alsaspdif: remove alsaspdifsink element

Remove alsaspdifsink, it's not needed any longer. alsasink in -base
has been able to handle SPDIF for a while now.

11 files changed, 3 insertions, 1151 deletions

diff --git a/Makefile.am b/Makefile.am
index 7b2b45688..ff0249420 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -43,11 +43,12 @@ include $(top_srcdir)/common/coverage/lcov.mak
 CRUFT_FILES = \
 	$(top_builddir)/common/shave \
 	$(top_builddir)/common/shave-libtool \
+	$(top_builddir)/ext/alsaspdif/.libs/*.{so,dll,DLL,dylib} \
+	$(top_builddir)/ext/ivorbis/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/shapewipe/.libs/*.{so,dll,DLL,dylib} \
-	$(top_builddir)/ext/ivorbis/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/gst/imagefreeze/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/sys/oss4/.libs/*.{so,dll,DLL,dylib} \
 	$(top_builddir)/tests/check/elements/capssetter \
@@ -62,6 +63,7 @@ CRUFT_DIRS = \
 	$(top_srcdir)/gst/imagefreeze \
 	$(top_srcdir)/gst/shapewipe \
 	$(top_srcdir)/tests/examples/shapewipe
+	$(top_srcdir)/ext/alsaspdif \
 	$(top_srcdir)/ext/ivorbis \
 	$(top_srcdir)/ext/metadata 
 
diff --git a/configure.ac b/configure.ac
index 3397fbcf2..33ca0ba99 100644
--- a/configure.ac
+++ b/configure.ac
@@ -499,18 +499,6 @@ dnl keep this list sorted alphabetically !
 
 if test "x$BUILD_EXTERNAL" = "xyes"; then
 
-dnl *** alsa ***
-translit(dnm, m, l) AM_CONDITIONAL(USE_ALSA, true)
-AG_GST_CHECK_FEATURE(ALSA, [alsa plug-ins], gstalsa, [
-  PKG_CHECK_MODULES(ALSA, alsa >= 0.9.1, [
-    HAVE_ALSA="yes"
-    AC_SUBST(ALSA_CFLAGS)
-    AC_SUBST(ALSA_LIBS)
-  ], [
-    AM_PATH_ALSA(0.9.1, HAVE_ALSA="yes", HAVE_ALSA="no")
-  ])
-])
-
 dnl *** assrender ***
 translit(dnm, m, l) AM_CONDITIONAL(USE_ASSRENDER, true)
 AG_GST_CHECK_FEATURE(ASSRENDER, [ASS/SSA renderer], assrender, [
@@ -1555,7 +1543,6 @@ else
 dnl not building plugins with external dependencies,
 dnl but we still need to set the conditionals
 
-AM_CONDITIONAL(USE_ALSA, false)
 AM_CONDITIONAL(USE_ASSRENDER, false)
 AM_CONDITIONAL(USE_AMRWB, false)
 AM_CONDITIONAL(USE_APEXSINK, false)
@@ -1777,7 +1764,6 @@ tests/examples/scaletempo/Makefile
 tests/examples/switch/Makefile
 tests/examples/jack/Makefile
 tests/icles/Makefile
-ext/alsaspdif/Makefile
 ext/amrwbenc/Makefile
 ext/assrender/Makefile
 ext/apexsink/Makefile
diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
index 387ba5d79..8a9b04eed 100644
--- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
@@ -138,7 +138,6 @@
     <xi:include href="xml/plugin-audioparsersbad.xml" />
     <xi:include href="xml/plugin-autoconvert.xml" />
     <xi:include href="xml/plugin-legacyresample.xml" />
-    <xi:include href="xml/plugin-alsaspdif.xml" />
     <xi:include href="xml/plugin-amrwbenc.xml" />
     <xi:include href="xml/plugin-assrender.xml" />
     <xi:include href="xml/plugin-bayer.xml" />
diff --git a/docs/plugins/inspect/plugin-alsaspdif.xml b/docs/plugins/inspect/plugin-alsaspdif.xml
deleted file mode 100644
index 92ae01a21..000000000
--- a/docs/plugins/inspect/plugin-alsaspdif.xml
+++ /dev/null
@@ -1,28 +0,0 @@
-<plugin>
-  <name>alsaspdif</name>
-  <description>Alsa plugin for S/PDIF output</description>
-  <filename>../../ext/alsaspdif/.libs/libgstalsaspdif.so</filename>
-  <basename>libgstalsaspdif.so</basename>
-  <version>0.10.20.1</version>
-  <license>LGPL</license>
-  <source>gst-plugins-bad</source>
-  <package>GStreamer Bad Plug-ins git</package>
-  <origin>Unknown package origin</origin>
-  <elements>
-    <element>
-      <name>alsaspdifsink</name>
-      <longname>S/PDIF ALSA audiosink</longname>
-      <class>Sink/Audio</class>
-      <description>Feeds audio to S/PDIF interfaces through the ALSA sound driver</description>
-      <author>Martin Soto &lt;martinsoto@users.sourceforge.net&gt;, Michael Smith &lt;msmith@fluendo.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>audio/x-iec958</details>
-        </caps>
-      </pads>
-    </element>
-  </elements>
-</plugin>
\ No newline at end of file
diff --git a/ext/Makefile.am b/ext/Makefile.am
index 3893c6274..a9a7c8b5f 100644
--- a/ext/Makefile.am
+++ b/ext/Makefile.am
@@ -1,9 +1,3 @@
-if USE_ALSA
-ALSASPDIF_DIR = alsaspdif
-else
-ALSASPDIF_DIR =
-endif
-
 if USE_ASSRENDER
 ASSRENDER_DIR = assrender
 else
@@ -380,7 +374,6 @@ endif
 
 
 SUBDIRS=\
-	$(ALSASPDIF_DIR) \
 	$(ASSRENDER_DIR) \
 	$(AMRWB_DIR) \
 	$(APEXSINK_DIR) \
@@ -444,7 +437,6 @@ SUBDIRS=\
 	$(RTMP_DIR)
 
 DIST_SUBDIRS = \
-	alsaspdif \
 	amrwbenc \
 	assrender \
 	apexsink \
diff --git a/ext/alsaspdif/Makefile.am b/ext/alsaspdif/Makefile.am
deleted file mode 100644
index b67470710..000000000
--- a/ext/alsaspdif/Makefile.am
+++ /dev/null
@@ -1,15 +0,0 @@
-plugin_LTLIBRARIES = libgstalsaspdif.la
-
-# sources used to compile this plugin
-libgstalsaspdif_la_SOURCES = alsaspdifsink.c
-
-# flags used to compile this plugin
-# we use the GST_LIBS flags because we might be using plug-in libs
-libgstalsaspdif_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(ALSA_CFLAGS)
-libgstalsaspdif_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(GST_BASE_LIBS) $(GST_LIBS) $(ALSA_LIBS)
-libgstalsaspdif_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-libgstalsaspdif_la_LIBTOOLFLAGS = --tag=disable-static
-
-# headers we need but don't want installed
-noinst_HEADERS = alsaspdifsink.h
-
diff --git a/ext/alsaspdif/alsaspdifsink.c b/ext/alsaspdif/alsaspdifsink.c
deleted file mode 100644
index c0f5533c2..000000000
--- a/ext/alsaspdif/alsaspdifsink.c
+++ /dev/null
@@ -1,846 +0,0 @@
-/* Based on a plugin from Martin Soto's Seamless DVD Player.
- * Copyright (C) 2003, 2004 Martin Soto <martinsoto@users.sourceforge.net>
- *               2005-6 Michael Smith <msmith@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.
- */
-
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-#include <unistd.h>
-
-#include <gst/gst.h>
-#include <gst/audio/gstaudioclock.h>
-#include <gst/base/gstbasesink.h>
-
-#include "alsaspdifsink.h"
-
-GST_DEBUG_CATEGORY_STATIC (alsaspdifsink_debug);
-#define GST_CAT_DEFAULT (alsaspdifsink_debug)
-
-/* The magic audio-type we pretend to be for AC3 output */
-#define AC3_CHANNELS 2
-#define AC3_BITS     16
-
-/* Define AC3 FORMAT as big endian. Fall back to swapping
- * on sound devices that don't support it */
-#define AC3_FORMAT_BE   SND_PCM_FORMAT_S16_BE
-#define AC3_FORMAT_LE   SND_PCM_FORMAT_S16_LE
-
-/* The size in bytes of an IEC958 frame. */
-#define IEC958_FRAME_SIZE 6144
-
-/* Size in bytes of an ALSA PCM frame (4, for this case). */
-#define ALSASPDIFSINK_BYTES_PER_FRAME ((AC3_BITS / 8) * AC3_CHANNELS)
-
-#define IEC958_SAMPLES_PER_FRAME (IEC958_FRAME_SIZE / ALSASPDIFSINK_BYTES_PER_FRAME)
-
-#if 0
-/* The duration of a single IEC958 frame. */
-#define IEC958_FRAME_DURATION (32 * GST_MSECOND)
-
-/* Maximal synchronization difference.  Measures will be taken if
-   block timestamps differ from actual playing time in more than this
-   value. */
-#define MAX_SYNC_DIFF (IEC958_FRAME_DURATION * 0.8)
-
-/* Playing time for the given number of ALSA PCM frames. */
-#define ALSASPDIFSINK_TIME_PER_FRAMES(sink, frames) \
-  (((GstClockTime) (frames) * GST_SECOND) / AC3_RATE)
-
-/* Number of ALSA PCM frames for the given playing time. */
-#define ALSASPDIFSINK_FRAMES_PER_TIME(sink, time) \
-  (((GstClockTime) AC3_RATE * (time)) / GST_SECOND)
-#endif
-
-/* AlsaSPDIFSink signals and args */
-enum
-{
-  LAST_SIGNAL
-};
-
-enum
-{
-  PROP_0,
-  PROP_CARD,
-  PROP_DEVICE
-};
-
-static GstStaticPadTemplate alsaspdifsink_sink_factory =
-GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-iec958")
-    );
-
-#define _do_init(bla) \
-    GST_DEBUG_CATEGORY_INIT (alsaspdifsink_debug, "alsaspdifsink", 0, \
-            "ALSA S/PDIF audio sink element");
-
-GST_BOILERPLATE_FULL (AlsaSPDIFSink, alsaspdifsink, GstBaseSink,
-    GST_TYPE_BASE_SINK, _do_init);
-
-static void alsaspdifsink_set_property (GObject * object,
-    guint prop_id, const GValue * value, GParamSpec * pspec);
-static void alsaspdifsink_get_property (GObject * object,
-    guint prop_id, GValue * value, GParamSpec * pspec);
-static gboolean alsaspdifsink_event (GstBaseSink * bsink, GstEvent * event);
-static GstFlowReturn alsaspdifsink_render (GstBaseSink * bsink,
-    GstBuffer * buf);
-static void alsaspdifsink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
-    GstClockTime * start, GstClockTime * end);
-static gboolean alsaspdifsink_set_caps (GstBaseSink * bsink, GstCaps * caps);
-
-static gboolean alsaspdifsink_open (AlsaSPDIFSink * sink);
-static void alsaspdifsink_close (AlsaSPDIFSink * sink);
-
-static GstClock *alsaspdifsink_provide_clock (GstElement * elem);
-static GstClockTime alsaspdifsink_get_time (GstClock * clock,
-    gpointer user_data);
-static void alsaspdifsink_dispose (GObject * object);
-static void alsaspdifsink_finalize (GObject * object);
-
-static GstStateChangeReturn alsaspdifsink_change_state (GstElement * element,
-    GstStateChange transition);
-static int alsaspdifsink_find_pcm_device (AlsaSPDIFSink * sink);
-static gboolean alsaspdifsink_set_params (AlsaSPDIFSink * sink);
-static snd_pcm_sframes_t alsaspdifsink_delay (AlsaSPDIFSink * sink);
-
-/* Alsa error handler to suppress messages from within the ALSA library */
-static void ignore_alsa_err (const char *file, int line, const char *function,
-    int err, const char *fmt, ...);
-
-static void
-alsaspdifsink_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class, "S/PDIF ALSA audiosink",
-      "Sink/Audio",
-      "Feeds audio to S/PDIF interfaces through the ALSA sound driver",
-      "Martin Soto <martinsoto@users.sourceforge.net>, "
-      "Michael Smith <msmith@fluendo.com>");
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&alsaspdifsink_sink_factory));
-}
-
-static void
-alsaspdifsink_class_init (AlsaSPDIFSinkClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-  GstBaseSinkClass *gstbasesink_class;
-
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-  gstbasesink_class = (GstBaseSinkClass *) klass;
-
-  gobject_class->set_property = alsaspdifsink_set_property;
-  gobject_class->get_property = alsaspdifsink_get_property;
-  gobject_class->dispose = alsaspdifsink_dispose;
-  gobject_class->finalize = alsaspdifsink_finalize;
-
-  gstelement_class->change_state = alsaspdifsink_change_state;
-  gstelement_class->provide_clock = alsaspdifsink_provide_clock;
-
-  gstbasesink_class->event = alsaspdifsink_event;
-  gstbasesink_class->render = alsaspdifsink_render;
-  gstbasesink_class->get_times = alsaspdifsink_get_times;
-  gstbasesink_class->set_caps = alsaspdifsink_set_caps;
-
-#if 0
-  /* We ignore the device property anyway, so don't install it
-   * we don't want the user supplying just any device string for us. 
-   * At most we might want a card number and an iec958.%d device name
-   * to attempt */
-  g_object_class_install_property (gobject_class, PROP_DEVICE,
-      g_param_spec_string ("device", "Device",
-          "ALSA device, as defined in an asound configuration file",
-          "default", G_PARAM_READWRITE));
-#endif
-  g_object_class_install_property (gobject_class, PROP_CARD,
-      g_param_spec_int ("card", "Card",
-          "ALSA card number for the SPDIF device to use",
-          0, G_MAXINT, 0, G_PARAM_READWRITE));
-
-  snd_lib_error_set_handler (ignore_alsa_err);
-}
-
-static void
-alsaspdifsink_init (AlsaSPDIFSink * sink, AlsaSPDIFSinkClass * g_class)
-{
-  /* Create the provided clock. */
-#if GST_CHECK_VERSION(0, 10, 31) || (GST_CHECK_VERSION(0, 10, 30) && GST_VERSION_NANO > 0)
-  sink->clock =
-      gst_audio_clock_new_full ("clock", alsaspdifsink_get_time,
-      gst_object_ref (sink), (GDestroyNotify) gst_object_unref);
-#else
-  sink->clock = gst_audio_clock_new ("clock", alsaspdifsink_get_time, sink);
-#endif
-
-  sink->card = 0;
-  sink->device = g_strdup ("default");
-}
-
-static void
-alsaspdifsink_dispose (GObject * object)
-{
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (object);
-
-  if (sink->clock)
-    gst_object_unref (sink->clock);
-  sink->clock = NULL;
-
-  G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
-alsaspdifsink_finalize (GObject * object)
-{
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (object);
-
-  g_free (sink->device);
-  sink->device = NULL;
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-alsaspdifsink_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  AlsaSPDIFSink *sink;
-
-  sink = ALSASPDIFSINK (object);
-
-  switch (prop_id) {
-      /*
-         case PROP_DEVICE:
-         if(sink->device)
-         g_free(sink->device);
-         sink->device = g_strdup(g_value_get_string(value));
-         break;
-       */
-    case PROP_CARD:
-      sink->card = g_value_get_int (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-
-static void
-alsaspdifsink_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  AlsaSPDIFSink *sink;
-
-  sink = ALSASPDIFSINK (object);
-
-  switch (prop_id) {
-      /*
-         case PROP_DEVICE:
-         g_value_set_string(value, sink->device);
-         break;
-       */
-    case PROP_CARD:
-      g_value_set_int (value, sink->card);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static gboolean
-alsaspdifsink_set_caps (GstBaseSink * bsink, GstCaps * caps)
-{
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
-
-  if (!gst_structure_get_int (gst_caps_get_structure (caps, 0), "rate",
-          &sink->rate))
-    sink->rate = 48000;
-
-  if (!alsaspdifsink_set_params (sink)) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("Cannot set ALSA hardware parameters"), GST_ERROR_SYSTEM);
-    return FALSE;
-  }
-
-  return TRUE;
-}
-
-static GstClock *
-alsaspdifsink_provide_clock (GstElement * elem)
-{
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (elem);
-
-  return GST_CLOCK (gst_object_ref (sink->clock));
-}
-
-static GstClockTime
-alsaspdifsink_get_time (GstClock * clock, gpointer user_data)
-{
-  GstClockTime result;
-  snd_pcm_sframes_t raw, delay, samples;
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (user_data);
-
-  raw = samples = sink->frames * IEC958_SAMPLES_PER_FRAME;
-  delay = alsaspdifsink_delay (sink);
-
-  if (samples > delay)
-    samples -= delay;
-  else
-    samples = 0;
-
-  result = gst_util_uint64_scale_int (samples, GST_SECOND, sink->rate);
-  GST_LOG_OBJECT (sink, "Samples raw: %d, delay: %d, real: %d, "
-      "Time: %" GST_TIME_FORMAT, (int) raw, (int) delay, (int) samples,
-      GST_TIME_ARGS (result));
-  return result;
-}
-
-static gboolean
-alsaspdifsink_open (AlsaSPDIFSink * sink)
-{
-  char *pcm_name = sink->device;
-  int err;
-  char devstr[256];             /* Storage for local 'default' device string */
-
-  /*
-   * Try and open our default iec958 device. Fall back to searching on card x
-   * if this fails, which should only happen on older alsa setups
-   */
-
-  /* The string will be one of these:
-   * SPDIF_CON: Non-audio flag not set:
-   *    spdif:{AES0 0x0 AES1 0x82 AES2 0x0 AES3 0x2}
-   * SPDIF_CON: Non-audio flag set:
-   *    spdif:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}
-   */
-  sprintf (devstr,
-      "iec958:{CARD %d AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x}",
-      sink->card,
-      IEC958_AES0_NONAUDIO,
-      IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
-      0, IEC958_AES3_CON_FS_48000);
-
-  GST_DEBUG_OBJECT (sink, "Generated device string \"%s\"", devstr);
-  pcm_name = devstr;
-
-  err = snd_pcm_open (&(sink->pcm), pcm_name, SND_PCM_STREAM_PLAYBACK, 0);
-  if (err < 0) {
-    GST_DEBUG_OBJECT (sink,
-        "Open failed for %s - searching for IEC958 manually\n", pcm_name);
-
-    err = alsaspdifsink_find_pcm_device (sink);
-    if (err == 0 && sink->pcm == NULL)
-      goto open_failed;
-  }
-  if (err < 0)
-    goto failed;
-
-  return TRUE;
-
-  /* ERRORS */
-open_failed:
-  {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("Could not open IEC958/SPDIF output device"), GST_ERROR_SYSTEM);
-    return FALSE;
-  }
-failed:
-  {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("snd_pcm_open: %s", snd_strerror (err)), GST_ERROR_SYSTEM);
-    return FALSE;
-  }
-}
-
-static gboolean
-alsaspdifsink_set_params (AlsaSPDIFSink * sink)
-{
-  snd_pcm_hw_params_t *params;
-  unsigned int rate;
-  int err;
-
-  snd_pcm_hw_params_malloc (&params);
-
-  err = snd_pcm_hw_params_any (sink->pcm, params);
-  if (err < 0) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("Broken configuration for this PCM: "
-            "no configurations available"), GST_ERROR_SYSTEM);
-    goto __error;
-  }
-
-  /* Set interleaved access. */
-  err = snd_pcm_hw_params_set_access (sink->pcm, params,
-      SND_PCM_ACCESS_RW_INTERLEAVED);
-  if (err < 0) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("Access type not available"), GST_ERROR_SYSTEM);
-    goto __error;
-  }
-
-  err = snd_pcm_hw_params_set_format (sink->pcm, params, AC3_FORMAT_BE);
-  if (err < 0) {
-    /* Try LE output and swap data */
-    GST_DEBUG_OBJECT (sink, "PCM format S16_BE not supported, trying S16_LE");
-    err = snd_pcm_hw_params_set_format (sink->pcm, params, AC3_FORMAT_LE);
-    sink->need_swap = TRUE;
-  } else
-    sink->need_swap = FALSE;
-
-  if (err < 0) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("Sample format not available"), GST_ERROR_SYSTEM);
-    goto __error;
-  }
-
-  err = snd_pcm_hw_params_set_channels (sink->pcm, params, AC3_CHANNELS);
-  if (err < 0) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("Channels count not available"), GST_ERROR_SYSTEM);
-    goto __error;
-  }
-
-  rate = sink->rate;
-  GST_DEBUG_OBJECT (sink, "Setting S/PDIF sample rate: %d", rate);
-  err = snd_pcm_hw_params_set_rate_near (sink->pcm, params, &rate, 0);
-  if (err != 0) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("Rate not available"), GST_ERROR_SYSTEM);
-    goto __error;
-  }
-
-  err = snd_pcm_hw_params (sink->pcm, params);
-  if (err < 0) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("PCM hw_params failed: %s", snd_strerror (err)), GST_ERROR_SYSTEM);
-    goto __error;
-  }
-
-  snd_pcm_hw_params_free (params);
-
-  return TRUE;
-
-  /* ERRORS */
-__error:
-  {
-    snd_pcm_hw_params_free (params);
-    return FALSE;
-  }
-}
-
-static void
-alsaspdifsink_close (AlsaSPDIFSink * sink)
-{
-  if (sink->pcm) {
-    snd_pcm_close (sink->pcm);
-    sink->pcm = NULL;
-  }
-}
-
-/* Try and find an IEC958 PCM device and mixer on card 0 and open it
- * This function is only used on older ALSA installs that don't have the
- * correct iec958 alias stuff set up, and relies on there being only
- * one IEC958 PCM device (relies IEC958 in the device name) and one IEC958
- * mixer control for doing the settings.
- */
-static int
-alsaspdifsink_find_pcm_device (AlsaSPDIFSink * sink)
-{
-  int err = -1, dev, idx, count;
-  const gchar *ctl_name = "hw:0";
-  const gchar *spdif_name = SND_CTL_NAME_IEC958 ("", PLAYBACK, NONE);
-  int card = sink->card;
-  gchar pcm_name[24];
-  snd_pcm_t *pcm = NULL;
-  snd_ctl_t *ctl = NULL;
-  snd_ctl_card_info_t *info = NULL;
-  snd_ctl_elem_list_t *clist = NULL;
-  snd_ctl_elem_id_t *cid = NULL;
-  snd_pcm_info_t *pinfo = NULL;
-
-  GST_WARNING ("Opening IEC958 named device failed. Trying to autodetect");
-
-  if ((err = snd_ctl_open (&ctl, ctl_name, card)) < 0)
-    return err;
-
-  snd_ctl_card_info_malloc (&info);
-  snd_pcm_info_malloc (&pinfo);
-
-  /* Find a mixer for IEC958 settings */
-  snd_ctl_elem_list_malloc (&clist);
-  if ((err = snd_ctl_elem_list (ctl, clist)) < 0)
-    goto beach;
-
-  if ((err =
-          snd_ctl_elem_list_alloc_space (clist,
-              snd_ctl_elem_list_get_count (clist))) < 0)
-    goto beach;
-  if ((err = snd_ctl_elem_list (ctl, clist)) < 0)
-    goto beach;
-
-  count = snd_ctl_elem_list_get_used (clist);
-  for (idx = 0; idx < count; idx++) {
-    if (strstr (snd_ctl_elem_list_get_name (clist, idx), spdif_name) != NULL)
-      break;
-  }
-  if (idx == count) {
-    /* No SPDIF mixer availble */
-    err = 0;
-    goto beach;
-  }
-  snd_ctl_elem_id_malloc (&cid);
-  snd_ctl_elem_list_get_id (clist, idx, cid);
-
-  /* Now find a PCM device for IEC 958 */
-  if ((err = snd_ctl_card_info (ctl, info)) < 0)
-    goto beach;
-  dev = -1;
-  do {
-    if (snd_ctl_pcm_next_device (ctl, &dev) < 0)
-      goto beach;
-    if (dev < 0)
-      break;                    /* No more devices */
-
-    /* Filter for playback devices */
-    snd_pcm_info_set_device (pinfo, dev);
-    snd_pcm_info_set_subdevice (pinfo, 0);
-    snd_pcm_info_set_stream (pinfo, SND_PCM_STREAM_PLAYBACK);
-    if ((err = snd_ctl_pcm_info (ctl, pinfo)) < 0) {
-      if (err != -ENOENT)
-        goto beach;             /* Genuine error */
-
-      /* Device has no playback streams */
-      continue;
-    }
-    if (strstr (snd_pcm_info_get_name (pinfo), "IEC958") == NULL)
-      continue;                 /* Not the device we are looking for */
-
-    count = snd_pcm_info_get_subdevices_count (pinfo);
-    GST_LOG_OBJECT (sink, "Device %d has %d subdevices\n", dev,
-        snd_pcm_info_get_subdevices_count (pinfo));
-    for (idx = 0; idx < count; idx++) {
-      snd_pcm_info_set_subdevice (pinfo, idx);
-
-      if ((err = snd_ctl_pcm_info (ctl, pinfo)) < 0)
-        goto beach;
-
-      g_assert (snd_pcm_info_get_stream (pinfo) == SND_PCM_STREAM_PLAYBACK);
-
-      GST_LOG_OBJECT (sink, "Found playback stream on dev %d sub-d %d\n", dev,
-          idx);
-
-      /* Found a suitable PCM device, let's open it */
-      g_snprintf (pcm_name, 24, "hw:%d,%d", card, dev);
-      if ((err =
-              snd_pcm_open (&(pcm), pcm_name, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
-        goto beach;
-
-      break;
-    }
-  } while (pcm == NULL);
-
-  if (pcm != NULL) {
-    snd_ctl_elem_value_t *cval;
-    snd_aes_iec958_t iec958;
-
-    /* Have a PCM device and a mixer, set things up */
-    snd_ctl_elem_value_malloc (&cval);
-    snd_ctl_elem_value_set_id (cval, cid);
-    snd_ctl_elem_value_get_iec958 (cval, &iec958);
-    iec958.status[0] = IEC958_AES0_NONAUDIO;
-    iec958.status[1] = IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER;
-    iec958.status[2] = 0;
-    iec958.status[3] = IEC958_AES3_CON_FS_48000;
-    snd_ctl_elem_value_set_iec958 (cval, &iec958);
-    snd_ctl_elem_value_free (cval);
-
-    sink->pcm = pcm;
-    pcm = NULL;
-    err = 0;
-  }
-
-beach:
-  if (pcm)
-    snd_pcm_close (pcm);
-  if (clist)
-    snd_ctl_elem_list_clear (clist);
-  if (ctl)
-    snd_ctl_close (ctl);
-  if (clist)
-    snd_ctl_elem_list_free (clist);
-  if (cid)
-    snd_ctl_elem_id_free (cid);
-  if (info)
-    snd_ctl_card_info_free (info);
-  if (pinfo)
-    snd_pcm_info_free (pinfo);
-  return err;
-}
-
-static void
-alsaspdifsink_write_frame (AlsaSPDIFSink * sink, guchar * buf)
-{
-  snd_pcm_sframes_t res;
-  int num_frames = IEC958_FRAME_SIZE / ALSASPDIFSINK_BYTES_PER_FRAME;
-
-  /* If we couldn't output big endian when we opened the devic, then
-   * we need to swap here */
-  if (sink->need_swap) {
-    int i;
-    guchar tmp;
-
-    for (i = 0; i < IEC958_FRAME_SIZE; i += 2) {
-      tmp = buf[i];
-      buf[i] = buf[i + 1];
-      buf[i + 1] = tmp;
-    }
-  }
-
-  res = 0;
-  do {
-    if (res == -EPIPE) {
-      /* Underrun. */
-      GST_INFO_OBJECT (sink, "buffer underrun");
-      res = snd_pcm_prepare (sink->pcm);
-    } else if (res == -ESTRPIPE) {
-      /* Suspend. */
-      while ((res = snd_pcm_resume (sink->pcm)) == -EAGAIN) {
-        GST_DEBUG_OBJECT (sink, "sleeping for suspend");
-        g_usleep (100000);
-      }
-
-      if (res < 0) {
-        res = snd_pcm_prepare (sink->pcm);
-      }
-    }
-
-    if (res >= 0) {
-      res = snd_pcm_writei (sink->pcm, (void *) buf, num_frames);
-    }
-
-    if (res > 0) {
-      num_frames -= res;
-    }
-
-  } while (res == -EPIPE || num_frames > 0);
-
-  sink->frames++;
-
-  if (res < 0) {
-    GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
-        ("writei returned error: %s", snd_strerror (res)), GST_ERROR_SYSTEM);
-    return;
-  }
-}
-
-static gboolean
-alsaspdifsink_event (GstBaseSink * bsink, GstEvent * event)
-{
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_FLUSH_START:
-      snd_pcm_drop (sink->pcm);
-      break;
-    case GST_EVENT_FLUSH_STOP:
-      snd_pcm_start (sink->pcm);
-      break;
-    default:
-      break;
-  }
-
-  return TRUE;
-}
-
-static void
-alsaspdifsink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
-    GstClockTime * start, GstClockTime * end)
-{
-  /* Like GstBaseAudioSink, we set these to NONE */
-  *start = GST_CLOCK_TIME_NONE;
-  *end = GST_CLOCK_TIME_NONE;
-}
-
-static snd_pcm_sframes_t
-alsaspdifsink_delay (AlsaSPDIFSink * sink)
-{
-  snd_pcm_sframes_t delay;
-  int err;
-
-  err = snd_pcm_delay (sink->pcm, &delay);
-  if (err < 0 || delay < 0) {
-    return 0;
-  }
-
-  return delay;
-}
-
-#if 0
-static void
-generate_iec958_zero_frame (guchar * buffer)
-{
-  /* 2 sync words, 16 bits each */
-  buffer[0] = 0xF8;
-  buffer[1] = 0x72;
-  buffer[2] = 0x4E;
-  buffer[3] = 0x1F;
-
-  /* 16-bit burst-info. Contains data type (zero here, for 'null data'),
-     stream number (we output '0' for this always), and a few other bits.
-     As it happens, all-zero is the correct value.
-   */
-  buffer[4] = 0;
-  buffer[5] = 0;
-
-  /* 16-bit frame size. Also zero */
-  buffer[6] = 0;
-  buffer[7] = 0;
-
-  memset (buffer + 8, 0, IEC958_FRAME_SIZE - 8);
-}
-#endif
-
-static GstFlowReturn
-alsaspdifsink_render (GstBaseSink * bsink, GstBuffer * buf)
-{
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
-
-#if 0
-  GstClockTime next_write;
-
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
-    sink->cur_ts = GST_BUFFER_TIMESTAMP (buf);
-
-  next_write = gst_element_get_time (GST_ELEMENT (sink)) +
-      alsaspdifsink_current_delay (sink);
-
-  /*     
-     fprintf (stderr, "Drift: % 0.6fs, delay: % 0.6fs\r",  
-     GST_TIME_ARGS (GST_CLOCK_DIFF (sink->cur_ts, next_write)),
-     GST_TIME_ARGS (alsaspdifsink_current_delay (sink)));
-   */
-
-  /* If we're too far behind, send empty IEC958 frames. */
-  if (sink->cur_ts > next_write + MAX_SYNC_DIFF) {
-    int frames = (int) (
-        ((double) (sink->cur_ts - next_write)) /
-        (double) IEC958_FRAME_DURATION + 0.5);
-    int i;
-
-    for (i = 0; i < frames; i++) {
-      static guchar frame[IEC958_FRAME_SIZE];
-
-      generate_iec958_zero_frame (frame);
-
-      alsaspdifsink_write_frame (sink, frame);
-    }
-  }
-  /* If we're too far ahead, just drop this buffer */
-  else if (sink->cur_ts + MAX_SYNC_DIFF < next_write) {
-    goto end;
-  }
-#endif
-
-  GST_LOG_OBJECT (sink, "Writing %d bytes to spdif out", GST_BUFFER_SIZE (buf));
-  if (GST_BUFFER_SIZE (buf) == IEC958_FRAME_SIZE)
-    alsaspdifsink_write_frame (sink, GST_BUFFER_DATA (buf));
-  else
-    GST_WARNING_OBJECT (sink, "Ignoring buffer of incorrect size");
-
-#if 0
-end:
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DURATION (buf)))
-    sink->cur_ts = GST_BUFFER_DURATION (buf);
-#endif
-
-  return GST_FLOW_OK;
-}
-
-/* Drop error output from within alsalib on the floor */
-static void
-ignore_alsa_err (const char *file, int line, const char *function,
-    int err, const char *fmt, ...)
-{
-}
-
-static GstStateChangeReturn
-alsaspdifsink_change_state (GstElement * element, GstStateChange transition)
-{
-  AlsaSPDIFSink *sink = ALSASPDIFSINK (element);
-  GstStateChangeReturn ret;
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-      sink->frames = 0;
-      gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->clock), 0);
-      break;
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      if (!alsaspdifsink_open (sink)) {
-        GST_WARNING_OBJECT (sink, "Failed to open alsa device");
-        return GST_STATE_CHANGE_FAILURE;
-      }
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      break;
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-  GST_INFO_OBJECT (sink, "Parent change_state returned %d", ret);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      alsaspdifsink_close (sink);
-      break;
-    default:
-      break;
-  }
-
-  return ret;
-}
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  /* no rank so it doesn't get autoplugged by autoaudiosink */
-  if (!gst_element_register (plugin, "alsaspdifsink", GST_RANK_NONE,
-          GST_TYPE_ALSASPDIFSINK)) {
-    return FALSE;
-  }
-  return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    "alsaspdif",
-    "Alsa plugin for S/PDIF output",
-    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
diff --git a/ext/alsaspdif/alsaspdifsink.h b/ext/alsaspdif/alsaspdifsink.h
deleted file mode 100644
index 925b1f778..000000000
--- a/ext/alsaspdif/alsaspdifsink.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/* Based on a plugin from Martin Soto's Seamless DVD Player.
- * Copyright (C) 2003, 2004 Martin Soto <martinsoto@users.sourceforge.net>
- *               2005-6 Michael Smith <msmith@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.
- */
-
-#ifndef __ALSASPDIFSINK_H__
-#define __ALSASPDIFSINK_H__
-
-#include <gst/gst.h>
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-#include <alsa/asoundlib.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_ALSASPDIFSINK \
-  (alsaspdifsink_get_type())
-#define ALSASPDIFSINK(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_ALSASPDIFSINK,AlsaSPDIFSink))
-#define ALSASPDIFSINK_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_ALSASPDIFSINK,AlsaSPDIFSinkClass))
-#define GST_IS_ALSASPDIFSINK(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_ALSASPDIFSINK))
-#define GST_IS_ALSASPDIFSINK_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_ALSASPDIFSINK))
-#define GST_TYPE_ALSASPDIFSINK (alsaspdifsink_get_type())
-
-typedef struct _AlsaSPDIFSink AlsaSPDIFSink;
-typedef struct _AlsaSPDIFSinkClass AlsaSPDIFSinkClass;
-
-typedef enum {
-  ALSASPDIFSINK_OPEN = GST_ELEMENT_FLAG_LAST,
-  ALSASPDIFSINK_FLAG_LAST  = GST_ELEMENT_FLAG_LAST + 2,
-} AlsaSPDIFSinkFlags;
-
-/* ALSA spdif types. */
-enum {
-  SPDIF_NONE = 0,
-  SPDIF_CON,
-  SPDIF_PRO,
-  SPDIF_PCM
-};
-
-struct _AlsaSPDIFSink {
-  GstBaseSink basesink;
-
-  GstClockTime cur_ts;		/* Current time stamp. */
-
-  snd_pcm_t *pcm;		/* ALSA output device. */
-
-  gint card;                    /* ALSA card number to use */
-  char *device;                 /* ALSA device name */
-
-  GstClock *clock;		/* The clock for this element. */
-
-  guint64 frames;               /* Number of complete frames written */
-  gboolean need_swap;           /* Whether to byte swap outgoing data */
-
-  gint rate;                    /* Sampling rate of data */
-};
-
-struct _AlsaSPDIFSinkClass {
-  GstBaseSinkClass parent_class;
-};
-
-extern GType	alsaspdifsink_get_type		(void);
-
-G_END_DECLS
-
-#endif /* __DXR3AUDIOINK_H__ */
diff --git a/gst-plugins-bad.spec.in b/gst-plugins-bad.spec.in
index 24043da41..8da39aae5 100644
--- a/gst-plugins-bad.spec.in
+++ b/gst-plugins-bad.spec.in
@@ -159,7 +159,6 @@ rm -rf $RPM_BUILD_ROOT
 @USE_XVID_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstxvid.so
 @USE_BZ2_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstbz2.so
 @USE_NEON_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstneonhttpsrc.so
-@USE_ALSA_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstalsaspdif.so
 @USE_MUSEPACK_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstmusepack.so
 @USE_GSM_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstgsm.so
 @USE_DTS_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstdtsdec.so
diff --git a/m4/Makefile.am b/m4/Makefile.am
index 094c3fdb8..e03a94a61 100644
--- a/m4/Makefile.am
+++ b/m4/Makefile.am
@@ -13,7 +13,6 @@ EXTRA_DIST = \
 	gettext.m4 \
 	glibc21.m4 \
 	glib.m4 \
-	gst-alsa.m4 \
 	gst-artsc.m4 \
 	gst-fionread.m4 \
 	gst-matroska.m4 \
diff --git a/m4/gst-alsa.m4 b/m4/gst-alsa.m4
deleted file mode 100644
index 4141d0664..000000000
--- a/m4/gst-alsa.m4
+++ /dev/null
@@ -1,150 +0,0 @@
-dnl Configure Paths for Alsa
-dnl Some modifications by Richard Boulton <richard-alsa@tartarus.org>
-dnl Christopher Lansdown <lansdoct@cs.alfred.edu>
-dnl Jaroslav Kysela <perex@suse.cz>
-dnl Last modification: alsa.m4,v 1.23 2004/01/16 18:14:22 tiwai Exp
-dnl AM_PATH_ALSA([MINIMUM-VERSION [, ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]])
-dnl Test for libasound, and define ALSA_CFLAGS and ALSA_LIBS as appropriate.
-dnl enables arguments --with-alsa-prefix=
-dnl                   --with-alsa-enc-prefix=
-dnl                   --disable-alsatest
-dnl
-dnl For backwards compatibility, if ACTION_IF_NOT_FOUND is not specified,
-dnl and the alsa libraries are not found, a fatal AC_MSG_ERROR() will result.
-dnl
-AC_DEFUN([AM_PATH_ALSA],
-[dnl Save the original CFLAGS, LDFLAGS, and LIBS
-alsa_save_CFLAGS="$CFLAGS"
-alsa_save_LDFLAGS="$LDFLAGS"
-alsa_save_LIBS="$LIBS"
-alsa_found=yes
-
-dnl
-dnl Get the cflags and libraries for alsa
-dnl
-AC_ARG_WITH(alsa-prefix,
-  AC_HELP_STRING([--with-alsa-prefix=PFX],
-                 [prefix where Alsa library is installed(optional)]),
-  [alsa_prefix="$withval"], [alsa_prefix=""])
-
-AC_ARG_WITH(alsa-inc-prefix,
-  AC_HELP_STRING([--with-alsa-inc-prefix=PFX],
-                 [prefix where include libraries are (optional)]),
-  [alsa_inc_prefix="$withval"], [alsa_inc_prefix=""])
-
-dnl FIXME: this is not yet implemented
-dnl AC_ARG_ENABLE(alsatest,
-dnl  AC_HELP_STRING([--disable-alsatest],
-dnl                 [do not try to compile and run a test Alsa program],
-dnl                 [enable_alsatest=no], [enable_alsatest=yes])
-
-dnl Add any special include directories
-AC_MSG_CHECKING(for ALSA CFLAGS)
-if test "$alsa_inc_prefix" != "" ; then
-	ALSA_CFLAGS="$ALSA_CFLAGS -I$alsa_inc_prefix"
-	CFLAGS="$CFLAGS -I$alsa_inc_prefix"
-fi
-AC_MSG_RESULT($ALSA_CFLAGS)
-
-dnl add any special lib dirs
-AC_MSG_CHECKING(for ALSA LDFLAGS)
-if test "$alsa_prefix" != "" ; then
-	ALSA_LIBS="$ALSA_LIBS -L$alsa_prefix"
-	LDFLAGS="$LDFLAGS $ALSA_LIBS"
-fi
-
-dnl add the alsa library
-ALSA_LIBS="$ALSA_LIBS -lasound -lm -ldl -lpthread"
-LIBS=`echo $LIBS | sed 's/-lm//'`
-LIBS=`echo $LIBS | sed 's/-ldl//'`
-LIBS=`echo $LIBS | sed 's/-lpthread//'`
-LIBS=`echo $LIBS | sed 's/  //'`
-LIBS="$ALSA_LIBS $LIBS"
-AC_MSG_RESULT($ALSA_LIBS)
-
-dnl Check for a working version of libasound that is of the right version.
-min_alsa_version=ifelse([$1], ,0.1.1,$1)
-AC_MSG_CHECKING(for libasound headers version >= $min_alsa_version)
-no_alsa=""
-    alsa_min_major_version=`echo $min_alsa_version | \
-           sed 's/\([[0-9]]*\).\([[0-9]]*\).\([[0-9]]*\)/\1/'`
-    alsa_min_minor_version=`echo $min_alsa_version | \
-           sed 's/\([[0-9]]*\).\([[0-9]]*\).\([[0-9]]*\)/\2/'`
-    alsa_min_micro_version=`echo $min_alsa_version | \
-           sed 's/\([[0-9]]*\).\([[0-9]]*\).\([[0-9]]*\)/\3/'`
-
-AC_LANG_SAVE
-AC_LANG_C
-AC_TRY_COMPILE([
-#include <alsa/asoundlib.h>
-], [
-void main(void)
-{
-/* ensure backward compatibility */
-#if !defined(SND_LIB_MAJOR) && defined(SOUNDLIB_VERSION_MAJOR)
-#define SND_LIB_MAJOR SOUNDLIB_VERSION_MAJOR
-#endif
-#if !defined(SND_LIB_MINOR) && defined(SOUNDLIB_VERSION_MINOR)
-#define SND_LIB_MINOR SOUNDLIB_VERSION_MINOR
-#endif
-#if !defined(SND_LIB_SUBMINOR) && defined(SOUNDLIB_VERSION_SUBMINOR)
-#define SND_LIB_SUBMINOR SOUNDLIB_VERSION_SUBMINOR
-#endif
-
-#  if(SND_LIB_MAJOR > $alsa_min_major_version)
-  exit(0);
-#  else
-#    if(SND_LIB_MAJOR < $alsa_min_major_version)
-#       error not present
-#    endif
-
-#   if(SND_LIB_MINOR > $alsa_min_minor_version)
-  exit(0);
-#   else
-#     if(SND_LIB_MINOR < $alsa_min_minor_version)
-#          error not present
-#      endif
-
-#      if(SND_LIB_SUBMINOR < $alsa_min_micro_version)
-#        error not present
-#      endif
-#    endif
-#  endif
-exit(0);
-}
-],
-  [AC_MSG_RESULT(found.)],
-  [AC_MSG_RESULT(not present.)
-   ifelse([$3], , [AC_MSG_ERROR(Sufficiently new version of libasound not found.)])
-   alsa_found=no]
-)
-AC_LANG_RESTORE
-
-dnl Now that we know that we have the right version, let's see if we have the library and not just the headers.
-if test "x$enable_alsatest" = "xyes"; then
-AC_CHECK_LIB([asound], [snd_ctl_open],,
-	[ifelse([$3], , [AC_MSG_ERROR(No linkable libasound was found.)])
-	 alsa_found=no]
-)
-fi
-
-if test "x$alsa_found" = "xyes" ; then
-   ifelse([$2], , :, [$2])
-   LIBS=`echo $LIBS | sed 's/-lasound//g'`
-   LIBS=`echo $LIBS | sed 's/  //'`
-   LIBS="-lasound $LIBS"
-fi
-if test "x$alsa_found" = "xno" ; then
-   ifelse([$3], , :, [$3])
-   CFLAGS="$alsa_save_CFLAGS"
-   LDFLAGS="$alsa_save_LDFLAGS"
-   LIBS="$alsa_save_LIBS"
-   ALSA_CFLAGS=""
-   ALSA_LIBS=""
-fi
-
-dnl That should be it.  Now just export out symbols:
-AC_SUBST(ALSA_CFLAGS)
-AC_SUBST(ALSA_LIBS)
-])
-