enumeration with no list available   */
+  gboolean             is_input   : 1; /* is an input-related control          */
+  gboolean             is_output  : 1; /* is an output-related control         */
+  gboolean             used       : 1; /* whether we know what to do with this */
+
+  gboolean             changed      : 1; /* transient flag used by watch thread */
+  gboolean             list_changed : 1; /* transient flag used by watch thread */
+};
+
+/* header says parent=-1 means root, but it can also be parent=ctrl */
+#define MIXEXT_IS_ROOT(me) ((me).parent == -1 || (me).parent == (me).ctrl)
+
+#define MIXEXT_IS_SLIDER(me) ((me).type == MIXT_MONOSLIDER ||            \
+    (me).type == MIXT_STEREOSLIDER || (me).type == MIXT_MONOSLIDER16 ||  \
+    (me).type == MIXT_STEREOSLIDER16 || (me).type == MIXT_SLIDER)
+
+#define MIXEXT_HAS_DESCRIPTION(me) (((me).flags & MIXF_DESCR) != 0)
+
+#define MIXEXT_ENUM_IS_AVAILABLE(me,num) \
+    (((me).enum_present[num/8]) & (1 << (num % 8)))
+
+
+GType     gst_oss4_mixer_get_type (void);
+
+gboolean  gst_oss4_mixer_get_control_val (GstOss4Mixer        * mixer,
+                                          GstOss4MixerControl * mc,
+                                          int                 * val);
+
+gboolean  gst_oss4_mixer_set_control_val (GstOss4Mixer        * mixer,
+                                          GstOss4MixerControl * mc,
+                                          int                   val);
+
+G_END_DECLS
+
+#endif /* OSS4_MIXER_H */
+
diff --git a/sys/oss4/oss4-property-probe.c b/sys/oss4/oss4-property-probe.c
new file mode 100644
index 00000000..e4b56679
--- /dev/null
+++ b/sys/oss4/oss4-property-probe.c
@@ -0,0 +1,396 @@
+/* GStreamer OSS4 audio property probe interface implementation
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-audio.h"
+#include "oss4-mixer.h"
+#include "oss4-sink.h"
+#include "oss4-source.h"
+#include "oss4-soundcard.h"
+#include "oss4-property-probe.h"
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+
+GST_DEBUG_CATEGORY_EXTERN (oss4_debug);
+#define GST_CAT_DEFAULT oss4_debug
+
+static const GList *
+gst_oss4_property_probe_get_properties (GstPropertyProbe * probe)
+{
+  GObjectClass *klass = G_OBJECT_GET_CLASS (probe);
+  GList *list;
+
+  GST_OBJECT_LOCK (GST_OBJECT (probe));
+
+  /* we create a new list and store it in the instance struct, since apparently
+   * we forgot to update the API for 0.10 (and why don't we mark probable
+   * properties with a flag instead anyway?). A bit hackish, but what can you
+   * do (can't really use a static variable since the pspec will be different
+   * for src and sink class); this isn't particularly pretty, but the best
+   * we can do given that we can't create a common base class (we could do
+   * fancy things with the interface, or use g_object_set_data instead, but
+   * it's not really going to make it much better) */
+  if (GST_IS_AUDIO_SINK_CLASS (klass)) {
+    list = GST_OSS4_SINK (probe)->property_probe_list;
+  } else if (GST_IS_AUDIO_SRC_CLASS (klass)) {
+    list = GST_OSS4_SOURCE (probe)->property_probe_list;
+  } else if (GST_IS_OSS4_MIXER_CLASS (klass)) {
+    list = GST_OSS4_MIXER (probe)->property_probe_list;
+  } else {
+    GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+    g_return_val_if_reached (NULL);
+  }
+
+  if (list == NULL) {
+    GParamSpec *pspec;
+
+    pspec = g_object_class_find_property (klass, "device");
+    list = g_list_prepend (NULL, pspec);
+
+    if (GST_IS_AUDIO_SINK_CLASS (klass)) {
+      GST_OSS4_SINK (probe)->property_probe_list = list;
+    } else if (GST_IS_AUDIO_SRC_CLASS (klass)) {
+      GST_OSS4_SOURCE (probe)->property_probe_list = list;
+    } else if (GST_IS_OSS4_MIXER_CLASS (klass)) {
+      GST_OSS4_MIXER (probe)->property_probe_list = list;
+    }
+  }
+
+  GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+
+  return list;
+}
+
+static void
+gst_oss4_property_probe_probe_property (GstPropertyProbe * probe,
+    guint prop_id, const GParamSpec * pspec)
+{
+  if (!g_str_equal (pspec->name, "device")) {
+    G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
+  }
+}
+
+static gboolean
+gst_oss4_property_probe_needs_probe (GstPropertyProbe * probe,
+    guint prop_id, const GParamSpec * pspec)
+{
+  /* don't cache probed data */
+  return TRUE;
+}
+
+static gint
+oss4_mixerinfo_priority_cmp (struct oss_mixerinfo *mi1,
+    struct oss_mixerinfo *mi2)
+{
+  /* return negative vaue if mi1 comes before mi2 */
+  if (mi1->priority != mi2->priority)
+    return mi2->priority - mi1->priority;
+
+  return strcmp (mi1->devnode, mi2->devnode);
+}
+
+/* caller must ensure LOCK is taken (e.g. if ioctls need to be serialised) */
+gboolean
+gst_oss4_property_probe_find_device_name (GstObject * obj, int fd,
+    const gchar * device_handle, gchar ** device_name)
+{
+  struct oss_sysinfo si = { {0,}, };
+  gchar *name = NULL;
+
+  if (ioctl (fd, SNDCTL_SYSINFO, &si) == 0) {
+    int i;
+
+    for (i = 0; i < si.numaudios; ++i) {
+      struct oss_audioinfo ai = { 0, };
+
+      ai.dev = i;
+      if (ioctl (fd, SNDCTL_AUDIOINFO, &ai) == -1) {
+        GST_DEBUG_OBJECT (obj, "AUDIOINFO ioctl for device %d failed", i);
+        continue;
+      }
+      if (strcmp (ai.devnode, device_handle) == 0) {
+        name = g_strdup (ai.name);
+        break;
+      }
+    }
+  } else {
+    GST_WARNING_OBJECT (obj, "SYSINFO ioctl failed: %s", g_strerror (errno));
+  }
+
+  /* try ENGINEINFO as fallback (which is better than nothing) */
+  if (name == NULL) {
+    struct oss_audioinfo ai = { 0, };
+
+    GST_LOG_OBJECT (obj, "device %s not listed in AUDIOINFO", device_handle);
+    ai.dev = -1;
+    if (ioctl (fd, SNDCTL_ENGINEINFO, &ai) == 0)
+      name = g_strdup (ai.name);
+  }
+
+  GST_DEBUG_OBJECT (obj, "Device name: %s", GST_STR_NULL (name));
+
+  if (name != NULL)
+    *device_name = name;
+
+  return (name != NULL);
+}
+
+static GList *
+gst_oss4_property_probe_get_mixer_devices (GstObject * obj, int fd,
+    struct oss_sysinfo *si)
+{
+  GList *m, *mixers = NULL;
+  GList *devices = NULL;
+
+  int i;
+
+  GST_LOG_OBJECT (obj, "%d mixer devices", si->nummixers);
+
+  /* first, find suitable mixer devices and sort by priority */
+  for (i = 0; i < si->nummixers; ++i) {
+    struct oss_mixerinfo mi = { 0, };
+
+    mi.dev = i;
+    if (ioctl (fd, SNDCTL_MIXERINFO, &mi) == -1) {
+      GST_DEBUG_OBJECT (obj, "MIXERINFO ioctl for device %d failed", i);
+      continue;
+    }
+
+    GST_INFO_OBJECT (obj, "mixer device %d:", i);
+    GST_INFO_OBJECT (obj, "  enabled  : %s", (mi.enabled) ? "yes" :
+        "no (powered off or unplugged)");
+    GST_INFO_OBJECT (obj, "  priority : %d", mi.priority);
+    GST_INFO_OBJECT (obj, "  devnode  : %s", mi.devnode);
+    GST_INFO_OBJECT (obj, "  handle   : %s", mi.handle);
+    GST_INFO_OBJECT (obj, "  caps     : 0x%02x", mi.caps);
+    GST_INFO_OBJECT (obj, "  name     : %s", mi.name);
+
+    if (!mi.enabled) {
+      GST_DEBUG_OBJECT (obj, "mixer device is not usable/enabled, skipping");
+      continue;
+    }
+
+    /* only want mixers that control hardware directly */
+    if ((mi.caps & MIXER_CAP_VIRTUAL)) {
+      GST_DEBUG_OBJECT (obj, "mixer device is a virtual device, skipping");
+      continue;
+    }
+
+    mixers = g_list_insert_sorted (mixers, g_memdup (&mi, sizeof (mi)),
+        (GCompareFunc) oss4_mixerinfo_priority_cmp);
+  }
+
+  /* then create device list according to priority */
+  for (m = mixers; m != NULL; m = m->next) {
+    struct oss_mixerinfo *mi = (struct oss_mixerinfo *) m->data;
+
+    GST_LOG_OBJECT (obj, "mixer device: '%s'", mi->devnode);
+    devices = g_list_prepend (devices, g_strdup (mi->devnode));
+    g_free (m->data);
+  }
+  g_list_free (mixers);
+  mixers = NULL;
+
+  return g_list_reverse (devices);
+}
+
+static GList *
+gst_oss4_property_probe_get_audio_devices (GstObject * obj, int fd,
+    struct oss_sysinfo *si, int cap_mask)
+{
+  GList *devices = NULL;
+  int i;
+
+  GST_LOG_OBJECT (obj, "%d audio/dsp devices", si->numaudios);
+
+  for (i = 0; i < si->numaudios; ++i) {
+    struct oss_audioinfo ai = { 0, };
+
+    ai.dev = i;
+    if (ioctl (fd, SNDCTL_AUDIOINFO, &ai) == -1) {
+      GST_DEBUG_OBJECT (obj, "AUDIOINFO ioctl for device %d failed", i);
+      continue;
+    }
+
+    if ((ai.caps & cap_mask) == 0) {
+      GST_DEBUG_OBJECT (obj, "audio device %d is not an %s device", i,
+          (cap_mask == PCM_CAP_OUTPUT) ? "output" : "input");
+      continue;
+    }
+
+    if (!ai.enabled) {
+      GST_DEBUG_OBJECT (obj, "audio device %d is not usable/enabled", i);
+      continue;
+    }
+
+    GST_DEBUG_OBJECT (obj, "audio device %d looks ok: %s (\"%s\")", i,
+        ai.devnode, ai.name);
+
+    devices = g_list_prepend (devices, g_strdup (ai.devnode));
+  }
+
+  return g_list_reverse (devices);
+}
+
+static GValueArray *
+gst_oss4_property_probe_get_values (GstPropertyProbe * probe,
+    guint prop_id, const GParamSpec * pspec)
+{
+  struct oss_sysinfo si = { {0,}, };
+  GstElementClass *klass;
+  GValueArray *array = NULL;
+  GstObject *obj;
+  GList *devices, *l;
+  int cap_mask, fd = -1;
+
+  if (!g_str_equal (pspec->name, "device")) {
+    G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
+    return NULL;
+  }
+
+  obj = GST_OBJECT (probe);
+
+  GST_OBJECT_LOCK (obj);
+
+  /* figure out whether the element is a source or sink */
+  klass = GST_ELEMENT_GET_CLASS (GST_ELEMENT (probe));
+  if (GST_IS_OSS4_SINK (probe)) {
+    GST_DEBUG_OBJECT (probe, "probing available output devices");
+    cap_mask = PCM_CAP_OUTPUT;
+    fd = GST_OSS4_SINK (probe)->fd;
+  } else if (GST_IS_OSS4_SOURCE (probe)) {
+    GST_DEBUG_OBJECT (probe, "probing available input devices");
+    cap_mask = PCM_CAP_INPUT;
+    fd = GST_OSS4_SOURCE (probe)->fd;
+  } else if (GST_IS_OSS4_MIXER (probe)) {
+    fd = GST_OSS4_MIXER (probe)->fd;
+    cap_mask = 0;
+  } else {
+    GST_OBJECT_UNLOCK (obj);
+    g_return_val_if_reached (NULL);
+  }
+
+  /* copy fd if it's open, so we can just unconditionally close() later */
+  if (fd != -1)
+    fd = dup (fd);
+
+  /* this will also catch the unlikely case where the above dup() failed */
+  if (fd == -1) {
+    fd = open ("/dev/mixer", O_RDONLY | O_NONBLOCK, 0);
+    if (fd < 0)
+      goto open_failed;
+    else if (!gst_oss4_audio_check_version (GST_OBJECT (probe), fd))
+      goto legacy_oss;
+  }
+
+  if (ioctl (fd, SNDCTL_SYSINFO, &si) == -1)
+    goto no_sysinfo;
+
+  if (cap_mask != 0) {
+    devices =
+        gst_oss4_property_probe_get_audio_devices (obj, fd, &si, cap_mask);
+  } else {
+    devices = gst_oss4_property_probe_get_mixer_devices (obj, fd, &si);
+  }
+
+  if (devices == NULL) {
+    GST_DEBUG_OBJECT (obj, "No devices found");
+    goto done;
+  }
+
+  array = g_value_array_new (1);
+
+  for (l = devices; l != NULL; l = l->next) {
+    GValue val = { 0, };
+
+    g_value_init (&val, G_TYPE_STRING);
+    g_value_take_string (&val, (gchar *) l->data);
+    l->data = NULL;
+    g_value_array_append (array, &val);
+    g_value_unset (&val);
+  }
+
+  GST_OBJECT_UNLOCK (obj);
+
+  g_list_free (devices);
+
+done:
+
+  close (fd);
+
+  return array;
+
+/* ERRORS */
+open_failed:
+  {
+    GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+    GST_WARNING_OBJECT (probe, "Can't open file descriptor to probe "
+        "available devices: %s", g_strerror (errno));
+    return NULL;
+  }
+legacy_oss:
+  {
+    close (fd);
+    GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+    GST_DEBUG_OBJECT (probe, "Legacy OSS (ie. not OSSv4), not supported");
+    return NULL;
+  }
+no_sysinfo:
+  {
+    close (fd);
+    GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+    GST_WARNING_OBJECT (probe, "Can't open file descriptor to probe "
+        "available devices: %s", g_strerror (errno));
+    return NULL;
+  }
+}
+
+static void
+gst_oss4_property_probe_interface_init (GstPropertyProbeInterface * iface)
+{
+  iface->get_properties = gst_oss4_property_probe_get_properties;
+  iface->probe_property = gst_oss4_property_probe_probe_property;
+  iface->needs_probe = gst_oss4_property_probe_needs_probe;
+  iface->get_values = gst_oss4_property_probe_get_values;
+}
+
+void
+gst_oss4_add_property_probe_interface (GType type)
+{
+  static const GInterfaceInfo probe_iface_info = {
+    (GInterfaceInitFunc) gst_oss4_property_probe_interface_init,
+    NULL,
+    NULL,
+  };
+
+  g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
+      &probe_iface_info);
+}
diff --git a/sys/oss4/oss4-property-probe.h b/sys/oss4/oss4-property-probe.h
new file mode 100644
index 00000000..26ee8b79
--- /dev/null
+++ b/sys/oss4/oss4-property-probe.h
@@ -0,0 +1,34 @@
+/* GStreamer OSS4 audio property probe interface implementation
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef GST_OSS4_PROPERTY_PROBE_H
+#define GST_OSS4_PROPERTY_PROBE_H
+
+#include <gst/interfaces/propertyprobe.h>
+
+void      gst_oss4_add_property_probe_interface (GType type);
+
+gboolean  gst_oss4_property_probe_find_device_name (GstObject   * obj,
+                                                    int           fd,
+                                                    const gchar * device_handle,
+                                                    gchar      ** device_name);
+
+#endif /* GST_OSS4_PROPERTY_PROBE_H */
+
+
diff --git a/sys/oss4/oss4-sink.c b/sys/oss4/oss4-sink.c
new file mode 100644
index 00000000..1cff22a5
--- /dev/null
+++ b/sys/oss4/oss4-sink.c
@@ -0,0 +1,571 @@
+/* GStreamer OSS4 audio sink
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+/**
+ * SECTION:element-oss4sink
+ * @short_description: output sound using OSS4
+ *
+ * <refsect2>
+ * <para>
+ * This element lets you output sound using the Open Sound System (OSS)
+ * version 4.
+ * </para>
+ * <para>
+ * Note that you should almost always use generic audio conversion elements
+ * like audioconvert and audioresample in front of an audiosink to make sure
+ * your pipeline works under all circumstances (those conversion elements will
+ * act in passthrough-mode if no conversion is necessary).
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! oss4sink
+ * </programlisting>
+ * will output a sine wave (continuous beep sound) to your sound card (with
+ * a very low volume as precaution).
+ * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! oss4sink
+ * </programlisting>
+ * will play an Ogg/Vorbis audio file and output it using the Open Sound System
+ * version 4.
+ * </para>
+ * </refsect2>
+ *
+ * Since: 0.10.7
+ */
+
+/* TODO: - add "volume" property for stream volume control and intercept tags
+ *         to set stream title
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-audio.h"
+#include "oss4-sink.h"
+#include "oss4-property-probe.h"
+#include "oss4-soundcard.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4sink_debug);
+#define GST_CAT_DEFAULT oss4sink_debug
+
+static void gst_oss4_sink_init_interfaces (GType type);
+static void gst_oss4_sink_dispose (GObject * object);
+static void gst_oss4_sink_finalise (GObject * object);
+
+static void gst_oss4_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_oss4_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_oss4_sink_getcaps (GstBaseSink * bsink);
+static gboolean gst_oss4_sink_open (GstAudioSink * asink,
+    gboolean silent_errors);
+static gboolean gst_oss4_sink_open_func (GstAudioSink * asink);
+static gboolean gst_oss4_sink_close (GstAudioSink * asink);
+static gboolean gst_oss4_sink_prepare (GstAudioSink * asink,
+    GstRingBufferSpec * spec);
+static gboolean gst_oss4_sink_unprepare (GstAudioSink * asink);
+static guint gst_oss4_sink_write (GstAudioSink * asink, gpointer data,
+    guint length);
+static guint gst_oss4_sink_delay (GstAudioSink * asink);
+static void gst_oss4_sink_reset (GstAudioSink * asink);
+
+#define DEFAULT_DEVICE      NULL
+#define DEFAULT_DEVICE_NAME NULL
+
+enum
+{
+  PROP_0,
+  PROP_DEVICE,
+  PROP_DEVICE_NAME
+};
+
+GST_BOILERPLATE_FULL (GstOss4Sink, gst_oss4_sink, GstAudioSink,
+    GST_TYPE_AUDIO_SINK, gst_oss4_sink_init_interfaces);
+
+static void
+gst_oss4_sink_dispose (GObject * object)
+{
+  GstOss4Sink *osssink = GST_OSS4_SINK (object);
+
+  if (osssink->probed_caps) {
+    gst_caps_unref (osssink->probed_caps);
+    osssink->probed_caps = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_oss4_sink_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  GstPadTemplate *templ;
+
+  gst_element_class_set_details_simple (element_class,
+      "OSS v4 Audio Sink", "Sink/Audio",
+      "Output to a sound card via OSS version 4",
+      "Tim-Philipp Müller <tim centricular net>");
+
+  templ = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+      gst_oss4_audio_get_template_caps ());
+  gst_element_class_add_pad_template (element_class, templ);
+}
+
+static void
+gst_oss4_sink_class_init (GstOss4SinkClass * klass)
+{
+  GstAudioSinkClass *audiosink_class = (GstAudioSinkClass *) klass;
+  GstBaseSinkClass *basesink_class = (GstBaseSinkClass *) klass;
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss4_sink_dispose);
+  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_oss4_sink_finalise);
+  gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss4_sink_get_property);
+  gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss4_sink_set_property);
+
+  g_object_class_install_property (gobject_class, PROP_DEVICE,
+      g_param_spec_string ("device", "Device",
+          "OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) "
+          "(NULL = use first available playback device)",
+          DEFAULT_DEVICE, G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+      g_param_spec_string ("device-name", "Device name",
+          "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+          G_PARAM_READABLE));
+
+  basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss4_sink_getcaps);
+
+  audiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss4_sink_open_func);
+  audiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss4_sink_close);
+  audiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss4_sink_prepare);
+  audiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss4_sink_unprepare);
+  audiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss4_sink_write);
+  audiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss4_sink_delay);
+  audiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss4_sink_reset);
+}
+
+static void
+gst_oss4_sink_init (GstOss4Sink * osssink, GstOss4SinkClass * klass)
+{
+  const gchar *device;
+
+  device = g_getenv ("AUDIODEV");
+  if (device == NULL)
+    device = DEFAULT_DEVICE;
+  osssink->device = g_strdup (device);
+  osssink->fd = -1;
+  osssink->bytes_per_sample = 0;
+  osssink->probed_caps = NULL;
+  osssink->device_name = NULL;
+}
+
+static void
+gst_oss4_sink_finalise (GObject * object)
+{
+  GstOss4Sink *osssink = GST_OSS4_SINK (object);
+
+  g_free (osssink->device);
+  osssink->device = NULL;
+
+  g_list_free (osssink->property_probe_list);
+  osssink->property_probe_list = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_oss4_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOss4Sink *oss = GST_OSS4_SINK (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      GST_OBJECT_LOCK (oss);
+      if (oss->fd == -1) {
+        g_free (oss->device);
+        oss->device = g_value_dup_string (value);
+        if (oss->probed_caps) {
+          gst_caps_unref (oss->probed_caps);
+          oss->probed_caps = NULL;
+        }
+        g_free (oss->device_name);
+        oss->device_name = NULL;
+      } else {
+        g_warning ("%s: can't change \"device\" property while audio sink "
+            "is open", GST_OBJECT_NAME (oss));
+      }
+      GST_OBJECT_UNLOCK (oss);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_oss4_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstOss4Sink *oss = GST_OSS4_SINK (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      GST_OBJECT_LOCK (oss);
+      g_value_set_string (value, oss->device);
+      GST_OBJECT_UNLOCK (oss);
+      break;
+    case PROP_DEVICE_NAME:
+      GST_OBJECT_LOCK (oss);
+      if (oss->fd == -1 && oss->device != NULL) {
+        /* If device is set, try to retrieve the name even if we're not open */
+        if (gst_oss4_sink_open (GST_AUDIO_SINK (oss), TRUE)) {
+          g_value_set_string (value, oss->device_name);
+          gst_oss4_sink_close (GST_AUDIO_SINK (oss));
+        } else {
+          g_value_set_string (value, oss->device_name);
+        }
+      } else {
+        g_value_set_string (value, oss->device_name);
+      }
+      GST_OBJECT_UNLOCK (oss);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_oss4_sink_getcaps (GstBaseSink * bsink)
+{
+  GstOss4Sink *oss;
+  GstCaps *caps;
+
+  oss = GST_OSS4_SINK (bsink);
+
+  if (oss->fd == -1) {
+    caps = gst_caps_copy (gst_oss4_audio_get_template_caps ());
+  } else if (oss->probed_caps) {
+    caps = gst_caps_copy (oss->probed_caps);
+  } else {
+    caps = gst_oss4_audio_probe_caps (GST_OBJECT (oss), oss->fd);
+    if (caps != NULL && !gst_caps_is_empty (caps)) {
+      oss->probed_caps = gst_caps_copy (caps);
+    }
+  }
+
+  return caps;
+}
+
+/* note: we must not take the object lock here unless we fix up get_property */
+static gboolean
+gst_oss4_sink_open (GstAudioSink * asink, gboolean silent_errors)
+{
+  GstOss4Sink *oss;
+  gchar *device;
+  int mode;
+
+  oss = GST_OSS4_SINK (asink);
+
+  if (oss->device)
+    device = g_strdup (oss->device);
+  else
+    device = gst_oss4_audio_find_device (GST_OBJECT_CAST (oss));
+
+  /* desperate times, desperate measures */
+  if (device == NULL)
+    device = g_strdup ("/dev/dsp0");
+
+  GST_INFO_OBJECT (oss, "Trying to open OSS4 device '%s'", device);
+
+  /* we open in non-blocking mode even if we don't really want to do writes
+   * non-blocking because we can't be sure that this is really a genuine
+   * OSS4 device with well-behaved drivers etc. We really don't want to
+   * hang forever under any circumstances. */
+  oss->fd = open (device, O_WRONLY | O_NONBLOCK, 0);
+  if (oss->fd == -1) {
+    switch (errno) {
+      case EBUSY:
+        goto busy;
+      case EACCES:
+        goto no_permission;
+      default:
+        goto open_failed;
+    }
+  }
+
+  GST_INFO_OBJECT (oss, "Opened device '%s'", device);
+
+  /* Make sure it's OSS4. If it's old OSS, let osssink handle it */
+  if (!gst_oss4_audio_check_version (GST_OBJECT_CAST (oss), oss->fd))
+    goto legacy_oss;
+
+  /* now remove the non-blocking flag. */
+  mode = fcntl (oss->fd, F_GETFL);
+  mode &= ~O_NONBLOCK;
+  if (fcntl (oss->fd, F_SETFL, mode) < 0) {
+    /* some drivers do no support unsetting the non-blocking flag, try to
+     * close/open the device then. This is racy but we error out properly. */
+    GST_WARNING_OBJECT (oss, "failed to unset O_NONBLOCK (buggy driver?), "
+        "will try to re-open device now");
+    gst_oss4_sink_close (asink);
+    if ((oss->fd = open (device, O_WRONLY, 0)) == -1)
+      goto non_block;
+  }
+
+  oss->open_device = device;
+
+  /* not using ENGINEINFO here because it sometimes returns a different and
+   * less useful name than AUDIOINFO for the same device */
+  if (!gst_oss4_property_probe_find_device_name (GST_OBJECT (oss), oss->fd,
+          oss->open_device, &oss->device_name)) {
+    oss->device_name = NULL;
+  }
+
+  /* list output routings, for informational purposes only so far */
+  {
+    oss_mixer_enuminfo routings = { 0, };
+    guint i;
+
+    if (ioctl (oss->fd, SNDCTL_DSP_GET_PLAYTGT_NAMES, &routings) != -1) {
+      GST_LOG_OBJECT (oss, "%u output routings (static list: %d)",
+          routings.nvalues, !!(routings.version == 0));
+      for (i = 0; i < routings.nvalues; ++i) {
+        GST_LOG_OBJECT (oss, "  output routing %d: %s", i,
+            &routings.strings[routings.strindex[i]]);
+      }
+    }
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+busy:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
+          (_("Could not open audio device for playback. "
+                  "Device is being used by another application.")), (NULL));
+    }
+    g_free (device);
+    return FALSE;
+  }
+no_permission:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
+          (_("Could not open audio device for playback."
+                  "You don't have permission to open the device.")),
+          GST_ERROR_SYSTEM);
+    }
+    g_free (device);
+    return FALSE;
+  }
+open_failed:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
+          (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
+    }
+    g_free (device);
+    return FALSE;
+  }
+legacy_oss:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
+          (_("Could not open audio device for playback."
+                  "This version of the Open Sound System is not supported by this "
+                  "element.")), ("Try the 'osssink' element instead"));
+    }
+    gst_oss4_sink_close (asink);
+    g_free (device);
+    return FALSE;
+  }
+non_block:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
+          ("Unable to set device %s into non-blocking mode: %s",
+              oss->device, g_strerror (errno)));
+    }
+    g_free (device);
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_oss4_sink_open_func (GstAudioSink * asink)
+{
+  return gst_oss4_sink_open (asink, FALSE);
+}
+
+static gboolean
+gst_oss4_sink_close (GstAudioSink * asink)
+{
+  GstOss4Sink *oss = GST_OSS4_SINK (asink);
+
+  if (oss->fd != -1) {
+    GST_DEBUG_OBJECT (oss, "closing device");
+    close (oss->fd);
+    oss->fd = -1;
+  }
+
+  oss->bytes_per_sample = 0;
+  /* we keep the probed caps cached, at least until the device changes */
+
+  g_free (oss->open_device);
+  oss->open_device = NULL;
+
+  g_free (oss->device_name);
+  oss->device_name = NULL;
+
+  if (oss->probed_caps) {
+    gst_caps_unref (oss->probed_caps);
+    oss->probed_caps = NULL;
+  }
+
+  return TRUE;
+}
+
+static gboolean
+gst_oss4_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
+{
+  GstOss4Sink *oss;
+
+  oss = GST_OSS4_SINK (asink);
+
+  if (!gst_oss4_audio_set_format (GST_OBJECT_CAST (oss), oss->fd, spec)) {
+    GST_WARNING_OBJECT (oss, "Couldn't set requested format %" GST_PTR_FORMAT,
+        spec->caps);
+    return FALSE;
+  }
+
+  oss->bytes_per_sample = spec->bytes_per_sample;
+  return TRUE;
+}
+
+static gboolean
+gst_oss4_sink_unprepare (GstAudioSink * asink)
+{
+  /* could do a SNDCTL_DSP_HALT, but the OSS manual recommends a close/open,
+   * since HALT won't properly reset some devices, apparently */
+
+  if (!gst_oss4_sink_close (asink))
+    goto couldnt_close;
+
+  if (!gst_oss4_sink_open_func (asink))
+    goto couldnt_reopen;
+
+  return TRUE;
+
+  /* ERRORS */
+couldnt_close:
+  {
+    GST_DEBUG_OBJECT (asink, "Couldn't close the audio device");
+    return FALSE;
+  }
+couldnt_reopen:
+  {
+    GST_DEBUG_OBJECT (asink, "Couldn't reopen the audio device");
+    return FALSE;
+  }
+}
+
+static guint
+gst_oss4_sink_write (GstAudioSink * asink, gpointer data, guint length)
+{
+  GstOss4Sink *oss;
+  int n;
+
+  oss = GST_OSS4_SINK_CAST (asink);
+
+  n = write (oss->fd, data, length);
+  GST_LOG_OBJECT (asink, "wrote %d/%d samples, %d bytes",
+      n / oss->bytes_per_sample, length / oss->bytes_per_sample, n);
+
+  if (G_UNLIKELY (n < 0)) {
+    switch (errno) {
+      case ENOTSUP:
+      case EACCES:{
+        /* This is the most likely cause, I think */
+        GST_ELEMENT_ERROR (asink, RESOURCE, WRITE,
+            (_("Playback is not supported by this audio device.")),
+            ("write: %s (device: %s) (maybe this is an input-only device?)",
+                g_strerror (errno), oss->open_device));
+        break;
+      }
+      default:{
+        GST_ELEMENT_ERROR (asink, RESOURCE, WRITE,
+            (_("Audio playback error.")),
+            ("write: %s (device: %s)", g_strerror (errno), oss->open_device));
+        break;
+      }
+    }
+  }
+
+  return n;
+}
+
+static guint
+gst_oss4_sink_delay (GstAudioSink * asink)
+{
+  GstOss4Sink *oss;
+  gint delay = -1;
+
+  oss = GST_OSS4_SINK_CAST (asink);
+
+  if (ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay) < 0 || delay < 0) {
+    GST_LOG_OBJECT (oss, "GETODELAY failed");
+    return 0;
+  }
+
+  return delay / oss->bytes_per_sample;
+}
+
+static void
+gst_oss4_sink_reset (GstAudioSink * asink)
+{
+  /* There's nothing we can do here really: OSS can't handle access to the
+   * same device/fd from multiple threads and might deadlock or blow up in
+   * other ways if we try an ioctl SNDCTL_DSP_HALT or similar */
+}
+
+static void
+gst_oss4_sink_init_interfaces (GType type)
+{
+  gst_oss4_add_property_probe_interface (type);
+}
diff --git a/sys/oss4/oss4-sink.h b/sys/oss4/oss4-sink.h
new file mode 100644
index 00000000..01205989
--- /dev/null
+++ b/sys/oss4/oss4-sink.h
@@ -0,0 +1,63 @@
+/* GStreamer OSS4 audio sink
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef GST_OSS4_SINK_H
+#define GST_OSS4_SINK_H
+
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosink.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_SINK            (gst_oss4_sink_get_type())
+#define GST_OSS4_SINK(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_SINK,GstOss4Sink))
+#define GST_OSS4_SINK_CAST(obj)       ((GstOss4Sink *)(obj))
+#define GST_OSS4_SINK_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_SINK,GstOss4SinkClass))
+#define GST_IS_OSS4_SINK(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_SINK))
+#define GST_IS_OSS4_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_SINK))
+
+typedef struct _GstOss4Sink GstOss4Sink;
+typedef struct _GstOss4SinkClass GstOss4SinkClass;
+
+struct _GstOss4Sink {
+  GstAudioSink  audio_sink;
+
+  gchar       * device;             /* NULL if none was set      */
+  gchar       * open_device;        /* the device we opened      */
+  gchar       * device_name;        /* set if the device is open */
+  gint          fd;                 /* -1 if not open            */
+  gint          bytes_per_sample;
+
+  GstCaps     * probed_caps;
+
+  GList       * property_probe_list;
+};
+
+struct _GstOss4SinkClass {
+  GstAudioSinkClass audio_sink_class;
+};
+
+GType  gst_oss4_sink_get_type (void);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_SINK_H */
+
+
diff --git a/sys/oss4/oss4-soundcard.h b/sys/oss4/oss4-soundcard.h
new file mode 100644
index 00000000..8b124892
--- /dev/null
+++ b/sys/oss4/oss4-soundcard.h
@@ -0,0 +1,2067 @@
+/*
+ * Purpose: The C/C++ header file that defines the OSS API.
+ * Description:
+ * This header file contains all the declarations required to compile OSS
+ * programs. The latest version is always installed together with OSS
+ * use of the latest version is strongly recommended.
+ *
+ * {!notice This header file contains many obsolete definitions
+ * (for compatibility with older applications that still ned them).
+ * Do not use this file as a reference manual of OSS.
+ * Please check the OSS Programmer's guide for descriptions
+ * of the supported API details (http://manuals.opensound.com/developer).}
+ */
+
+#ifndef SOUNDCARD_H
+#define SOUNDCARD_H
+
+#define COPYING40 Copyright (C) 4Front Technologies 2000-2006. Released under the BSD license.
+
+#if defined(__cplusplus)
+#define EXTERNC extern "C"
+#else
+#define EXTERNC extern
+#endif /* EXTERN_C_WRAPPERS */
+
+#define OSS_VERSION	0x040090 // Pre 4.1
+
+#define SOUND_VERSION	OSS_VERSION
+#define OPEN_SOUND_SYSTEM
+
+#if defined(__hpux) && !defined(_HPUX_SOURCE)
+#	error "-D_HPUX_SOURCE must be used when compiling OSS applications"
+#endif
+
+#ifdef __hpux
+#include <sys/ioctl.h>
+#endif
+
+#ifdef linux
+/* In Linux we need to be prepared for cross compiling */
+#include <linux/ioctl.h>
+#else
+# ifdef __FreeBSD__
+#    include <sys/ioccom.h>
+# else
+#    include <sys/ioctl.h>
+# endif
+#endif
+
+#ifndef __SIOWR
+#if defined(__hpux) || (defined(_IOWR) && (defined(_AIX) || (!defined(sun) && !defined(sparc) && !defined(__INCioctlh) && !defined(__Lynx__))))
+
+/* 
+ * Make sure the ioctl macros are compatible with the ones already used
+ * by this operating system.
+ */
+#define	SIOCPARM_MASK	IOCPARM_MASK
+#define	SIOC_VOID	IOC_VOID
+#define	SIOC_OUT	IOC_OUT
+#define	SIOC_IN		IOC_IN
+#define	SIOC_INOUT	IOC_INOUT
+#define __SIOC_SIZE	_IOC_SIZE
+#define __SIOC_DIR	_IOC_DIR
+#define __SIOC_NONE	_IOC_NONE
+#define __SIOC_READ	_IOC_READ
+#define __SIOC_WRITE	_IOC_WRITE
+#define	__SIO		_IO
+#define	__SIOR		_IOR
+#define	__SIOW		_IOW
+#define	__SIOWR		_IOWR
+#else
+
+/* #define	SIOCTYPE		(0xff<<8) */
+#define	SIOCPARM_MASK	0x1fff	/* parameters must be < 8192 bytes */
+#define	SIOC_VOID	0x00000000	/* no parameters */
+#define	SIOC_OUT	0x20000000	/* copy out parameters */
+#define	SIOC_IN		0x40000000	/* copy in parameters */
+#define	SIOC_INOUT	(SIOC_IN|SIOC_OUT)
+
+#define	__SIO(x,y)	((int)(SIOC_VOID|(x<<8)|y))
+#define	__SIOR(x,y,t)	((int)(SIOC_OUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y))
+#define	__SIOW(x,y,t)	((int)(SIOC_IN|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y))
+#define	__SIOWR(x,y,t)	((int)(SIOC_INOUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y))
+#define __SIOC_SIZE(x)	((x>>16)&SIOCPARM_MASK)
+#define __SIOC_DIR(x)	(x & 0xf0000000)
+#define __SIOC_NONE	SIOC_VOID
+#define __SIOC_READ	SIOC_OUT
+#define __SIOC_WRITE	SIOC_IN
+#  endif /* _IOWR */
+#endif /* !__SIOWR */
+
+#define OSS_LONGNAME_SIZE	64
+#define OSS_LABEL_SIZE		16
+#define OSS_DEVNODE_SIZE	32
+typedef char oss_longname_t[OSS_LONGNAME_SIZE];
+typedef char oss_label_t[OSS_LABEL_SIZE];
+typedef char oss_devnode_t[OSS_DEVNODE_SIZE];
+
+#ifndef DISABLE_SEQUENCER
+/*
+ ****************************************************************************
+ * IOCTL Commands for /dev/sequencer and /dev/music (AKA /dev/sequencer2)
+ *
+ * Note that this interface is obsolete and no longer developed. New
+ * applications should use /dev/midi instead.
+ ****************************************************************************/
+#define SNDCTL_SEQ_RESET		__SIO  ('Q', 0)
+#define SNDCTL_SEQ_SYNC			__SIO  ('Q', 1)
+#define SNDCTL_SYNTH_INFO		__SIOWR('Q', 2, struct synth_info)
+#define SNDCTL_SEQ_CTRLRATE		__SIOWR('Q', 3, int)	/* Set/get timer resolution (HZ) */
+#define SNDCTL_SEQ_GETOUTCOUNT		__SIOR ('Q', 4, int)
+#define SNDCTL_SEQ_GETINCOUNT		__SIOR ('Q', 5, int)
+#define SNDCTL_SEQ_PERCMODE		__SIOW ('Q', 6, int)
+#define SNDCTL_FM_LOAD_INSTR		__SIOW ('Q', 7, struct sbi_instrument)	/* Obsolete. Don't use!!!!!! */
+#define SNDCTL_SEQ_TESTMIDI		__SIOW ('Q', 8, int)
+#define SNDCTL_SEQ_RESETSAMPLES		__SIOW ('Q', 9, int)
+#define SNDCTL_SEQ_NRSYNTHS		__SIOR ('Q',10, int)
+#define SNDCTL_SEQ_NRMIDIS		__SIOR ('Q',11, int)
+#define SNDCTL_MIDI_INFO		__SIOWR('Q',12, struct midi_info)	/* OBSOLETE - use SNDCTL_MIDIINFO instead */
+#define SNDCTL_SEQ_THRESHOLD		__SIOW ('Q',13, int)
+#define SNDCTL_SYNTH_MEMAVL		__SIOWR('Q',14, int)	/* in=dev#, out=memsize */
+#define SNDCTL_FM_4OP_ENABLE		__SIOW ('Q',15, int)	/* in=dev# */
+#define SNDCTL_SEQ_PANIC		__SIO  ('Q',17)
+#define SNDCTL_SEQ_OUTOFBAND		__SIOW ('Q',18, struct seq_event_rec)
+#define SNDCTL_SEQ_GETTIME		__SIOR ('Q',19, int)
+#define SNDCTL_SYNTH_ID			__SIOWR('Q',20, struct synth_info)
+#define SNDCTL_SYNTH_CONTROL		__SIOWR('Q',21, struct synth_control)
+#define SNDCTL_SYNTH_REMOVESAMPLE	__SIOWR('Q',22, struct remove_sample)	/* Reserved for future use */
+#define SNDCTL_SEQ_TIMING_ENABLE	__SIO  ('Q', 23)	/* Enable incoming MIDI timing messages */
+#define SNDCTL_SEQ_ACTSENSE_ENABLE	__SIO  ('Q', 24)	/* Enable incoming active sensing messages */
+#define SNDCTL_SEQ_RT_ENABLE		__SIO  ('Q', 25)	/* Enable other incoming realtime messages */
+
+typedef struct synth_control
+{
+  int devno;			/* Synthesizer # */
+  char data[4000];		/* Device spesific command/data record */
+} synth_control;
+
+typedef struct remove_sample
+{
+  int devno;			/* Synthesizer # */
+  int bankno;			/* MIDI bank # (0=General MIDI) */
+  int instrno;			/* MIDI instrument number */
+} remove_sample;
+
+typedef struct seq_event_rec
+{
+  unsigned char arr[8];
+} seq_event_rec;
+
+#define SNDCTL_TMR_TIMEBASE		__SIOWR('T', 1, int)
+#define SNDCTL_TMR_START		__SIO  ('T', 2)
+#define SNDCTL_TMR_STOP			__SIO  ('T', 3)
+#define SNDCTL_TMR_CONTINUE		__SIO  ('T', 4)
+#define SNDCTL_TMR_TEMPO		__SIOWR('T', 5, int)
+#define SNDCTL_TMR_SOURCE		__SIOWR('T', 6, int)
+#	define TMR_INTERNAL		0x00000001
+#	define TMR_EXTERNAL		0x00000002
+#		define TMR_MODE_MIDI	0x00000010
+#		define TMR_MODE_FSK	0x00000020
+#		define TMR_MODE_CLS	0x00000040
+#		define TMR_MODE_SMPTE	0x00000080
+#define SNDCTL_TMR_METRONOME		__SIOW ('T', 7, int)
+#define SNDCTL_TMR_SELECT		__SIOW ('T', 8, int)
+
+/*
+ * Sample loading mechanism for internal synthesizers (/dev/sequencer)
+ * (for the .PAT format).
+ */
+
+struct patch_info
+{
+  unsigned short key;		/* Use WAVE_PATCH here */
+#define WAVE_PATCH	_PATCHKEY(0x04)
+#define GUS_PATCH	WAVE_PATCH
+#define WAVEFRONT_PATCH _PATCHKEY(0x06)
+
+  short device_no;		/* Synthesizer number */
+  short instr_no;		/* Midi pgm# */
+
+  unsigned int mode;
+/*
+ * The least significant byte has the same format than the GUS .PAT
+ * files
+ */
+#define WAVE_16_BITS	0x01	/* bit 0 = 8 or 16 bit wave data. */
+#define WAVE_UNSIGNED	0x02	/* bit 1 = Signed - Unsigned data. */
+#define WAVE_LOOPING	0x04	/* bit 2 = looping enabled-1. */
+#define WAVE_BIDIR_LOOP	0x08	/* bit 3 = Set is bidirectional looping. */
+#define WAVE_LOOP_BACK	0x10	/* bit 4 = Set is looping backward. */
+#define WAVE_SUSTAIN_ON	0x20	/* bit 5 = Turn sustaining on. (Env. pts. 3) */
+#define WAVE_ENVELOPES	0x40	/* bit 6 = Enable envelopes - 1 */
+#define WAVE_FAST_RELEASE 0x80	/* bit 7 = Shut off immediately after note off */
+  /*  (use the env_rate/env_offs fields). */
+/* Linux specific bits */
+#define WAVE_VIBRATO	0x00010000	/* The vibrato info is valid */
+#define WAVE_TREMOLO	0x00020000	/* The tremolo info is valid */
+#define WAVE_SCALE	0x00040000	/* The scaling info is valid */
+#define WAVE_FRACTIONS	0x00080000	/* Fraction information is valid */
+/* Reserved bits */
+#define WAVE_ROM	0x40000000	/* For future use */
+#define WAVE_MULAW	0x20000000	/* For future use */
+/* Other bits must be zeroed */
+
+  int len;			/* Size of the wave data in bytes */
+  int loop_start, loop_end;	/* Byte offsets from the beginning */
+
+/* 
+ * The base_freq and base_note fields are used when computing the
+ * playback speed for a note. The base_note defines the tone frequency
+ * which is heard if the sample is played using the base_freq as the
+ * playback speed.
+ *
+ * The low_note and high_note fields define the minimum and maximum note
+ * frequencies for which this sample is valid. It is possible to define
+ * more than one samples for an instrument number at the same time. The
+ * low_note and high_note fields are used to select the most suitable one.
+ *
+ * The fields base_note, high_note and low_note should contain
+ * the note frequency multiplied by 1000. For example value for the
+ * middle A is 440*1000.
+ */
+
+  unsigned int base_freq;
+  unsigned int base_note;
+  unsigned int high_note;
+  unsigned int low_note;
+  int panning;			/* -128=left, 127=right */
+  int detuning;
+
+  /* Envelope. Enabled by mode bit WAVE_ENVELOPES  */
+  unsigned char env_rate[6];	/* GUS HW ramping rate */
+  unsigned char env_offset[6];	/* 255 == 100% */
+
+  /* 
+   * The tremolo, vibrato and scale info are not supported yet.
+   * Enable by setting the mode bits WAVE_TREMOLO, WAVE_VIBRATO or
+   * WAVE_SCALE
+   */
+
+  unsigned char tremolo_sweep;
+  unsigned char tremolo_rate;
+  unsigned char tremolo_depth;
+
+  unsigned char vibrato_sweep;
+  unsigned char vibrato_rate;
+  unsigned char vibrato_depth;
+
+  int scale_frequency;
+  unsigned int scale_factor;	/* from 0 to 2048 or 0 to 2 */
+
+  int volume;
+  int fractions;
+  int reserved1;
+  int spare[2];
+  char data[1];			/* The waveform data starts here */
+};
+
+struct sysex_info
+{
+  short key;			/* Use SYSEX_PATCH or MAUI_PATCH here */
+#define SYSEX_PATCH	_PATCHKEY(0x05)
+#define MAUI_PATCH	_PATCHKEY(0x06)
+  short device_no;		/* Synthesizer number */
+  int len;			/* Size of the sysex data in bytes */
+  unsigned char data[1];	/* Sysex data starts here */
+};
+
+/*
+ * /dev/sequencer input events.
+ *
+ * The data written to the /dev/sequencer is a stream of events. Events
+ * are records of 4 or 8 bytes. The first byte defines the size. 
+ * Any number of events can be written with a write call. There
+ * is a set of macros for sending these events. Use these macros if you
+ * want to maximize portability of your program.
+ *
+ * Events SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO. Are also input events.
+ * (All input events are currently 4 bytes long. Be prepared to support
+ * 8 byte events also. If you receive any event having first byte >= 128,
+ * it's a 8 byte event.
+ *
+ * The events are documented at the end of this file.
+ *
+ * Normal events (4 bytes)
+ * There is also a 8 byte version of most of the 4 byte events. The
+ * 8 byte one is recommended.
+ *
+ * NOTE! All 4 byte events are now obsolete. Applications should not write
+ *       them. However 4 byte events are still used as inputs from
+ *       /dev/sequencer (/dev/music uses only 8 byte ones).
+ */
+#define SEQ_NOTEOFF		0
+#define SEQ_FMNOTEOFF		SEQ_NOTEOFF	/* Just old name */
+#define SEQ_NOTEON		1
+#define	SEQ_FMNOTEON		SEQ_NOTEON
+#define SEQ_WAIT		TMR_WAIT_ABS
+#define SEQ_PGMCHANGE		3
+#define SEQ_FMPGMCHANGE		SEQ_PGMCHANGE
+#define SEQ_SYNCTIMER		TMR_START
+#define SEQ_MIDIPUTC		5
+#define SEQ_DRUMON		6		/*** OBSOLETE ***/
+#define SEQ_DRUMOFF		7		/*** OBSOLETE ***/
+#define SEQ_ECHO		TMR_ECHO	/* For synching programs with output */
+#define SEQ_AFTERTOUCH		9
+#define SEQ_CONTROLLER		10
+#define SEQ_BALANCE		11
+#define SEQ_VOLMODE             12
+
+/************************************
+ *	Midi controller numbers	    *
+ ************************************/
+/*
+ * Controllers 0 to 31 (0x00 to 0x1f) and
+ * 32 to 63 (0x20 to 0x3f) are continuous
+ * controllers.
+ * In the MIDI 1.0 these controllers are sent using
+ * two messages. Controller numbers 0 to 31 are used
+ * to send the MSB and the controller numbers 32 to 63
+ * are for the LSB. Note that just 7 bits are used in MIDI bytes.
+ */
+
+#define	   CTL_BANK_SELECT		0x00
+#define	   CTL_MODWHEEL			0x01
+#define    CTL_BREATH			0x02
+/*		undefined		0x03 */
+#define    CTL_FOOT			0x04
+#define    CTL_PORTAMENTO_TIME		0x05
+#define    CTL_DATA_ENTRY		0x06
+#define    CTL_MAIN_VOLUME		0x07
+#define    CTL_BALANCE			0x08
+/*		undefined		0x09 */
+#define    CTL_PAN			0x0a
+#define    CTL_EXPRESSION		0x0b
+/*		undefined		0x0c */
+/*		undefined		0x0d */
+/*		undefined		0x0e */
+/*		undefined		0x0f */
+#define    CTL_GENERAL_PURPOSE1		0x10
+#define    CTL_GENERAL_PURPOSE2		0x11
+#define    CTL_GENERAL_PURPOSE3		0x12
+#define    CTL_GENERAL_PURPOSE4		0x13
+/*		undefined		0x14 - 0x1f */
+
+/*		undefined		0x20 */
+/* The controller numbers 0x21 to 0x3f are reserved for the */
+/* least significant bytes of the controllers 0x00 to 0x1f. */
+/* These controllers are not recognised by the driver. */
+
+/* Controllers 64 to 69 (0x40 to 0x45) are on/off switches. */
+/* 0=OFF and 127=ON (intermediate values are possible) */
+#define    CTL_DAMPER_PEDAL		0x40
+#define    CTL_SUSTAIN			0x40	/* Alias */
+#define    CTL_HOLD			0x40	/* Alias */
+#define    CTL_PORTAMENTO		0x41
+#define    CTL_SOSTENUTO		0x42
+#define    CTL_SOFT_PEDAL		0x43
+/*		undefined		0x44 */
+#define    CTL_HOLD2			0x45
+/*		undefined		0x46 - 0x4f */
+
+#define    CTL_GENERAL_PURPOSE5		0x50
+#define    CTL_GENERAL_PURPOSE6		0x51
+#define    CTL_GENERAL_PURPOSE7		0x52
+#define    CTL_GENERAL_PURPOSE8		0x53
+/*		undefined		0x54 - 0x5a */
+#define    CTL_EXT_EFF_DEPTH		0x5b
+#define    CTL_TREMOLO_DEPTH		0x5c
+#define    CTL_CHORUS_DEPTH		0x5d
+#define    CTL_DETUNE_DEPTH		0x5e
+#define    CTL_CELESTE_DEPTH		0x5e	/* Alias for the above one */
+#define    CTL_PHASER_DEPTH		0x5f
+#define    CTL_DATA_INCREMENT		0x60
+#define    CTL_DATA_DECREMENT		0x61
+#define    CTL_NONREG_PARM_NUM_LSB	0x62
+#define    CTL_NONREG_PARM_NUM_MSB	0x63
+#define    CTL_REGIST_PARM_NUM_LSB	0x64
+#define    CTL_REGIST_PARM_NUM_MSB	0x65
+/*		undefined		0x66 - 0x78 */
+/*		reserved		0x79 - 0x7f */
+
+/* Pseudo controllers (not midi compatible) */
+#define    CTRL_PITCH_BENDER		255
+#define    CTRL_PITCH_BENDER_RANGE	254
+#define    CTRL_EXPRESSION		253	/* Obsolete */
+#define    CTRL_MAIN_VOLUME		252	/* Obsolete */
+
+/*
+ * Volume mode defines how volumes are used
+ */
+
+#define VOL_METHOD_ADAGIO	1
+#define VOL_METHOD_LINEAR	2
+
+/*
+ * Note! SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO are used also as
+ *	 input events.
+ */
+
+/*
+ * Event codes 0xf0 to 0xfc are reserved for future extensions.
+ */
+
+#define SEQ_FULLSIZE		0xfd	/* Long events */
+/*
+ * SEQ_FULLSIZE events are used for loading patches/samples to the
+ * synthesizer devices. These events are passed directly to the driver
+ * of the associated synthesizer device. There is no limit to the size
+ * of the extended events. These events are not queued but executed
+ * immediately when the write() is called (execution can take several
+ * seconds of time). 
+ *
+ * When a SEQ_FULLSIZE message is written to the device, it must
+ * be written using exactly one write() call. Other events cannot
+ * be mixed to the same write.
+ *	
+ * For FM synths (YM3812/OPL3) use struct sbi_instrument and write it to the 
+ * /dev/sequencer. Don't write other data together with the instrument structure
+ * Set the key field of the structure to FM_PATCH. The device field is used to
+ * route the patch to the corresponding device.
+ *
+ * For wave table use struct patch_info. Initialize the key field
+ * to WAVE_PATCH.
+ */
+#define SEQ_PRIVATE		0xfe	/* Low level HW dependent events (8 bytes) */
+#define SEQ_EXTENDED		0xff	/* Extended events (8 bytes) OBSOLETE */
+
+/*
+ * Record for FM patches
+ */
+
+typedef unsigned char sbi_instr_data[32];
+
+struct sbi_instrument
+{
+  unsigned short key;		/* FM_PATCH or OPL3_PATCH */
+#define FM_PATCH	_PATCHKEY(0x01)
+#define OPL3_PATCH	_PATCHKEY(0x03)
+  short device;			/*  Synth# (0-4)    */
+  int channel;			/*  Program# to be initialized  */
+  sbi_instr_data operators;	/*  Register settings for operator cells (.SBI format)  */
+};
+
+struct synth_info
+{				/* Read only */
+  char name[30];
+  int device;			/* 0-N. INITIALIZE BEFORE CALLING */
+  int synth_type;
+#define SYNTH_TYPE_FM			0
+#define SYNTH_TYPE_SAMPLE		1
+#define SYNTH_TYPE_MIDI			2	/* Midi interface */
+
+  int synth_subtype;
+#define FM_TYPE_ADLIB			0x00
+#define FM_TYPE_OPL3			0x01
+#define MIDI_TYPE_MPU401		0x401
+
+#define SAMPLE_TYPE_BASIC		0x10
+#define SAMPLE_TYPE_GUS			SAMPLE_TYPE_BASIC
+#define SAMPLE_TYPE_WAVEFRONT   	0x11
+
+  int perc_mode;		/* No longer supported */
+  int nr_voices;
+  int nr_drums;			/* Obsolete field */
+  int instr_bank_size;
+  unsigned int capabilities;
+#define SYNTH_CAP_PERCMODE	0x00000001	/* No longer used */
+#define SYNTH_CAP_OPL3		0x00000002	/* Set if OPL3 supported */
+#define SYNTH_CAP_INPUT		0x00000004	/* Input (MIDI) device */
+  int dummies[19];		/* Reserve space */
+};
+
+struct sound_timer_info
+{
+  char name[32];
+  int caps;
+};
+
+struct midi_info		/* OBSOLETE */
+{
+  char name[30];
+  int device;			/* 0-N. INITIALIZE BEFORE CALLING */
+  unsigned int capabilities;	/* To be defined later */
+  int dev_type;
+  int dummies[18];		/* Reserve space */
+};
+
+/*
+ * Level 2 event types for /dev/sequencer
+ */
+
+/*
+ * The 4 most significant bits of byte 0 specify the class of
+ * the event: 
+ *
+ *	0x8X = system level events,
+ *	0x9X = device/port specific events, event[1] = device/port,
+ *		The last 4 bits give the subtype:
+ *			0x02	= Channel event (event[3] = chn).
+ *			0x01	= note event (event[4] = note).
+ *			(0x01 is not used alone but always with bit 0x02).
+ *	       event[2] = MIDI message code (0x80=note off etc.)
+ *
+ */
+
+#define EV_SEQ_LOCAL		0x80
+#define EV_TIMING		0x81
+#define EV_CHN_COMMON		0x92
+#define EV_CHN_VOICE		0x93
+#define EV_SYSEX		0x94
+#define EV_SYSTEM		0x95	/* MIDI system and real time messages (input only) */
+/*
+ * Event types 200 to 220 are reserved for application use.
+ * These numbers will not be used by the driver.
+ */
+
+/*
+ * Events for event type EV_CHN_VOICE
+ */
+
+#define MIDI_NOTEOFF		0x80
+#define MIDI_NOTEON		0x90
+#define MIDI_KEY_PRESSURE	0xA0
+
+/*
+ * Events for event type EV_CHN_COMMON
+ */
+
+#define MIDI_CTL_CHANGE		0xB0
+#define MIDI_PGM_CHANGE		0xC0
+#define MIDI_CHN_PRESSURE	0xD0
+#define MIDI_PITCH_BEND		0xE0
+
+#define MIDI_SYSTEM_PREFIX	0xF0
+
+/*
+ * Timer event types
+ */
+#define TMR_WAIT_REL		1	/* Time relative to the prev time */
+#define TMR_WAIT_ABS		2	/* Absolute time since TMR_START */
+#define TMR_STOP		3
+#define TMR_START		4
+#define TMR_CONTINUE		5
+#define TMR_TEMPO		6
+#define TMR_ECHO		8
+#define TMR_CLOCK		9	/* MIDI clock */
+#define TMR_SPP			10	/* Song position pointer */
+#define TMR_TIMESIG		11	/* Time signature */
+
+/*
+ *	Local event types
+ */
+#define LOCL_STARTAUDIO		1
+#define LOCL_STARTAUDIO2	2
+#define LOCL_STARTAUDIO3	3
+#define LOCL_STARTAUDIO4	4
+
+#if (!defined(__KERNEL__) && !defined(KERNEL) && !defined(INKERNEL) && !defined(_KERNEL)) || defined(USE_SEQ_MACROS)
+/*
+ * Some convenience macros to simplify programming of the
+ * /dev/sequencer interface
+ *
+ * These macros define the API which should be used when possible.
+ */
+#define SEQ_DECLAREBUF()		SEQ_USE_EXTBUF()
+
+void seqbuf_dump (void);	/* This function must be provided by programs */
+
+EXTERNC int OSS_init (int seqfd, int buflen);
+EXTERNC void OSS_seqbuf_dump (int fd, unsigned char *buf, int buflen);
+EXTERNC void OSS_seq_advbuf (int len, int fd, unsigned char *buf, int buflen);
+EXTERNC void OSS_seq_needbuf (int len, int fd, unsigned char *buf,
+			      int buflen);
+EXTERNC void OSS_patch_caching (int dev, int chn, int patch, int fd,
+				unsigned char *buf, int buflen);
+EXTERNC void OSS_drum_caching (int dev, int chn, int patch, int fd,
+			       unsigned char *buf, int buflen);
+EXTERNC void OSS_write_patch (int fd, unsigned char *buf, int len);
+EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len);
+
+#define SEQ_PM_DEFINES int __foo_bar___
+#ifdef OSSLIB
+#  define SEQ_USE_EXTBUF() \
+		EXTERNC unsigned char *_seqbuf; \
+		EXTERNC int _seqbuflen;EXTERNC int _seqbufptr
+#  define SEQ_DEFINEBUF(len) SEQ_USE_EXTBUF();static int _requested_seqbuflen=len
+#  define _SEQ_ADVBUF(len) OSS_seq_advbuf(len, seqfd, _seqbuf, _seqbuflen)
+#  define _SEQ_NEEDBUF(len) OSS_seq_needbuf(len, seqfd, _seqbuf, _seqbuflen)
+#  define SEQ_DUMPBUF() OSS_seqbuf_dump(seqfd, _seqbuf, _seqbuflen)
+
+#  define SEQ_LOAD_GMINSTR(dev, instr) \
+		OSS_patch_caching(dev, -1, instr, seqfd, _seqbuf, _seqbuflen)
+#  define SEQ_LOAD_GMDRUM(dev, drum) \
+		OSS_drum_caching(dev, -1, drum, seqfd, _seqbuf, _seqbuflen)
+#else /* !OSSLIB */
+
+#  define SEQ_LOAD_GMINSTR(dev, instr)
+#  define SEQ_LOAD_GMDRUM(dev, drum)
+
+#  define SEQ_USE_EXTBUF() \
+		EXTERNC unsigned char _seqbuf[]; \
+		EXTERNC int _seqbuflen;EXTERNC int _seqbufptr
+
+#ifndef USE_SIMPLE_MACROS
+/* Sample seqbuf_dump() implementation:
+ *
+ *	SEQ_DEFINEBUF (2048);	-- Defines a buffer for 2048 bytes
+ *
+ *	int seqfd;		-- The file descriptor for /dev/sequencer.
+ *
+ *	void
+ *	seqbuf_dump ()
+ *	{
+ *	  if (_seqbufptr)
+ *	    if (write (seqfd, _seqbuf, _seqbufptr) == -1)
+ *	      {
+ *		perror ("write /dev/sequencer");
+ *		exit (-1);
+ *	      }
+ *	  _seqbufptr = 0;
+ *	}
+ */
+
+#define SEQ_DEFINEBUF(len) \
+	unsigned char _seqbuf[len]; int _seqbuflen = len;int _seqbufptr = 0
+#define _SEQ_NEEDBUF(len) \
+	if ((_seqbufptr+(len)) > _seqbuflen) seqbuf_dump()
+#define _SEQ_ADVBUF(len) _seqbufptr += len
+#define SEQ_DUMPBUF seqbuf_dump
+#else
+/*
+ * This variation of the sequencer macros is used just to format one event
+ * using fixed buffer.
+ * 
+ * The program using the macro library must define the following macros before
+ * using this library.
+ *
+ * #define _seqbuf 		 name of the buffer (unsigned char[]) 
+ * #define _SEQ_ADVBUF(len)	 If the applic needs to know the exact
+ *				 size of the event, this macro can be used.
+ *				 Otherwise this must be defined as empty.
+ * #define _seqbufptr		 Define the name of index variable or 0 if
+ *				 not required. 
+ */
+#define _SEQ_NEEDBUF(len)	/* empty */
+#endif
+#endif /* !OSSLIB */
+
+#define SEQ_VOLUME_MODE(dev, mode) \
+				{_SEQ_NEEDBUF(8);\
+				_seqbuf[_seqbufptr] = SEQ_EXTENDED;\
+				_seqbuf[_seqbufptr+1] = SEQ_VOLMODE;\
+				_seqbuf[_seqbufptr+2] = (dev);\
+				_seqbuf[_seqbufptr+3] = (mode);\
+				_seqbuf[_seqbufptr+4] = 0;\
+				_seqbuf[_seqbufptr+5] = 0;\
+				_seqbuf[_seqbufptr+6] = 0;\
+				_seqbuf[_seqbufptr+7] = 0;\
+				_SEQ_ADVBUF(8);}
+
+/*
+ * Midi voice messages
+ */
+
+#define _CHN_VOICE(dev, event, chn, note, parm) \
+				{_SEQ_NEEDBUF(8);\
+				_seqbuf[_seqbufptr] = EV_CHN_VOICE;\
+				_seqbuf[_seqbufptr+1] = (dev);\
+				_seqbuf[_seqbufptr+2] = (event);\
+				_seqbuf[_seqbufptr+3] = (chn);\
+				_seqbuf[_seqbufptr+4] = (note);\
+				_seqbuf[_seqbufptr+5] = (parm);\
+				_seqbuf[_seqbufptr+6] = (0);\
+				_seqbuf[_seqbufptr+7] = 0;\
+				_SEQ_ADVBUF(8);}
+
+#define SEQ_START_NOTE(dev, chn, note, vol) \
+			_CHN_VOICE(dev, MIDI_NOTEON, chn, note, vol)
+
+#define SEQ_STOP_NOTE(dev, chn, note, vol) \
+			_CHN_VOICE(dev, MIDI_NOTEOFF, chn, note, vol)
+
+#define SEQ_KEY_PRESSURE(dev, chn, note, pressure) \
+			_CHN_VOICE(dev, MIDI_KEY_PRESSURE, chn, note, pressure)
+
+/*
+ * Midi channel messages
+ */
+
+#define _CHN_COMMON(dev, event, chn, p1, p2, w14) \
+				{_SEQ_NEEDBUF(8);\
+				_seqbuf[_seqbufptr] = EV_CHN_COMMON;\
+				_seqbuf[_seqbufptr+1] = (dev);\
+				_seqbuf[_seqbufptr+2] = (event);\
+				_seqbuf[_seqbufptr+3] = (chn);\
+				_seqbuf[_seqbufptr+4] = (p1);\
+				_seqbuf[_seqbufptr+5] = (p2);\
+				*(short *)&_seqbuf[_seqbufptr+6] = (w14);\
+				_SEQ_ADVBUF(8);}
+/*
+ * SEQ_SYSEX permits sending of sysex messages. (It may look that it permits
+ * sending any MIDI bytes but it's absolutely not possible. Trying to do
+ * so _will_ cause problems with MPU401 intelligent mode).
+ *
+ * Sysex messages are sent in blocks of 1 to 6 bytes. Longer messages must be 
+ * sent by calling SEQ_SYSEX() several times (there must be no other events
+ * between them). First sysex fragment must have 0xf0 in the first byte
+ * and the last byte (buf[len-1] of the last fragment must be 0xf7. No byte
+ * between these sysex start and end markers cannot be larger than 0x7f. Also
+ * lengths of each fragments (except the last one) must be 6.
+ *
+ * Breaking the above rules may work with some MIDI ports but is likely to
+ * cause fatal problems with some other devices (such as MPU401).
+ */
+#define SEQ_SYSEX(dev, buf, len) \
+				{int ii, ll=(len); \
+				 unsigned char *bufp=buf;\
+				 if (ll>6)ll=6;\
+				_SEQ_NEEDBUF(8);\
+				_seqbuf[_seqbufptr] = EV_SYSEX;\
+				_seqbuf[_seqbufptr+1] = (dev);\
+				for(ii=0;ii<ll;ii++)\
+				   _seqbuf[_seqbufptr+ii+2] = bufp[ii];\
+				for(ii=ll;ii<6;ii++)\
+				   _seqbuf[_seqbufptr+ii+2] = 0xff;\
+				_SEQ_ADVBUF(8);}
+
+#define SEQ_CHN_PRESSURE(dev, chn, pressure) \
+		_CHN_COMMON(dev, MIDI_CHN_PRESSURE, chn, pressure, 0, 0)
+
+#define SEQ_SET_PATCH SEQ_PGM_CHANGE
+#ifdef OSSLIB
+#   define SEQ_PGM_CHANGE(dev, chn, patch) \
+		{OSS_patch_caching(dev, chn, patch, seqfd, _seqbuf, _seqbuflen); \
+		 _CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0);}
+#else
+#   define SEQ_PGM_CHANGE(dev, chn, patch) \
+		_CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0)
+#endif
+
+#define SEQ_CONTROL(dev, chn, controller, value) \
+		_CHN_COMMON(dev, MIDI_CTL_CHANGE, chn, controller, 0, value)
+
+#define SEQ_BENDER(dev, chn, value) \
+		_CHN_COMMON(dev, MIDI_PITCH_BEND, chn, 0, 0, value)
+
+#define SEQ_V2_X_CONTROL(dev, voice, controller, value)	\
+				{_SEQ_NEEDBUF(8);\
+				_seqbuf[_seqbufptr] = SEQ_EXTENDED;\
+				_seqbuf[_seqbufptr+1] = SEQ_CONTROLLER;\
+				_seqbuf[_seqbufptr+2] = (dev);\
+				_seqbuf[_seqbufptr+3] = (voice);\
+				_seqbuf[_seqbufptr+4] = (controller);\
+				_seqbuf[_seqbufptr+5] = ((value)&0xff);\
+				_seqbuf[_seqbufptr+6] = ((value>>8)&0xff);\
+				_seqbuf[_seqbufptr+7] = 0;\
+				_SEQ_ADVBUF(8);}
+/*
+ * The following 5 macros are incorrectly implemented and obsolete.
+ * Use SEQ_BENDER and SEQ_CONTROL (with proper controller) instead.
+ */
+#define SEQ_PITCHBEND(dev, voice, value) \
+	SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER, value)
+#define SEQ_BENDER_RANGE(dev, voice, value) \
+	SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER_RANGE, value)
+#define SEQ_EXPRESSION(dev, voice, value) \
+	SEQ_CONTROL(dev, voice, CTL_EXPRESSION, value*128)
+#define SEQ_MAIN_VOLUME(dev, voice, value) \
+	SEQ_CONTROL(dev, voice, CTL_MAIN_VOLUME, (value*16383)/100)
+#define SEQ_PANNING(dev, voice, pos) \
+	SEQ_CONTROL(dev, voice, CTL_PAN, (pos+128) / 2)
+
+/*
+ * Timing and syncronization macros
+ */
+
+#define _TIMER_EVENT(ev, parm)	{_SEQ_NEEDBUF(8);\
+			 	_seqbuf[_seqbufptr+0] = EV_TIMING; \
+			 	_seqbuf[_seqbufptr+1] = (ev); \
+				_seqbuf[_seqbufptr+2] = 0;\
+				_seqbuf[_seqbufptr+3] = 0;\
+			 	*(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \
+				_SEQ_ADVBUF(8);}
+
+#define SEQ_START_TIMER()		_TIMER_EVENT(TMR_START, 0)
+#define SEQ_STOP_TIMER()		_TIMER_EVENT(TMR_STOP, 0)
+#define SEQ_CONTINUE_TIMER()		_TIMER_EVENT(TMR_CONTINUE, 0)
+#define SEQ_WAIT_TIME(ticks)		_TIMER_EVENT(TMR_WAIT_ABS, ticks)
+#define SEQ_DELTA_TIME(ticks)		_TIMER_EVENT(TMR_WAIT_REL, ticks)
+#define SEQ_ECHO_BACK(key)		_TIMER_EVENT(TMR_ECHO, key)
+#define SEQ_SET_TEMPO(value)		_TIMER_EVENT(TMR_TEMPO, value)
+#define SEQ_SONGPOS(pos)		_TIMER_EVENT(TMR_SPP, pos)
+#define SEQ_TIME_SIGNATURE(sig)		_TIMER_EVENT(TMR_TIMESIG, sig)
+
+/*
+ * Local control events
+ */
+
+#define _LOCAL_EVENT(ev, parm)		{_SEQ_NEEDBUF(8);\
+				 	_seqbuf[_seqbufptr+0] = EV_SEQ_LOCAL; \
+				 	_seqbuf[_seqbufptr+1] = (ev); \
+					_seqbuf[_seqbufptr+2] = 0;\
+					_seqbuf[_seqbufptr+3] = 0;\
+				 	*(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \
+					_SEQ_ADVBUF(8);}
+
+#define SEQ_PLAYAUDIO(devmask)		_LOCAL_EVENT(LOCL_STARTAUDIO, devmask)
+#define SEQ_PLAYAUDIO2(devmask)		_LOCAL_EVENT(LOCL_STARTAUDIO2, devmask)
+#define SEQ_PLAYAUDIO3(devmask)		_LOCAL_EVENT(LOCL_STARTAUDIO3, devmask)
+#define SEQ_PLAYAUDIO4(devmask)		_LOCAL_EVENT(LOCL_STARTAUDIO4, devmask)
+/*
+ * Events for the level 1 interface only 
+ */
+
+#define SEQ_MIDIOUT(device, byte)	{_SEQ_NEEDBUF(4);\
+					_seqbuf[_seqbufptr] = SEQ_MIDIPUTC;\
+					_seqbuf[_seqbufptr+1] = (byte);\
+					_seqbuf[_seqbufptr+2] = (device);\
+					_seqbuf[_seqbufptr+3] = 0;\
+					_SEQ_ADVBUF(4);}
+
+/*
+ * Patch loading.
+ */
+#ifdef OSSLIB
+#   define SEQ_WRPATCH(patchx, len) \
+		OSS_write_patch(seqfd, (char*)(patchx), len)
+#   define SEQ_WRPATCH2(patchx, len) \
+		OSS_write_patch2(seqfd, (char*)(patchx), len)
+#else
+#   define SEQ_WRPATCH(patchx, len) \
+		{if (_seqbufptr) SEQ_DUMPBUF();\
+		 if (write(seqfd, (char*)(patchx), len)==-1) \
+		    perror("Write patch: /dev/sequencer");}
+#   define SEQ_WRPATCH2(patchx, len) \
+		(SEQ_DUMPBUF(), write(seqfd, (char*)(patchx), len))
+#endif
+
+#endif
+#endif /* ifndef DISABLE_SEQUENCER */
+
+/*
+ ****************************************************************************
+ * ioctl commands for the /dev/midi## 
+ ****************************************************************************/
+#define SNDCTL_MIDI_PRETIME	__SIOWR('m', 0, int)
+
+#if 0
+/*
+ * The SNDCTL_MIDI_MPUMODE and SNDCTL_MIDI_MPUCMD calls
+ * are completely obsolete. The hardware device (MPU-401 "intelligent mode"
+ * and compatibles) has disappeared from the market 10 years ago so there 
+ * is no need for this stuff. The MPU-401 "UART" mode devices don't support
+ * this stuff.
+ */
+typedef struct
+{
+  unsigned char cmd;
+  char nr_args, nr_returns;
+  unsigned char data[30];
+} mpu_command_rec;
+
+#define SNDCTL_MIDI_MPUMODE	__SIOWR('m', 1, int)
+#define SNDCTL_MIDI_MPUCMD	__SIOWR('m', 2, mpu_command_rec)
+#endif
+
+/*
+ * SNDCTL_MIDI_MTCINPUT turns on a mode where OSS automatically inserts
+ * MTC quarter frame messages (F1 xx) to the input.
+ * The argument is the MTC mode:
+ *
+ * 	-1 = Turn MTC messages OFF (default)
+ *	24 = 24 FPS 
+ *	25 = 25 FPS 
+ *	29 = 30 FPS drop frame
+ *	30 = 30 FPS 
+ *
+ * Note that 25 FPS mode is probably the only mode that is supported. Other
+ * modes may be supported in the future versions of OSS, 25 FPS is handy 
+ * because it generates 25*4=100 quarter frame messages per second which
+ * matches the usual 100 HZ system timer rate).
+ *
+ * The quarter frame timer will be reset to 0:00:00:00.0 at the moment this
+ * ioctl is made.
+ */
+#define SNDCTL_MIDI_MTCINPUT	__SIOWR('m', 3, int)
+
+/*
+ * MTC/SMPTE time code record (for future use)
+ */
+typedef struct
+{
+  unsigned char hours, minutes, seconds, frames, qframes;
+  char direction;
+#define MTC_DIR_STOPPED		 0
+#define MTC_DIR_FORWARD		 1
+#define MTC_DIR_BACKWARD	-1
+  unsigned char time_code_type;
+  unsigned int flags;
+} oss_mtc_data_t;
+
+#define SNDCTL_MIDI_SETMODE	__SIOWR('m', 6, int)
+#	define MIDI_MODE_TRADITIONAL	0
+#	define MIDI_MODE_TIMED		1	/* Input times are in MIDI ticks */
+#	define MIDI_MODE_TIMED_ABS  	2	/* Input times are absolute (usecs) */
+
+/*
+ * Packet header for MIDI_MODE_TIMED and MIDI_MODE_TIMED_ABS
+ */
+typedef unsigned long long oss_midi_time_t;	/* Variable type for MIDI time (clock ticks) */
+
+typedef struct
+{
+  int magic;			/* Initialize to MIDI_HDR_MAGIC */
+#define MIDI_HDR_MAGIC	-1
+  unsigned short event_type;
+#define MIDI_EV_WRITE			0	/* Write or read (with payload) */
+#define MIDI_EV_TEMPO			1
+#define MIDI_EV_ECHO			2
+#define MIDI_EV_START			3
+#define MIDI_EV_STOP			4
+#define MIDI_EV_CONTINUE		5
+#define MIDI_EV_XPRESSWRITE		6
+#define MIDI_EV_TIMEBASE		7
+#define MIDI_EV_DEVCTL			8	/* Device control read/write */
+  unsigned short options;
+#define MIDI_OPT_NONE			0x0000
+#define MIDI_OPT_TIMED			0x0001
+#define MIDI_OPT_CONTINUATION		0x0002
+#define MIDI_OPT_USECTIME		0x0004	/* Time is absolute (in usecs) */
+#define MIDI_OPT_BUSY			0x0008	/* Reserved for internal use */
+  oss_midi_time_t time;
+  int parm;
+  int filler[3];		/* Fur future expansion - init to zeros */
+} midi_packet_header_t;
+/* 
+ * MIDI_PAYLOAD_SIZE is the maximum size of one MIDI input chunk. It must be
+ * less (or equal) than 1024 which is the read size recommended in the 
+ * documentation. TODO: Explain this better.
+ */
+#define MIDI_PAYLOAD_SIZE		1000
+
+typedef struct
+{
+  midi_packet_header_t hdr;
+  unsigned char payload[MIDI_PAYLOAD_SIZE];
+} midi_packet_t;
+
+#define SNDCTL_MIDI_TIMEBASE		__SIOWR('m', 7, int)
+#define SNDCTL_MIDI_TEMPO		__SIOWR('m', 8, int)
+/*
+ * User land MIDI servers (synths) can use SNDCTL_MIDI_SET_LATENCY
+ * to request MIDI events to be sent to them in advance. The parameter
+ * (in microseconds) tells how much before the events are submitted.
+ *
+ * This feature is only valid for loopback devices and possibly some other
+ * types of virtual devices.
+ */
+#define SNDCTL_MIDI_SET_LATENCY		__SIOW ('m', 9, int)
+/*
+ ****************************************************************************
+ * IOCTL commands for /dev/dsp
+ ****************************************************************************/
+
+#define SNDCTL_DSP_HALT			__SIO  ('P', 0)
+#define SNDCTL_DSP_RESET		SNDCTL_DSP_HALT	/* Old name */
+#define SNDCTL_DSP_SYNC			__SIO  ('P', 1)
+#define SNDCTL_DSP_SPEED		__SIOWR('P', 2, int)
+
+/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */
+#define SNDCTL_DSP_STEREO		__SIOWR('P', 3, int)
+/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */
+
+#define SNDCTL_DSP_GETBLKSIZE		__SIOWR('P', 4, int)
+#define SNDCTL_DSP_SAMPLESIZE		SNDCTL_DSP_SETFMT
+#define SNDCTL_DSP_CHANNELS		__SIOWR('P', 6, int)
+#define SNDCTL_DSP_POST			__SIO  ('P', 8)
+#define SNDCTL_DSP_SUBDIVIDE		__SIOWR('P', 9, int)
+#define SNDCTL_DSP_SETFRAGMENT		__SIOWR('P',10, int)
+
+/* Audio data formats (Note! U8=8 and S16_LE=16 for compatibility) */
+#define SNDCTL_DSP_GETFMTS		__SIOR ('P',11, int)	/* Returns a mask */
+#define SNDCTL_DSP_SETFMT		__SIOWR('P',5, int)	/* Selects ONE fmt */
+#	define AFMT_QUERY	0x00000000	/* Return current fmt */
+#	define AFMT_MU_LAW	0x00000001
+#	define AFMT_A_LAW	0x00000002
+#	define AFMT_IMA_ADPCM	0x00000004
+#	define AFMT_U8		0x00000008
+#	define AFMT_S16_LE	0x00000010	/* Little endian signed 16 */
+#	define AFMT_S16_BE	0x00000020	/* Big endian signed 16 */
+#	define AFMT_S8		0x00000040
+#	define AFMT_U16_LE	0x00000080	/* Little endian U16 */
+#	define AFMT_U16_BE	0x00000100	/* Big endian U16 */
+#	define AFMT_MPEG	0x00000200	/* MPEG (2) audio */
+
+/* AC3 _compressed_ bitstreams (See Programmer's Guide for details). */
+#	define AFMT_AC3		0x00000400
+/* Ogg Vorbis _compressed_ bit streams */
+#	define AFMT_VORBIS	0x00000800
+
+/* 32 bit formats (MSB aligned) formats */
+#	define AFMT_S32_LE	0x00001000
+#	define AFMT_S32_BE	0x00002000
+
+/* Reserved for _native_ endian double precision IEEE floating point */
+#	define AFMT_FLOAT	0x00004000
+
+/* 24 bit formats (LSB aligned in 32 bit word) formats */
+#	define AFMT_S24_LE	0x00008000
+#	define AFMT_S24_BE	0x00010000
+
+/*
+ * S/PDIF raw format. In this format the S/PDIF frames (including all
+ * control and user bits) are included in the data stream. Each sample
+ * is stored in a 32 bit frame (see IEC-958 for more info). This format
+ * is supported by very few devices and it's only usable for purposes
+ * where full access to the control/user bits is required (real time control).
+ */
+#	define AFMT_SPDIF_RAW	0x00020000
+
+/* 24 bit packed (3 byte) little endian format (USB compatibility) */
+#	define AFMT_S24_PACKED	0x00040000
+
+
+/*
+ * Some big endian/little endian handling macros (native endian and opposite
+ * endian formats). The usage of these macros is described in the OSS
+ * Programmer's Manual.
+ */
+
+#if defined(_AIX) || defined(AIX) || defined(sparc) || defined(__hppa) || defined(PPC) || defined(__powerpc__) && !defined(i386) && !defined(__i386) && !defined(__i386__)
+
+/* Big endian machines */
+#  define _PATCHKEY(id) (0xfd00|id)
+#  define AFMT_S16_NE AFMT_S16_BE
+#  define AFMT_U16_NE AFMT_U16_BE
+#  define AFMT_S32_NE AFMT_S32_BE
+#  define AFMT_S24_NE AFMT_S24_BE
+#  define AFMT_S16_OE AFMT_S16_LE
+#  define AFMT_S32_OE AFMT_S32_LE
+#  define AFMT_S24_OE AFMT_S24_LE
+#else
+#  define _PATCHKEY(id) ((id<<8)|0xfd)
+#  define AFMT_S16_NE AFMT_S16_LE
+#  define AFMT_U16_NE AFMT_U16_LE
+#  define AFMT_S32_NE AFMT_S32_LE
+#  define AFMT_S24_NE AFMT_S24_LE
+#  define AFMT_S16_OE AFMT_S16_BE
+#  define AFMT_S32_OE AFMT_S32_BE
+#  define AFMT_S24_OE AFMT_S24_BE
+#endif
+/*
+ * Buffer status queries.
+ */
+typedef struct audio_buf_info
+{
+  int fragments;		/* # of available fragments (partially usend ones not counted) */
+  int fragstotal;		/* Total # of fragments allocated */
+  int fragsize;			/* Size of a fragment in bytes */
+  int bytes;			/* Available space in bytes (includes partially used fragments) */
+  /* Note! 'bytes' could be more than fragments*fragsize */
+} audio_buf_info;
+
+#define SNDCTL_DSP_GETOSPACE		__SIOR ('P',12, audio_buf_info)
+#define SNDCTL_DSP_GETISPACE		__SIOR ('P',13, audio_buf_info)
+#define SNDCTL_DSP_GETCAPS		__SIOR ('P',15, int)
+#	define PCM_CAP_REVISION		0x000000ff	/* Bits for revision level (0 to 255) */
+#	define PCM_CAP_DUPLEX		0x00000100	/* Full duplex record/playback */
+#	define PCM_CAP_REALTIME		0x00000200	/* Not in use */
+#	define PCM_CAP_BATCH		0x00000400	/* Device has some kind of */
+							/* internal buffers which may */
+							/* cause some delays and */
+							/* decrease precision of timing */
+#	define PCM_CAP_COPROC		0x00000800	/* Has a coprocessor */
+							/* Sometimes it's a DSP */
+							/* but usually not */
+#	define PCM_CAP_TRIGGER		0x00001000	/* Supports SETTRIGGER */
+#	define PCM_CAP_MMAP		0x00002000	/* Supports mmap() */
+#	define PCM_CAP_MULTI		0x00004000	/* Supports multiple open */
+#	define PCM_CAP_BIND		0x00008000	/* Supports binding to front/rear/center/lfe */
+#   	define PCM_CAP_INPUT		0x00010000	/* Supports recording */
+#   	define PCM_CAP_OUTPUT		0x00020000	/* Supports playback */
+#	define PCM_CAP_VIRTUAL		0x00040000	/* Virtual device */
+/* 0x00040000 and 0x00080000 reserved for future use */
+
+/* Analog/digital control capabilities */
+#	define PCM_CAP_ANALOGOUT	0x00100000
+#	define PCM_CAP_ANALOGIN		0x00200000
+#	define PCM_CAP_DIGITALOUT	0x00400000
+#	define PCM_CAP_DIGITALIN	0x00800000
+#	define PCM_CAP_ADMASK		0x00f00000
+/*
+ * NOTE! (capabilities & PCM_CAP_ADMASK)==0 means just that the
+ * digital/analog interface control features are not supported by the 
+ * device/driver. However the device still supports analog, digital or
+ * both inputs/outputs (depending on the device). See the OSS Programmer's
+ * Guide for full details.
+ */
+#	define PCM_CAP_SHADOW		0x01000000	/* "Shadow" device */
+
+/*
+ * Preferred channel usage. These bits can be used to
+ * give recommendations to the application. Used by few drivers.
+ * For example if ((caps & DSP_CH_MASK) == DSP_CH_MONO) means that
+ * the device works best in mono mode. However it doesn't necessarily mean
+ * that the device cannot be used in stereo. These bits should only be used
+ * by special applications such as multi track hard disk recorders to find
+ * out the initial setup. However the user should be able to override this
+ * selection.
+ *
+ * To find out which modes are actually supported the application should 
+ * try to select them using SNDCTL_DSP_CHANNELS.
+ */
+#	define DSP_CH_MASK		0x06000000	/* Mask */
+#	define DSP_CH_ANY		0x00000000	/* No preferred mode */
+#	define DSP_CH_MONO		0x02000000
+#	define DSP_CH_STEREO		0x04000000
+#	define DSP_CH_MULTI		0x06000000	/* More than two channels */
+
+#	define PCM_CAP_HIDDEN		0x08000000	/* Hidden device */
+#	define PCM_CAP_FREERATE		0x10000000
+#	define PCM_CAP_MODEM		0x20000000	/* Modem device */
+#	define PCM_CAP_DEFAULT		0x40000000	/* "Default" device */
+
+/*
+ * The PCM_CAP_* capability names were known as DSP_CAP_* prior OSS 4.0
+ * so it's necessary to define the older names too.
+ */
+#define DSP_CAP_ADMASK		PCM_CAP_ADMASK
+#define DSP_CAP_ANALOGIN	PCM_CAP_ANALOGIN
+#define DSP_CAP_ANALOGOUT	PCM_CAP_ANALOGOUT
+#define DSP_CAP_BATCH		PCM_CAP_BATCH
+#define DSP_CAP_BIND		PCM_CAP_BIND
+#define DSP_CAP_COPROC		PCM_CAP_COPROC
+#define DSP_CAP_DEFAULT		PCM_CAP_DEFAULT
+#define DSP_CAP_DIGITALIN	PCM_CAP_DIGITALIN
+#define DSP_CAP_DIGITALOUT	PCM_CAP_DIGITALOUT
+#define DSP_CAP_DUPLEX		PCM_CAP_DUPLEX
+#define DSP_CAP_FREERATE	PCM_CAP_FREERATE
+#define DSP_CAP_HIDDEN		PCM_CAP_HIDDEN
+#define DSP_CAP_INPUT		PCM_CAP_INPUT
+#define DSP_CAP_MMAP		PCM_CAP_MMAP
+#define DSP_CAP_MODEM		PCM_CAP_MODEM
+#define DSP_CAP_MULTI		PCM_CAP_MULTI
+#define DSP_CAP_OUTPUT		PCM_CAP_OUTPUT
+#define DSP_CAP_REALTIME	PCM_CAP_REALTIME
+#define DSP_CAP_REVISION	PCM_CAP_REVISION
+#define DSP_CAP_SHADOW		PCM_CAP_SHADOW
+#define DSP_CAP_TRIGGER		PCM_CAP_TRIGGER
+#define DSP_CAP_VIRTUAL		PCM_CAP_VIRTUAL
+
+#define SNDCTL_DSP_GETTRIGGER		__SIOR ('P',16, int)
+#define SNDCTL_DSP_SETTRIGGER		__SIOW ('P',16, int)
+#	define PCM_ENABLE_INPUT		0x00000001
+#	define PCM_ENABLE_OUTPUT	0x00000002
+
+typedef struct count_info
+{
+  unsigned int bytes;		/* Total # of bytes processed */
+  int blocks;			/* # of fragment transitions since last time */
+  int ptr;			/* Current DMA pointer value */
+} count_info;
+
+#define SNDCTL_DSP_GETIPTR		__SIOR ('P',17, count_info)
+#define SNDCTL_DSP_GETOPTR		__SIOR ('P',18, count_info)
+
+typedef struct buffmem_desc
+{
+  unsigned *buffer;
+  int size;
+} buffmem_desc;
+#define SNDCTL_DSP_SETSYNCRO		__SIO  ('P', 21)
+#define SNDCTL_DSP_SETDUPLEX		__SIO  ('P', 22)
+
+#define SNDCTL_DSP_PROFILE		__SIOW ('P', 23, int)	/* OBSOLETE */
+#define	  APF_NORMAL	0	/* Normal applications */
+#define	  APF_NETWORK	1	/* Underruns probably caused by an "external" delay */
+#define   APF_CPUINTENS 2	/* Underruns probably caused by "overheating" the CPU */
+
+#define SNDCTL_DSP_GETODELAY		__SIOR ('P', 23, int)
+
+typedef struct audio_errinfo
+{
+  int play_underruns;
+  int rec_overruns;
+  unsigned int play_ptradjust;
+  unsigned int rec_ptradjust;
+  int play_errorcount;
+  int rec_errorcount;
+  int play_lasterror;
+  int rec_lasterror;
+  int play_errorparm;
+  int rec_errorparm;
+  int filler[16];
+} audio_errinfo;
+
+#define SNDCTL_DSP_GETPLAYVOL		__SIOR ('P', 24, int)
+#define SNDCTL_DSP_SETPLAYVOL		__SIOWR('P', 24, int)
+#define SNDCTL_DSP_GETERROR		__SIOR ('P', 25, audio_errinfo)
+/*
+ ****************************************************************************
+ * Digital interface (S/PDIF) control interface
+ */
+
+typedef struct oss_digital_control
+{
+  unsigned int caps;
+#define DIG_CBITIN_NONE			0x00000000
+#define DIG_CBITIN_LIMITED		0x00000001
+#define DIG_CBITIN_DATA 		0x00000002
+#define DIG_CBITIN_BYTE0		0x00000004
+#define DIG_CBITIN_FULL 		0x00000008
+#define DIG_CBITIN_MASK 		0x0000000f
+#define DIG_CBITOUT_NONE		0x00000000
+#define DIG_CBITOUT_LIMITED		0x00000010
+#define DIG_CBITOUT_BYTE0		0x00000020
+#define DIG_CBITOUT_FULL 		0x00000040
+#define DIG_CBITOUT_DATA 		0x00000080
+#define DIG_CBITOUT_MASK 		0x000000f0
+#define DIG_UBITIN			0x00000100
+#define DIG_UBITOUT			0x00000200
+#define DIG_VBITOUT			0x00000400
+#define DIG_OUTRATE			0x00000800
+#define DIG_INRATE			0x00001000
+#define DIG_INBITS			0x00002000
+#define DIG_OUTBITS			0x00004000
+#define DIG_EXACT			0x00010000
+#define DIG_PRO				0x00020000
+#define DIG_CONSUMER			0x00040000
+#define DIG_PASSTHROUGH			0x00080000
+#define DIG_OUTSEL			0x00100000
+
+  unsigned int valid;
+#define VAL_CBITIN			0x00000001
+#define VAL_UBITIN			0x00000002
+#define VAL_CBITOUT			0x00000004
+#define VAL_UBITOUT			0x00000008
+#define VAL_ISTATUS			0x00000010
+#define VAL_IRATE			0x00000020
+#define VAL_ORATE			0x00000040
+#define VAL_INBITS			0x00000080
+#define VAL_OUTBITS			0x00000100
+#define VAL_REQUEST			0x00000200
+#define VAL_OUTSEL			0x00000400
+
+#define VAL_OUTMASK (VAL_CBITOUT|VAL_UBITOUT|VAL_ORATE|VAL_OUTBITS|VAL_OUTSEL)
+
+  unsigned int request, param;
+#define SPD_RQ_PASSTHROUGH				1
+
+  unsigned char cbitin[24];
+  unsigned char ubitin[24];
+  unsigned char cbitout[24];
+  unsigned char ubitout[24];
+
+  unsigned int outsel;
+#define OUTSEL_DIGITAL		1
+#define OUTSEL_ANALOG		2
+#define OUTSEL_BOTH		(OUTSEL_DIGITAL|OUTSEL_ANALOG)
+
+  int in_data;			/* Audio/data if autodetectable by the receiver */
+#define IND_UNKNOWN		0
+#define IND_AUDIO		1
+#define IND_DATA		2
+
+  int in_locked;		/* Receiver locked */
+#define LOCK_NOT_INDICATED	0
+#define LOCK_UNLOCKED		1
+#define LOCK_LOCKED		2
+
+  int in_quality;		/* Input signal quality */
+#define IN_QUAL_NOT_INDICATED	0
+#define IN_QUAL_POOR		1
+#define IN_QUAL_GOOD		2
+
+  int in_vbit, out_vbit;	/* V bits */
+#define VBIT_NOT_INDICATED	0
+#define VBIT_OFF		1
+#define VBIT_ON			2
+
+  unsigned int in_errors;	/* Various input error conditions */
+#define INERR_CRC		0x0001
+#define INERR_QCODE_CRC		0x0002
+#define INERR_PARITY		0x0004
+#define INERR_BIPHASE		0x0008
+
+  int srate_in, srate_out;
+  int bits_in, bits_out;
+
+  int filler[32];
+} oss_digital_control;
+
+#define SNDCTL_DSP_READCTL		__SIOWR('P', 26, oss_digital_control)
+#define SNDCTL_DSP_WRITECTL		__SIOWR('P', 27, oss_digital_control)
+
+/*
+ ****************************************************************************
+ * Sync groups for audio devices
+ */
+typedef struct oss_syncgroup
+{
+  int id;
+  int mode;
+  int filler[16];
+} oss_syncgroup;
+
+#define SNDCTL_DSP_SYNCGROUP		__SIOWR('P', 28, oss_syncgroup)
+#define SNDCTL_DSP_SYNCSTART		__SIOW ('P', 29, int)
+
+/*
+ **************************************************************************
+ * "cooked" mode enables software based conversions for sample rate, sample
+ * format (bits) and number of channels (mono/stereo). These conversions are
+ * required with some devices that support only one sample rate or just stereo
+ * to let the applications to use other formats. The cooked mode is enabled by
+ * default. However it's necessary to disable this mode when mmap() is used or
+ * when very deterministic timing is required. SNDCTL_DSP_COOKEDMODE is an
+ * optional call introduced in OSS 3.9.6f. It's _error return must be ignored_
+ * since normally this call will return erno=EINVAL.
+ *
+ * SNDCTL_DSP_COOKEDMODE must be called immediately after open before doing
+ * anything else. Otherwise the call will not have any effect.
+ */
+#define SNDCTL_DSP_COOKEDMODE		__SIOW ('P', 30, int)
+
+/*
+ **************************************************************************
+ * SNDCTL_DSP_SILENCE and SNDCTL_DSP_SKIP are new calls in OSS 3.99.0
+ * that can be used to implement pause/continue during playback (no effect
+ * on recording).
+ */
+#define SNDCTL_DSP_SILENCE		__SIO  ('P', 31)
+#define SNDCTL_DSP_SKIP			__SIO  ('P', 32)
+/*
+ ****************************************************************************
+ * Abort transfer (reset) functions for input and output
+ */
+#define SNDCTL_DSP_HALT_INPUT		__SIO  ('P', 33)
+#define SNDCTL_DSP_RESET_INPUT	SNDCTL_DSP_HALT_INPUT	/* Old name */
+#define SNDCTL_DSP_HALT_OUTPUT		__SIO  ('P', 34)
+#define SNDCTL_DSP_RESET_OUTPUT	SNDCTL_DSP_HALT_OUTPUT	/* Old name */
+/*
+ ****************************************************************************
+ * Low water level control
+ */
+#define SNDCTL_DSP_LOW_WATER		__SIOW ('P', 34, int)
+
+/*
+ ****************************************************************************
+ * 64 bit pointer support. Only available in environments that support
+ * the 64 bit (long long) integer type.
+ */
+#ifndef OSS_NO_LONG_LONG
+typedef struct
+{
+  long long samples;
+  int fifo_samples;
+  int filler[32];		/* For future use */
+} oss_count_t;
+
+#define SNDCTL_DSP_CURRENT_IPTR		__SIOR ('P', 35, oss_count_t)
+#define SNDCTL_DSP_CURRENT_OPTR		__SIOR ('P', 36, oss_count_t)
+#endif
+
+/*
+ ****************************************************************************
+ * Interface for selecting recording sources and playback output routings.
+ */
+#define SNDCTL_DSP_GET_RECSRC_NAMES	__SIOR ('P', 37, oss_mixer_enuminfo)
+#define SNDCTL_DSP_GET_RECSRC		__SIOR ('P', 38, int)
+#define SNDCTL_DSP_SET_RECSRC		__SIOWR('P', 38, int)
+
+#define SNDCTL_DSP_GET_PLAYTGT_NAMES	__SIOR ('P', 39, oss_mixer_enuminfo)
+#define SNDCTL_DSP_GET_PLAYTGT		__SIOR ('P', 40, int)
+#define SNDCTL_DSP_SET_PLAYTGT		__SIOWR('P', 40, int)
+#define SNDCTL_DSP_GETRECVOL		__SIOR ('P', 41, int)
+#define SNDCTL_DSP_SETRECVOL		__SIOWR('P', 41, int)
+
+/*
+ ***************************************************************************
+ * Some calls for setting the channel assignment with multi channel devices
+ * (see the manual for details).
+ */
+#ifndef OSS_NO_LONG_LONG
+#define SNDCTL_DSP_GET_CHNORDER		__SIOR ('P', 42, unsigned long long)
+#define SNDCTL_DSP_SET_CHNORDER		__SIOWR('P', 42, unsigned long long)
+#	define CHID_UNDEF	0
+#	define CHID_L		1
+#	define CHID_R		2
+#	define CHID_C		3
+#	define CHID_LFE		4
+#	define CHID_LS		5
+#	define CHID_RS		6
+#	define CHID_LR		7
+#	define CHID_RR		8
+#define CHNORDER_UNDEF		0x0000000000000000ULL
+#define CHNORDER_NORMAL		0x0000000087654321ULL
+#endif
+
+#define MAX_PEAK_CHANNELS	128
+typedef unsigned short oss_peaks_t[MAX_PEAK_CHANNELS];
+#define SNDCTL_DSP_GETIPEAKS		__SIOR('P', 43, oss_peaks_t)
+#define SNDCTL_DSP_GETOPEAKS		__SIOR('P', 44, oss_peaks_t)
+
+#define SNDCTL_DSP_POLICY		__SIOW('P', 45, int)	/* See the manual */
+
+/*
+ ****************************************************************************
+ * Few ioctl calls that are not official parts of OSS. They have been used
+ * by few freeware implementations of OSS.
+ */
+#define SNDCTL_DSP_GETCHANNELMASK	__SIOWR('P', 64, int)
+#define SNDCTL_DSP_BIND_CHANNEL		__SIOWR('P', 65, int)
+#     define DSP_BIND_QUERY           0x00000000
+#     define DSP_BIND_FRONT           0x00000001
+#     define DSP_BIND_SURR            0x00000002
+#     define DSP_BIND_CENTER_LFE      0x00000004
+#     define DSP_BIND_HANDSET         0x00000008
+#     define DSP_BIND_MIC             0x00000010
+#     define DSP_BIND_MODEM1          0x00000020
+#     define DSP_BIND_MODEM2          0x00000040
+#     define DSP_BIND_I2S             0x00000080
+#     define DSP_BIND_SPDIF           0x00000100
+#     define DSP_BIND_REAR            0x00000200
+
+#ifdef sun
+/* Not part of OSS. Reserved for internal use by Solaris */
+#define X_SADA_GET_PLAYTGT_MASK	__SIOR ('P', 66, int)
+#define X_SADA_GET_PLAYTGT	__SIOR ('P', 67, int)
+#define X_SADA_SET_PLAYTGT	__SIOWR('P', 68, int)
+#endif
+
+#ifndef NO_LEGACY_MIXER
+/*
+ ****************************************************************************
+ * IOCTL commands for the "legacy " /dev/mixer API (obsolete)
+ *
+ * Mixer controls
+ *
+ * There can be up to 20 different analog mixer channels. The
+ * SOUND_MIXER_NRDEVICES gives the currently supported maximum. 
+ * The SOUND_MIXER_READ_DEVMASK returns a bitmask which tells
+ * the devices supported by the particular mixer.
+ *
+ * {!notice This "legacy" mixer API is obsolete. It has been superceded
+ * by a new one (see below).
+ */
+
+#define SOUND_MIXER_NRDEVICES	28
+#define SOUND_MIXER_VOLUME	0
+#define SOUND_MIXER_BASS	1
+#define SOUND_MIXER_TREBLE	2
+#define SOUND_MIXER_SYNTH	3
+#define SOUND_MIXER_PCM		4
+#define SOUND_MIXER_SPEAKER	5
+#define SOUND_MIXER_LINE	6
+#define SOUND_MIXER_MIC		7
+#define SOUND_MIXER_CD		8
+#define SOUND_MIXER_IMIX	9	/*  Recording monitor  */
+#define SOUND_MIXER_ALTPCM	10
+#define SOUND_MIXER_RECLEV	11	/* Recording level */
+#define SOUND_MIXER_IGAIN	12	/* Input gain */
+#define SOUND_MIXER_OGAIN	13	/* Output gain */
+/* 
+ * Some soundcards have three line level inputs (line, aux1 and aux2). 
+ * Since each card manufacturer has assigned different meanings to 
+ * these inputs, it's impractical to assign specific meanings 
+ * (eg line, cd, synth etc.) to them.
+ */
+#define SOUND_MIXER_LINE1	14	/* Input source 1  (aux1) */
+#define SOUND_MIXER_LINE2	15	/* Input source 2  (aux2) */
+#define SOUND_MIXER_LINE3	16	/* Input source 3  (line) */
+#define SOUND_MIXER_DIGITAL1	17	/* Digital I/O 1 */
+#define SOUND_MIXER_DIGITAL2	18	/* Digital I/O 2 */
+#define SOUND_MIXER_DIGITAL3	19	/* Digital I/O 3 */
+#define SOUND_MIXER_PHONE	20	/* Phone */
+#define SOUND_MIXER_MONO	21	/* Mono Output */
+#define SOUND_MIXER_VIDEO	22	/* Video/TV (audio) in */
+#define SOUND_MIXER_RADIO	23	/* Radio in */
+#define SOUND_MIXER_DEPTH	24	/* Surround depth */
+#define SOUND_MIXER_REARVOL	25	/* Rear/Surround speaker vol */
+#define SOUND_MIXER_CENTERVOL	26	/* Center/LFE speaker vol */
+#define SOUND_MIXER_SIDEVOL	27	/* Side-Surround (8speaker) vol */
+
+/*
+ * Warning: SOUND_MIXER_SURRVOL is an old name of SOUND_MIXER_SIDEVOL.
+ *          They are both assigned to the same mixer control. Don't
+ *          use both control names in the same program/driver.
+ */
+#define SOUND_MIXER_SURRVOL	SOUND_MIXER_SIDEVOL
+
+/* Some on/off settings (SOUND_SPECIAL_MIN - SOUND_SPECIAL_MAX) */
+/* Not counted to SOUND_MIXER_NRDEVICES, but use the same number space */
+#define SOUND_ONOFF_MIN		28
+#define SOUND_ONOFF_MAX		30
+
+/* Note!	Number 31 cannot be used since the sign bit is reserved */
+#define SOUND_MIXER_NONE	31
+
+/*
+ * The following unsupported macros are no longer functional.
+ * Use SOUND_MIXER_PRIVATE# macros in future.
+ */
+#define SOUND_MIXER_ENHANCE	SOUND_MIXER_NONE
+#define SOUND_MIXER_MUTE	SOUND_MIXER_NONE
+#define SOUND_MIXER_LOUD	SOUND_MIXER_NONE
+
+#define SOUND_DEVICE_LABELS \
+	{"Vol  ", "Bass ", "Treble", "Synth", "Pcm  ", "Speaker ", "Line ", \
+	 "Mic  ", "CD   ", "Mix  ", "Pcm2 ", "Rec  ", "IGain", "OGain", \
+	 "Aux1", "Aux2", "Aux3", "Digital1", "Digital2", "Digital3", \
+	 "Phone", "Mono", "Video", "Radio", "Depth", \
+	 "Rear", "Center", "Side"}
+
+#define SOUND_DEVICE_NAMES \
+	{"vol", "bass", "treble", "synth", "pcm", "speaker", "line", \
+	 "mic", "cd", "mix", "pcm2", "rec", "igain", "ogain", \
+	 "aux1", "aux2", "aux3", "dig1", "dig2", "dig3", \
+	 "phone", "mono", "video", "radio", "depth", \
+	 "rear", "center", "side"}
+
+/*	Device bitmask identifiers	*/
+
+#define SOUND_MIXER_RECSRC	0xff	/* Arg contains a bit for each recording source */
+#define SOUND_MIXER_DEVMASK	0xfe	/* Arg contains a bit for each supported device */
+#define SOUND_MIXER_RECMASK	0xfd	/* Arg contains a bit for each supported recording source */
+#define SOUND_MIXER_CAPS	0xfc
+#	define SOUND_CAP_EXCL_INPUT	0x00000001	/* Only one recording source at a time */
+#	define SOUND_CAP_NOLEGACY	0x00000004	/* For internal use only */
+#	define SOUND_CAP_NORECSRC	0x00000008
+#define SOUND_MIXER_STEREODEVS	0xfb	/* Mixer channels supporting stereo */
+
+/* OSS/Free ONLY */
+#define SOUND_MIXER_OUTSRC    0xfa	/* Arg contains a bit for each input source to output */
+#define SOUND_MIXER_OUTMASK   0xf9	/* Arg contains a bit for each supported input source to output */
+/* OSS/Free ONLY */
+
+/*	Device mask bits	*/
+
+#define SOUND_MASK_VOLUME	(1 << SOUND_MIXER_VOLUME)
+#define SOUND_MASK_BASS		(1 << SOUND_MIXER_BASS)
+#define SOUND_MASK_TREBLE	(1 << SOUND_MIXER_TREBLE)
+#define SOUND_MASK_SYNTH	(1 << SOUND_MIXER_SYNTH)
+#define SOUND_MASK_PCM		(1 << SOUND_MIXER_PCM)
+#define SOUND_MASK_SPEAKER	(1 << SOUND_MIXER_SPEAKER)
+#define SOUND_MASK_LINE		(1 << SOUND_MIXER_LINE)
+#define SOUND_MASK_MIC		(1 << SOUND_MIXER_MIC)
+#define SOUND_MASK_CD		(1 << SOUND_MIXER_CD)
+#define SOUND_MASK_IMIX		(1 << SOUND_MIXER_IMIX)
+#define SOUND_MASK_ALTPCM	(1 << SOUND_MIXER_ALTPCM)
+#define SOUND_MASK_RECLEV	(1 << SOUND_MIXER_RECLEV)
+#define SOUND_MASK_IGAIN	(1 << SOUND_MIXER_IGAIN)
+#define SOUND_MASK_OGAIN	(1 << SOUND_MIXER_OGAIN)
+#define SOUND_MASK_LINE1	(1 << SOUND_MIXER_LINE1)
+#define SOUND_MASK_LINE2	(1 << SOUND_MIXER_LINE2)
+#define SOUND_MASK_LINE3	(1 << SOUND_MIXER_LINE3)
+#define SOUND_MASK_DIGITAL1	(1 << SOUND_MIXER_DIGITAL1)
+#define SOUND_MASK_DIGITAL2	(1 << SOUND_MIXER_DIGITAL2)
+#define SOUND_MASK_DIGITAL3	(1 << SOUND_MIXER_DIGITAL3)
+#define SOUND_MASK_MONO		(1 << SOUND_MIXER_MONO)
+#define SOUND_MASK_PHONE	(1 << SOUND_MIXER_PHONE)
+#define SOUND_MASK_RADIO	(1 << SOUND_MIXER_RADIO)
+#define SOUND_MASK_VIDEO	(1 << SOUND_MIXER_VIDEO)
+#define SOUND_MASK_DEPTH	(1 << SOUND_MIXER_DEPTH)
+#define SOUND_MASK_REARVOL	(1 << SOUND_MIXER_REARVOL)
+#define SOUND_MASK_CENTERVOL	(1 << SOUND_MIXER_CENTERVOL)
+#define SOUND_MASK_SIDEVOL	(1 << SOUND_MIXER_SIDEVOL)
+
+/* Note! SOUND_MASK_SURRVOL is alias of SOUND_MASK_SIDEVOL */
+#define SOUND_MASK_SURRVOL	(1 << SOUND_MIXER_SIDEVOL)
+
+/* Obsolete macros */
+#define SOUND_MASK_MUTE		(1 << SOUND_MIXER_MUTE)
+#define SOUND_MASK_ENHANCE	(1 << SOUND_MIXER_ENHANCE)
+#define SOUND_MASK_LOUD		(1 << SOUND_MIXER_LOUD)
+
+#define MIXER_READ(dev)			__SIOR('M', dev, int)
+#define SOUND_MIXER_READ_VOLUME		MIXER_READ(SOUND_MIXER_VOLUME)
+#define SOUND_MIXER_READ_BASS		MIXER_READ(SOUND_MIXER_BASS)
+#define SOUND_MIXER_READ_TREBLE		MIXER_READ(SOUND_MIXER_TREBLE)
+#define SOUND_MIXER_READ_SYNTH		MIXER_READ(SOUND_MIXER_SYNTH)
+#define SOUND_MIXER_READ_PCM		MIXER_READ(SOUND_MIXER_PCM)
+#define SOUND_MIXER_READ_SPEAKER	MIXER_READ(SOUND_MIXER_SPEAKER)
+#define SOUND_MIXER_READ_LINE		MIXER_READ(SOUND_MIXER_LINE)
+#define SOUND_MIXER_READ_MIC		MIXER_READ(SOUND_MIXER_MIC)
+#define SOUND_MIXER_READ_CD		MIXER_READ(SOUND_MIXER_CD)
+#define SOUND_MIXER_READ_IMIX		MIXER_READ(SOUND_MIXER_IMIX)
+#define SOUND_MIXER_READ_ALTPCM		MIXER_READ(SOUND_MIXER_ALTPCM)
+#define SOUND_MIXER_READ_RECLEV		MIXER_READ(SOUND_MIXER_RECLEV)
+#define SOUND_MIXER_READ_IGAIN		MIXER_READ(SOUND_MIXER_IGAIN)
+#define SOUND_MIXER_READ_OGAIN		MIXER_READ(SOUND_MIXER_OGAIN)
+#define SOUND_MIXER_READ_LINE1		MIXER_READ(SOUND_MIXER_LINE1)
+#define SOUND_MIXER_READ_LINE2		MIXER_READ(SOUND_MIXER_LINE2)
+#define SOUND_MIXER_READ_LINE3		MIXER_READ(SOUND_MIXER_LINE3)
+
+/* Obsolete macros */
+#define SOUND_MIXER_READ_MUTE		MIXER_READ(SOUND_MIXER_MUTE)
+#define SOUND_MIXER_READ_ENHANCE	MIXER_READ(SOUND_MIXER_ENHANCE)
+#define SOUND_MIXER_READ_LOUD		MIXER_READ(SOUND_MIXER_LOUD)
+
+#define SOUND_MIXER_READ_RECSRC		MIXER_READ(SOUND_MIXER_RECSRC)
+#define SOUND_MIXER_READ_DEVMASK	MIXER_READ(SOUND_MIXER_DEVMASK)
+#define SOUND_MIXER_READ_RECMASK	MIXER_READ(SOUND_MIXER_RECMASK)
+#define SOUND_MIXER_READ_STEREODEVS	MIXER_READ(SOUND_MIXER_STEREODEVS)
+#define SOUND_MIXER_READ_CAPS		MIXER_READ(SOUND_MIXER_CAPS)
+
+#define MIXER_WRITE(dev)		__SIOWR('M', dev, int)
+#define SOUND_MIXER_WRITE_VOLUME	MIXER_WRITE(SOUND_MIXER_VOLUME)
+#define SOUND_MIXER_WRITE_BASS		MIXER_WRITE(SOUND_MIXER_BASS)
+#define SOUND_MIXER_WRITE_TREBLE	MIXER_WRITE(SOUND_MIXER_TREBLE)
+#define SOUND_MIXER_WRITE_SYNTH		MIXER_WRITE(SOUND_MIXER_SYNTH)
+#define SOUND_MIXER_WRITE_PCM		MIXER_WRITE(SOUND_MIXER_PCM)
+#define SOUND_MIXER_WRITE_SPEAKER	MIXER_WRITE(SOUND_MIXER_SPEAKER)
+#define SOUND_MIXER_WRITE_LINE		MIXER_WRITE(SOUND_MIXER_LINE)
+#define SOUND_MIXER_WRITE_MIC		MIXER_WRITE(SOUND_MIXER_MIC)
+#define SOUND_MIXER_WRITE_CD		MIXER_WRITE(SOUND_MIXER_CD)
+#define SOUND_MIXER_WRITE_IMIX		MIXER_WRITE(SOUND_MIXER_IMIX)
+#define SOUND_MIXER_WRITE_ALTPCM	MIXER_WRITE(SOUND_MIXER_ALTPCM)
+#define SOUND_MIXER_WRITE_RECLEV	MIXER_WRITE(SOUND_MIXER_RECLEV)
+#define SOUND_MIXER_WRITE_IGAIN		MIXER_WRITE(SOUND_MIXER_IGAIN)
+#define SOUND_MIXER_WRITE_OGAIN		MIXER_WRITE(SOUND_MIXER_OGAIN)
+#define SOUND_MIXER_WRITE_LINE1		MIXER_WRITE(SOUND_MIXER_LINE1)
+#define SOUND_MIXER_WRITE_LINE2		MIXER_WRITE(SOUND_MIXER_LINE2)
+#define SOUND_MIXER_WRITE_LINE3		MIXER_WRITE(SOUND_MIXER_LINE3)
+
+/* Obsolete macros */
+#define SOUND_MIXER_WRITE_MUTE		MIXER_WRITE(SOUND_MIXER_MUTE)
+#define SOUND_MIXER_WRITE_ENHANCE	MIXER_WRITE(SOUND_MIXER_ENHANCE)
+#define SOUND_MIXER_WRITE_LOUD		MIXER_WRITE(SOUND_MIXER_LOUD)
+
+#define SOUND_MIXER_WRITE_RECSRC	MIXER_WRITE(SOUND_MIXER_RECSRC)
+
+typedef struct mixer_info	/* OBSOLETE */
+{
+  char id[16];
+  char name[32];
+  int modify_counter;
+  int card_number;
+  int port_number;
+  char handle[32];
+} mixer_info;
+
+/* SOUND_MIXER_INFO is obsolete - use SNDCTL_MIXERINFO instead */
+#define SOUND_MIXER_INFO		__SIOR ('M', 101, mixer_info)
+
+/*
+ * Two ioctls for special souncard function (OSS/Free only)
+ */
+#define SOUND_MIXER_AGC  _SIOWR('M', 103, int)
+#define SOUND_MIXER_3DSE  _SIOWR('M', 104, int)
+/*
+ * The SOUND_MIXER_PRIVATE# commands can be redefined by low level drivers.
+ * These features can be used when accessing device specific features.
+ */
+#define SOUND_MIXER_PRIVATE1		__SIOWR('M', 111, int)
+#define SOUND_MIXER_PRIVATE2		__SIOWR('M', 112, int)
+#define SOUND_MIXER_PRIVATE3		__SIOWR('M', 113, int)
+#define SOUND_MIXER_PRIVATE4		__SIOWR('M', 114, int)
+#define SOUND_MIXER_PRIVATE5		__SIOWR('M', 115, int)
+
+/* The following two controls were never implemented and they should not be used. */
+#define SOUND_MIXER_READ_MAINVOL		__SIOR ('M', 116, int)
+#define SOUND_MIXER_WRITE_MAINVOL		__SIOWR('M', 116, int)
+
+/*
+ * SOUND_MIXER_GETLEVELS and SOUND_MIXER_SETLEVELS calls can be used
+ * for querying current mixer settings from the driver and for loading
+ * default volume settings _prior_ activating the mixer (loading
+ * doesn't affect current state of the mixer hardware). These calls
+ * are for internal use by the driver software only.
+ */
+
+typedef struct mixer_vol_table
+{
+  int num;			/* Index to volume table */
+  char name[32];
+  int levels[32];
+} mixer_vol_table;
+
+#define SOUND_MIXER_GETLEVELS		__SIOWR('M', 116, mixer_vol_table)
+#define SOUND_MIXER_SETLEVELS		__SIOWR('M', 117, mixer_vol_table)
+
+#define OSS_GETVERSION			__SIOR ('M', 118, int)
+
+/*
+ * Calls to set/get the recording gain for the currently active
+ * recording source. These calls automatically map to the right control.
+ * Note that these calls are not supported by all drivers. In this case
+ * the call will return -1 with errno set to EINVAL
+ *
+ * The _MONGAIN work in similar way but set/get the monitoring gain for
+ * the currently selected recording source.
+ */
+#define SOUND_MIXER_READ_RECGAIN	__SIOR ('M', 119, int)
+#define SOUND_MIXER_WRITE_RECGAIN	__SIOWR('M', 119, int)
+#define SOUND_MIXER_READ_MONGAIN	__SIOR ('M', 120, int)
+#define SOUND_MIXER_WRITE_MONGAIN	__SIOWR('M', 120, int)
+
+/* The following call is for driver development time purposes. It's not
+ * present in any released drivers.
+ */
+typedef unsigned char oss_reserved_t[512];
+#define SOUND_MIXER_RESERVED		__SIOWR('M', 121, oss_reserved_t)
+#endif /* ifndef NO_LEGACY_MIXER */
+
+/*
+ *************************************************************************
+ * The "new" mixer API of OSS 4.0 and later.
+ *
+ * This improved mixer API makes it possible to access every possible feature
+ * of every possible device. However you should read the mixer programming
+ * section of the OSS API Developer's Manual. There is no chance that you
+ * could use this interface correctly just by examining this header.
+ */
+
+typedef struct oss_sysinfo
+{
+  char product[32];		/* For example OSS/Free, OSS/Linux or OSS/Solaris */
+  char version[32];		/* For example 4.0a */
+  int versionnum;		/* See OSS_GETVERSION */
+  char options[128];		/* Reserved */
+
+  int numaudios;		/* # of audio/dsp devices */
+  int openedaudio[8];		/* Bit mask telling which audio devices are busy */
+
+  int numsynths;		/* # of availavle synth devices */
+  int nummidis;			/* # of available MIDI ports */
+  int numtimers;		/* # of available timer devices */
+  int nummixers;		/* # of mixer devices */
+
+  int openedmidi[8];		/* Bit mask telling which midi devices are busy */
+  int numcards;			/* Number of sound cards in the system */
+  int numaudioengines;		/* Number of audio engines in the system */
+  char license[16];		/* For example "GPL" or "CDDL" */
+  int filler[236];		/* For future expansion (set to -1) */
+} oss_sysinfo;
+
+typedef struct oss_mixext
+{
+  int dev;			/* Mixer device number */
+  int ctrl;			/* Controller number */
+  int type;			/* Entry type */
+#	define MIXT_DEVROOT	 0	/* Device root entry */
+#	define MIXT_GROUP	 1	/* Controller group */
+#	define MIXT_ONOFF	 2	/* OFF (0) or ON (1) */
+#	define MIXT_ENUM	 3	/* Enumerated (0 to maxvalue) */
+#	define MIXT_MONOSLIDER	 4	/* Mono slider (0 to 255) */
+#	define MIXT_STEREOSLIDER 5	/* Stereo slider (dual 0 to 255) */
+#	define MIXT_MESSAGE	 6	/* (Readable) textual message */
+#	define MIXT_MONOVU	 7	/* VU meter value (mono) */
+#	define MIXT_STEREOVU	 8	/* VU meter value (stereo) */
+#	define MIXT_MONOPEAK	 9	/* VU meter peak value (mono) */
+#	define MIXT_STEREOPEAK	10	/* VU meter peak value (stereo) */
+#	define MIXT_RADIOGROUP	11	/* Radio button group */
+#	define MIXT_MARKER	12	/* Separator between normal and extension entries */
+#	define MIXT_VALUE	13	/* Decimal value entry */
+#	define MIXT_HEXVALUE	14	/* Hexadecimal value entry */
+#	define MIXT_MONODB	15	/* OBSOLETE */
+#	define MIXT_STEREODB	16	/* OBSOLETE */
+#	define MIXT_SLIDER	17	/* Slider (mono) with full (31 bit) postitive integer range */
+#	define MIXT_3D		18
+
+/*
+ * Sliders with range expanded to 15 bits per channel (0-32767)
+ */
+#	define MIXT_MONOSLIDER16	19
+#	define MIXT_STEREOSLIDER16	20
+#	define MIXT_MUTE	21	/* Mute=1, unmute=0 */
+
+  /**************************************************************/
+
+  /* Possible value range (minvalue to maxvalue) */
+  /* Note that maxvalue may also be smaller than minvalue */
+  int maxvalue;
+  int minvalue;
+
+  int flags;
+#	define MIXF_READABLE	0x00000001	/* Has readable value */
+#	define MIXF_WRITEABLE	0x00000002	/* Has writeable value */
+#	define MIXF_POLL	0x00000004	/* May change itself */
+#	define MIXF_HZ		0x00000008	/* Herz scale */
+#	define MIXF_STRING	0x00000010	/* Use dynamic extensions for value */
+#	define MIXF_DYNAMIC	0x00000010	/* Supports dynamic extensions */
+#	define MIXF_OKFAIL	0x00000020	/* Interpret value as 1=OK, 0=FAIL */
+#	define MIXF_FLAT	0x00000040	/* Flat vertical space requirements */
+#	define MIXF_LEGACY	0x00000080	/* Legacy mixer control group */
+#	define MIXF_CENTIBEL	0x00000100	/* Centibel (0.1 dB) step size */
+#	define MIXF_DECIBEL	0x00000200	/* Step size of 1 dB */
+#	define MIXF_MAINVOL	0x00000400	/* Main volume control */
+#	define MIXF_PCMVOL	0x00000800	/* PCM output volume control */
+#	define MIXF_RECVOL	0x00001000	/* PCM recording volume control */
+#	define MIXF_MONVOL	0x00002000	/* Input->output monitor volume */
+#	define MIXF_WIDE	0x00004000	/* Enum control has wide labels */
+#	define MIXF_DESCR	0x00008000	/* Description (tooltip) available */
+  char id[16];			/* Mnemonic ID (mainly for internal use) */
+  int parent;			/* Entry# of parent (group) node (-1 if root) */
+
+  int dummy;			/* Internal use */
+
+  int timestamp;
+
+  char data[64];		/* Misc data (entry type dependent) */
+  unsigned char enum_present[32];	/* Mask of allowed enum values */
+  int control_no;		/* SOUND_MIXER_VOLUME..SOUND_MIXER_MIDI */
+  /* (-1 means not indicated) */
+
+/*
+ * The desc field is reserved for internal purposes of OSS. It should not be 
+ * used by applications.
+ */
+  unsigned int desc;
+#define MIXEXT_SCOPE_MASK			0x0000003f
+#define MIXEXT_SCOPE_OTHER			0x00000000
+#define MIXEXT_SCOPE_INPUT			0x00000001
+#define MIXEXT_SCOPE_OUTPUT			0x00000002
+#define MIXEXT_SCOPE_MONITOR			0x00000003
+#define MIXEXT_SCOPE_RECSWITCH			0x00000004
+
+  char extname[32];
+  int update_counter;
+  int rgbcolor;		/* 0 means default color (not black) . Otherwise 24 bit RGB color */
+  int filler[6];
+} oss_mixext;
+
+/*
+ * Recommended colors to be used in the rgbcolor field. These match the
+ * colors used as the audio jack colors in HD audio motherboards.
+ */
+#define OSS_RGB_BLUE	0x7aabde		// Light blue
+#define OSS_RGB_GREEN	0xb3c98c		// Lime green
+#define OSS_RGB_PINK	0xe88c99
+#define OSS_RGB_GRAY	0xd1ccc4
+#define OSS_RGB_BLACK	0x2b2926		// Light black
+#define OSS_RGB_ORANGE	0xe89e47
+#define OSS_RGB_RED	0xff0000
+#define OSS_RGB_YELLOW	0xffff00
+#define OSS_RGB_PURPLE	0x800080
+#define OSS_RGB_WHITE	0xf8f8ff
+
+typedef struct oss_mixext_root
+{
+  char id[16];
+  char name[48];
+} oss_mixext_root;
+
+typedef struct oss_mixer_value
+{
+  int dev;
+  int ctrl;
+  int value;
+  int flags;			/* Reserved for future use. Initialize to 0 */
+  int timestamp;		/* Must be set to oss_mixext.timestamp */
+  int filler[8];		/* Reserved for future use. Initialize to 0 */
+} oss_mixer_value;
+
+#define OSS_ENUM_MAXVALUE	255
+#define OSS_ENUM_STRINGSIZE	3000
+typedef struct oss_mixer_enuminfo
+{
+  int dev;
+  int ctrl;
+  int nvalues;
+  int version;			/* Read the manual */
+  short strindex[OSS_ENUM_MAXVALUE];
+  char strings[OSS_ENUM_STRINGSIZE];
+} oss_mixer_enuminfo;
+
+#define OPEN_READ	PCM_ENABLE_INPUT
+#define OPEN_WRITE	PCM_ENABLE_OUTPUT
+#define OPEN_READWRITE	(OPEN_READ|OPEN_WRITE)
+
+typedef struct oss_audioinfo
+{
+  int dev;			/* Audio device number */
+  char name[64];
+  int busy;			/* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */
+  int pid;
+  int caps;			/* PCM_CAP_INPUT, PCM_CAP_OUTPUT */
+  int iformats, oformats;
+  int magic;			/* Reserved for internal use */
+  char cmd[64];			/* Command using the device (if known) */
+  int card_number;
+  int port_number;
+  int mixer_dev;
+  int legacy_device;		/* Obsolete field. Replaced by devnode */
+  int enabled;			/* 1=enabled, 0=device not ready at this moment */
+  int flags;			/* For internal use only - no practical meaning */
+  int min_rate, max_rate;	/* Sample rate limits */
+  int min_channels, max_channels;	/* Number of channels supported */
+  int binding;			/* DSP_BIND_FRONT, etc. 0 means undefined */
+  int rate_source;
+  char handle[32];
+#define OSS_MAX_SAMPLE_RATES	20	/* Cannot be changed  */
+  unsigned int nrates, rates[OSS_MAX_SAMPLE_RATES];	/* Please read the manual before using these */
+  oss_longname_t song_name;	/* Song name (if given) */
+  oss_label_t label;		/* Device label (if given) */
+  int latency;			/* In usecs, -1=unknown */
+  oss_devnode_t devnode;	/* Device special file name (absolute path) */
+  int next_play_engine;		/* Read the documentation for more info */
+  int next_rec_engine;		/* Read the documentation for more info */
+  int filler[184];
+} oss_audioinfo;
+
+typedef struct oss_mixerinfo
+{
+  int dev;
+  char id[16];
+  char name[32];
+  int modify_counter;
+  int card_number;
+  int port_number;
+  char handle[32];
+  int magic;			/* Reserved */
+  int enabled;			/* Reserved */
+  int caps;
+#define MIXER_CAP_VIRTUAL	0x00000001
+#define MIXER_CAP_LAYOUT_B	0x00000002	/* For internal use only */
+#define MIXER_CAP_NARROW	0x00000004	/* Conserve horiz space */
+  int flags;			/* Reserved */
+  int nrext;
+  /*
+   * The priority field can be used to select the default (motherboard)
+   * mixer device. The mixer with the highest priority is the
+   * most preferred one. -2 or less means that this device cannot be used
+   * as the default mixer.
+   */
+  int priority;
+  oss_devnode_t devnode;	/* Device special file name (absolute path) */
+  int legacy_device;
+  int filler[245];		/* Reserved */
+} oss_mixerinfo;
+
+typedef struct oss_midi_info
+{
+  int dev;			/* Midi device number */
+  char name[64];
+  int busy;			/* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */
+  int pid;
+  char cmd[64];			/* Command using the device (if known) */
+  int caps;
+#define MIDI_CAP_MPU401		0x00000001	/**** OBSOLETE ****/
+#define MIDI_CAP_INPUT		0x00000002
+#define MIDI_CAP_OUTPUT		0x00000004
+#define MIDI_CAP_INOUT		(MIDI_CAP_INPUT|MIDI_CAP_OUTPUT)
+#define MIDI_CAP_VIRTUAL	0x00000008	/* Pseudo device */
+#define MIDI_CAP_MTCINPUT	0x00000010	/* Supports SNDCTL_MIDI_MTCINPUT */
+#define MIDI_CAP_CLIENT		0x00000020	/* Virtual client side device */
+#define MIDI_CAP_SERVER		0x00000040	/* Virtual server side device */
+#define MIDI_CAP_INTERNAL	0x00000080	/* Internal (synth) device */
+#define MIDI_CAP_EXTERNAL	0x00000100	/* external (MIDI port) device */
+#define MIDI_CAP_PTOP		0x00000200	/* Point to point link to one device */
+#define MIDI_CAP_MTC		0x00000400	/* MTC/SMPTE (control) device */
+  int magic;			/* Reserved for internal use */
+  int card_number;
+  int port_number;
+  int enabled;			/* 1=enabled, 0=device not ready at this moment */
+  int flags;			/* For internal use only - no practical meaning */
+  char handle[32];
+  oss_longname_t song_name;	/* Song name (if known) */
+  oss_label_t label;		/* Device label (if given) */
+  int latency;			/* In usecs, -1=unknown */
+  oss_devnode_t devnode;	/* Device special file name (absolute path) */
+  int legacy_device;		/* Legacy device mapping */
+  int filler[235];
+} oss_midi_info;
+
+typedef struct oss_card_info
+{
+  int card;
+  char shortname[16];
+  char longname[128];
+  int flags;
+  char hw_info[400];
+  int intr_count, ack_count;
+  int filler[154];
+} oss_card_info;
+
+#define SNDCTL_SYSINFO		__SIOR ('X', 1, oss_sysinfo)
+#define OSS_SYSINFO		SNDCTL_SYSINFO	/* Old name */
+
+#define SNDCTL_MIX_NRMIX	__SIOR ('X', 2, int)
+#define SNDCTL_MIX_NREXT	__SIOWR('X', 3, int)
+#define SNDCTL_MIX_EXTINFO	__SIOWR('X', 4, oss_mixext)
+#define SNDCTL_MIX_READ		__SIOWR('X', 5, oss_mixer_value)
+#define SNDCTL_MIX_WRITE	__SIOWR('X', 6, oss_mixer_value)
+
+#define SNDCTL_AUDIOINFO	__SIOWR('X', 7, oss_audioinfo)
+#define SNDCTL_MIX_ENUMINFO	__SIOWR('X', 8, oss_mixer_enuminfo)
+#define SNDCTL_MIDIINFO		__SIOWR('X', 9, oss_midi_info)
+#define SNDCTL_MIXERINFO	__SIOWR('X',10, oss_mixerinfo)
+#define SNDCTL_CARDINFO		__SIOWR('X',11, oss_card_info)
+#define SNDCTL_ENGINEINFO	__SIOWR('X',12, oss_audioinfo)
+#define SNDCTL_AUDIOINFO_EX	__SIOWR('X',13, oss_audioinfo)
+
+#define SNDCTL_MIX_DESCRIPTION	__SIOWR('X',14, oss_mixer_enuminfo)
+
+/* ioctl codes 'X', 200-255 are reserved for internal use */
+
+/*
+ * Few more "globally" available ioctl calls.
+ */
+#define SNDCTL_SETSONG		__SIOW ('Y', 2, oss_longname_t)
+#define SNDCTL_GETSONG		__SIOR ('Y', 2, oss_longname_t)
+#define SNDCTL_SETNAME		__SIOW ('Y', 3, oss_longname_t)
+#define SNDCTL_SETLABEL		__SIOW ('Y', 4, oss_label_t)
+#define SNDCTL_GETLABEL		__SIOR ('Y', 4, oss_label_t)
+/*
+ * The "new" mixer API definitions end here.
+ ***************************************
+ */
+
+/*
+ *********************************************************
+ * Few routines that are included in -lOSSlib
+ *
+ * At this moment this interface is not used. OSSlib contains just
+ * stubs that call the related system calls directly.
+ */
+#ifdef OSSLIB
+extern int osslib_open (const char *path, int flags, int dummy);
+extern void osslib_close (int fd);
+extern int osslib_write (int fd, const void *buf, int count);
+extern int osslib_read (int fd, void *buf, int count);
+extern int osslib_ioctl (int fd, unsigned int request, void *arg);
+#else
+#  define osslib_open	open
+#  define osslib_close	close
+#  define osslib_write	write
+#  define osslib_read	read
+#  define osslib_ioctl	ioctl
+#endif
+
+#if 1
+#define SNDCTL_DSP_NONBLOCK		__SIO  ('P',14)	/* Obsolete. Not supported any more */
+#endif
+
+#if 1
+/*
+ * Some obsolete macros that are not part of Open Sound System API.
+ */
+#define SOUND_PCM_READ_RATE             SOUND_PCM_READ_RATE_is_obsolete
+#define SOUND_PCM_READ_BITS             SOUND_PCM_READ_BITS_is_obsolete
+#define SOUND_PCM_READ_CHANNELS         SOUND_PCM_READ_CHANNELS_is_obsolete
+#define SOUND_PCM_WRITE_RATE            SOUND_PCM_WRITE_RATE_is_obsolet_use_SNDCTL_DSP_SPEED_instead
+#define SOUND_PCM_WRITE_CHANNELS        SOUND_PCM_WRITE_CHANNELS_is_obsolete_use_SNDCTL_DSP_CHANNELS_instead
+#define SOUND_PCM_WRITE_BITS            SOUND_PCM_WRITE_BITS_is_obsolete_use_SNDCTL_DSP_SETFMT_instead
+#define SOUND_PCM_POST                  SOUND_PCM_POST_is_obsolete_use_SNDCTL_DSP_POST_instead
+#define SOUND_PCM_RESET                 SOUND_PCM_RESET_is_obsolete_use_SNDCTL_DSP_HALT_instead
+#define SOUND_PCM_SYNC                  SOUND_PCM_SYNC_is_obsolete_use_SNDCTL_DSP_SYNC_instead
+#define SOUND_PCM_SUBDIVIDE             SOUND_PCM_SUBDIVIDE_is_obsolete_use_SNDCTL_DSP_SUBDIVIDE_instead
+#define SOUND_PCM_SETFRAGMENT           SOUND_PCM_SETFRAGMENT_is_obsolete_use_SNDCTL_DSP_SETFRAGMENT_instead
+#define SOUND_PCM_GETFMTS               SOUND_PCM_GETFMTS_is_obsolete_use_SNDCTL_DSP_GETFMTS_instead
+#define SOUND_PCM_SETFMT                SOUND_PCM_SETFMT_is_obsolete_use_SNDCTL_DSP_SETFMT_instead
+#define SOUND_PCM_GETOSPACE             SOUND_PCM_GETOSPACE_is_obsolete_use_SNDCTL_DSP_GETOSPACE_instead
+#define SOUND_PCM_GETISPACE             SOUND_PCM_GETISPACE_is_obsolete_use_SNDCTL_DSP_GETISPACE_instead
+#define SOUND_PCM_NONBLOCK              SOUND_PCM_NONBLOCK_is_obsolete_use_SNDCTL_DSP_NONBLOCK_instead
+#define SOUND_PCM_GETCAPS               SOUND_PCM_GETCAPS_is_obsolete_use_SNDCTL_DSP_GETCAPS_instead
+#define SOUND_PCM_GETTRIGGER            SOUND_PCM_GETTRIGGER_is_obsolete_use_SNDCTL_DSP_GETTRIGGER_instead
+#define SOUND_PCM_SETTRIGGER            SOUND_PCM_SETTRIGGER_is_obsolete_use_SNDCTL_DSP_SETTRIGGER_instead
+#define SOUND_PCM_SETSYNCRO             SOUND_PCM_SETSYNCRO_is_obsolete_use_SNDCTL_DSP_SETSYNCRO_instead
+#define SOUND_PCM_GETIPTR               SOUND_PCM_GETIPTR_is_obsolete_use_SNDCTL_DSP_GETIPTR_instead
+#define SOUND_PCM_GETOPTR               SOUND_PCM_GETOPTR_is_obsolete_use_SNDCTL_DSP_GETOPTR_instead
+#define SOUND_PCM_MAPINBUF              SOUND_PCM_MAPINBUF_is_obsolete_use_SNDCTL_DSP_MAPINBUF_instead
+#define SOUND_PCM_MAPOUTBUF             SOUND_PCM_MAPOUTBUF_is_obsolete_use_SNDCTL_DSP_MAPOUTBUF_instead
+#endif
+
+#endif
diff --git a/sys/oss4/oss4-source.c b/sys/oss4/oss4-source.c
new file mode 100644
index 00000000..1cf2328f
--- /dev/null
+++ b/sys/oss4/oss4-source.c
@@ -0,0 +1,1004 @@
+/* GStreamer OSS4 audio source
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-oss4src
+ * @short_description: record sound from your sound card using OSS4
+ *
+ * <refsect2>
+ * <para>
+ * This element lets you record sound using the Open Sound System (OSS)
+ * version 4.
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v oss4src ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
+ * </programlisting>
+ * will record sound from your sound card using OSS4 and encode it to an
+ * Ogg/Vorbis file (this will only work if your mixer settings are right
+ * and the right inputs areenabled etc.)
+ * </para>
+ * </refsect2>
+ *
+ * Since: 0.10.7
+ */
+
+/* FIXME: make sure we're not doing ioctls from the app thread (e.g. via the
+ * mixer interface) while recording */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+
+#include <gst/interfaces/mixer.h>
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-audio.h"
+#include "oss4-source.h"
+#include "oss4-property-probe.h"
+#include "oss4-soundcard.h"
+
+#define GST_OSS4_SOURCE_IS_OPEN(src)  (GST_OSS4_SOURCE(src)->fd != -1)
+
+GST_DEBUG_CATEGORY_EXTERN (oss4src_debug);
+#define GST_CAT_DEFAULT oss4src_debug
+
+#define DEFAULT_DEVICE       NULL
+#define DEFAULT_DEVICE_NAME  NULL
+
+enum
+{
+  PROP_0,
+  PROP_DEVICE,
+  PROP_DEVICE_NAME
+};
+
+static void gst_oss4_source_init_interfaces (GType type);
+
+GST_BOILERPLATE_FULL (GstOss4Source, gst_oss4_source, GstAudioSrc,
+    GST_TYPE_AUDIO_SRC, gst_oss4_source_init_interfaces);
+
+static void gst_oss4_source_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_oss4_source_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+
+static void gst_oss4_source_dispose (GObject * object);
+static void gst_oss4_source_finalize (GstOss4Source * osssrc);
+
+static GstCaps *gst_oss4_source_getcaps (GstBaseSrc * bsrc);
+
+static gboolean gst_oss4_source_open (GstAudioSrc * asrc,
+    gboolean silent_errors);
+static gboolean gst_oss4_source_open_func (GstAudioSrc * asrc);
+static gboolean gst_oss4_source_close (GstAudioSrc * asrc);
+static gboolean gst_oss4_source_prepare (GstAudioSrc * asrc,
+    GstRingBufferSpec * spec);
+static gboolean gst_oss4_source_unprepare (GstAudioSrc * asrc);
+static guint gst_oss4_source_read (GstAudioSrc * asrc, gpointer data,
+    guint length);
+static guint gst_oss4_source_delay (GstAudioSrc * asrc);
+static void gst_oss4_source_reset (GstAudioSrc * asrc);
+
+static void
+gst_oss4_source_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  GstPadTemplate *templ;
+
+  gst_element_class_set_details_simple (element_class,
+      "OSS v4 Audio Source", "Source/Audio",
+      "Capture from a sound card via OSS version 4",
+      "Tim-Philipp Müller <tim centricular net>");
+
+  templ = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+      gst_oss4_audio_get_template_caps ());
+  gst_element_class_add_pad_template (element_class, templ);
+}
+static void
+gst_oss4_source_class_init (GstOss4SourceClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSrcClass *gstbasesrc_class;
+  GstBaseAudioSrcClass *gstbaseaudiosrc_class;
+  GstAudioSrcClass *gstaudiosrc_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasesrc_class = (GstBaseSrcClass *) klass;
+  gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
+  gstaudiosrc_class = (GstAudioSrcClass *) klass;
+
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss4_source_dispose);
+  gobject_class->finalize =
+      (GObjectFinalizeFunc) GST_DEBUG_FUNCPTR (gst_oss4_source_finalize);
+  gobject_class->get_property =
+      GST_DEBUG_FUNCPTR (gst_oss4_source_get_property);
+  gobject_class->set_property =
+      GST_DEBUG_FUNCPTR (gst_oss4_source_set_property);
+
+  gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss4_source_getcaps);
+
+  gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss4_source_open_func);
+  gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss4_source_prepare);
+  gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss4_source_unprepare);
+  gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss4_source_close);
+  gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss4_source_read);
+  gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss4_source_delay);
+  gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss4_source_reset);
+
+  g_object_class_install_property (gobject_class, PROP_DEVICE,
+      g_param_spec_string ("device", "Device",
+          "OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) "
+          "(NULL = use first available device)",
+          DEFAULT_DEVICE, G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+      g_param_spec_string ("device-name", "Device name",
+          "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+          G_PARAM_READABLE));
+}
+
+static void
+gst_oss4_source_init (GstOss4Source * osssrc, GstOss4SourceClass * g_class)
+{
+  const gchar *device;
+
+  device = g_getenv ("AUDIODEV");
+  if (device == NULL)
+    device = DEFAULT_DEVICE;
+
+  osssrc->fd = -1;
+  osssrc->device = g_strdup (device);
+  osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME);
+  osssrc->device_name = NULL;
+}
+
+static void
+gst_oss4_source_finalize (GstOss4Source * oss)
+{
+  g_free (oss->device);
+  oss->device = NULL;
+
+  g_list_free (oss->property_probe_list);
+  oss->property_probe_list = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (oss));
+}
+
+static void
+gst_oss4_source_dispose (GObject * object)
+{
+  GstOss4Source *oss = GST_OSS4_SOURCE (object);
+
+  if (oss->probed_caps) {
+    gst_caps_unref (oss->probed_caps);
+    oss->probed_caps = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_oss4_source_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOss4Source *oss;
+
+  oss = GST_OSS4_SOURCE (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      GST_OBJECT_LOCK (oss);
+      if (oss->fd == -1) {
+        g_free (oss->device);
+        oss->device = g_value_dup_string (value);
+        g_free (oss->device_name);
+        oss->device_name = NULL;
+      } else {
+        g_warning ("%s: can't change \"device\" property while audio source "
+            "is open", GST_OBJECT_NAME (oss));
+      }
+      GST_OBJECT_UNLOCK (oss);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_oss4_source_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstOss4Source *oss;
+
+  oss = GST_OSS4_SOURCE (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      GST_OBJECT_LOCK (oss);
+      g_value_set_string (value, oss->device);
+      GST_OBJECT_UNLOCK (oss);
+      break;
+    case PROP_DEVICE_NAME:
+      GST_OBJECT_LOCK (oss);
+      /* If device is set, try to retrieve the name even if we're not open */
+      if (oss->fd == -1 && oss->device != NULL) {
+        if (gst_oss4_source_open (GST_AUDIO_SRC (oss), TRUE)) {
+          g_value_set_string (value, oss->device_name);
+          gst_oss4_source_close (GST_AUDIO_SRC (oss));
+        } else {
+          g_value_set_string (value, NULL);
+        }
+      } else {
+        g_value_set_string (value, oss->device_name);
+      }
+      GST_OBJECT_UNLOCK (oss);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_oss4_source_getcaps (GstBaseSrc * bsrc)
+{
+  GstOss4Source *oss;
+  GstCaps *caps;
+
+  oss = GST_OSS4_SOURCE (bsrc);
+
+  if (oss->fd == -1) {
+    caps = gst_caps_copy (gst_oss4_audio_get_template_caps ());
+  } else if (oss->probed_caps) {
+    caps = gst_caps_copy (oss->probed_caps);
+  } else {
+    caps = gst_oss4_audio_probe_caps (GST_OBJECT (oss), oss->fd);
+    if (caps != NULL && !gst_caps_is_empty (caps)) {
+      oss->probed_caps = gst_caps_copy (caps);
+    }
+  }
+
+  return caps;
+}
+
+/* note: we must not take the object lock here unless we fix up get_property */
+static gboolean
+gst_oss4_source_open (GstAudioSrc * asrc, gboolean silent_errors)
+{
+  GstOss4Source *oss;
+  gchar *device;
+  int mode;
+
+  oss = GST_OSS4_SOURCE (asrc);
+
+  if (oss->device)
+    device = g_strdup (oss->device);
+  else
+    device = gst_oss4_audio_find_device (GST_OBJECT_CAST (oss));
+
+  /* desperate times, desperate measures */
+  if (device == NULL)
+    device = g_strdup ("/dev/dsp0");
+
+  GST_INFO_OBJECT (oss, "Trying to open OSS4 device '%s'", device);
+
+  /* we open in non-blocking mode even if we don't really want to do writes
+   * non-blocking because we can't be sure that this is really a genuine
+   * OSS4 device with well-behaved drivers etc. We really don't want to
+   * hang forever under any circumstances. */
+  oss->fd = open (device, O_RDONLY | O_NONBLOCK, 0);
+  if (oss->fd == -1) {
+    switch (errno) {
+      case EBUSY:
+        goto busy;
+      case EACCES:
+        goto no_permission;
+      default:
+        goto open_failed;
+    }
+  }
+
+  GST_INFO_OBJECT (oss, "Opened device");
+
+  /* Make sure it's OSS4. If it's old OSS, let osssink handle it */
+  if (!gst_oss4_audio_check_version (GST_OBJECT_CAST (oss), oss->fd))
+    goto legacy_oss;
+
+  /* now remove the non-blocking flag. */
+  mode = fcntl (oss->fd, F_GETFL);
+  mode &= ~O_NONBLOCK;
+  if (fcntl (oss->fd, F_SETFL, mode) < 0) {
+    /* some drivers do no support unsetting the non-blocking flag, try to
+     * close/open the device then. This is racy but we error out properly. */
+    GST_WARNING_OBJECT (oss, "failed to unset O_NONBLOCK (buggy driver?), "
+        "will try to re-open device now");
+    gst_oss4_source_close (asrc);
+    if ((oss->fd = open (device, O_RDONLY, 0)) == -1)
+      goto non_block;
+  }
+
+  oss->open_device = device;
+
+  /* not using ENGINEINFO here because it sometimes returns a different and
+   * less useful name than AUDIOINFO for the same device */
+  if (!gst_oss4_property_probe_find_device_name (GST_OBJECT (oss), oss->fd,
+          oss->open_device, &oss->device_name)) {
+    oss->device_name = NULL;
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+busy:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
+          (_("Could not open audio device for playback. "
+                  "Device is being used by another application.")), (NULL));
+    }
+    g_free (device);
+    return FALSE;
+  }
+no_permission:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+          (_("Could not open audio device for playback."
+                  "You don't have permission to open the device.")),
+          GST_ERROR_SYSTEM);
+    }
+    g_free (device);
+    return FALSE;
+  }
+open_failed:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+          (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
+    }
+    g_free (device);
+    return FALSE;
+  }
+legacy_oss:
+  {
+    gst_oss4_source_close (asrc);
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+          (_("Could not open audio device for playback."
+                  "This version of the Open Sound System is not supported by this "
+                  "element.")), ("Try the 'osssink' element instead"));
+    }
+    g_free (device);
+    return FALSE;
+  }
+non_block:
+  {
+    if (!silent_errors) {
+      GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
+          ("Unable to set device %s into non-blocking mode: %s",
+              oss->device, g_strerror (errno)));
+    }
+    g_free (device);
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_oss4_source_open_func (GstAudioSrc * asrc)
+{
+  return gst_oss4_source_open (asrc, FALSE);
+}
+
+static void
+gst_oss4_source_free_mixer_tracks (GstOss4Source * oss)
+{
+  g_list_foreach (oss->tracks, (GFunc) g_object_unref, NULL);
+  g_list_free (oss->tracks);
+  oss->tracks = NULL;
+}
+
+static gboolean
+gst_oss4_source_close (GstAudioSrc * asrc)
+{
+  GstOss4Source *oss;
+
+  oss = GST_OSS4_SOURCE (asrc);
+
+  if (oss->fd != -1) {
+    GST_DEBUG_OBJECT (oss, "closing device");
+    close (oss->fd);
+    oss->fd = -1;
+  }
+
+  oss->bytes_per_sample = 0;
+
+  gst_caps_replace (&oss->probed_caps, NULL);
+
+  g_free (oss->open_device);
+  oss->open_device = NULL;
+
+  g_free (oss->device_name);
+  oss->device_name = NULL;
+
+  gst_oss4_source_free_mixer_tracks (oss);
+
+  return TRUE;
+}
+
+static gboolean
+gst_oss4_source_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
+{
+  GstOss4Source *oss;
+
+  oss = GST_OSS4_SOURCE (asrc);
+
+  if (!gst_oss4_audio_set_format (GST_OBJECT_CAST (oss), oss->fd, spec)) {
+    GST_WARNING_OBJECT (oss, "Couldn't set requested format %" GST_PTR_FORMAT,
+        spec->caps);
+    return FALSE;
+  }
+
+  oss->bytes_per_sample = spec->bytes_per_sample;
+  return TRUE;
+}
+
+static gboolean
+gst_oss4_source_unprepare (GstAudioSrc * asrc)
+{
+  /* could do a SNDCTL_DSP_HALT, but the OSS manual recommends a close/open,
+   * since HALT won't properly reset some devices, apparently */
+
+  if (!gst_oss4_source_close (asrc))
+    goto couldnt_close;
+
+  if (!gst_oss4_source_open_func (asrc))
+    goto couldnt_reopen;
+
+  return TRUE;
+
+  /* ERRORS */
+couldnt_close:
+  {
+    GST_DEBUG_OBJECT (asrc, "Couldn't close the audio device");
+    return FALSE;
+  }
+couldnt_reopen:
+  {
+    GST_DEBUG_OBJECT (asrc, "Couldn't reopen the audio device");
+    return FALSE;
+  }
+}
+
+static guint
+gst_oss4_source_read (GstAudioSrc * asrc, gpointer data, guint length)
+{
+  GstOss4Source *oss;
+  int n;
+
+  oss = GST_OSS4_SOURCE_CAST (asrc);
+
+  n = read (oss->fd, data, length);
+  GST_LOG_OBJECT (asrc, "%u bytes, %u samples", n, n / oss->bytes_per_sample);
+
+  if (G_UNLIKELY (n < 0)) {
+    switch (errno) {
+      case ENOTSUP:
+      case EACCES:{
+        /* This is the most likely cause, I think */
+        GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
+            (_("Recording is not supported by this audio device.")),
+            ("read: %s (device: %s) (maybe this is an output-only device?)",
+                g_strerror (errno), oss->open_device));
+        break;
+      }
+      default:{
+        GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
+            (_("Error recording from audio device.")),
+            ("read: %s (device: %s)", g_strerror (errno), oss->open_device));
+        break;
+      }
+    }
+  }
+
+  return (guint) n;
+}
+
+static guint
+gst_oss4_source_delay (GstAudioSrc * asrc)
+{
+  audio_buf_info info = { 0, };
+  GstOss4Source *oss;
+  guint delay;
+
+  oss = GST_OSS4_SOURCE_CAST (asrc);
+
+  if (ioctl (oss->fd, SNDCTL_DSP_GETISPACE, &info) == -1) {
+    GST_LOG_OBJECT (oss, "GETISPACE failed: %s", g_strerror (errno));
+    return 0;
+  }
+
+  delay = (info.fragstotal * info.fragsize) - info.bytes;
+  GST_LOG_OBJECT (oss, "fragstotal:%d, fragsize:%d, bytes:%d, delay:%d");
+  return delay;
+}
+
+static void
+gst_oss4_source_reset (GstAudioSrc * asrc)
+{
+  /* There's nothing we can do here really: OSS can't handle access to the
+   * same device/fd from multiple threads and might deadlock or blow up in
+   * other ways if we try an ioctl SNDCTL_DSP_HALT or similar */
+}
+
+/* GstMixer interface, which we abuse here for input selection, because we
+ * don't have a proper interface for that and because that's what
+ * gnome-sound-recorder does. */
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4SourceInput, gst_oss4_source_input, GST_TYPE_MIXER_TRACK);
+
+static void
+gst_oss4_source_input_class_init (GstOss4SourceInputClass * klass)
+{
+  /* nothing to do here */
+}
+
+static void
+gst_oss4_source_input_init (GstOss4SourceInput * i)
+{
+  /* nothing to do here */
+}
+
+#if 0
+
+static void
+gst_ossmixer_ensure_track_list (GstOssMixer * mixer)
+{
+  gint i, master = -1;
+
+  g_return_if_fail (mixer->fd != -1);
+
+  if (mixer->tracklist)
+    return;
+
+  /* find master volume */
+  if (mixer->devmask & SOUND_MASK_VOLUME)
+    master = SOUND_MIXER_VOLUME;
+  else if (mixer->devmask & SOUND_MASK_PCM)
+    master = SOUND_MIXER_PCM;
+  else if (mixer->devmask & SOUND_MASK_SPEAKER)
+    master = SOUND_MIXER_SPEAKER;       /* doubtful... */
+  /* else: no master, so we won't set any */
+
+  /* build track list */
+  for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
+    if (mixer->devmask & (1 << i)) {
+      GstMixerTrack *track;
+      gboolean input = FALSE, stereo = FALSE, record = FALSE;
+
+      /* track exists, make up capabilities */
+      if (MASK_BIT_IS_SET (mixer->stereomask, i))
+        stereo = TRUE;
+      if (MASK_BIT_IS_SET (mixer->recmask, i))
+        input = TRUE;
+      if (MASK_BIT_IS_SET (mixer->recdevs, i))
+        record = TRUE;
+
+      /* do we want mixer in our list? */
+      if (!((mixer->dir & GST_OSS_MIXER_CAPTURE && input == TRUE) ||
+              (mixer->dir & GST_OSS_MIXER_PLAYBACK && i != SOUND_MIXER_PCM)))
+        /* the PLAYBACK case seems hacky, but that's how 0.8 had it */
+        continue;
+
+      /* add track to list */
+      track = gst_ossmixer_track_new (mixer->fd, i, stereo ? 2 : 1,
+          (record ? GST_MIXER_TRACK_RECORD : 0) |
+          (input ? GST_MIXER_TRACK_INPUT :
+              GST_MIXER_TRACK_OUTPUT) |
+          ((master != i) ? 0 : GST_MIXER_TRACK_MASTER));
+      mixer->tracklist = g_list_append (mixer->tracklist, track);
+    }
+  }
+}
+
+/* unused with G_DISABLE_* */
+static G_GNUC_UNUSED gboolean
+gst_ossmixer_contains_track (GstOssMixer * mixer, GstOssMixerTrack * osstrack)
+{
+  const GList *item;
+
+  for (item = mixer->tracklist; item != NULL; item = item->next)
+    if (item->data == osstrack)
+      return TRUE;
+
+  return FALSE;
+}
+
+const GList *
+gst_ossmixer_list_tracks (GstOssMixer * mixer)
+{
+  gst_ossmixer_ensure_track_list (mixer);
+
+  return (const GList *) mixer->tracklist;
+}
+
+void
+gst_ossmixer_get_volume (GstOssMixer * mixer,
+    GstMixerTrack * track, gint * volumes)
+{
+  gint volume;
+  GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
+
+  g_return_if_fail (mixer->fd != -1);
+  g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
+
+  if (track->flags & GST_MIXER_TRACK_MUTE) {
+    volumes[0] = osstrack->lvol;
+    if (track->num_channels == 2) {
+      volumes[1] = osstrack->rvol;
+    }
+  } else {
+    /* get */
+    if (ioctl (mixer->fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
+      g_warning ("Error getting recording device (%d) volume: %s",
+          osstrack->track_num, g_strerror (errno));
+      volume = 0;
+    }
+
+    osstrack->lvol = volumes[0] = (volume & 0xff);
+    if (track->num_channels == 2) {
+      osstrack->rvol = volumes[1] = ((volume >> 8) & 0xff);
+    }
+  }
+}
+
+void
+gst_ossmixer_set_mute (GstOssMixer * mixer, GstMixerTrack * track,
+    gboolean mute)
+{
+  int volume;
+  GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
+
+  g_return_if_fail (mixer->fd != -1);
+  g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
+
+  if (mute) {
+    volume = 0;
+  } else {
+    volume = (osstrack->lvol & 0xff);
+    if (MASK_BIT_IS_SET (mixer->stereomask, osstrack->track_num)) {
+      volume |= ((osstrack->rvol & 0xff) << 8);
+    }
+  }
+
+  if (ioctl (mixer->fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
+    g_warning ("Error setting mixer recording device volume (0x%x): %s",
+        volume, g_strerror (errno));
+    return;
+  }
+
+  if (mute) {
+    track->flags |= GST_MIXER_TRACK_MUTE;
+  } else {
+    track->flags &= ~GST_MIXER_TRACK_MUTE;
+  }
+}
+#endif
+
+static gint
+gst_oss4_source_mixer_get_current_input (GstOss4Source * oss)
+{
+  int cur = -1;
+
+  if (ioctl (oss->fd, SNDCTL_DSP_GET_RECSRC, &cur) == -1 || cur < 0)
+    return -1;
+
+  return cur;
+}
+
+static const gchar *
+gst_oss4_source_mixer_update_record_flags (GstOss4Source * oss, gint cur_route)
+{
+  const gchar *cur_name = "";
+  GList *t;
+
+  for (t = oss->tracks; t != NULL; t = t->next) {
+    GstMixerTrack *track = t->data;
+
+    if (GST_OSS4_SOURCE_INPUT (track)->route == cur_route) {
+      if (!GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) {
+        track->flags |= GST_MIXER_TRACK_RECORD;
+        /* no point in sending a mixer-record-changes message here */
+      }
+      cur_name = track->label;
+    } else {
+      if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) {
+        track->flags &= ~GST_MIXER_TRACK_RECORD;
+        /* no point in sending a mixer-record-changes message here */
+      }
+    }
+  }
+
+  return cur_name;
+}
+
+static const GList *
+gst_oss4_source_mixer_list_tracks (GstMixer * mixer)
+{
+  oss_mixer_enuminfo names = { 0, };
+  GstOss4Source *oss;
+  const gchar *cur_name;
+  GList *tracks = NULL;
+  gint i, cur;
+
+  g_return_val_if_fail (mixer != NULL, NULL);
+  g_return_val_if_fail (GST_IS_OSS4_SOURCE (mixer), NULL);
+  g_return_val_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer), NULL);
+
+  oss = GST_OSS4_SOURCE (mixer);
+
+  if (oss->tracks != NULL && oss->tracks_static)
+    goto done;
+
+  if (ioctl (oss->fd, SNDCTL_DSP_GET_RECSRC_NAMES, &names) == -1)
+    goto get_recsrc_names_error;
+
+  oss->tracks_static = (names.version == 0);
+
+  GST_INFO_OBJECT (oss, "%d inputs (list is static: %s):", names.nvalues,
+      (oss->tracks_static) ? "yes" : "no");
+
+  for (i = 0; i < MIN (names.nvalues, OSS_ENUM_MAXVALUE + 1); ++i) {
+    GstMixerTrack *track;
+
+    track = g_object_new (GST_TYPE_OSS4_SOURCE_INPUT, NULL);
+    track->label = g_strdup (&names.strings[names.strindex[i]]);
+    track->flags = GST_MIXER_TRACK_INPUT;
+    track->num_channels = 2;
+    track->min_volume = 0;
+    track->max_volume = 100;
+    GST_OSS4_SOURCE_INPUT (track)->route = i;
+
+    GST_INFO_OBJECT (oss, " [%d] %s", i, track->label);
+    tracks = g_list_append (tracks, track);
+  }
+
+  gst_oss4_source_free_mixer_tracks (oss);
+  oss->tracks = tracks;
+
+done:
+
+  /* update RECORD flags */
+  cur = gst_oss4_source_mixer_get_current_input (oss);
+  cur_name = gst_oss4_source_mixer_update_record_flags (oss, cur);
+  GST_DEBUG_OBJECT (oss, "current input route: %d (%s)", cur, cur_name);
+
+  return (const GList *) oss->tracks;
+
+/* ERRORS */
+get_recsrc_names_error:
+  {
+    GST_WARNING_OBJECT (oss, "GET_RECSRC_NAMES failed: %s", g_strerror (errno));
+    return NULL;
+  }
+}
+
+static void
+gst_oss4_source_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
+    gint * volumes)
+{
+  GstOss4Source *oss;
+  int new_vol, cur;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (track != NULL);
+  g_return_if_fail (GST_IS_MIXER_TRACK (track));
+  g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+  g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+  oss = GST_OSS4_SOURCE (mixer);
+
+  cur = gst_oss4_source_mixer_get_current_input (oss);
+  if (cur != GST_OSS4_SOURCE_INPUT (track)->route) {
+    GST_DEBUG_OBJECT (oss, "track not selected input route, ignoring request");
+    return;
+  }
+
+  new_vol = (volumes[1] << 8) | volumes[0];
+  if (ioctl (oss->fd, SNDCTL_DSP_SETRECVOL, &new_vol) == -1) {
+    GST_WARNING_OBJECT (oss, "SETRECVOL failed: %s", g_strerror (errno));
+  }
+}
+
+static void
+gst_oss4_source_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
+    gint * volumes)
+{
+  GstOss4Source *oss;
+  int cur;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+  g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+  oss = GST_OSS4_SOURCE (mixer);
+
+  cur = gst_oss4_source_mixer_get_current_input (oss);
+  if (cur != GST_OSS4_SOURCE_INPUT (track)->route) {
+    volumes[0] = 0;
+    volumes[1] = 0;
+  } else {
+    int vol = -1;
+
+    if (ioctl (oss->fd, SNDCTL_DSP_GETRECVOL, &vol) == -1 || vol < 0) {
+      GST_WARNING_OBJECT (oss, "GETRECVOL failed: %s", g_strerror (errno));
+      volumes[0] = 100;
+      volumes[1] = 100;
+    } else {
+      volumes[0] = MIN (100, vol & 0xff);
+      volumes[1] = MIN (100, (vol >> 8) & 0xff);
+    }
+  }
+}
+
+static void
+gst_oss4_source_mixer_set_record (GstMixer * mixer, GstMixerTrack * track,
+    gboolean record)
+{
+  GstOss4Source *oss;
+  const gchar *cur_name;
+  gint cur;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (track != NULL);
+  g_return_if_fail (GST_IS_MIXER_TRACK (track));
+  g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+  g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+  oss = GST_OSS4_SOURCE (mixer);
+
+  cur = gst_oss4_source_mixer_get_current_input (oss);
+
+  /* stop recording for an input that's not selected anyway => nothing to do */
+  if (!record && cur != GST_OSS4_SOURCE_INPUT (track)->route)
+    goto done;
+
+  /* select recording for an input that's already selected => nothing to do
+   * (or should we mess with the recording volume in this case maybe?) */
+  if (record && cur == GST_OSS4_SOURCE_INPUT (track)->route)
+    goto done;
+
+  /* make current input stop recording: we can't really make an input stop
+   * recording, we can only select an input FOR recording, so we'll just ignore
+   * all requests to stop for now */
+  if (!record) {
+    GST_WARNING_OBJECT (oss, "Can't un-select an input as such, only switch "
+        "to a different input source");
+    /* FIXME: set recording volume to 0 maybe? */
+  } else {
+    int new_route = GST_OSS4_SOURCE_INPUT (track)->route;
+
+    /* select this input for recording */
+
+    if (ioctl (oss->fd, SNDCTL_DSP_SET_RECSRC, &new_route) == -1) {
+      GST_WARNING_OBJECT (oss, "Could not select input %d for recording: %s",
+          new_route, g_strerror (errno));
+    } else {
+      cur = new_route;
+    }
+  }
+
+done:
+
+  cur_name = gst_oss4_source_mixer_update_record_flags (oss, cur);
+  GST_DEBUG_OBJECT (oss, "active input route: %d (%s)", cur, cur_name);
+}
+
+static void
+gst_oss4_source_mixer_set_mute (GstMixer * mixer, GstMixerTrack * track,
+    gboolean mute)
+{
+  GstOss4Source *oss;
+
+  g_return_if_fail (mixer != NULL);
+  g_return_if_fail (track != NULL);
+  g_return_if_fail (GST_IS_MIXER_TRACK (track));
+  g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+  g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+  oss = GST_OSS4_SOURCE (mixer);
+
+  /* FIXME: implement gst_oss4_source_mixer_set_mute() - what to do here? */
+  /* oss4_mixer_set_mute (mixer->mixer, track, mute); */
+}
+
+static void
+gst_oss4_source_mixer_interface_init (GstMixerClass * klass)
+{
+  GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE;
+
+  klass->list_tracks = gst_oss4_source_mixer_list_tracks;
+  klass->set_volume = gst_oss4_source_mixer_set_volume;
+  klass->get_volume = gst_oss4_source_mixer_get_volume;
+  klass->set_mute = gst_oss4_source_mixer_set_mute;
+  klass->set_record = gst_oss4_source_mixer_set_record;
+}
+
+/* Implement the horror that is GstImplementsInterface */
+
+static gboolean
+gst_oss4_source_mixer_supported (GstImplementsInterface * iface,
+    GType iface_type)
+{
+  GstOss4Source *oss;
+  gboolean is_open;
+
+  g_return_val_if_fail (GST_IS_OSS4_SOURCE (iface), FALSE);
+  g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
+
+  oss = GST_OSS4_SOURCE (iface);
+
+  GST_OBJECT_LOCK (oss);
+  is_open = GST_OSS4_SOURCE_IS_OPEN (iface);
+  GST_OBJECT_UNLOCK (oss);
+
+  return is_open;
+}
+
+static void
+gst_oss4_source_mixer_implements_interface_init (GstImplementsInterfaceClass *
+    klass)
+{
+  klass->supported = gst_oss4_source_mixer_supported;
+}
+
+static void
+gst_oss4_source_init_interfaces (GType type)
+{
+  static const GInterfaceInfo implements_iface_info = {
+    (GInterfaceInitFunc) gst_oss4_source_mixer_implements_interface_init,
+    NULL,
+    NULL,
+  };
+  static const GInterfaceInfo mixer_iface_info = {
+    (GInterfaceInitFunc) gst_oss4_source_mixer_interface_init,
+    NULL,
+    NULL,
+  };
+
+  g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
+      &implements_iface_info);
+  g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
+
+  gst_oss4_add_property_probe_interface (type);
+}
diff --git a/sys/oss4/oss4-source.h b/sys/oss4/oss4-source.h
new file mode 100644
index 00000000..3a86b43a
--- /dev/null
+++ b/sys/oss4/oss4-source.h
@@ -0,0 +1,89 @@
+/* GStreamer OSS4 audio source
+ * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef GST_OSS4_SOURCE_H
+#define GST_OSS4_SOURCE_H
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosrc.h>
+#include <gst/interfaces/mixertrack.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_SOURCE            (gst_oss4_source_get_type())
+#define GST_OSS4_SOURCE(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_SOURCE,GstOss4Source))
+#define GST_OSS4_SOURCE_CAST(obj)       ((GstOss4Source *)(obj))
+#define GST_OSS4_SOURCE_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_SOURCE,GstOss4SourceClass))
+#define GST_IS_OSS4_SOURCE(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_SOURCE))
+#define GST_IS_OSS4_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_SOURCE))
+
+typedef struct _GstOss4Source GstOss4Source;
+typedef struct _GstOss4SourceClass GstOss4SourceClass;
+
+struct _GstOss4Source {
+  GstAudioSrc     audiosrc;
+
+  gchar         * device;             /* NULL if none was set      */
+  gchar         * open_device;        /* the device we opened      */
+  gchar         * device_name;        /* set if the device is open */
+  gint            fd;                 /* -1 if not open            */
+  gint            bytes_per_sample;
+
+  GstCaps       * probed_caps;
+
+  /* property probe interface */
+  GList         * property_probe_list;
+
+  /* mixer interface */
+  GList         * tracks;
+  gboolean        tracks_static;  /* FALSE if the list of inputs may change */
+};
+
+struct _GstOss4SourceClass {
+  GstAudioSrcClass audiosrc_class;
+};
+
+GType  gst_oss4_source_get_type (void);
+
+/* our mixer track for input selection */
+#define GST_TYPE_OSS4_SOURCE_INPUT            (gst_oss4_source_input_get_type())
+#define GST_OSS4_SOURCE_INPUT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_SOURCE_INPUT,GstOss4SourceInput))
+#define GST_OSS4_SOURCE_INPUT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_SOURCE_INPUT,GstOss4SourceInputClass))
+#define GST_IS_OSS4_SOURCE_INPUT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_SOURCE_INPUT))
+#define GST_IS_OSS4_SOURCE_INPUT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_SOURCE_INPUT))
+
+typedef struct _GstOss4SourceInput GstOss4SourceInput;
+typedef struct _GstOss4SourceInputClass GstOss4SourceInputClass;
+
+struct _GstOss4SourceInput {
+  GstMixerTrack mixer_track;
+
+  int           route; /* number for SNDCTL_DSP_SET_RECSRC etc. */
+};
+
+struct _GstOss4SourceInputClass {
+  GstMixerTrackClass mixer_track_class;
+};
+
+GType  gst_oss4_source_input_get_type (void);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_SOURCE_H */
+