New DTS decoder.

Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/dts/Makefile.am:
* ext/dts/gstdtsdec.c: (gst_dtsdec_get_type),
(gst_dtsdec_base_init), (gst_dtsdec_class_init), (gst_dtsdec_init),
(gst_dtsdec_channels), (gst_dtsdec_renegotiate),
(gst_dtsdec_handle_event), (gst_dtsdec_update_streaminfo),
(gst_dtsdec_loop), (gst_dtsdec_change_state),
(gst_dtsdec_set_property), (gst_dtsdec_get_property),
(plugin_init):
* ext/dts/gstdtsdec.h:
New DTS decoder.
* ext/faad/gstfaad.c: (gst_faad_sinkconnect),
(gst_faad_srcconnect):
Add ESDS atom handling (.m4a).

ChangeLog

@@ -1,3 +1,21 @@
+2004-04-28  Ronald Bultje  <rbultje@ronald.bitfreak.net>
+
+	* configure.ac:
+	* ext/Makefile.am:
+	* ext/dts/Makefile.am:
+	* ext/dts/gstdtsdec.c: (gst_dtsdec_get_type),
+	(gst_dtsdec_base_init), (gst_dtsdec_class_init), (gst_dtsdec_init),
+	(gst_dtsdec_channels), (gst_dtsdec_renegotiate),
+	(gst_dtsdec_handle_event), (gst_dtsdec_update_streaminfo),
+	(gst_dtsdec_loop), (gst_dtsdec_change_state),
+	(gst_dtsdec_set_property), (gst_dtsdec_get_property),
+	(plugin_init):
+	* ext/dts/gstdtsdec.h:
+	  New DTS decoder.
+	* ext/faad/gstfaad.c: (gst_faad_sinkconnect),
+	(gst_faad_srcconnect):
+	  Add ESDS atom handling (.m4a).
+
 2004-04-27  Ronald Bultje  <rbultje@ronald.bitfreak.net>
 
 	* ext/divx/gstdivxdec.c: (plugin_init):

configure.ac

@@ -808,6 +808,13 @@ return 0;
   fi
 ])
 
+dnl *** DTS ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_DTS, true)
+GST_CHECK_FEATURE(DTS, [dts library], dtsdec, [
+  GST_CHECK_LIBHEADER(DTS, dts_pic, dts_init, -lm, dts.h, DTS_LIBS="-ldts_pic -lm")
+  AC_SUBST(DTS_LIBS)
+])
+
 dnl *** dvdread ***
 translit(dnm, m, l) AM_CONDITIONAL(USE_DVDREAD, true)
 GST_CHECK_FEATURE(DVDREAD, [dvdread library], dvdreadsrc, [
@@ -1778,6 +1785,7 @@ ext/artsd/Makefile
 ext/audiofile/Makefile
 ext/cdparanoia/Makefile
 ext/divx/Makefile
+ext/dts/Makefile
 ext/dv/Makefile
 ext/dvdread/Makefile
 ext/dvdnav/Makefile

ext/Makefile.am

@@ -46,6 +46,12 @@ else
 DIVX_DIR=
 endif
 
+if USE_DTS
+DTS_DIR=dvdread
+else
+DTS_DIR=
+endif
+
 if USE_DVDREAD
 DVDREAD_DIR=dvdread
 else
@@ -331,6 +337,7 @@ SUBDIRS=\
 	$(AUDIOFILE_DIR) \
 	$(CDPARANOIA_DIR) \
 	$(DIVX_DIR) \
+	$(DTS_DIR) \
 	$(DVDREAD_DIR) \
 	$(DVDNAV_DIR) \
 	$(ESD_DIR) \
@@ -386,6 +393,7 @@ DIST_SUBDIRS=\
 	audiofile \
 	cdparanoia \
 	divx \
+	dts \
 	dv \
 	dvdread \
 	dvdnav \

ext/dts/Makefile.am

@@ -0,0 +1,8 @@
+plugin_LTLIBRARIES = libgstdtsdec.la
+
+libgstdtsdec_la_SOURCES = gstdtsdec.c
+libgstdtsdec_la_CFLAGS = $(GST_CFLAGS)
+libgstdtsdec_la_LIBADD = $(DTS_LIBS)
+libgstdtsdec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstdtsdec.h

ext/dts/gstdtsdec.c

@@ -0,0 +1,514 @@
+/* GStreamer DTS decoder plugin based on libdtsdec
+ * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include "_stdint.h"
+#include <stdlib.h>
+
+#include <gst/gst.h>
+#include <dts.h>
+
+#include "gstdtsdec.h"
+
+GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
+#define GST_CAT_DEFAULT (dtsdec_debug)
+
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  ARG_0,
+  ARG_DRC
+      /* FILL ME */
+};
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-dts")
+    );
+
+#if defined(LIBDTS_FIXED)
+#define DTS_CAPS "audio/x-raw-int, " \
+    "endianness = (int) BYTE_ORDER, " \
+    "signed = (boolean) true, " \
+    "width = (int) 16, " \
+    "depth = (int) 16"
+#define SAMPLE_WIDTH 16
+#elif defined(LIBDTS_DOUBLE)
+#define DTS_CAPS "audio/x-raw-float, " \
+    "endianness = (int) BYTE_ORDER, " \
+    "width = (int) 64"
+#define SAMPLE_WIDTH 64
+#else
+#define DTS_CAPS "audio/x-raw-float, " \
+    "endianness = (int) BYTE_ORDER, " \
+    "width = (int) 32"
+#define SAMPLE_WIDTH 32
+#endif
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (DTS_CAPS ", "
+        "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
+    );
+
+static void gst_dtsdec_base_init (GstDtsDecClass * klass);
+static void gst_dtsdec_class_init (GstDtsDecClass * klass);
+static void gst_dtsdec_init (GstDtsDec * dtsdec);
+
+static void gst_dtsdec_loop (GstElement * element);
+static GstElementStateReturn gst_dtsdec_change_state (GstElement * element);
+
+static void gst_dtsdec_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_dtsdec_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstElementClass *parent_class = NULL;
+
+/* static guint gst_dtsdec_signals[LAST_SIGNAL] = { 0 }; */
+
+GType
+gst_dtsdec_get_type (void)
+{
+  static GType dtsdec_type = 0;
+
+  if (!dtsdec_type) {
+    static const GTypeInfo dtsdec_info = {
+      sizeof (GstDtsDecClass),
+      (GBaseInitFunc) gst_dtsdec_base_init,
+      NULL, (GClassInitFunc) gst_dtsdec_class_init,
+      NULL,
+      NULL,
+      sizeof (GstDtsDec),
+      0,
+      (GInstanceInitFunc) gst_dtsdec_init,
+    };
+
+    dtsdec_type =
+        g_type_register_static (GST_TYPE_ELEMENT, "GstDtsDec", &dtsdec_info, 0);
+
+    GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
+  }
+  return dtsdec_type;
+}
+
+static void
+gst_dtsdec_base_init (GstDtsDecClass * klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  static GstElementDetails gst_dtsdec_details = {
+    "DTS audio decoder",
+    "Codec/Audio/Decoder",
+    "Decodes DTS audio streams",
+    "Ronald Bultje <rbultje@ronald.bitfreak.net>"
+  };
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_factory));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_factory));
+  gst_element_class_set_details (element_class, &gst_dtsdec_details);
+}
+
+static void
+gst_dtsdec_class_init (GstDtsDecClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+  g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
+      g_param_spec_boolean ("drc", "Dynamic Range Compression",
+          "Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
+
+  gobject_class->set_property = gst_dtsdec_set_property;
+  gobject_class->get_property = gst_dtsdec_get_property;
+
+  gstelement_class->change_state = gst_dtsdec_change_state;
+}
+
+static void
+gst_dtsdec_init (GstDtsDec * dtsdec)
+{
+  GstElement *element = GST_ELEMENT (dtsdec);
+
+  /* create the sink and src pads */
+  dtsdec->sinkpad =
+      gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
+          (dtsdec), "sink"), "sink");
+  gst_element_add_pad (element, dtsdec->sinkpad);
+  gst_element_set_loop_function (element, gst_dtsdec_loop);
+
+  dtsdec->srcpad =
+      gst_pad_new_from_template (gst_element_get_pad_template (element,
+          "src"), "src");
+  gst_pad_use_explicit_caps (dtsdec->srcpad);
+  gst_element_add_pad (element, dtsdec->srcpad);
+
+  GST_FLAG_SET (element, GST_ELEMENT_EVENT_AWARE);
+  dtsdec->dynamic_range_compression = FALSE;
+}
+
+static gint
+gst_dtsdec_channels (uint32_t flags)
+{
+  gint chans = 0;
+
+  switch (flags & DTS_CHANNEL_MASK) {
+    case DTS_MONO:
+      chans = 1;
+      break;
+    case DTS_CHANNEL:
+    case DTS_STEREO:
+    case DTS_STEREO_SUMDIFF:
+    case DTS_STEREO_TOTAL:
+    case DTS_DOLBY:
+      chans = 2;
+      break;
+    case DTS_3F:
+    case DTS_2F1R:
+      chans = 3;
+      break;
+    case DTS_3F1R:
+    case DTS_2F2R:
+      chans = 4;
+      break;
+    case DTS_3F2R:
+      chans = 5;
+      break;
+    case DTS_4F2R:
+      chans = 6;
+      break;
+    default:
+      /* error */
+      g_warning ("dtsdec: invalid flags 0x%x", flags);
+      return 0;
+  }
+  if (flags & DTS_LFE)
+    chans += 1;
+
+  return chans;
+}
+
+static gboolean
+gst_dtsdec_renegotiate (GstDtsDec * dts)
+{
+  GstCaps *caps = gst_caps_from_string (DTS_CAPS);
+  gint channels = gst_dtsdec_channels (dts->using_channels);
+
+  GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
+      channels, dts->sample_rate);
+
+  gst_caps_set_simple (caps,
+      "channels", G_TYPE_INT, channels,
+      "rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
+
+  return gst_pad_set_explicit_caps (dts->srcpad, caps);
+}
+
+static void
+gst_dtsdec_handle_event (GstDtsDec * dts)
+{
+  guint32 remaining;
+  GstEvent *event;
+
+  gst_bytestream_get_status (dts->bs, &remaining, &event);
+
+  if (!event) {
+    GST_ELEMENT_ERROR (dts, RESOURCE, READ, (NULL), (NULL));
+    return;
+  }
+
+  GST_LOG ("Handling event of type %d timestamp %llu", GST_EVENT_TYPE (event),
+      GST_EVENT_TIMESTAMP (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_DISCONTINUOUS:
+    case GST_EVENT_FLUSH:
+      gst_bytestream_flush_fast (dts->bs, remaining);
+      break;
+    default:
+      break;
+  }
+
+  gst_pad_event_default (dts->sinkpad, event);
+}
+
+static void
+gst_dtsdec_update_streaminfo (GstDtsDec * dts)
+{
+  GstTagList *taglist;
+
+  taglist = gst_tag_list_new ();
+
+  gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
+      GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
+
+  gst_element_found_tags_for_pad (GST_ELEMENT (dts),
+      dts->srcpad, dts->current_ts, taglist);
+}
+
+static void
+gst_dtsdec_loop (GstElement * element)
+{
+  GstDtsDec *dts = GST_DTSDEC (element);
+  guint8 *data;
+  GstBuffer *buf, *out;
+  sample_t *samples;
+  gint i, length, flags, sample_rate, bit_rate, frame_length, s, c, num_c;
+  gint channels, skipped = 0, num_blocks;
+  guint32 got_bytes;
+  gboolean need_renegotiation = FALSE;
+  GstClockTime timestamp = 0;
+
+  /* find sync. Don't know what 3840 is based on...  */
+#define MAX_SKIP 3840
+  while (skipped < MAX_SKIP) {
+    got_bytes = gst_bytestream_peek_bytes (dts->bs, &data, 7);
+    if (got_bytes < 7) {
+      gst_dtsdec_handle_event (dts);
+      return;
+    }
+    length = dts_syncinfo (dts->state, data, &flags,
+        &sample_rate, &bit_rate, &frame_length);
+    if (length == 0) {
+      /* shift window to re-find sync */
+      gst_bytestream_flush_fast (dts->bs, 1);
+      skipped++;
+      GST_LOG ("Skipped");
+    } else
+      break;
+  }
+
+  if (skipped >= MAX_SKIP) {
+    GST_ELEMENT_ERROR (dts, RESOURCE, SYNC, (NULL), (NULL));
+    return;
+  }
+
+  /* go over stream properties, update caps/streaminfo if needed */
+  if (dts->sample_rate != sample_rate) {
+    need_renegotiation = TRUE;
+    dts->sample_rate = sample_rate;
+  }
+
+  dts->stream_channels = flags;
+
+  if (bit_rate != dts->bit_rate) {
+    dts->bit_rate = bit_rate;
+    gst_dtsdec_update_streaminfo (dts);
+  }
+
+  /* read the header + rest of frame */
+  got_bytes = gst_bytestream_read (dts->bs, &buf, length);
+  if (got_bytes < length) {
+    gst_dtsdec_handle_event (dts);
+    return;
+  }
+
+  data = GST_BUFFER_DATA (buf);
+  timestamp = gst_bytestream_get_timestamp (dts->bs);
+  if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
+    if (timestamp == dts->last_ts) {
+      timestamp = dts->current_ts;
+    } else {
+      dts->last_ts = timestamp;
+    }
+  }
+
+  /* process */
+  flags = dts->request_channels | DTS_ADJUST_LEVEL;
+  dts->level = 1;
+
+  if (dts_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
+    GST_WARNING ("dts_frame error");
+    goto end;
+  }
+
+  channels = flags & (DTS_CHANNEL_MASK | DTS_LFE);
+
+  if (dts->using_channels != channels) {
+    need_renegotiation = TRUE;
+    dts->using_channels = channels;
+  }
+
+  if (need_renegotiation == TRUE) {
+    GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
+        dts->sample_rate, dts->stream_channels, dts->using_channels);
+    if (!gst_dtsdec_renegotiate (dts))
+      goto end;
+  }
+
+  if (dts->dynamic_range_compression == FALSE) {
+    dts_dynrng (dts->state, NULL, NULL);
+  }
+
+  /* handle decoded data, one block is 256 samples */
+  num_blocks = dts_blocks_num (dts->state);
+  for (i = 0; i < num_blocks; i++) {
+    if (dts_block (dts->state)) {
+      GST_WARNING ("dts_block error %d", i);
+      continue;
+    }
+
+    samples = dts_samples (dts->state);
+    num_c = gst_dtsdec_channels (dts->using_channels);
+    out = gst_buffer_new_and_alloc ((SAMPLE_WIDTH / 8) * 256 * num_c);
+    GST_BUFFER_TIMESTAMP (out) = timestamp;
+    GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
+
+    /* libdts returns buffers in 256-sample-blocks per channel,
+     * we want interleaved. And we need to copy anyway... */
+    data = GST_BUFFER_DATA (out);
+    for (s = 0; s < 256; s++) {
+      for (c = 0; c < num_c; c++) {
+        *(sample_t *) data = samples[s + c * 256];
+        data += (SAMPLE_WIDTH / 8);
+      }
+    }
+
+    /* push on */
+    gst_pad_push (dts->srcpad, GST_DATA (out));
+
+    timestamp += GST_SECOND * 256 / dts->sample_rate;
+  }
+
+  dts->current_ts = timestamp;
+
+end:
+  gst_buffer_unref (buf);
+}
+
+static GstElementStateReturn
+gst_dtsdec_change_state (GstElement * element)
+{
+  GstDtsDec *dts = GST_DTSDEC (element);
+
+  switch (GST_STATE_TRANSITION (element)) {
+    case GST_STATE_NULL_TO_READY:{
+      GstCPUFlags cpuflags;
+      uint32_t mm_accel = 0;
+
+      dts->bs = gst_bytestream_new (dts->sinkpad);
+      cpuflags = gst_cpu_get_flags ();
+      if (cpuflags & GST_CPU_FLAG_MMX)
+        mm_accel |= MM_ACCEL_X86_MMX;
+      if (cpuflags & GST_CPU_FLAG_3DNOW)
+        mm_accel |= MM_ACCEL_X86_3DNOW;
+      if (cpuflags & GST_CPU_FLAG_MMXEXT)
+        mm_accel |= MM_ACCEL_X86_MMXEXT;
+
+      dts->state = dts_init (mm_accel);
+      break;
+    }
+    case GST_STATE_READY_TO_PAUSED:
+      dts->samples = dts_samples (dts->state);
+      dts->bit_rate = -1;
+      dts->sample_rate = -1;
+      dts->stream_channels = 0;
+      /* FIXME force stereo for now */
+      dts->request_channels = DTS_STEREO;
+      dts->using_channels = 0;
+      dts->level = 1;
+      dts->bias = 0;
+      dts->last_ts = 0;
+      dts->current_ts = 0;
+      break;
+    case GST_STATE_PAUSED_TO_READY:
+      dts->samples = NULL;
+      break;
+    case GST_STATE_READY_TO_NULL:
+      gst_bytestream_destroy (dts->bs);
+      dts->bs = NULL;
+      dts_free (dts->state);
+      dts->state = NULL;
+      break;
+    default:
+      break;
+  }
+
+  if (GST_ELEMENT_CLASS (parent_class)->change_state)
+    return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+  return GST_STATE_SUCCESS;
+}
+
+static void
+gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
+    GParamSpec * pspec)
+{
+  GstDtsDec *dts = GST_DTSDEC (object);
+
+  switch (prop_id) {
+    case ARG_DRC:
+      dts->dynamic_range_compression = g_value_get_boolean (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstDtsDec *dts = GST_DTSDEC (object);
+
+  switch (prop_id) {
+    case ARG_DRC:
+      g_value_set_boolean (value, dts->dynamic_range_compression);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_library_load ("gstbytestream"))
+    return FALSE;
+
+  if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
+          GST_TYPE_DTSDEC))
+    return FALSE;
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "dtsdec",
+    "Decodes DTS audio streams",
+    plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN);

ext/dts/gstdtsdec.h

@@ -0,0 +1,77 @@
+/* GStreamer DTS decoder plugin based on libdtsdec
+ * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+
+#ifndef __GST_DTSDEC_H__
+#define __GST_DTSDEC_H__
+
+#include <gst/gst.h>
+#include <gst/bytestream/bytestream.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_DTSDEC \
+  (gst_dtsdec_get_type())
+#define GST_DTSDEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DTSDEC,GstDtsDec))
+#define GST_DTSDEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DTSDEC,GstDtsDecClass))
+#define GST_IS_DTSDEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DTSDEC))
+#define GST_IS_DTSDEC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DTSDEC))
+
+typedef struct _GstDtsDec GstDtsDec;
+typedef struct _GstDtsDecClass GstDtsDecClass;
+
+struct _GstDtsDec {
+  GstElement 	 element;
+
+  /* pads */
+  GstPad 	*sinkpad,
+  		*srcpad;
+
+  /* stream properties */
+  gint 		 bit_rate;
+  gint 		 sample_rate;
+  gint 		 stream_channels;
+  gint 		 request_channels;
+  gint 		 using_channels;
+
+  /* decoding properties */
+  sample_t 	 level;
+  sample_t 	 bias;
+  gboolean 	 dynamic_range_compression;
+  sample_t 	*samples;
+  dts_state_t 	*state;
+
+  GstByteStream *bs;
+
+  /* keep track of time */
+  GstClockTime	last_ts;
+  GstClockTime	current_ts;
+};
+
+struct _GstDtsDecClass {
+  GstElementClass parent_class;
+};
+
+G_END_DECLS
+
+#endif /* __GST_DTSDEC_H__ */

ext/faad/gstfaad.c

@@ -146,8 +146,33 @@ gst_faad_init (GstFaad * faad)
 static GstPadLinkReturn
 gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
 {
-  /* oh, we really don't care what's in here. We'll
-   * get AAC audio (MPEG-2/4) anyway, so why bother? */
+  GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
+  GstStructure *str = gst_caps_get_structure (caps, 0);
+  const GValue *value;
+  GstBuffer *buf;
+
+  if ((value = gst_structure_get_value (str, "codec_data"))) {
+    GstPadLinkReturn ret;
+    gulong samplerate;
+    guchar channels;
+
+    buf = g_value_get_boxed (value);
+    if (faacDecInit2 (faad->handle, GST_BUFFER_DATA (buf),
+            GST_BUFFER_SIZE (buf), &samplerate, &channels) < 0)
+      return GST_PAD_LINK_REFUSED;
+
+    faad->samplerate = samplerate;
+    faad->channels = channels;
+
+    ret = gst_pad_renegotiate (faad->srcpad);
+    if (ret == GST_PAD_LINK_DELAYED)
+      ret = GST_PAD_LINK_OK;
+
+    return ret;
+  }
+
+  /* if there's no decoderspecificdata, it's all fine. We cannot know
+   * much more at this point... */
   return GST_PAD_LINK_OK;
 }
 
@@ -229,7 +254,7 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
 {
   GstStructure *structure;
   const gchar *mimetype;
-  gint fmt = 0;
+  gint fmt = -1;
   gint depth, rate, channels;
   GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
 
@@ -282,7 +307,7 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
     }
   }
 
-  if (fmt) {
+  if (fmt != -1) {
     faacDecConfiguration *conf;
 
     conf = faacDecGetCurrentConfiguration (faad->handle);