dtsdec: Reconcile element code with a52dec changes

Re-work the dtsdec element code to unify it with changes made it a52dec,
including support for reverse playback and dynamic channel negotiation
on the source pad.

2 files changed, 348 insertions, 173 deletions

diff --git a/ext/dts/gstdtsdec.c b/ext/dts/gstdtsdec.c
index 5b85a8032..08695936f 100644
--- a/ext/dts/gstdtsdec.c
+++ b/ext/dts/gstdtsdec.c
@@ -1,5 +1,6 @@
 /* GStreamer DTS decoder plugin based on libdtsdec
  * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * Copyright (C) 2009 Jan Schmidt <thaytan@noraisin.net>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -84,43 +85,46 @@ typedef struct dts_state_s dca_state_t;
 #include <liboil/liboilcpu.h>
 #include <liboil/liboilfunction.h>
 
-GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
-#define GST_CAT_DEFAULT (dtsdec_debug)
-
 static const GstElementDetails gst_dtsdec_details =
 GST_ELEMENT_DETAILS ("DTS audio decoder",
     "Codec/Decoder/Audio",
     "Decodes DTS audio streams",
+    "Jan Schmidt <thaytan@noraisin.net>\n"
     "Ronald Bultje <rbultje@ronald.bitfreak.net>");
 
+#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
+#define SAMPLE_WIDTH 16
+#elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
+#define SAMPLE_WIDTH 64
+#else
+#define SAMPLE_WIDTH 32
+#endif
+
+GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
+#define GST_CAT_DEFAULT (dtsdec_debug)
+
 enum
 {
   ARG_0,
   ARG_DRC
 };
+
 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-dts;" "audio/x-private1-dts")
+    GST_STATIC_CAPS ("audio/x-dts; audio/x-private1-dts")
     );
 
 #if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
 #define DTS_CAPS "audio/x-raw-int, " \
     "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
     "signed = (boolean) true, " \
-    "width = (int) 16, " \
+    "width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", " \
     "depth = (int) 16"
-#define SAMPLE_WIDTH 16
-#elif defined(LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
-#define DTS_CAPS "audio/x-raw-float, " \
-    "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
-    "width = (int) 64"
-#define SAMPLE_WIDTH 64
 #else
 #define DTS_CAPS "audio/x-raw-float, " \
     "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
-    "width = (int) 32"
-#define SAMPLE_WIDTH 32
+    "width = (int) " G_STRINGIFY (SAMPLE_WIDTH)
 #endif
 
 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
@@ -135,6 +139,7 @@ GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstElement, GST_TYPE_ELEMENT);
 static gboolean gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps);
 static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event);
 static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf);
+static GstFlowReturn gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf);
 static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element,
     GstStateChange transition);
 
@@ -155,7 +160,7 @@ gst_dtsdec_base_init (gpointer g_class)
       gst_static_pad_template_get (&src_factory));
   gst_element_class_set_details (element_class, &gst_dtsdec_details);
 
-  GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
+  GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder");
 }
 
 static void
@@ -202,6 +207,7 @@ gst_dtsdec_class_init (GstDtsDecClass * klass)
 static void
 gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
 {
+  /* create the sink and src pads */
   dtsdec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
   gst_pad_set_setcaps_function (dtsdec->sinkpad,
       GST_DEBUG_FUNCPTR (gst_dtsdec_sink_setcaps));
@@ -212,10 +218,12 @@ gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
   gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad);
 
   dtsdec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
-  gst_pad_use_fixed_caps (dtsdec->srcpad);
   gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->srcpad);
 
+  dtsdec->request_channels = DCA_CHANNEL;
   dtsdec->dynamic_range_compression = FALSE;
+
+  gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
 }
 
 static gint
@@ -317,6 +325,105 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
   return chans;
 }
 
+static void
+clear_queued (GstDtsDec * dec)
+{
+  g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (dec->queued);
+  dec->queued = NULL;
+}
+
+static GstFlowReturn
+flush_queued (GstDtsDec * dec)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+
+  while (dec->queued) {
+    GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
+
+    GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
+        GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+    /* iterate ouput queue an push downstream */
+    ret = gst_pad_push (dec->srcpad, buf);
+
+    dec->queued = g_list_delete_link (dec->queued, dec->queued);
+  }
+  return ret;
+}
+
+static GstFlowReturn
+gst_dtsdec_drain (GstDtsDec * dec)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+
+  if (dec->segment.rate < 0.0) {
+    /* if we have some queued frames for reverse playback, flush
+     * them now */
+    ret = flush_queued (dec);
+  }
+  return ret;
+}
+
+static GstFlowReturn
+gst_dtsdec_push (GstDtsDec * dtsdec,
+    GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
+{
+  GstBuffer *buf;
+  int chans, n, c;
+  GstFlowReturn result;
+
+  flags &= (DCA_CHANNEL_MASK | DCA_LFE);
+  chans = gst_dtsdec_channels (flags, NULL);
+  if (!chans) {
+    GST_ELEMENT_ERROR (GST_ELEMENT (dtsdec), STREAM, DECODE, (NULL),
+        ("Invalid channel flags: %d", flags));
+    return GST_FLOW_ERROR;
+  }
+
+  result =
+      gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
+      256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
+  if (result != GST_FLOW_OK)
+    return result;
+
+  for (n = 0; n < 256; n++) {
+    for (c = 0; c < chans; c++) {
+      ((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
+          samples[c * 256 + n];
+    }
+  }
+  GST_BUFFER_TIMESTAMP (buf) = timestamp;
+  GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / dtsdec->sample_rate;
+
+  result = GST_FLOW_OK;
+  if ((buf = gst_audio_buffer_clip (buf, &dtsdec->segment,
+              dtsdec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
+    /* set discont when needed */
+    if (dtsdec->discont) {
+      GST_LOG_OBJECT (dtsdec, "marking DISCONT");
+      GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+      dtsdec->discont = FALSE;
+    }
+
+    if (dtsdec->segment.rate > 0.0) {
+      GST_DEBUG_OBJECT (dtsdec,
+          "Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
+          GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+          GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+
+      result = gst_pad_push (srcpad, buf);
+    } else {
+      /* reverse playback, queue frame till later when we get a discont. */
+      GST_DEBUG_OBJECT (dtsdec, "queued frame");
+      dtsdec->queued = g_list_prepend (dtsdec->queued, buf);
+    }
+  }
+  return result;
+}
+
 static gboolean
 gst_dtsdec_renegotiate (GstDtsDec * dts)
 {
@@ -360,32 +467,57 @@ gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_NEWSEGMENT:{
       GstFormat format;
-      gint64 val;
-
-      gst_event_parse_new_segment (event, NULL, NULL, &format, &val, NULL,
-          NULL);
-      if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (val)) {
-        GST_WARNING ("No time in newsegment event %p", event);
+      gboolean update;
+      gint64 start, end, pos;
+      gdouble rate;
+
+      gst_event_parse_new_segment (event, &update, &rate, &format, &start, &end,
+          &pos);
+
+      /* drain queued buffers before activating the segment so that we can clip
+       * against the old segment first */
+      gst_dtsdec_drain (dtsdec);
+
+      if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) {
+        GST_WARNING ("No time in newsegment event %p (format is %s)",
+            event, gst_format_get_name (format));
+        gst_event_unref (event);
+        dtsdec->sent_segment = FALSE;
+        /* set some dummy values, FIXME: do proper conversion */
+        dtsdec->time = start = pos = 0;
+        format = GST_FORMAT_TIME;
+        end = -1;
       } else {
-        dtsdec->current_ts = val;
+        dtsdec->time = start;
+        dtsdec->sent_segment = TRUE;
+        ret = gst_pad_push_event (dtsdec->srcpad, event);
       }
 
-      if (dtsdec->cache) {
-        gst_buffer_unref (dtsdec->cache);
-        dtsdec->cache = NULL;
-      }
-      ret = gst_pad_event_default (pad, event);
+      gst_segment_set_newsegment (&dtsdec->segment, update, rate, format, start,
+          end, pos);
       break;
     }
+    case GST_EVENT_TAG:
+      ret = gst_pad_push_event (dtsdec->srcpad, event);
+      break;
+    case GST_EVENT_EOS:
+      gst_dtsdec_drain (dtsdec);
+      ret = gst_pad_push_event (dtsdec->srcpad, event);
+      break;
+    case GST_EVENT_FLUSH_START:
+      ret = gst_pad_push_event (dtsdec->srcpad, event);
+      break;
     case GST_EVENT_FLUSH_STOP:
       if (dtsdec->cache) {
         gst_buffer_unref (dtsdec->cache);
         dtsdec->cache = NULL;
       }
-      ret = gst_pad_event_default (pad, event);
+      clear_queued (dtsdec);
+      gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
+      ret = gst_pad_push_event (dtsdec->srcpad, event);
       break;
     default:
-      ret = gst_pad_event_default (pad, event);
+      ret = gst_pad_push_event (dtsdec->srcpad, event);
       break;
   }
 
@@ -393,24 +525,6 @@ gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
   return ret;
 }
 
-static gboolean
-gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
-{
-  GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
-  GstStructure *structure;
-
-  structure = gst_caps_get_structure (caps, 0);
-
-  if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
-    dts->dvdmode = TRUE;
-  else
-    dts->dvdmode = FALSE;
-
-  gst_object_unref (dts);
-
-  return TRUE;
-}
-
 static void
 gst_dtsdec_update_streaminfo (GstDtsDec * dts)
 {
@@ -419,6 +533,7 @@ gst_dtsdec_update_streaminfo (GstDtsDec * dts)
   taglist = gst_tag_list_new ();
 
   gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
+      GST_TAG_AUDIO_CODEC, "DTS DCA",
       GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
 
   gst_element_found_tags_for_pad (GST_ELEMENT (dts), dts->srcpad, taglist);
@@ -428,29 +543,37 @@ static GstFlowReturn
 gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
     guint length, gint flags, gint sample_rate, gint bit_rate)
 {
+  gint channels, i, num_blocks;
   gboolean need_renegotiation = FALSE;
-  gint channels, num_blocks;
-  GstBuffer *out;
-  gint i, s, c, num_c;
-  sample_t *samples;
-  GstFlowReturn result = GST_FLOW_OK;
 
-  /* go over stream properties, update caps/streaminfo if needed */
+  /* go over stream properties, renegotiate or update streaminfo if needed */
   if (dts->sample_rate != sample_rate) {
     need_renegotiation = TRUE;
     dts->sample_rate = sample_rate;
   }
 
-  dts->stream_channels = flags;
+  if (flags) {
+    dts->stream_channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
+  }
 
   if (bit_rate != dts->bit_rate) {
     dts->bit_rate = bit_rate;
     gst_dtsdec_update_streaminfo (dts);
   }
 
-  if (dts->request_channels == DCA_CHANNEL) {
+  /* If we haven't had an explicit number of channels chosen through properties
+   * at this point, choose what to downmix to now, based on what the peer will 
+   * accept - this allows a52dec to do downmixing in preference to a 
+   * downstream element such as audioconvert.
+   * FIXME: Add the property back in for forcing output channels.
+   */
+  if (dts->request_channels != DCA_CHANNEL) {
+    flags = dts->request_channels;
+  } else if (dts->flag_update) {
     GstCaps *caps;
 
+    dts->flag_update = FALSE;
+
     caps = gst_pad_get_allowed_caps (dts->srcpad);
     if (caps && gst_caps_get_size (caps) > 0) {
       GstCaps *copy = gst_caps_copy_nth (caps, 0);
@@ -472,38 +595,38 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
           flags ? gst_dtsdec_channels (flags, NULL) : 6);
       gst_structure_get_int (structure, "channels", &channels);
       if (channels <= 6)
-        dts->request_channels = dts_channels[channels - 1];
+        flags = dts_channels[channels - 1];
       else
-        dts->request_channels = dts_channels[5];
+        flags = dts_channels[5];
 
       gst_caps_unref (copy);
     } else if (flags) {
-      dts->request_channels = dts->stream_channels;
+      flags = dts->stream_channels;
     } else {
-      dts->request_channels = DCA_3F2R | DCA_LFE;
+      flags = DCA_3F2R | DCA_LFE;
     }
 
     if (caps)
       gst_caps_unref (caps);
+  } else {
+    flags = dts->using_channels;
   }
-
   /* process */
-  flags = dts->request_channels | DCA_ADJUST_LEVEL;
+  flags |= DCA_ADJUST_LEVEL;
   dts->level = 1;
-
   if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
-    GST_WARNING ("dts_frame error");
+    GST_WARNING_OBJECT (dts, "dts_frame error");
+    dts->discont = TRUE;
     return GST_FLOW_OK;
   }
-
   channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
-
   if (dts->using_channels != channels) {
     need_renegotiation = TRUE;
     dts->using_channels = channels;
   }
 
-  if (need_renegotiation == TRUE) {
+  /* negotiate if required */
+  if (need_renegotiation) {
     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)) {
@@ -520,107 +643,60 @@ gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
   num_blocks = dca_blocks_num (dts->state);
   for (i = 0; i < num_blocks; i++) {
     if (dca_block (dts->state)) {
-      GST_WARNING ("dts_block error %d", i);
-      continue;
-    }
-
-    samples = dca_samples (dts->state);
-    num_c = gst_dtsdec_channels (dts->using_channels, NULL);
-
-    result = gst_pad_alloc_buffer_and_set_caps (dts->srcpad, 0,
-        (SAMPLE_WIDTH / 8) * 256 * num_c, GST_PAD_CAPS (dts->srcpad), &out);
-
-    if (result != GST_FLOW_OK)
-      break;
+      /* Ignore errors, but mark a discont */
+      GST_WARNING_OBJECT (dts, "dts_block error %d", i);
+      dts->discont = TRUE;
+    } else {
+      GstFlowReturn ret;
 
-    GST_BUFFER_TIMESTAMP (out) = dts->current_ts;
-    GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
-    dts->current_ts += GST_BUFFER_DURATION (out);
-
-    /* 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 */
+      ret = gst_dtsdec_push (dts, dts->srcpad, dts->using_channels,
+          dts->samples, dts->time);
+      if (ret != GST_FLOW_OK)
+        return ret;
     }
-
-    /* push on */
-    result = gst_pad_push (dts->srcpad, out);
-
-    if (result != GST_FLOW_OK)
-      break;
+    dts->time += GST_SECOND * 256 / dts->sample_rate;
   }
 
-  return result;
+  return GST_FLOW_OK;
 }
 
-static GstFlowReturn
-gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
+static gboolean
+gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
 {
-  GstDtsDec *dts;
-  guint8 *data;
-  gint size;
-  gint length, flags, sample_rate, bit_rate, frame_length;
-  GstFlowReturn result = GST_FLOW_OK;
-
-  dts = GST_DTSDEC (GST_PAD_PARENT (pad));
-
-  if (dts->cache) {
-    buf = gst_buffer_join (dts->cache, buf);
-    dts->cache = NULL;
-  }
+  GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
+  GstStructure *structure;
 
-  data = GST_BUFFER_DATA (buf);
-  size = GST_BUFFER_SIZE (buf);
-  length = 0;
-  while (size >= 7) {
-    length = dca_syncinfo (dts->state, data, &flags,
-        &sample_rate, &bit_rate, &frame_length);
-    if (length == 0) {
-      /* shift window to re-find sync */
-      data++;
-      size--;
-    } else if (length <= size) {
-      GST_DEBUG ("Sync: frame size %d", length);
-      result = gst_dtsdec_handle_frame (dts, data, length,
-          flags, sample_rate, bit_rate);
-      if (result != GST_FLOW_OK) {
-        size = 0;
-        break;
-      }
-      size -= length;
-      data += length;
-    } else {
-      GST_LOG ("Not enough data available (needed %d had %d)", length, size);
-      break;
-    }
-  }
+  structure = gst_caps_get_structure (caps, 0);
 
-  /* keep cache */
-  if (length == 0) {
-    GST_LOG ("No sync found");
-  }
-  if (size > 0) {
-    dts->cache = gst_buffer_create_sub (buf,
-        GST_BUFFER_SIZE (buf) - size, size);
-  }
+  if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
+    dts->dvdmode = TRUE;
+  else
+    dts->dvdmode = FALSE;
 
-  gst_buffer_unref (buf);
+  gst_object_unref (dts);
 
-  return result;
+  return TRUE;
 }
 
-
 static GstFlowReturn
 gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
 {
-  GstFlowReturn res = GST_FLOW_OK;
+  GstFlowReturn ret = GST_FLOW_OK;
   GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
   gint first_access;
 
+  if (GST_BUFFER_IS_DISCONT (buf)) {
+    GST_LOG_OBJECT (dts, "received DISCONT");
+    gst_dtsdec_drain (dts);
+    /* clear cache on discont and mark a discont in the element */
+    if (dts->cache) {
+      gst_buffer_unref (dts->cache);
+      dts->cache = NULL;
+    }
+    dts->discont = TRUE;
+  }
+
   if (dts->dvdmode) {
     gint size = GST_BUFFER_SIZE (buf);
     guint8 *data = GST_BUFFER_DATA (buf);
@@ -644,8 +720,8 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
 
       subbuf = gst_buffer_create_sub (buf, offset, len);
       GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
-      res = gst_dtsdec_chain_raw (pad, subbuf);
-      if (res != GST_FLOW_OK)
+      ret = gst_dtsdec_chain_raw (pad, subbuf);
+      if (ret != GST_FLOW_OK)
         goto done;
 
       offset += len;
@@ -655,21 +731,20 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
         subbuf = gst_buffer_create_sub (buf, offset, len);
         GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
 
-        res = gst_dtsdec_chain_raw (pad, subbuf);
+        ret = gst_dtsdec_chain_raw (pad, subbuf);
       }
     } else {
-      /* first_access = 0 or 1, so if there's a timestamp it applies
-       * to the first byte */
+      /* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
       subbuf = gst_buffer_create_sub (buf, offset, size - offset);
       GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
-      res = gst_dtsdec_chain_raw (pad, subbuf);
+      ret = gst_dtsdec_chain_raw (pad, subbuf);
     }
   } else {
-    res = gst_dtsdec_chain_raw (pad, buf);
+    ret = gst_dtsdec_chain_raw (pad, buf);
   }
 
 done:
-  return res;
+  return ret;
 
 /* ERRORS */
 not_enough_data:
@@ -684,7 +759,97 @@ bad_first_access_parameter:
         ("Bad first_access parameter (%d) in buffer", first_access));
     return GST_FLOW_ERROR;
   }
+}
+
+static GstFlowReturn
+gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
+{
+  GstDtsDec *dts;
+  guint8 *data;
+  gint size;
+  gint length = 0, flags, sample_rate, bit_rate, frame_length;
+  GstFlowReturn result = GST_FLOW_OK;
+
+  dts = GST_DTSDEC (GST_PAD_PARENT (pad));
+
+  if (!dts->sent_segment) {
+    GstSegment segment;
+
+    /* Create a basic segment. Usually, we'll get a new-segment sent by 
+     * another element that will know more information (a demuxer). If we're
+     * just looking at a raw AC3 stream, we won't - so we need to send one
+     * here, but we don't know much info, so just send a minimal TIME 
+     * new-segment event
+     */
+    gst_segment_init (&segment, GST_FORMAT_TIME);
+    gst_pad_push_event (dts->srcpad, gst_event_new_new_segment (FALSE,
+            segment.rate, segment.format, segment.start,
+            segment.duration, segment.start));
+    dts->sent_segment = TRUE;
+  }
+
+  /* merge with cache, if any. Also make sure timestamps match */
+  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+    dts->time = GST_BUFFER_TIMESTAMP (buf);
+    GST_DEBUG_OBJECT (dts,
+        "Received buffer with ts %" GST_TIME_FORMAT " duration %"
+        GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+  }
+
+  if (dts->cache) {
+    buf = gst_buffer_join (dts->cache, buf);
+    dts->cache = NULL;
+  }
+  data = GST_BUFFER_DATA (buf);
+  size = GST_BUFFER_SIZE (buf);
+
+  /* find and read header */
+  bit_rate = dts->bit_rate;
+  sample_rate = dts->sample_rate;
+  flags = 0;
+  while (size >= 7) {
+    length = dca_syncinfo (dts->state, data, &flags,
+        &sample_rate, &bit_rate, &frame_length);
+
+    if (length == 0) {
+      /* shift window to re-find sync */
+      data++;
+      size--;
+    } else if (length <= size) {
+      GST_DEBUG ("Sync: frame size %d", length);
 
+      if (flags != dts->prev_flags)
+        dts->flag_update = TRUE;
+      dts->prev_flags = flags;
+
+      result = gst_dtsdec_handle_frame (dts, data, length,
+          flags, sample_rate, bit_rate);
+      if (result != GST_FLOW_OK) {
+        size = 0;
+        break;
+      }
+      size -= length;
+      data += length;
+    } else {
+      GST_LOG ("Not enough data available (needed %d had %d)", length, size);
+      break;
+    }
+  }
+
+  /* keep cache */
+  if (length == 0) {
+    GST_LOG ("No sync found");
+  }
+
+  if (size > 0) {
+    dts->cache = gst_buffer_create_sub (buf,
+        GST_BUFFER_SIZE (buf) - size, size);
+  }
+
+  gst_buffer_unref (buf);
+
+  return result;
 }
 
 static GstStateChangeReturn
@@ -705,13 +870,14 @@ gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
       dts->samples = dca_samples (dts->state);
       dts->bit_rate = -1;
       dts->sample_rate = -1;
-      dts->stream_channels = 0;
-      /* FIXME force stereo for now */
-      dts->request_channels = DCA_CHANNEL;
-      dts->using_channels = 0;
+      dts->stream_channels = DCA_CHANNEL;
+      dts->using_channels = DCA_CHANNEL;
       dts->level = 1;
       dts->bias = 0;
-      dts->current_ts = 0;
+      dts->time = 0;
+      dts->sent_segment = FALSE;
+      dts->flag_update = TRUE;
+      gst_segment_init (&dts->segment, GST_FORMAT_UNDEFINED);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
@@ -730,6 +896,7 @@ gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
         gst_buffer_unref (dts->cache);
         dts->cache = NULL;
       }
+      clear_queued (dts);
       break;
     case GST_STATE_CHANGE_READY_TO_NULL:
       dca_free (dts->state);
diff --git a/ext/dts/gstdtsdec.h b/ext/dts/gstdtsdec.h
index 5222c687a..a7c8f7180 100644
--- a/ext/dts/gstdtsdec.h
+++ b/ext/dts/gstdtsdec.h
@@ -43,15 +43,22 @@ struct _GstDtsDec {
   GstElement 	 element;
 
   /* pads */
-  GstPad 	*sinkpad;
-  GstPad 	*srcpad;
+  GstPad        *sinkpad;
+  GstPad        *srcpad;
+  GstSegment     segment;
+
+  gboolean       dvdmode;
+  gboolean       sent_segment;
+  gboolean       discont;
+  gboolean       flag_update;
+  gboolean       prev_flags;
 
   /* stream properties */
-  gint 		 bit_rate;
-  gint 		 sample_rate;
-  gint 		 stream_channels;
-  gint 		 request_channels;
-  gint 		 using_channels;
+  gint 	         bit_rate;
+  gint 	         sample_rate;
+  gint 	         stream_channels;
+  gint 	         request_channels;
+  gint 	         using_channels;
 
   /* decoding properties */
   sample_t 	 level;
@@ -63,13 +70,14 @@ struct _GstDtsDec {
 #else
   dts_state_t 	*state;
 #endif
-  gboolean       dvdmode;
+
 
   /* Data left over from the previous buffer */
-  GstBuffer	*cache;
-  
-  /* keep track of time */
-  GstClockTime	current_ts;
+  GstBuffer		*cache;
+  GstClockTime	time;
+
+  /* reverse playback */
+  GList *queued;
 };
 
 struct _GstDtsDecClass {