Git init

17 files changed, 3834 insertions, 0 deletions

diff --git a/sys/oss/Makefile.am b/sys/oss/Makefile.am
new file mode 100644
index 0000000..c42dfea
--- /dev/null
+++ b/sys/oss/Makefile.am
@@ -0,0 +1,28 @@
+plugin_LTLIBRARIES = libgstossaudio.la
+
+libgstossaudio_la_SOURCES = gstossaudio.c	 \
+		            gstosshelper.c	 \
+			    gstossmixer.c	 \
+			    gstossmixerelement.c \
+			    gstossmixertrack.c	 \
+			    gstosssink.c	 \
+			    gstosssrc.c
+
+libgstossaudio_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
+libgstossaudio_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) \
+        -lgstinterfaces-$(GST_MAJORMINOR) \
+	-lgstaudio-$(GST_MAJORMINOR) \
+	$(GST_BASE_LIBS) \
+	$(GST_LIBS)
+libgstossaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) 
+libgstossaudio_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = common.h            \
+		gstosssink.h	     \
+		gstosssrc.h	     \
+                gstosshelper.h	     \
+		gstossdmabuffer.h    \
+		gstossmixer.h        \
+		gstossmixerelement.h \
+		gstossmixertrack.h
diff --git a/sys/oss/Makefile.in b/sys/oss/Makefile.in
new file mode 100644
index 0000000..1b1a00e
--- /dev/null
+++ b/sys/oss/Makefile.in
@@ -0,0 +1,884 @@
+# Makefile.in generated by automake 1.11.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
+# 2003, 2004, 2005, 2006, 2007, 2008, 2009  Free Software Foundation,
+# Inc.
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
+
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+transform = $(program_transform_name)
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+POST_INSTALL = :
+NORMAL_UNINSTALL = :
+PRE_UNINSTALL = :
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+host_triplet = @host@
+subdir = sys/oss
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+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1)
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+	libgstossaudio_la-gstosshelper.lo \
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+	libgstossaudio_la-gstossmixerelement.lo \
+	libgstossaudio_la-gstossmixertrack.lo \
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+			    gstossmixertrack.c	 \
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+	$(GST_BASE_LIBS) \
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+		gstossdmabuffer.h    \
+		gstossmixer.h        \
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+		gstossmixertrack.h
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+	      exit 1;; \
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+	  test "$$dir" != "$$p" || dir=.; \
+	  echo "rm -f \"$${dir}/so_locations\""; \
+	  rm -f "$${dir}/so_locations"; \
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+	  done | \
+	  $(AWK) '{ files[$$0] = 1; nonempty = 1; } \
+	      END { if (nonempty) { for (i in files) print i; }; }'`; \
+	shift; \
+	if test -z "$(ETAGS_ARGS)$$*$$unique"; then :; else \
+	  test -n "$$unique" || unique=$$empty_fix; \
+	  if test $$# -gt 0; then \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      "$$@" $$unique; \
+	  else \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      $$unique; \
+	  fi; \
+	fi
+ctags: CTAGS
+CTAGS:  $(HEADERS) $(SOURCES)  $(TAGS_DEPENDENCIES) \
+		$(TAGS_FILES) $(LISP)
+	list='$(SOURCES) $(HEADERS)  $(LISP) $(TAGS_FILES)'; \
+	unique=`for i in $$list; do \
+	    if test -f "$$i"; then echo $$i; else echo $(srcdir)/$$i; fi; \
+	  done | \
+	  $(AWK) '{ files[$$0] = 1; nonempty = 1; } \
+	      END { if (nonempty) { for (i in files) print i; }; }'`; \
+	test -z "$(CTAGS_ARGS)$$unique" \
+	  || $(CTAGS) $(CTAGSFLAGS) $(AM_CTAGSFLAGS) $(CTAGS_ARGS) \
+	     $$unique
+
+GTAGS:
+	here=`$(am__cd) $(top_builddir) && pwd` \
+	  && $(am__cd) $(top_srcdir) \
+	  && gtags -i $(GTAGS_ARGS) "$$here"
+
+distclean-tags:
+	-rm -f TAGS ID GTAGS GRTAGS GSYMS GPATH tags
+
+distdir: $(DISTFILES)
+	@srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	list='$(DISTFILES)'; \
+	  dist_files=`for file in $$list; do echo $$file; done | \
+	  sed -e "s|^$$srcdirstrip/||;t" \
+	      -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+	case $$dist_files in \
+	  */*) $(MKDIR_P) `echo "$$dist_files" | \
+			   sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+			   sort -u` ;; \
+	esac; \
+	for file in $$dist_files; do \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+check-am: all-am
+check: check-am
+all-am: Makefile $(LTLIBRARIES) $(HEADERS)
+installdirs:
+	for dir in "$(DESTDIR)$(plugindir)"; do \
+	  test -z "$$dir" || $(MKDIR_P) "$$dir"; \
+	done
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+	$(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	  install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	  `test -z '$(STRIP)' || \
+	    echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool clean-pluginLTLIBRARIES \
+	mostlyclean-am
+
+distclean: distclean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+distclean-am: clean-am distclean-compile distclean-generic \
+	distclean-tags
+
+dvi: dvi-am
+
+dvi-am:
+
+html: html-am
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am: install-pluginLTLIBRARIES
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-pluginLTLIBRARIES
+
+.MAKE: install-am install-strip
+
+.PHONY: CTAGS GTAGS all all-am check check-am clean clean-generic \
+	clean-libtool clean-pluginLTLIBRARIES ctags distclean \
+	distclean-compile distclean-generic distclean-libtool \
+	distclean-tags distdir dvi dvi-am html html-am info info-am \
+	install install-am install-data install-data-am install-dvi \
+	install-dvi-am install-exec install-exec-am install-html \
+	install-html-am install-info install-info-am install-man \
+	install-pdf install-pdf-am install-pluginLTLIBRARIES \
+	install-ps install-ps-am install-strip installcheck \
+	installcheck-am installdirs maintainer-clean \
+	maintainer-clean-generic mostlyclean mostlyclean-compile \
+	mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \
+	tags uninstall uninstall-am uninstall-pluginLTLIBRARIES
+
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/sys/oss/common.h b/sys/oss/common.h
new file mode 100644
index 0000000..2445c3d
--- /dev/null
+++ b/sys/oss/common.h
@@ -0,0 +1,43 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *               2000,2005 Wim Taymans <wim@fluendo.com>
+ *
+ * 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.
+ */
+
+#define SET_PARAM(_oss, _name, _val, _detail)   \
+G_STMT_START {                                  \
+  int _tmp = _val;                              \
+  if (ioctl(_oss->fd, _name, &_tmp) == -1) {    \
+    GST_ELEMENT_ERROR (_oss, RESOURCE, SETTINGS,\
+        (NULL),					\
+        ("Unable to set param " _detail ": %s", \
+                   g_strerror (errno)));        \
+    return FALSE;                               \
+  }                                             \
+  GST_DEBUG_OBJECT(_oss, _detail " %d", _tmp);  \
+} G_STMT_END
+
+#define GET_PARAM(_oss, _name, _val, _detail)   \
+G_STMT_START {                                  \
+  if (ioctl(oss->fd, _name, _val) == -1) {      \
+    GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, \
+        (NULL),					\
+        ("Unable to get param " _detail ": %s", \
+                   g_strerror (errno)));        \
+    return FALSE;                               \
+  }                                             \
+} G_STMT_END
diff --git a/sys/oss/gstossaudio.c b/sys/oss/gstossaudio.c
new file mode 100644
index 0000000..2a181e8
--- /dev/null
+++ b/sys/oss/gstossaudio.c
@@ -0,0 +1,61 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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-i18n-plugin.h"
+
+#include "gstossmixerelement.h"
+#include "gstosssink.h"
+#include "gstosssrc.h"
+
+GST_DEBUG_CATEGORY (oss_debug);
+#define GST_CAT_DEFAULT oss_debug
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "ossmixer", GST_RANK_NONE,
+          GST_TYPE_OSS_MIXER_ELEMENT) ||
+      !gst_element_register (plugin, "osssrc", GST_RANK_SECONDARY,
+          GST_TYPE_OSS_SRC) ||
+      !gst_element_register (plugin, "osssink", GST_RANK_SECONDARY,
+          GST_TYPE_OSSSINK)) {
+    return FALSE;
+  }
+
+  GST_DEBUG_CATEGORY_INIT (oss_debug, "oss", 0, "OSS elements");
+
+#ifdef ENABLE_NLS
+  GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
+      LOCALEDIR);
+  bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+  bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
+#endif /* ENABLE_NLS */
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "ossaudio",
+    "OSS (Open Sound System) support for GStreamer",
+    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/sys/oss/gstossdmabuffer.h b/sys/oss/gstossdmabuffer.h
new file mode 100644
index 0000000..c38ef6f
--- /dev/null
+++ b/sys/oss/gstossdmabuffer.h
@@ -0,0 +1,77 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2005 Wim Taymans <wim@fluendo.com>
+ *
+ * gstossdmabuffer.h: 
+ *
+ * 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_OSSDMABUFFER_H__
+#define __GST_OSSDMABUFFER_H__
+
+#include <gst/gst.h>
+
+#include "gstosshelper.h"
+#include <gst/audio/gstringbuffer.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSSDMABUFFER            (gst_ossdmabuffer_get_type())
+#define GST_OSSDMABUFFER(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSDMABUFFER,GstOssDMABuffer))
+#define GST_OSSDMABUFFER_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSDMABUFFER,GstOssDMABufferClass))
+#define GST_IS_OSSDMABUFFER(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSDMABUFFER))
+#define GST_IS_OSSDMABUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSDMABUFFER))
+
+#define GST_OSSELEMENT_GET(obj)  GST_OSSELEMENT (obj->element)
+
+typedef enum {
+  GST_OSSDMABUFFER_OPEN         = (1 << 0),
+} GstOssDMABufferFlags;
+
+typedef struct _GstOssDMABuffer GstOssDMABuffer;
+typedef struct _GstOssDMABufferClass GstOssDMABufferClass;
+
+#define GST_OSSDMABUFFER_THREAD(buf)   (GST_OSSDMABUFFER(buf)->thread)
+#define GST_OSSDMABUFFER_LOCK          GST_OBJECT_LOCK
+#define GST_OSSDMABUFFER_UNLOCK        GST_OBJECT_UNLOCK
+#define GST_OSSDMABUFFER_COND(buf)     (GST_OSSDMABUFFER(buf)->cond)
+#define GST_OSSDMABUFFER_SIGNAL(buf)   (g_cond_signal (GST_OSSDMABUFFER_COND (buf)))
+#define GST_OSSDMABUFFER_WAIT(buf)     (g_cond_wait (GST_OSSDMABUFFER_COND (buf), GST_OBJECT_GET_LOCK (buf)))
+
+struct _GstOssDMABuffer {
+  GstRingBuffer  buffer;
+
+  GstOssElement *element;
+
+  int            fd;
+  int            caps;
+  int            frag;
+
+  GThread       *thread;
+  GCond         *cond;
+  gboolean       running;
+};
+
+struct _GstOssDMABufferClass {
+  GstRingBufferClass parent_class;
+};
+
+GType gst_ossdmabuffer_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_OSSDMABUFFER_H__ */
diff --git a/sys/oss/gstosshelper.c b/sys/oss/gstosshelper.c
new file mode 100644
index 0000000..6d7e6bd
--- /dev/null
+++ b/sys/oss/gstosshelper.c
@@ -0,0 +1,407 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wim.taymans@chello.be>
+ *
+ * gstosshelper.c: OSS helper routines
+ *
+ * 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-i18n-plugin.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>
+
+#ifdef HAVE_OSS_INCLUDE_IN_SYS
+# include <sys/soundcard.h>
+#else
+# ifdef HAVE_OSS_INCLUDE_IN_ROOT
+#  include <soundcard.h>
+# else
+#  ifdef HAVE_OSS_INCLUDE_IN_MACHINE
+#   include <machine/soundcard.h>
+#  else
+#   error "What to include?"
+#  endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
+# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
+#endif /* HAVE_OSS_INCLUDE_IN_SYS */
+
+#include <gst/interfaces/propertyprobe.h>
+
+#include "gstosshelper.h"
+#include "gstossmixer.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss_debug);
+#define GST_CAT_DEFAULT oss_debug
+
+typedef struct _GstOssProbe GstOssProbe;
+struct _GstOssProbe
+{
+  int fd;
+  int format;
+  int n_channels;
+  GArray *rates;
+  int min;
+  int max;
+};
+
+typedef struct _GstOssRange GstOssRange;
+struct _GstOssRange
+{
+  int min;
+  int max;
+};
+
+static GstStructure *gst_oss_helper_get_format_structure (unsigned int
+    format_bit);
+static gboolean gst_oss_helper_rate_probe_check (GstOssProbe * probe);
+static int gst_oss_helper_rate_check_rate (GstOssProbe * probe, int irate);
+static void gst_oss_helper_rate_add_range (GQueue * queue, int min, int max);
+static void gst_oss_helper_rate_add_rate (GArray * array, int rate);
+static int gst_oss_helper_rate_int_compare (gconstpointer a, gconstpointer b);
+
+GstCaps *
+gst_oss_helper_probe_caps (gint fd)
+{
+  GstOssProbe *probe;
+  int i;
+  gboolean ret;
+  GstStructure *structure;
+  unsigned int format_bit;
+  unsigned int format_mask;
+  GstCaps *caps;
+
+  /* FIXME test make sure we're not currently playing */
+  /* FIXME test both mono and stereo */
+
+  format_mask = AFMT_U8 | AFMT_S8;
+
+  if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+    format_mask |= AFMT_S16_LE | AFMT_U16_LE;
+  else
+    format_mask |= AFMT_S16_BE | AFMT_U16_BE;
+
+  caps = gst_caps_new_empty ();
+
+  /* assume that the most significant bit of format_mask is 0 */
+  for (format_bit = 1 << 31; format_bit > 0; format_bit >>= 1) {
+    if (format_bit & format_mask) {
+      GValue rate_value = { 0 };
+
+      probe = g_new0 (GstOssProbe, 1);
+      probe->fd = fd;
+      probe->format = format_bit;
+      /* FIXME: this is not working for all cards, see bug #518474 */
+      probe->n_channels = 2;
+
+      ret = gst_oss_helper_rate_probe_check (probe);
+      if (probe->min == -1 || probe->max == -1) {
+        g_array_free (probe->rates, TRUE);
+        g_free (probe);
+        continue;
+      }
+
+      if (ret) {
+        GValue value = { 0 };
+
+        g_array_sort (probe->rates, gst_oss_helper_rate_int_compare);
+
+        g_value_init (&rate_value, GST_TYPE_LIST);
+        g_value_init (&value, G_TYPE_INT);
+
+        for (i = 0; i < probe->rates->len; i++) {
+          g_value_set_int (&value, g_array_index (probe->rates, int, i));
+
+          gst_value_list_append_value (&rate_value, &value);
+        }
+
+        g_value_unset (&value);
+      } else {
+        /* one big range */
+        g_value_init (&rate_value, GST_TYPE_INT_RANGE);
+        gst_value_set_int_range (&rate_value, probe->min, probe->max);
+      }
+
+      g_array_free (probe->rates, TRUE);
+      g_free (probe);
+
+      structure = gst_oss_helper_get_format_structure (format_bit);
+      gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
+      gst_structure_set_value (structure, "rate", &rate_value);
+      g_value_unset (&rate_value);
+
+      gst_caps_append_structure (caps, structure);
+    }
+  }
+
+  if (gst_caps_is_empty (caps)) {
+    /* fixme: make user-visible */
+    GST_WARNING ("Your OSS device could not be probed correctly");
+  }
+
+  GST_DEBUG ("probed caps: %" GST_PTR_FORMAT, caps);
+
+  return caps;
+}
+
+static GstStructure *
+gst_oss_helper_get_format_structure (unsigned int format_bit)
+{
+  GstStructure *structure;
+  int endianness;
+  gboolean sign;
+  int width;
+
+  switch (format_bit) {
+    case AFMT_U8:
+      endianness = 0;
+      sign = FALSE;
+      width = 8;
+      break;
+    case AFMT_S16_LE:
+      endianness = G_LITTLE_ENDIAN;
+      sign = TRUE;
+      width = 16;
+      break;
+    case AFMT_S16_BE:
+      endianness = G_BIG_ENDIAN;
+      sign = TRUE;
+      width = 16;
+      break;
+    case AFMT_S8:
+      endianness = 0;
+      sign = TRUE;
+      width = 8;
+      break;
+    case AFMT_U16_LE:
+      endianness = G_LITTLE_ENDIAN;
+      sign = FALSE;
+      width = 16;
+      break;
+    case AFMT_U16_BE:
+      endianness = G_BIG_ENDIAN;
+      sign = FALSE;
+      width = 16;
+      break;
+    default:
+      g_assert_not_reached ();
+      return NULL;
+  }
+
+  structure = gst_structure_new ("audio/x-raw-int",
+      "width", G_TYPE_INT, width,
+      "depth", G_TYPE_INT, width, "signed", G_TYPE_BOOLEAN, sign, NULL);
+
+  if (endianness) {
+    gst_structure_set (structure, "endianness", G_TYPE_INT, endianness, NULL);
+  }
+
+  return structure;
+}
+
+static gboolean
+gst_oss_helper_rate_probe_check (GstOssProbe * probe)
+{
+  GstOssRange *range;
+  GQueue *ranges;
+  int exact_rates = 0;
+  gboolean checking_exact_rates = TRUE;
+  int n_checks = 0;
+  gboolean result = TRUE;
+
+  ranges = g_queue_new ();
+
+  probe->rates = g_array_new (FALSE, FALSE, sizeof (int));
+
+  probe->min = gst_oss_helper_rate_check_rate (probe, 1000);
+  n_checks++;
+  probe->max = gst_oss_helper_rate_check_rate (probe, 100000);
+  /* a little bug workaround */
+  {
+    int max;
+
+    max = gst_oss_helper_rate_check_rate (probe, 48000);
+    if (max > probe->max) {
+      GST_ERROR
+          ("Driver bug recognized (driver does not round rates correctly).  Please file a bug report.");
+      probe->max = max;
+    }
+  }
+  n_checks++;
+  if (probe->min == -1 || probe->max == -1) {
+    /* This is a workaround for drivers that return -EINVAL (or another
+     * error) for rates outside of [8000,48000].  If this fails, the
+     * driver is seriously buggy, and probably doesn't work with other
+     * media libraries/apps.  */
+    probe->min = gst_oss_helper_rate_check_rate (probe, 8000);
+    probe->max = gst_oss_helper_rate_check_rate (probe, 48000);
+  }
+  if (probe->min == -1 || probe->max == -1) {
+    GST_DEBUG ("unexpected check_rate error");
+    return FALSE;
+  }
+  gst_oss_helper_rate_add_range (ranges, probe->min + 1, probe->max - 1);
+
+  while ((range = g_queue_pop_head (ranges))) {
+    int min1;
+    int max1;
+    int mid;
+    int mid_ret;
+
+    GST_DEBUG ("checking [%d,%d]", range->min, range->max);
+
+    mid = (range->min + range->max) / 2;
+    mid_ret = gst_oss_helper_rate_check_rate (probe, mid);
+    if (mid_ret == -1) {
+      /* FIXME ioctl returned an error.  do something */
+      GST_DEBUG ("unexpected check_rate error");
+    }
+    n_checks++;
+
+    if (mid == mid_ret && checking_exact_rates) {
+      int max_exact_matches = 20;
+
+      exact_rates++;
+      if (exact_rates > max_exact_matches) {
+        GST_DEBUG ("got %d exact rates, assuming all are exact",
+            max_exact_matches);
+        result = FALSE;
+        g_free (range);
+        break;
+      }
+    } else {
+      checking_exact_rates = FALSE;
+    }
+
+    /* Assume that the rate is arithmetically rounded to the nearest
+     * supported rate. */
+    if (mid == mid_ret) {
+      min1 = mid - 1;
+      max1 = mid + 1;
+    } else {
+      if (mid < mid_ret) {
+        min1 = mid - (mid_ret - mid);
+        max1 = mid_ret + 1;
+      } else {
+        min1 = mid_ret - 1;
+        max1 = mid + (mid - mid_ret);
+      }
+    }
+
+    gst_oss_helper_rate_add_range (ranges, range->min, min1);
+    gst_oss_helper_rate_add_range (ranges, max1, range->max);
+
+    g_free (range);
+  }
+
+  while ((range = g_queue_pop_head (ranges))) {
+    g_free (range);
+  }
+  g_queue_free (ranges);
+
+  return result;
+}
+
+static void
+gst_oss_helper_rate_add_range (GQueue * queue, int min, int max)
+{
+  if (min <= max) {
+    GstOssRange *range = g_new0 (GstOssRange, 1);
+
+    range->min = min;
+    range->max = max;
+
+    g_queue_push_tail (queue, range);
+    /* push_head also works, but has different probing behavior */
+    /*g_queue_push_head (queue, range); */
+  }
+}
+
+static int
+gst_oss_helper_rate_check_rate (GstOssProbe * probe, int irate)
+{
+  int rate;
+  int format;
+  int n_channels;
+  int ret;
+
+  rate = irate;
+  format = probe->format;
+  n_channels = probe->n_channels;
+
+  GST_LOG ("checking format %d, channels %d, rate %d",
+      format, n_channels, rate);
+  ret = ioctl (probe->fd, SNDCTL_DSP_SETFMT, &format);
+  if (ret < 0 || format != probe->format) {
+    GST_DEBUG ("unsupported format: %d (%d)", probe->format, format);
+    return -1;
+  }
+  ret = ioctl (probe->fd, SNDCTL_DSP_CHANNELS, &n_channels);
+  if (ret < 0 || n_channels != probe->n_channels) {
+    GST_DEBUG ("unsupported channels: %d (%d)", probe->n_channels, n_channels);
+    return -1;
+  }
+  ret = ioctl (probe->fd, SNDCTL_DSP_SPEED, &rate);
+  if (ret < 0) {
+    GST_DEBUG ("unsupported rate: %d (%d)", irate, rate);
+    return -1;
+  }
+
+  GST_DEBUG ("rate %d -> %d", irate, rate);
+
+  if (rate == irate - 1 || rate == irate + 1) {
+    rate = irate;
+  }
+  gst_oss_helper_rate_add_rate (probe->rates, rate);
+  return rate;
+}
+
+static void
+gst_oss_helper_rate_add_rate (GArray * array, int rate)
+{
+  int i;
+  int val;
+
+  for (i = 0; i < array->len; i++) {
+    val = g_array_index (array, int, i);
+
+    if (val == rate)
+      return;
+  }
+  GST_DEBUG ("supported rate: %d", rate);
+  g_array_append_val (array, rate);
+}
+
+static int
+gst_oss_helper_rate_int_compare (gconstpointer a, gconstpointer b)
+{
+  const int *va = (const int *) a;
+  const int *vb = (const int *) b;
+
+  if (*va < *vb)
+    return -1;
+  if (*va > *vb)
+    return 1;
+  return 0;
+}
diff --git a/sys/oss/gstosshelper.h b/sys/oss/gstosshelper.h
new file mode 100644
index 0000000..1593fe3
--- /dev/null
+++ b/sys/oss/gstosshelper.h
@@ -0,0 +1,43 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wim.taymans@chello.be>
+ *
+ * gstosshelper.h: OSS helper routines.
+ *
+ * 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_OSS_HELPER_H__
+#define __GST_OSS_HELPER_H__
+
+
+#include <gst/gst.h>
+#include <sys/types.h>
+
+#include "gstosshelper.h"
+
+
+G_BEGIN_DECLS
+
+
+GstCaps* gst_oss_helper_probe_caps (gint fd);
+
+
+G_END_DECLS
+
+
+#endif /* __GST_OSS_HELPER_H__ */
diff --git a/sys/oss/gstossmixer.c b/sys/oss/gstossmixer.c
new file mode 100644
index 0000000..e563031
--- /dev/null
+++ b/sys/oss/gstossmixer.c
@@ -0,0 +1,378 @@
+/* GStreamer OSS Mixer implementation
+ * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * gstossmixer.c: mixer interface implementation for OSS
+ *
+ * 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 <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+
+#ifdef HAVE_OSS_INCLUDE_IN_SYS
+# include <sys/soundcard.h>
+#else
+# ifdef HAVE_OSS_INCLUDE_IN_ROOT
+#  include <soundcard.h>
+# else
+#  ifdef HAVE_OSS_INCLUDE_IN_MACHINE
+#   include <machine/soundcard.h>
+#  else
+#   error "What to include?"
+#  endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
+# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
+#endif /* HAVE_OSS_INCLUDE_IN_SYS */
+
+#include <gst/gst-i18n-plugin.h>
+
+#include "gstossmixer.h"
+#include "gstossmixertrack.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss_debug);
+#define GST_CAT_DEFAULT oss_debug
+
+#define MASK_BIT_IS_SET(mask, bit) \
+  (mask & (1 << bit))
+
+static gboolean
+gst_ossmixer_open (GstOssMixer * mixer)
+{
+#ifdef SOUND_MIXER_INFO
+  struct mixer_info minfo;
+#endif
+
+  g_return_val_if_fail (mixer->mixer_fd == -1, FALSE);
+
+  mixer->mixer_fd = open (mixer->device, O_RDWR);
+  if (mixer->mixer_fd == -1)
+    goto open_failed;
+
+  /* get masks */
+  if (ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECMASK, &mixer->recmask) < 0
+      || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECSRC, &mixer->recdevs) < 0
+      || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_STEREODEVS,
+          &mixer->stereomask) < 0
+      || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_DEVMASK, &mixer->devmask) < 0
+      || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_CAPS, &mixer->mixcaps) < 0)
+    goto masks_failed;
+
+  /* get name, not fatal */
+  g_free (mixer->cardname);
+#ifdef SOUND_MIXER_INFO
+  if (ioctl (mixer->mixer_fd, SOUND_MIXER_INFO, &minfo) == 0) {
+    mixer->cardname = g_strdup (minfo.name);
+    GST_INFO ("Card name = %s", GST_STR_NULL (mixer->cardname));
+  } else
+#endif
+  {
+    mixer->cardname = g_strdup ("Unknown");
+    GST_INFO ("Unknown card name");
+  }
+  GST_INFO ("Opened mixer for device %s", mixer->device);
+
+  return TRUE;
+
+  /* ERRORS */
+open_failed:
+  {
+    /* this is valid. OSS devices don't need to expose a mixer */
+    GST_DEBUG ("Failed to open mixer device %s, mixing disabled: %s",
+        mixer->device, strerror (errno));
+    return FALSE;
+  }
+masks_failed:
+  {
+    GST_DEBUG ("Failed to get device masks");
+    close (mixer->mixer_fd);
+    mixer->mixer_fd = -1;
+    return FALSE;
+  }
+}
+
+static void
+gst_ossmixer_ensure_track_list (GstOssMixer * mixer)
+{
+  gint i, master = -1;
+
+  g_return_if_fail (mixer->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 this 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->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);
+    }
+  }
+}
+
+GstOssMixer *
+gst_ossmixer_new (const char *device, GstOssMixerDirection dir)
+{
+  GstOssMixer *ret = NULL;
+
+  g_return_val_if_fail (device != NULL, NULL);
+
+  ret = g_new0 (GstOssMixer, 1);
+
+  ret->device = g_strdup (device);
+  ret->dir = dir;
+  ret->mixer_fd = -1;
+
+  if (!gst_ossmixer_open (ret))
+    goto error;
+
+  return ret;
+
+  /* ERRORS */
+error:
+  {
+    gst_ossmixer_free (ret);
+    return NULL;
+  }
+}
+
+void
+gst_ossmixer_free (GstOssMixer * mixer)
+{
+  g_return_if_fail (mixer != NULL);
+
+  if (mixer->device) {
+    g_free (mixer->device);
+    mixer->device = NULL;
+  }
+
+  if (mixer->cardname) {
+    g_free (mixer->cardname);
+    mixer->cardname = NULL;
+  }
+
+  if (mixer->tracklist) {
+    g_list_foreach (mixer->tracklist, (GFunc) g_object_unref, NULL);
+    g_list_free (mixer->tracklist);
+    mixer->tracklist = NULL;
+  }
+
+  if (mixer->mixer_fd != -1) {
+    close (mixer->mixer_fd);
+    mixer->mixer_fd = -1;
+  }
+
+  g_free (mixer);
+}
+
+/* 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->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->mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
+      g_warning ("Error getting recording device (%d) volume: %s",
+          osstrack->track_num, 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_volume (GstOssMixer * mixer,
+    GstMixerTrack * track, gint * volumes)
+{
+  gint volume;
+  GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
+
+  g_return_if_fail (mixer->mixer_fd != -1);
+  g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
+
+  /* prepare the value for ioctl() */
+  if (!(track->flags & GST_MIXER_TRACK_MUTE)) {
+    volume = (volumes[0] & 0xff);
+    if (track->num_channels == 2) {
+      volume |= ((volumes[1] & 0xff) << 8);
+    }
+
+    /* set */
+    if (ioctl (mixer->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
+      g_warning ("Error setting recording device (%d) volume (0x%x): %s",
+          osstrack->track_num, volume, strerror (errno));
+      return;
+    }
+  }
+
+  osstrack->lvol = volumes[0];
+  if (track->num_channels == 2) {
+    osstrack->rvol = volumes[1];
+  }
+}
+
+void
+gst_ossmixer_set_mute (GstOssMixer * mixer, GstMixerTrack * track,
+    gboolean mute)
+{
+  int volume;
+  GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
+
+  g_return_if_fail (mixer->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->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
+    g_warning ("Error setting mixer recording device volume (0x%x): %s",
+        volume, strerror (errno));
+    return;
+  }
+
+  if (mute) {
+    track->flags |= GST_MIXER_TRACK_MUTE;
+  } else {
+    track->flags &= ~GST_MIXER_TRACK_MUTE;
+  }
+}
+
+void
+gst_ossmixer_set_record (GstOssMixer * mixer,
+    GstMixerTrack * track, gboolean record)
+{
+  GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
+
+  g_return_if_fail (mixer->mixer_fd != -1);
+  g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
+
+  /* if there's nothing to do... */
+  if ((record && GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) ||
+      (!record && !GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)))
+    return;
+
+  /* if we're exclusive, then we need to unset the current one(s) */
+  if (mixer->mixcaps & SOUND_CAP_EXCL_INPUT) {
+    GList *track;
+
+    for (track = mixer->tracklist; track != NULL; track = track->next) {
+      GstMixerTrack *turn = (GstMixerTrack *) track->data;
+
+      turn->flags &= ~GST_MIXER_TRACK_RECORD;
+    }
+    mixer->recdevs = 0;
+  }
+
+  /* set new record bit, if needed */
+  if (record) {
+    mixer->recdevs |= (1 << osstrack->track_num);
+  } else {
+    mixer->recdevs &= ~(1 << osstrack->track_num);
+  }
+
+  /* set it to the device */
+  if (ioctl (mixer->mixer_fd, SOUND_MIXER_WRITE_RECSRC, &mixer->recdevs) < 0) {
+    g_warning ("Error setting mixer recording devices (0x%x): %s",
+        mixer->recdevs, strerror (errno));
+    return;
+  }
+
+  if (record) {
+    track->flags |= GST_MIXER_TRACK_RECORD;
+  } else {
+    track->flags &= ~GST_MIXER_TRACK_RECORD;
+  }
+}
diff --git a/sys/oss/gstossmixer.h b/sys/oss/gstossmixer.h
new file mode 100644
index 0000000..d2e06fe
--- /dev/null
+++ b/sys/oss/gstossmixer.h
@@ -0,0 +1,171 @@
+/* GStreamer OSS Mixer implementation
+ * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * gstossmixer.h: mixer interface implementation for OSS
+ *
+ * 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_OSS_MIXER_H__
+#define __GST_OSS_MIXER_H__
+
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "gstosshelper.h"
+
+
+G_BEGIN_DECLS
+
+
+#define GST_OSS_MIXER(obj)              ((GstOssMixer*)(obj))
+
+
+typedef enum {
+  GST_OSS_MIXER_CAPTURE = 1<<0,
+  GST_OSS_MIXER_PLAYBACK = 1<<1,
+  GST_OSS_MIXER_ALL = GST_OSS_MIXER_CAPTURE | GST_OSS_MIXER_PLAYBACK
+} GstOssMixerDirection;
+
+
+typedef struct _GstOssMixer GstOssMixer;
+
+
+struct _GstOssMixer {
+  GList *               tracklist;      /* list of available tracks */
+
+  gint                  mixer_fd;
+
+  gchar *               device;
+  gchar *               cardname;
+
+  gint                  recmask;
+  gint                  recdevs;
+  gint                  stereomask;
+  gint                  devmask;
+  gint                  mixcaps;
+
+  GstOssMixerDirection dir;
+};
+
+
+GstOssMixer*    gst_ossmixer_new                (const gchar *device,
+                                                 GstOssMixerDirection dir);
+void            gst_ossmixer_free               (GstOssMixer *mixer);
+
+const GList*    gst_ossmixer_list_tracks        (GstOssMixer * mixer);
+void            gst_ossmixer_set_volume         (GstOssMixer * mixer,
+                                                 GstMixerTrack * track,
+                                                 gint * volumes);
+void            gst_ossmixer_get_volume         (GstOssMixer * mixer,
+                                                 GstMixerTrack * track,
+                                                 gint * volumes);
+void            gst_ossmixer_set_record         (GstOssMixer * mixer,
+                                                 GstMixerTrack * track,
+                                                 gboolean record);
+void            gst_ossmixer_set_mute           (GstOssMixer * mixer,
+                                                 GstMixerTrack * track,
+                                                 gboolean mute);
+
+
+#define GST_IMPLEMENT_OSS_MIXER_METHODS(Type, interface_as_function)            \
+static gboolean                                                                 \
+interface_as_function ## _supported (Type *this, GType iface_type)              \
+{                                                                               \
+  g_assert (iface_type == GST_TYPE_MIXER);                                      \
+                                                                                \
+  return (this->mixer != NULL);                                                 \
+}                                                                               \
+                                                                                \
+static const GList*                                                             \
+interface_as_function ## _list_tracks (GstMixer * mixer)                        \
+{                                                                               \
+  Type *this = (Type*) mixer;                                                   \
+                                                                                \
+  g_return_val_if_fail (this != NULL, NULL);                                    \
+  g_return_val_if_fail (this->mixer != NULL, NULL);                             \
+                                                                                \
+  return gst_ossmixer_list_tracks (this->mixer);                                \
+}                                                                               \
+                                                                                \
+static void                                                                     \
+interface_as_function ## _set_volume (GstMixer * mixer, GstMixerTrack * track,  \
+    gint * volumes)                                                             \
+{                                                                               \
+  Type *this = (Type*) mixer;                                                   \
+                                                                                \
+  g_return_if_fail (this != NULL);                                              \
+  g_return_if_fail (this->mixer != NULL);                                       \
+                                                                                \
+  gst_ossmixer_set_volume (this->mixer, track, volumes);                        \
+}                                                                               \
+                                                                                \
+static void                                                                     \
+interface_as_function ## _get_volume (GstMixer * mixer, GstMixerTrack * track,  \
+    gint * volumes)                                                             \
+{                                                                               \
+  Type *this = (Type*) mixer;                                                   \
+                                                                                \
+  g_return_if_fail (this != NULL);                                              \
+  g_return_if_fail (this->mixer != NULL);                                       \
+                                                                                \
+  gst_ossmixer_get_volume (this->mixer, track, volumes);                        \
+}                                                                               \
+                                                                                \
+static void                                                                     \
+interface_as_function ## _set_record (GstMixer * mixer, GstMixerTrack * track,  \
+    gboolean record)                                                            \
+{                                                                               \
+  Type *this = (Type*) mixer;                                                   \
+                                                                                \
+  g_return_if_fail (this != NULL);                                              \
+  g_return_if_fail (this->mixer != NULL);                                       \
+                                                                                \
+  gst_ossmixer_set_record (this->mixer, track, record);                         \
+}                                                                               \
+                                                                                \
+static void                                                                     \
+interface_as_function ## _set_mute (GstMixer * mixer, GstMixerTrack * track,    \
+    gboolean mute)                                                              \
+{                                                                               \
+  Type *this = (Type*) mixer;                                                   \
+                                                                                \
+  g_return_if_fail (this != NULL);                                              \
+  g_return_if_fail (this->mixer != NULL);                                       \
+                                                                                \
+  gst_ossmixer_set_mute (this->mixer, track, mute);                             \
+}                                                                               \
+                                                                                \
+static void                                                                     \
+interface_as_function ## _interface_init (GstMixerClass * klass)                \
+{                                                                               \
+  GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE;                                  \
+                                                                                \
+  /* set up the interface hooks */                                              \
+  klass->list_tracks = interface_as_function ## _list_tracks;                   \
+  klass->set_volume = interface_as_function ## _set_volume;                     \
+  klass->get_volume = interface_as_function ## _get_volume;                     \
+  klass->set_mute = interface_as_function ## _set_mute;                         \
+  klass->set_record = interface_as_function ## _set_record;                     \
+}
+
+
+G_END_DECLS
+
+
+#endif /* __GST_OSS_MIXER_H__ */
diff --git a/sys/oss/gstossmixerelement.c b/sys/oss/gstossmixerelement.c
new file mode 100644
index 0000000..7ceb2b8
--- /dev/null
+++ b/sys/oss/gstossmixerelement.c
@@ -0,0 +1,218 @@
+/* OSS mixer interface element.
+ * Copyright (C) 2005 Andrew Vander Wingo <wingo@pobox.com>
+ *
+ * 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-ossmixer
+ *
+ * This element lets you adjust sound input and output levels with the
+ * Open Sound System (OSS). 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.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * <para>
+ * ossmixer can't be used in a sensible way in gst-launch.
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstossmixerelement.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss_debug);
+#define GST_CAT_DEFAULT oss_debug
+
+#define DEFAULT_DEVICE       "/dev/mixer"
+#define DEFAULT_DEVICE_NAME  NULL
+
+enum
+{
+  PROP_0,
+  PROP_DEVICE,
+  PROP_DEVICE_NAME
+};
+
+GST_BOILERPLATE_WITH_INTERFACE (GstOssMixerElement, gst_oss_mixer_element,
+    GstElement, GST_TYPE_ELEMENT, GstMixer, GST_TYPE_MIXER,
+    gst_oss_mixer_element);
+
+GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssMixerElement, gst_oss_mixer_element);
+
+static GstStateChangeReturn gst_oss_mixer_element_change_state (GstElement *
+    element, GstStateChange transition);
+
+static void gst_oss_mixer_element_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_oss_mixer_element_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+static void gst_oss_mixer_element_finalize (GObject * object);
+
+static void
+gst_oss_mixer_element_base_init (gpointer klass)
+{
+  gst_element_class_set_details_simple (GST_ELEMENT_CLASS (klass), "OSS Mixer",
+      "Generic/Audio",
+      "Control sound input and output levels with OSS",
+      "Andrew Vander Wingo <wingo@pobox.com>");
+}
+
+static void
+gst_oss_mixer_element_class_init (GstOssMixerElementClass * klass)
+{
+  GstElementClass *element_class;
+  GObjectClass *gobject_class;
+
+  element_class = (GstElementClass *) klass;
+  gobject_class = (GObjectClass *) klass;
+
+  gobject_class->finalize = gst_oss_mixer_element_finalize;
+  gobject_class->set_property = gst_oss_mixer_element_set_property;
+  gobject_class->get_property = gst_oss_mixer_element_get_property;
+
+  /**
+   * GstOssMixerElement:device
+   *
+   * OSS mixer device (usually /dev/mixer)
+   *
+   * Since: 0.10.5
+   **/
+  g_object_class_install_property (gobject_class, PROP_DEVICE,
+      g_param_spec_string ("device", "Device",
+          "OSS mixer device (usually /dev/mixer)", DEFAULT_DEVICE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  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 | G_PARAM_STATIC_STRINGS));
+
+  element_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_oss_mixer_element_change_state);
+}
+
+static void
+gst_oss_mixer_element_finalize (GObject * obj)
+{
+  GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (obj);
+
+  g_free (this->device);
+
+  G_OBJECT_CLASS (parent_class)->finalize (obj);
+}
+
+static void
+gst_oss_mixer_element_init (GstOssMixerElement * this,
+    GstOssMixerElementClass * g_class)
+{
+  this->mixer = NULL;
+  this->device = g_strdup (DEFAULT_DEVICE);
+}
+
+static void
+gst_oss_mixer_element_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      g_free (this->device);
+      this->device = g_value_dup_string (value);
+      /* make sure we never set NULL */
+      if (this->device == NULL) {
+        this->device = g_strdup (DEFAULT_DEVICE);
+      }
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_oss_mixer_element_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      g_value_set_string (value, this->device);
+      break;
+    case PROP_DEVICE_NAME:
+      if (this->mixer) {
+        g_value_set_string (value, this->mixer->cardname);
+      } else {
+        g_value_set_string (value, NULL);
+      }
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_oss_mixer_element_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+  GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (!this->mixer) {
+        this->mixer = gst_ossmixer_new (this->device, GST_OSS_MIXER_ALL);
+        if (!this->mixer)
+          goto open_failed;
+      }
+      break;
+      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:
+      if (this->mixer) {
+        gst_ossmixer_free (this->mixer);
+        this->mixer = NULL;
+      }
+      break;
+    default:
+      break;
+  }
+  return ret;
+
+  /* ERRORS */
+open_failed:
+  {
+    GST_ELEMENT_ERROR (element, RESOURCE, OPEN_READ_WRITE, (NULL),
+        ("Failed to open oss mixer device '%s'", this->device));
+    return GST_STATE_CHANGE_FAILURE;
+  }
+}
diff --git a/sys/oss/gstossmixerelement.h b/sys/oss/gstossmixerelement.h
new file mode 100644
index 0000000..2d47e0e
--- /dev/null
+++ b/sys/oss/gstossmixerelement.h
@@ -0,0 +1,59 @@
+/* OSS mixer interface element.
+ * Copyright (C) 2005 Andrew Vander Wingo <wingo@pobox.com>
+ *
+ * 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., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+
+#ifndef __GST_OSS_MIXER_ELEMENT_H__
+#define __GST_OSS_MIXER_ELEMENT_H__
+
+
+#include "gstossmixer.h"
+
+
+G_BEGIN_DECLS
+
+
+#define GST_OSS_MIXER_ELEMENT(obj)              (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElement))
+#define GST_OSS_MIXER_ELEMENT_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElementClass))
+#define GST_IS_OSS_MIXER_ELEMENT(obj)           (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS_MIXER_ELEMENT))
+#define GST_IS_OSS_MIXER_ELEMENT_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS_MIXER_ELEMENT))
+#define GST_TYPE_OSS_MIXER_ELEMENT              (gst_oss_mixer_element_get_type())
+
+
+typedef struct _GstOssMixerElement GstOssMixerElement;
+typedef struct _GstOssMixerElementClass GstOssMixerElementClass;
+
+
+struct _GstOssMixerElement {
+  GstElement            parent;
+
+  gchar                 *device;
+  GstOssMixer           *mixer;
+};
+
+struct _GstOssMixerElementClass {
+  GstElementClass       parent;
+};
+
+
+GType           gst_oss_mixer_element_get_type          (void);
+
+
+G_END_DECLS
+
+
+#endif /* __GST_OSS_MIXER_ELEMENT_H__ */
diff --git a/sys/oss/gstossmixertrack.c b/sys/oss/gstossmixertrack.c
new file mode 100644
index 0000000..3bc8cde
--- /dev/null
+++ b/sys/oss/gstossmixertrack.c
@@ -0,0 +1,172 @@
+/* GStreamer OSS Mixer implementation
+ * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * gstossmixer.c: mixer interface implementation for OSS
+ *
+ * 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 <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+
+#ifdef HAVE_OSS_INCLUDE_IN_SYS
+# include <sys/soundcard.h>
+#else
+# ifdef HAVE_OSS_INCLUDE_IN_ROOT
+#  include <soundcard.h>
+# else
+#  ifdef HAVE_OSS_INCLUDE_IN_MACHINE
+#   include <machine/soundcard.h>
+#  else
+#   error "What to include?"
+#  endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
+# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
+#endif /* HAVE_OSS_INCLUDE_IN_SYS */
+
+#include <gst/gst-i18n-plugin.h>
+
+#include "gstossmixertrack.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss_debug);
+#define GST_CAT_DEFAULT oss_debug
+
+#define MASK_BIT_IS_SET(mask, bit) \
+  (mask & (1 << bit))
+
+G_DEFINE_TYPE (GstOssMixerTrack, gst_ossmixer_track, GST_TYPE_MIXER_TRACK);
+
+static void
+gst_ossmixer_track_class_init (GstOssMixerTrackClass * klass)
+{
+  /* nop */
+}
+
+static void
+gst_ossmixer_track_init (GstOssMixerTrack * track)
+{
+  track->lvol = track->rvol = 0;
+  track->track_num = 0;
+}
+
+static const gchar **labels = NULL;
+
+/* three functions: firstly, OSS has the nasty habit of inserting
+ * spaces in the labels, we want to get rid of them. Secondly,
+ * i18n is impossible with OSS' way of providing us with mixer
+ * labels, so we make a 'given' list of i18n'ed labels. Thirdly, I
+ * personally don't like the "1337" names that OSS gives to their
+ * labels ("Vol", "Mic", "Rec"), I'd rather see full names. */
+
+static void
+fill_labels (void)
+{
+  gint i, pos;
+  const gchar *origs[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_LABELS;
+  const struct
+  {
+    const gchar *given;
+    const gchar *wanted;
+  }
+  cases[] = {
+    /* Note: this list is simply ripped from soundcard.h. For
+     * some people, some values might be missing (3D surround,
+     * etc.) - feel free to add them. That's the reason why
+     * I'm doing this in such a horribly complicated way. */
+    {
+    "Vol  ", _("Volume")}, {
+    "Bass ", _("Bass")}, {
+    "Trebl", _("Treble")}, {
+    "Synth", _("Synth")}, {
+    "Pcm  ", _("PCM")}, {
+    "Spkr ", _("Speaker")}, {
+    "Line ", _("Line-in")}, {
+    "Mic  ", _("Microphone")}, {
+    "CD   ", _("CD")}, {
+    "Mix  ", _("Mixer")}, {
+    "Pcm2 ", _("PCM-2")}, {
+    "Rec  ", _("Record")}, {
+    "IGain", _("In-gain")}, {
+    "OGain", _("Out-gain")}, {
+    "Line1", _("Line-1")}, {
+    "Line2", _("Line-2")}, {
+    "Line3", _("Line-3")}, {
+    "Digital1", _("Digital-1")}, {
+    "Digital2", _("Digital-2")}, {
+    "Digital3", _("Digital-3")}, {
+    "PhoneIn", _("Phone-in")}, {
+    "PhoneOut", _("Phone-out")}, {
+    "Video", _("Video")}, {
+    "Radio", _("Radio")}, {
+    "Monitor", _("Monitor")}, {
+    NULL, NULL}
+  };
+
+  labels = g_malloc (sizeof (gchar *) * SOUND_MIXER_NRDEVICES);
+
+  for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
+    for (pos = 0; cases[pos].given != NULL; pos++) {
+      if (!strcmp (cases[pos].given, origs[i])) {
+        labels[i] = g_strdup (cases[pos].wanted);
+        break;
+      }
+    }
+    if (cases[pos].given == NULL)
+      labels[i] = g_strdup (origs[i]);
+  }
+}
+
+GstMixerTrack *
+gst_ossmixer_track_new (gint mixer_fd,
+    gint track_num, gint max_chans, gint flags)
+{
+  GstOssMixerTrack *osstrack;
+  GstMixerTrack *track;
+  gint volume;
+
+  if (!labels)
+    fill_labels ();
+
+  osstrack = g_object_new (GST_TYPE_OSSMIXER_TRACK, NULL);
+  track = GST_MIXER_TRACK (osstrack);
+  track->label = g_strdup (labels[track_num]);
+  track->num_channels = max_chans;
+  track->flags = flags;
+  track->min_volume = 0;
+  track->max_volume = 100;
+  osstrack->track_num = track_num;
+
+  /* volume */
+  if (ioctl (mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
+    g_warning ("Error getting device (%d) volume: %s",
+        osstrack->track_num, strerror (errno));
+    volume = 0;
+  }
+  osstrack->lvol = (volume & 0xff);
+  if (track->num_channels == 2) {
+    osstrack->rvol = ((volume >> 8) & 0xff);
+  }
+
+  return track;
+}
diff --git a/sys/oss/gstossmixertrack.h b/sys/oss/gstossmixertrack.h
new file mode 100644
index 0000000..2e62ec8
--- /dev/null
+++ b/sys/oss/gstossmixertrack.h
@@ -0,0 +1,62 @@
+/* GStreamer OSS Mixer implementation
+ * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * gstossmixertrack.h: OSS mixer tracks
+ *
+ * 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_OSS_MIXER_TRACK_H__
+#define __GST_OSS_MIXER_TRACK_H__
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "gstosshelper.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSSMIXER_TRACK \
+  (gst_ossmixer_track_get_type ())
+#define GST_OSSMIXER_TRACK(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_OSSMIXER_TRACK, \
+                               GstOssMixerTrack))
+#define GST_OSSMIXER_TRACK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_OSSMIXER_TRACK, \
+                            GstOssMixerTrackClass))
+#define GST_IS_OSSMIXER_TRACK(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_OSSMIXER_TRACK))
+#define GST_IS_OSSMIXER_TRACK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_OSSMIXER_TRACK))
+
+typedef struct _GstOssMixerTrack {
+  GstMixerTrack parent;
+
+  gint          lvol, rvol;
+  gint          track_num;
+} GstOssMixerTrack;
+
+typedef struct _GstOssMixerTrackClass {
+  GstMixerTrackClass parent;
+} GstOssMixerTrackClass;
+
+GType           gst_ossmixer_track_get_type     (void);
+GstMixerTrack*  gst_ossmixer_track_new          (gint mixer_fd,
+    gint track_num, gint max_chans, gint flags);
+
+G_END_DECLS
+
+#endif /* __GST_OSS_MIXER_TRACK_H__ */
diff --git a/sys/oss/gstosssink.c b/sys/oss/gstosssink.c
new file mode 100644
index 0000000..9f9b137
--- /dev/null
+++ b/sys/oss/gstosssink.c
@@ -0,0 +1,565 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *               2000,2005 Wim Taymans <wim@fluendo.com>
+ *
+ * gstosssink.c: 
+ *
+ * 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-osssink
+ *
+ * This element lets you output sound using the Open Sound System (OSS).
+ *
+ * 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).
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink
+ * ]| will output a sine wave (continuous beep sound) to your sound card (with
+ * a very low volume as precaution).
+ * |[
+ * gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink
+ * ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+
+#ifdef HAVE_OSS_INCLUDE_IN_SYS
+# include <sys/soundcard.h>
+#else
+# ifdef HAVE_OSS_INCLUDE_IN_ROOT
+#  include <soundcard.h>
+# else
+#  ifdef HAVE_OSS_INCLUDE_IN_MACHINE
+#   include <machine/soundcard.h>
+#  else
+#   error "What to include?"
+#  endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
+# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
+#endif /* HAVE_OSS_INCLUDE_IN_SYS */
+
+#include "common.h"
+#include "gstosssink.h"
+
+#include <gst/gst-i18n-plugin.h>
+
+GST_DEBUG_CATEGORY_EXTERN (oss_debug);
+#define GST_CAT_DEFAULT oss_debug
+
+static void gst_oss_sink_base_init (gpointer g_class);
+static void gst_oss_sink_class_init (GstOssSinkClass * klass);
+static void gst_oss_sink_init (GstOssSink * osssink);
+
+static void gst_oss_sink_dispose (GObject * object);
+static void gst_oss_sink_finalise (GObject * object);
+
+static void gst_oss_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_oss_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink);
+
+static gboolean gst_oss_sink_open (GstAudioSink * asink);
+static gboolean gst_oss_sink_close (GstAudioSink * asink);
+static gboolean gst_oss_sink_prepare (GstAudioSink * asink,
+    GstRingBufferSpec * spec);
+static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);
+static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
+    guint length);
+static guint gst_oss_sink_delay (GstAudioSink * asink);
+static void gst_oss_sink_reset (GstAudioSink * asink);
+
+/* OssSink signals and args */
+enum
+{
+  LAST_SIGNAL
+};
+
+#define DEFAULT_DEVICE  "/dev/dsp"
+enum
+{
+  PROP_0,
+  PROP_DEVICE,
+};
+
+static GstStaticPadTemplate osssink_sink_factory =
+    GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
+        "signed = (boolean) { TRUE, FALSE }, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
+        "audio/x-raw-int, "
+        "signed = (boolean) { TRUE, FALSE }, "
+        "width = (int) 8, "
+        "depth = (int) 8, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
+    );
+
+static GstElementClass *parent_class = NULL;
+
+/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */
+
+GType
+gst_oss_sink_get_type (void)
+{
+  static GType osssink_type = 0;
+
+  if (!osssink_type) {
+    static const GTypeInfo osssink_info = {
+      sizeof (GstOssSinkClass),
+      gst_oss_sink_base_init,
+      NULL,
+      (GClassInitFunc) gst_oss_sink_class_init,
+      NULL,
+      NULL,
+      sizeof (GstOssSink),
+      0,
+      (GInstanceInitFunc) gst_oss_sink_init,
+    };
+
+    osssink_type =
+        g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink",
+        &osssink_info, 0);
+  }
+
+  return osssink_type;
+}
+
+static void
+gst_oss_sink_dispose (GObject * object)
+{
+  GstOssSink *osssink = GST_OSSSINK (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_oss_sink_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_set_details_simple (element_class, "Audio Sink (OSS)",
+      "Sink/Audio",
+      "Output to a sound card via OSS",
+      "Erik Walthinsen <omega@cse.ogi.edu>, "
+      "Wim Taymans <wim.taymans@chello.be>");
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&osssink_sink_factory));
+}
+
+static void
+gst_oss_sink_class_init (GstOssSinkClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseSinkClass *gstbasesink_class;
+  GstAudioSinkClass *gstaudiosink_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstbasesink_class = (GstBaseSinkClass *) klass;
+  gstaudiosink_class = (GstAudioSinkClass *) klass;
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  gobject_class->dispose = gst_oss_sink_dispose;
+  gobject_class->finalize = gst_oss_sink_finalise;
+  gobject_class->get_property = gst_oss_sink_get_property;
+  gobject_class->set_property = gst_oss_sink_set_property;
+
+  g_object_class_install_property (gobject_class, PROP_DEVICE,
+      g_param_spec_string ("device", "Device",
+          "OSS device (usually /dev/dspN)", DEFAULT_DEVICE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps);
+
+  gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open);
+  gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close);
+  gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare);
+  gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare);
+  gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);
+  gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);
+  gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);
+}
+
+static void
+gst_oss_sink_init (GstOssSink * osssink)
+{
+  const gchar *device;
+
+  GST_DEBUG_OBJECT (osssink, "initializing osssink");
+
+  device = g_getenv ("AUDIODEV");
+  if (device == NULL)
+    device = DEFAULT_DEVICE;
+  osssink->device = g_strdup (device);
+  osssink->fd = -1;
+}
+
+static void
+gst_oss_sink_finalise (GObject * object)
+{
+  GstOssSink *osssink = GST_OSSSINK (object);
+
+  g_free (osssink->device);
+
+  G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object));
+}
+
+static void
+gst_oss_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOssSink *sink;
+
+  sink = GST_OSSSINK (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      g_free (sink->device);
+      sink->device = g_value_dup_string (value);
+      if (sink->probed_caps) {
+        gst_caps_unref (sink->probed_caps);
+        sink->probed_caps = NULL;
+      }
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_oss_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstOssSink *sink;
+
+  sink = GST_OSSSINK (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      g_value_set_string (value, sink->device);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_oss_sink_getcaps (GstBaseSink * bsink)
+{
+  GstOssSink *osssink;
+  GstCaps *caps;
+
+  osssink = GST_OSSSINK (bsink);
+
+  if (osssink->fd == -1) {
+    caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
+            (bsink)));
+  } else if (osssink->probed_caps) {
+    caps = gst_caps_copy (osssink->probed_caps);
+  } else {
+    caps = gst_oss_helper_probe_caps (osssink->fd);
+    if (caps && !gst_caps_is_empty (caps)) {
+      osssink->probed_caps = gst_caps_copy (caps);
+    }
+  }
+
+  return caps;
+}
+
+static gint
+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;
+}
+
+static gint
+gst_oss_sink_get_format (GstBufferFormat fmt)
+{
+  gint result;
+
+  switch (fmt) {
+    case GST_MU_LAW:
+      result = AFMT_MU_LAW;
+      break;
+    case GST_A_LAW:
+      result = AFMT_A_LAW;
+      break;
+    case GST_IMA_ADPCM:
+      result = AFMT_IMA_ADPCM;
+      break;
+    case GST_U8:
+      result = AFMT_U8;
+      break;
+    case GST_S16_LE:
+      result = AFMT_S16_LE;
+      break;
+    case GST_S16_BE:
+      result = AFMT_S16_BE;
+      break;
+    case GST_S8:
+      result = AFMT_S8;
+      break;
+    case GST_U16_LE:
+      result = AFMT_U16_LE;
+      break;
+    case GST_U16_BE:
+      result = AFMT_U16_BE;
+      break;
+    case GST_MPEG:
+      result = AFMT_MPEG;
+      break;
+    default:
+      result = 0;
+      break;
+  }
+  return result;
+}
+
+static gboolean
+gst_oss_sink_open (GstAudioSink * asink)
+{
+  GstOssSink *oss;
+  int mode;
+
+  oss = GST_OSSSINK (asink);
+
+  mode = O_WRONLY;
+  mode |= O_NONBLOCK;
+
+  oss->fd = open (oss->device, mode, 0);
+  if (oss->fd == -1) {
+    switch (errno) {
+      case EBUSY:
+        goto busy;
+      case EACCES:
+        goto no_permission;
+      default:
+        goto open_failed;
+    }
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+busy:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
+        (_("Could not open audio device for playback. "
+                "Device is being used by another application.")), (NULL));
+    return FALSE;
+  }
+no_permission:
+  {
+    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);
+    return FALSE;
+  }
+open_failed:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
+        (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_oss_sink_close (GstAudioSink * asink)
+{
+  close (GST_OSSSINK (asink)->fd);
+  GST_OSSSINK (asink)->fd = -1;
+  return TRUE;
+}
+
+static gboolean
+gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
+{
+  GstOssSink *oss;
+  struct audio_buf_info info;
+  int mode;
+  int tmp;
+
+  oss = GST_OSSSINK (asink);
+
+  /* we opened non-blocking so that we can detect if the device is available
+   * without hanging forever. We now want to remove the non-blocking flag. */
+  mode = fcntl (oss->fd, F_GETFL);
+  mode &= ~O_NONBLOCK;
+  if (fcntl (oss->fd, F_SETFL, mode) == -1) {
+    /* 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_oss_sink_close (asink);
+    if ((oss->fd = open (oss->device, O_WRONLY, 0)) == -1)
+      goto non_block;
+  }
+
+  tmp = gst_oss_sink_get_format (spec->format);
+  if (tmp == 0)
+    goto wrong_format;
+
+  if (spec->width != 16 && spec->width != 8)
+    goto dodgy_width;
+
+  SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT");
+  if (spec->channels == 2)
+    SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
+  SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
+  SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
+
+  tmp = ilog2 (spec->segsize);
+  tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
+  GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
+      spec->segsize, spec->segtotal, tmp);
+
+  SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
+  GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE");
+
+  spec->segsize = info.fragsize;
+  spec->segtotal = info.fragstotal;
+
+  spec->bytes_per_sample = (spec->width / 8) * spec->channels;
+  oss->bytes_per_sample = (spec->width / 8) * spec->channels;
+
+  GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
+      spec->segsize, spec->segtotal, tmp);
+
+  return TRUE;
+
+  /* ERRORS */
+non_block:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
+        ("Unable to set device %s in non blocking mode: %s",
+            oss->device, g_strerror (errno)));
+    return FALSE;
+  }
+wrong_format:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
+        ("Unable to get format %d", spec->format));
+    return FALSE;
+  }
+dodgy_width:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
+        ("unexpected width %d", spec->width));
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_oss_sink_unprepare (GstAudioSink * asink)
+{
+  /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
+
+  if (!gst_oss_sink_close (asink))
+    goto couldnt_close;
+
+  if (!gst_oss_sink_open (asink))
+    goto couldnt_reopen;
+
+  return TRUE;
+
+  /* ERRORS */
+couldnt_close:
+  {
+    GST_DEBUG_OBJECT (asink, "Could not close the audio device");
+    return FALSE;
+  }
+couldnt_reopen:
+  {
+    GST_DEBUG_OBJECT (asink, "Could not reopen the audio device");
+    return FALSE;
+  }
+}
+
+static guint
+gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)
+{
+  return write (GST_OSSSINK (asink)->fd, data, length);
+}
+
+static guint
+gst_oss_sink_delay (GstAudioSink * asink)
+{
+  GstOssSink *oss;
+  gint delay = 0;
+  gint ret;
+
+  oss = GST_OSSSINK (asink);
+
+#ifdef SNDCTL_DSP_GETODELAY
+  ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
+#else
+  ret = -1;
+#endif
+  if (ret < 0) {
+    audio_buf_info info;
+
+    ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
+
+    delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
+  }
+  return delay / oss->bytes_per_sample;
+}
+
+static void
+gst_oss_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_RESET or similar */
+}
diff --git a/sys/oss/gstosssink.h b/sys/oss/gstosssink.h
new file mode 100644
index 0000000..bb2ca93
--- /dev/null
+++ b/sys/oss/gstosssink.h
@@ -0,0 +1,62 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * gstosssink.h: 
+ *
+ * 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_OSSSINK_H__
+#define __GST_OSSSINK_H__
+
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosink.h>
+
+#include "gstosshelper.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSSSINK            (gst_oss_sink_get_type())
+#define GST_OSSSINK(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSSINK,GstOssSink))
+#define GST_OSSSINK_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSSINK,GstOssSinkClass))
+#define GST_IS_OSSSINK(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSSINK))
+#define GST_IS_OSSSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSSINK))
+
+typedef struct _GstOssSink GstOssSink;
+typedef struct _GstOssSinkClass GstOssSinkClass;
+
+struct _GstOssSink {
+  GstAudioSink    sink;
+
+  gchar *device;
+  gint   fd;
+  gint   bytes_per_sample;
+
+  GstCaps *probed_caps;
+};
+
+struct _GstOssSinkClass {
+  GstAudioSinkClass parent_class;
+};
+
+GType gst_oss_sink_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_OSSSINK_H__ */
diff --git a/sys/oss/gstosssrc.c b/sys/oss/gstosssrc.c
new file mode 100644
index 0000000..33f585d
--- /dev/null
+++ b/sys/oss/gstosssrc.c
@@ -0,0 +1,537 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *               2000,2005 Wim Taymans <wim@fluendo.com>
+ *
+ * gstosssrc.c: 
+ *
+ * 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-osssrc
+ *
+ * This element lets you record sound using the Open Sound System (OSS).
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
+ * ]| will record sound from your sound card using OSS and encode it to an
+ * Ogg/Vorbis file (this will only work if your mixer settings are right
+ * and the right inputs enabled etc.)
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+
+#ifdef HAVE_OSS_INCLUDE_IN_SYS
+# include <sys/soundcard.h>
+#else
+# ifdef HAVE_OSS_INCLUDE_IN_ROOT
+#  include <soundcard.h>
+# else
+#  ifdef HAVE_OSS_INCLUDE_IN_MACHINE
+#   include <machine/soundcard.h>
+#  else
+#   error "What to include?"
+#  endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
+# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
+#endif /* HAVE_OSS_INCLUDE_IN_SYS */
+
+#include "gstosssrc.h"
+#include "common.h"
+
+#include <gst/gst-i18n-plugin.h>
+
+GST_DEBUG_CATEGORY_EXTERN (oss_debug);
+#define GST_CAT_DEFAULT oss_debug
+
+#define DEFAULT_DEVICE          "/dev/dsp"
+#define DEFAULT_DEVICE_NAME     ""
+
+enum
+{
+  PROP_0,
+  PROP_DEVICE,
+  PROP_DEVICE_NAME,
+};
+
+GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc,
+    GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
+
+GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer);
+
+static void gst_oss_src_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_oss_src_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+
+static void gst_oss_src_dispose (GObject * object);
+static void gst_oss_src_finalize (GstOssSrc * osssrc);
+
+static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc);
+
+static gboolean gst_oss_src_open (GstAudioSrc * asrc);
+static gboolean gst_oss_src_close (GstAudioSrc * asrc);
+static gboolean gst_oss_src_prepare (GstAudioSrc * asrc,
+    GstRingBufferSpec * spec);
+static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc);
+static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length);
+static guint gst_oss_src_delay (GstAudioSrc * asrc);
+static void gst_oss_src_reset (GstAudioSrc * asrc);
+
+
+
+static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
+        "signed = (boolean) { TRUE, FALSE }, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
+        "audio/x-raw-int, "
+        "signed = (boolean) { TRUE, FALSE }, "
+        "width = (int) 8, "
+        "depth = (int) 8, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
+    );
+
+
+static void
+gst_oss_src_dispose (GObject * object)
+{
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_oss_src_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_set_details_simple (element_class, "Audio Source (OSS)",
+      "Source/Audio",
+      "Capture from a sound card via OSS",
+      "Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&osssrc_src_factory));
+}
+
+static void
+gst_oss_src_class_init (GstOssSrcClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseSrcClass *gstbasesrc_class;
+  GstAudioSrcClass *gstaudiosrc_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstbasesrc_class = (GstBaseSrcClass *) klass;
+  gstaudiosrc_class = (GstAudioSrcClass *) klass;
+
+  gobject_class->dispose = gst_oss_src_dispose;
+  gobject_class->finalize = (GObjectFinalizeFunc) gst_oss_src_finalize;
+  gobject_class->get_property = gst_oss_src_get_property;
+  gobject_class->set_property = gst_oss_src_set_property;
+
+  gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps);
+
+  gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open);
+  gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare);
+  gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare);
+  gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close);
+  gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read);
+  gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay);
+  gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_reset);
+
+  g_object_class_install_property (gobject_class, PROP_DEVICE,
+      g_param_spec_string ("device", "Device",
+          "OSS device (usually /dev/dspN)", DEFAULT_DEVICE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  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 | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_oss_src_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOssSrc *src;
+
+  src = GST_OSS_SRC (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      if (src->device)
+        g_free (src->device);
+      src->device = g_value_dup_string (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_oss_src_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstOssSrc *src;
+
+  src = GST_OSS_SRC (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+      g_value_set_string (value, src->device);
+      break;
+    case PROP_DEVICE_NAME:
+      g_value_set_string (value, src->device_name);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_oss_src_init (GstOssSrc * osssrc, GstOssSrcClass * g_class)
+{
+  const gchar *device;
+
+  GST_DEBUG ("initializing osssrc");
+
+  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->probed_caps = NULL;
+}
+
+static void
+gst_oss_src_finalize (GstOssSrc * osssrc)
+{
+  g_free (osssrc->device);
+  g_free (osssrc->device_name);
+
+  G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (osssrc));
+}
+
+static GstCaps *
+gst_oss_src_getcaps (GstBaseSrc * bsrc)
+{
+  GstOssSrc *osssrc;
+  GstCaps *caps;
+
+  osssrc = GST_OSS_SRC (bsrc);
+
+  if (osssrc->fd == -1) {
+    GST_DEBUG_OBJECT (osssrc, "device not open, using template caps");
+    return NULL;                /* base class will get template caps for us */
+  }
+
+  if (osssrc->probed_caps) {
+    GST_LOG_OBJECT (osssrc, "Returning cached caps");
+    return gst_caps_ref (osssrc->probed_caps);
+  }
+
+  caps = gst_oss_helper_probe_caps (osssrc->fd);
+
+  if (caps) {
+    osssrc->probed_caps = gst_caps_ref (caps);
+  }
+
+  GST_INFO_OBJECT (osssrc, "returning caps %" GST_PTR_FORMAT, caps);
+
+  return caps;
+}
+
+static gint
+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;
+}
+
+static gint
+gst_oss_src_get_format (GstBufferFormat fmt)
+{
+  gint result;
+
+  switch (fmt) {
+    case GST_MU_LAW:
+      result = AFMT_MU_LAW;
+      break;
+    case GST_A_LAW:
+      result = AFMT_A_LAW;
+      break;
+    case GST_IMA_ADPCM:
+      result = AFMT_IMA_ADPCM;
+      break;
+    case GST_U8:
+      result = AFMT_U8;
+      break;
+    case GST_S16_LE:
+      result = AFMT_S16_LE;
+      break;
+    case GST_S16_BE:
+      result = AFMT_S16_BE;
+      break;
+    case GST_S8:
+      result = AFMT_S8;
+      break;
+    case GST_U16_LE:
+      result = AFMT_U16_LE;
+      break;
+    case GST_U16_BE:
+      result = AFMT_U16_BE;
+      break;
+    case GST_MPEG:
+      result = AFMT_MPEG;
+      break;
+    default:
+      result = 0;
+      break;
+  }
+  return result;
+}
+
+static gboolean
+gst_oss_src_open (GstAudioSrc * asrc)
+{
+  GstOssSrc *oss;
+  int mode;
+
+  oss = GST_OSS_SRC (asrc);
+
+  mode = O_RDONLY;
+  mode |= O_NONBLOCK;
+
+  oss->fd = open (oss->device, mode, 0);
+  if (oss->fd == -1) {
+    switch (errno) {
+      case EACCES:
+        goto no_permission;
+      default:
+        goto open_failed;
+    }
+  }
+
+  if (!oss->mixer) {
+    oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
+
+    if (oss->mixer) {
+      g_free (oss->device_name);
+      oss->device_name = g_strdup (oss->mixer->cardname);
+    }
+  }
+  return TRUE;
+
+no_permission:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+        (_("Could not open audio device for recording. "
+                "You don't have permission to open the device.")),
+        GST_ERROR_SYSTEM);
+    return FALSE;
+  }
+open_failed:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+        (_("Could not open audio device for recording.")),
+        ("Unable to open device %s for recording: %s",
+            oss->device, g_strerror (errno)));
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_oss_src_close (GstAudioSrc * asrc)
+{
+  GstOssSrc *oss;
+
+  oss = GST_OSS_SRC (asrc);
+
+  close (oss->fd);
+
+  if (oss->mixer) {
+    gst_ossmixer_free (oss->mixer);
+    oss->mixer = NULL;
+  }
+
+  gst_caps_replace (&oss->probed_caps, NULL);
+
+  return TRUE;
+}
+
+static gboolean
+gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
+{
+  GstOssSrc *oss;
+  struct audio_buf_info info;
+  int mode;
+  int fmt, tmp;
+
+  oss = GST_OSS_SRC (asrc);
+
+  mode = fcntl (oss->fd, F_GETFL);
+  mode &= ~O_NONBLOCK;
+  if (fcntl (oss->fd, F_SETFL, mode) == -1)
+    goto non_block;
+
+  fmt = gst_oss_src_get_format (spec->format);
+  if (fmt == 0)
+    goto wrong_format;
+
+  tmp = ilog2 (spec->segsize);
+  tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
+  GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
+      spec->segsize, spec->segtotal, tmp);
+
+  SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
+
+  SET_PARAM (oss, SNDCTL_DSP_RESET, 0, "RESET");
+
+  SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt, "SETFMT");
+  if (spec->channels == 2)
+    SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
+  SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
+  SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
+
+  GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE");
+
+  spec->segsize = info.fragsize;
+  spec->segtotal = info.fragstotal;
+
+  if (spec->width != 16 && spec->width != 8)
+    goto dodgy_width;
+
+  spec->bytes_per_sample = (spec->width / 8) * spec->channels;
+  oss->bytes_per_sample = (spec->width / 8) * spec->channels;
+  memset (spec->silence_sample, 0, spec->bytes_per_sample);
+
+  GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
+      spec->segsize, spec->segtotal, tmp);
+
+  return TRUE;
+
+non_block:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+        ("Unable to set device %s in non blocking mode: %s",
+            oss->device, g_strerror (errno)), (NULL));
+    return FALSE;
+  }
+wrong_format:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+        ("Unable to get format %d", spec->format), (NULL));
+    return FALSE;
+  }
+dodgy_width:
+  {
+    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+        ("Unexpected width %d", spec->width), (NULL));
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_oss_src_unprepare (GstAudioSrc * asrc)
+{
+  /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
+
+  if (!gst_oss_src_close (asrc))
+    goto couldnt_close;
+
+  if (!gst_oss_src_open (asrc))
+    goto couldnt_reopen;
+
+  return TRUE;
+
+couldnt_close:
+  {
+    GST_DEBUG_OBJECT (asrc, "Could not close the audio device");
+    return FALSE;
+  }
+couldnt_reopen:
+  {
+    GST_DEBUG_OBJECT (asrc, "Could not reopen the audio device");
+    return FALSE;
+  }
+}
+
+static guint
+gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length)
+{
+  return read (GST_OSS_SRC (asrc)->fd, data, length);
+}
+
+static guint
+gst_oss_src_delay (GstAudioSrc * asrc)
+{
+  GstOssSrc *oss;
+  gint delay = 0;
+  gint ret;
+
+  oss = GST_OSS_SRC (asrc);
+
+#ifdef SNDCTL_DSP_GETODELAY
+  ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
+#else
+  ret = -1;
+#endif
+  if (ret < 0) {
+    audio_buf_info info;
+
+    ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
+
+    delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
+  }
+  return delay / oss->bytes_per_sample;
+}
+
+static void
+gst_oss_src_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_RESET or similar */
+}
diff --git a/sys/oss/gstosssrc.h b/sys/oss/gstosssrc.h
new file mode 100644
index 0000000..0d14613
--- /dev/null
+++ b/sys/oss/gstosssrc.h
@@ -0,0 +1,67 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * gstosssrc.h: 
+ *
+ * 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_OSS_SRC_H__
+#define __GST_OSS_SRC_H__
+
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosrc.h>
+
+#include "gstosshelper.h"
+#include "gstossmixer.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS_SRC           (gst_oss_src_get_type())
+#define GST_OSS_SRC(obj)           (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS_SRC,GstOssSrc))
+#define GST_OSS_SRC_CLASS(klass)   (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS_SRC,GstOssSrcClass))
+#define GST_IS_OSS_SRC(obj)        (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS_SRC))
+#define GST_IS_OSS_SRC_CLASS(klas) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS_SRC))
+
+typedef struct _GstOssSrc GstOssSrc;
+typedef struct _GstOssSrcClass GstOssSrcClass;
+
+struct _GstOssSrc {
+  GstAudioSrc    src;
+
+  gint fd;
+  gint           bytes_per_sample;
+
+  gchar *device;
+  gchar *device_name;
+
+  GstCaps       *probed_caps;
+
+  GstOssMixer *mixer;
+};
+
+struct _GstOssSrcClass {
+  GstAudioSrcClass parent_class;
+};
+
+GType gst_oss_src_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_OSS_SRC_H__ */