a52dec: port to audiodecoder

ext/a52dec/Makefile.am

@@ -3,11 +3,14 @@ plugin_LTLIBRARIES = libgsta52dec.la
 libgsta52dec_la_SOURCES = gsta52dec.c
 libgsta52dec_la_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) \
+        $(GST_BASE_CFLAGS) \
         $(GST_CFLAGS) \
 	$(ORC_CFLAGS) \
 	$(A52DEC_CFLAGS)
 libgsta52dec_la_LIBADD = \
 	$(GST_PLUGINS_BASE_LIBS) \
+        $(GST_BASE_LIBS) \
+        $(GST_LIBS) \
         -lgstaudio-$(GST_MAJORMINOR) \
 	$(ORC_LIBS) \
 	$(A52DEC_LIBS)

ext/a52dec/gsta52dec.c

@@ -86,14 +86,20 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
         "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
     );
 
-GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstElement, GST_TYPE_ELEMENT);
+GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstAudioDecoder,
+    GST_TYPE_AUDIO_DECODER);
+
+static gboolean gst_a52dec_start (GstAudioDecoder * dec);
+static gboolean gst_a52dec_stop (GstAudioDecoder * dec);
+static gboolean gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
+static gboolean gst_a52dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
+    gint * offset, gint * length);
+static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * dec,
+    GstBuffer * buffer);
+static GstFlowReturn gst_a52dec_pre_push (GstAudioDecoder * bdec,
+    GstBuffer ** buffer);
 
 static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buffer);
-static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf);
-static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps);
-static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event);
-static GstStateChangeReturn gst_a52dec_change_state (GstElement * element,
-    GstStateChange transition);
 
 static void gst_a52dec_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -143,16 +149,21 @@ static void
 gst_a52dec_class_init (GstA52DecClass * klass)
 {
   GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
+  GstAudioDecoderClass *gstbase_class;
   guint cpuflags;
 
   gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
+  gstbase_class = (GstAudioDecoderClass *) klass;
 
   gobject_class->set_property = gst_a52dec_set_property;
   gobject_class->get_property = gst_a52dec_get_property;
 
-  gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state);
+  gstbase_class->start = GST_DEBUG_FUNCPTR (gst_a52dec_start);
+  gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_a52dec_stop);
+  gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_a52dec_set_format);
+  gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_a52dec_parse);
+  gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_a52dec_handle_frame);
+  gstbase_class->pre_push = GST_DEBUG_FUNCPTR (gst_a52dec_pre_push);
 
   /**
    * GstA52Dec::drc
@@ -213,27 +224,106 @@ gst_a52dec_class_init (GstA52DecClass * klass)
 static void
 gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class)
 {
-  /* create the sink and src pads */
-  a52dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
-  gst_pad_set_setcaps_function (a52dec->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_a52dec_sink_setcaps));
-  gst_pad_set_chain_function (a52dec->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_a52dec_chain));
-  gst_pad_set_event_function (a52dec->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_a52dec_sink_event));
-  gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
-
-  a52dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
-  gst_pad_use_fixed_caps (a52dec->srcpad);
-  gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
-
   a52dec->request_channels = A52_CHANNEL;
   a52dec->dynamic_range_compression = FALSE;
 
   a52dec->state = NULL;
   a52dec->samples = NULL;
 
-  gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
+  /* retrieve and intercept base class chain.
+   * Quite HACKish, but that's dvd specs/caps for you,
+   * since one buffer needs to be split into 2 frames */
+  a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
+  gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
+      GST_DEBUG_FUNCPTR (gst_a52dec_chain));
+}
+
+static gboolean
+gst_a52dec_start (GstAudioDecoder * dec)
+{
+  GstA52Dec *a52dec = GST_A52DEC (dec);
+  GstA52DecClass *klass;
+
+  GST_DEBUG_OBJECT (dec, "start");
+
+  klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
+  a52dec->state = a52_init (klass->a52_cpuflags);
+  a52dec->samples = a52_samples (a52dec->state);
+  a52dec->bit_rate = -1;
+  a52dec->sample_rate = -1;
+  a52dec->stream_channels = A52_CHANNEL;
+  a52dec->using_channels = A52_CHANNEL;
+  a52dec->level = 1;
+  a52dec->bias = 0;
+  a52dec->flag_update = TRUE;
+
+  /* call upon legacy upstream byte support (e.g. seeking) */
+  gst_audio_decoder_set_byte_time (dec, TRUE);
+
+  return TRUE;
+}
+
+static gboolean
+gst_a52dec_stop (GstAudioDecoder * dec)
+{
+  GstA52Dec *a52dec = GST_A52DEC (dec);
+
+  GST_DEBUG_OBJECT (dec, "stop");
+
+  a52dec->samples = NULL;
+  if (a52dec->state) {
+    a52_free (a52dec->state);
+    a52dec->state = NULL;
+  }
+  if (a52dec->pending_tags) {
+    gst_tag_list_free (a52dec->pending_tags);
+    a52dec->pending_tags = NULL;
+  }
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
+    gint * _offset, gint * len)
+{
+  GstA52Dec *a52dec;
+  guint8 *data;
+  gint av, size;
+  gint length = 0, flags, sample_rate, bit_rate;
+  GstFlowReturn result = GST_FLOW_UNEXPECTED;
+
+  a52dec = GST_A52DEC (bdec);
+
+  size = av = gst_adapter_available (adapter);
+  data = (guint8 *) gst_adapter_peek (adapter, av);
+
+  /* find and read header */
+  bit_rate = a52dec->bit_rate;
+  sample_rate = a52dec->sample_rate;
+  flags = 0;
+  while (av >= 7) {
+    length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
+
+    if (length == 0) {
+      /* shift window to re-find sync */
+      data++;
+      size--;
+    } else if (length <= size) {
+      GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
+      result = GST_FLOW_OK;
+      break;
+    } else {
+      GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
+          length, size);
+      break;
+    }
+  }
+
+  *_offset = av - size;
+  *len = length;
+
+  return result;
 }
 
 static gint
@@ -324,106 +414,6 @@ gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos)
   return chans;
 }
 
-static void
-clear_queued (GstA52Dec * dec)
-{
-  g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
-  g_list_free (dec->queued);
-  dec->queued = NULL;
-}
-
-static GstFlowReturn
-flush_queued (GstA52Dec * 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_a52dec_drain (GstA52Dec * 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_a52dec_push (GstA52Dec * a52dec,
-    GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
-{
-  GstBuffer *buf;
-  int chans, n, c;
-  GstFlowReturn result;
-
-  flags &= (A52_CHANNEL_MASK | A52_LFE);
-  chans = gst_a52dec_channels (flags, NULL);
-  if (!chans) {
-    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), 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 / a52dec->sample_rate;
-
-  result = GST_FLOW_OK;
-  if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment,
-              a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
-    /* set discont when needed */
-    if (a52dec->discont) {
-      GST_LOG_OBJECT (a52dec, "marking DISCONT");
-      GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
-      a52dec->discont = FALSE;
-    }
-
-    if (a52dec->segment.rate > 0.0) {
-      GST_DEBUG_OBJECT (a52dec,
-          "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 (a52dec, "queued frame");
-      a52dec->queued = g_list_prepend (a52dec->queued, buf);
-    }
-  }
-  return result;
-}
-
 static gboolean
 gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
 {
@@ -457,102 +447,69 @@ done:
   return result;
 }
 
-static gboolean
-gst_a52dec_sink_event (GstPad * pad, GstEvent * event)
-{
-  GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
-  gboolean ret = FALSE;
-
-  GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_NEWSEGMENT:
-    {
-      GstFormat fmt;
-      gboolean update;
-      gint64 start, end, pos;
-      gdouble rate, arate;
-
-      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &fmt,
-          &start, &end, &pos);
-
-      /* drain queued buffers before activating the segment so that we can clip
-       * against the old segment first */
-      gst_a52dec_drain (a52dec);
-
-      if (fmt != 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 (fmt));
-        gst_event_unref (event);
-        a52dec->sent_segment = FALSE;
-        /* set some dummy values, FIXME: do proper conversion */
-        a52dec->time = start = pos = 0;
-        fmt = GST_FORMAT_TIME;
-        end = -1;
-      } else {
-        a52dec->time = start;
-        a52dec->sent_segment = TRUE;
-        GST_DEBUG_OBJECT (a52dec,
-            "Pushing newseg rate %g, applied rate %g, format %d, start %"
-            G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT ", pos %"
-            G_GINT64_FORMAT, rate, arate, fmt, start, end, pos);
-
-        ret = gst_pad_push_event (a52dec->srcpad, event);
-      }
-
-      gst_segment_set_newsegment (&a52dec->segment, update, rate, fmt, start,
-          end, pos);
-      break;
-    }
-    case GST_EVENT_TAG:
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    case GST_EVENT_EOS:
-      gst_a52dec_drain (a52dec);
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    case GST_EVENT_FLUSH_START:
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    case GST_EVENT_FLUSH_STOP:
-      if (a52dec->cache) {
-        gst_buffer_unref (a52dec->cache);
-        a52dec->cache = NULL;
-      }
-      clear_queued (a52dec);
-      gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    default:
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-  }
-
-  gst_object_unref (a52dec);
-  return ret;
-}
-
 static void
 gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
 {
   GstTagList *taglist;
 
   taglist = gst_tag_list_new ();
+  gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
+      (guint) a52dec->bit_rate, NULL);
 
-  gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
+  if (a52dec->pending_tags) {
-      GST_TAG_AUDIO_CODEC, "Dolby Digital (AC-3)",
+    gst_tag_list_free (a52dec->pending_tags);
-      GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
+    a52dec->pending_tags = NULL;
+  }
 
-  gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
+  a52dec->pending_tags = taglist;
-      GST_PAD (a52dec->srcpad), taglist);
 }
 
 static GstFlowReturn
-gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
+gst_a52dec_pre_push (GstAudioDecoder * bdec, GstBuffer ** buffer)
-    guint length, gint flags, gint sample_rate, gint bit_rate)
 {
+  GstA52Dec *a52dec = GST_A52DEC (bdec);
+
+  if (G_UNLIKELY (a52dec->pending_tags)) {
+    gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
+        GST_AUDIO_DECODER_SRC_PAD (a52dec), a52dec->pending_tags);
+    a52dec->pending_tags = NULL;
+  }
+
+  return GST_FLOW_OK;
+}
+
+static GstFlowReturn
+gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
+{
+  GstA52Dec *a52dec;
   gint channels, i;
   gboolean need_reneg = FALSE;
+  gint size, chans;
+  gint length = 0, flags, sample_rate, bit_rate;
+  guint8 *data;
+  GstFlowReturn result = GST_FLOW_OK;
+  GstBuffer *outbuf;
+  const gint num_blocks = 6;
+
+  a52dec = GST_A52DEC (bdec);
+
+  /* no fancy draining */
+  if (G_UNLIKELY (!buffer))
+    return GST_FLOW_OK;
+
+  /* parsed stuff already, so this should work out fine */
+  data = GST_BUFFER_DATA (buffer);
+  size = GST_BUFFER_SIZE (buffer);
+  g_assert (size >= 7);
+
+  /* re-obtain some sync header info,
+   * should be same as during _parse and could also be cached there,
+   * but anyway ... */
+  bit_rate = a52dec->bit_rate;
+  sample_rate = a52dec->sample_rate;
+  flags = 0;
+  length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
+  g_assert (length == size);
 
   /* update stream information, renegotiate or re-streaminfo if needed */
   need_reneg = FALSE;
@@ -582,7 +539,7 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
 
     a52dec->flag_update = FALSE;
 
-    caps = gst_pad_get_allowed_caps (a52dec->srcpad);
+    caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
     if (caps && gst_caps_get_size (caps) > 0) {
       GstCaps *copy = gst_caps_copy_nth (caps, 0);
       GstStructure *structure = gst_caps_get_structure (copy, 0);
@@ -618,14 +575,16 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
   } else {
     flags = a52dec->using_channels;
   }
+
   /* process */
   flags |= A52_ADJUST_LEVEL;
   a52dec->level = 1;
   if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
-    GST_WARNING ("a52_frame error");
+    GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
-    a52dec->discont = TRUE;
+        ("a52_frame error"), result);
-    return GST_FLOW_OK;
+    goto exit;
   }
+
   channels = flags & (A52_CHANNEL_MASK | A52_LFE);
   if (a52dec->using_channels != channels) {
     need_reneg = TRUE;
@@ -634,43 +593,74 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
 
   /* negotiate if required */
   if (need_reneg) {
-    GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
+    GST_DEBUG_OBJECT (a52dec,
+        "a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
         a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
-    if (!gst_a52dec_reneg (a52dec, a52dec->srcpad)) {
+    if (!gst_a52dec_reneg (a52dec, GST_AUDIO_DECODER_SRC_PAD (a52dec)))
-      GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
+      goto failed_negotiation;
-      return GST_FLOW_ERROR;
-    }
   }
 
   if (a52dec->dynamic_range_compression == FALSE) {
     a52_dynrng (a52dec->state, NULL, NULL);
   }
 
-  /* each frame consists of 6 blocks */
+  flags &= (A52_CHANNEL_MASK | A52_LFE);
-  for (i = 0; i < 6; i++) {
+  chans = gst_a52dec_channels (flags, NULL);
+  if (!chans)
+    goto invalid_flags;
+
+  /* handle decoded data;
+   * each frame has 6 blocks, one block is 256 samples, ea */
+  result =
+      gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec), 0,
+      256 * chans * (SAMPLE_WIDTH / 8) * num_blocks,
+      GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (a52dec)), &outbuf);
+  if (result != GST_FLOW_OK)
+    goto exit;
+
+  data = GST_BUFFER_DATA (outbuf);
+  for (i = 0; i < num_blocks; i++) {
     if (a52_block (a52dec->state)) {
-      /* ignore errors but mark a discont */
+      /* also marks discont */
-      GST_WARNING ("a52_block error %d", i);
+      GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
-      a52dec->discont = TRUE;
+          ("error decoding block %d", i), result);
+      if (result != GST_FLOW_OK)
+        goto exit;
     } else {
-      GstFlowReturn ret;
+      gint n, c;
 
-      /* push on */
+      for (n = 0; n < 256; n++) {
-      ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
+        for (c = 0; c < chans; c++) {
-          a52dec->samples, a52dec->time);
+          ((sample_t *) data)[n * chans + c] = a52dec->samples[c * 256 + n];
-      if (ret != GST_FLOW_OK)
-        return ret;
         }
-    a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
+      }
+    }
+    data += 256 * chans * (SAMPLE_WIDTH / 8);
   }
 
-  return GST_FLOW_OK;
+  result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
+
+exit:
+  return result;
+
+  /* ERRORS */
+failed_negotiation:
+  {
+    GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
+    return GST_FLOW_ERROR;
+  }
+invalid_flags:
+  {
+    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
+        ("Invalid channel flags: %d", flags));
+    return GST_FLOW_ERROR;
+  }
 }
 
 static gboolean
-gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
 {
-  GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
+  GstA52Dec *a52dec = GST_A52DEC (bdec);
   GstStructure *structure;
 
   structure = gst_caps_get_structure (caps, 0);
@@ -680,8 +670,6 @@ gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
   else
     a52dec->dvdmode = FALSE;
 
-  gst_object_unref (a52dec);
-
   return TRUE;
 }
 
@@ -689,20 +677,9 @@ static GstFlowReturn
 gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
 {
   GstA52Dec *a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
-  GstFlowReturn ret;
+  GstFlowReturn ret = GST_FLOW_OK;
   gint first_access;
 
-  if (GST_BUFFER_IS_DISCONT (buf)) {
-    GST_LOG_OBJECT (a52dec, "received DISCONT");
-    gst_a52dec_drain (a52dec);
-    /* clear cache on discont and mark a discont in the element */
-    if (a52dec->cache) {
-      gst_buffer_unref (a52dec->cache);
-      a52dec->cache = NULL;
-    }
-    a52dec->discont = TRUE;
-  }
-
   if (a52dec->dvdmode) {
     gint size = GST_BUFFER_SIZE (buf);
     guchar *data = GST_BUFFER_DATA (buf);
@@ -726,33 +703,36 @@ gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
         goto bad_first_access_parameter;
 
       subbuf = gst_buffer_create_sub (buf, offset, len);
+      gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
       GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
-      ret = gst_a52dec_chain_raw (pad, subbuf);
+      ret = a52dec->base_chain (pad, subbuf);
-      if (ret != GST_FLOW_OK)
+      if (ret != GST_FLOW_OK) {
+        gst_buffer_unref (buf);
         goto done;
+      }
 
       offset += len;
       len = size - offset;
 
       if (len > 0) {
         subbuf = gst_buffer_create_sub (buf, offset, len);
+        gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
         GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
 
-        ret = gst_a52dec_chain_raw (pad, subbuf);
+        ret = a52dec->base_chain (pad, subbuf);
       }
+      gst_buffer_unref (buf);
     } else {
       /* 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);
+      gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
-      ret = gst_a52dec_chain_raw (pad, subbuf);
+      ret = a52dec->base_chain (pad, subbuf);
     }
   } else {
-    gst_buffer_ref (buf);
+    ret = a52dec->base_chain (pad, buf);
-    ret = gst_a52dec_chain_raw (pad, buf);
   }
 
 done:
-  gst_buffer_unref (buf);
   return ret;
 
 /* ERRORS */
@@ -772,162 +752,6 @@ bad_first_access_parameter:
   }
 }
 
-static GstFlowReturn
-gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
-{
-  GstA52Dec *a52dec;
-  guint8 *data;
-  guint size;
-  gint length = 0, flags, sample_rate, bit_rate;
-  GstFlowReturn result = GST_FLOW_OK;
-
-  a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
-
-  if (!a52dec->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 (a52dec->srcpad, gst_event_new_new_segment (FALSE,
-            segment.rate, segment.format, segment.start,
-            segment.duration, segment.start));
-    a52dec->sent_segment = TRUE;
-  }
-
-  /* merge with cache, if any. Also make sure timestamps match */
-  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
-    a52dec->time = GST_BUFFER_TIMESTAMP (buf);
-    GST_DEBUG_OBJECT (a52dec,
-        "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 (a52dec->cache) {
-    buf = gst_buffer_join (a52dec->cache, buf);
-    a52dec->cache = NULL;
-  }
-  data = GST_BUFFER_DATA (buf);
-  size = GST_BUFFER_SIZE (buf);
-
-  /* find and read header */
-  bit_rate = a52dec->bit_rate;
-  sample_rate = a52dec->sample_rate;
-  flags = 0;
-  while (size >= 7) {
-    length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
-
-    if (length == 0) {
-      /* no sync */
-      data++;
-      size--;
-    } else if (length <= size) {
-      GST_DEBUG ("Sync: %d", length);
-
-      if (flags != a52dec->prev_flags)
-        a52dec->flag_update = TRUE;
-      a52dec->prev_flags = flags;
-
-      result = gst_a52dec_handle_frame (a52dec, data,
-          length, flags, sample_rate, bit_rate);
-      if (result != GST_FLOW_OK) {
-        size = 0;
-        break;
-      }
-      size -= length;
-      data += length;
-    } else {
-      /* not enough data */
-      GST_LOG ("Not enough data available");
-      break;
-    }
-  }
-
-  /* keep cache */
-  if (length == 0) {
-    GST_LOG ("No sync found");
-  }
-
-  if (size > 0) {
-    a52dec->cache = gst_buffer_create_sub (buf,
-        GST_BUFFER_SIZE (buf) - size, size);
-  }
-
-  gst_buffer_unref (buf);
-
-  return result;
-}
-
-static GstStateChangeReturn
-gst_a52dec_change_state (GstElement * element, GstStateChange transition)
-{
-  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
-  GstA52Dec *a52dec = GST_A52DEC (element);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:{
-      GstA52DecClass *klass;
-
-      klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
-      a52dec->state = a52_init (klass->a52_cpuflags);
-
-      if (!a52dec->state) {
-        GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
-            ("Failed to initialize a52 state"));
-        ret = GST_STATE_CHANGE_FAILURE;
-      }
-      break;
-    }
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      a52dec->samples = a52_samples (a52dec->state);
-      a52dec->bit_rate = -1;
-      a52dec->sample_rate = -1;
-      a52dec->stream_channels = A52_CHANNEL;
-      a52dec->using_channels = A52_CHANNEL;
-      a52dec->level = 1;
-      a52dec->bias = 0;
-      a52dec->time = 0;
-      a52dec->sent_segment = FALSE;
-      a52dec->flag_update = TRUE;
-      gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      break;
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      a52dec->samples = NULL;
-      if (a52dec->cache) {
-        gst_buffer_unref (a52dec->cache);
-        a52dec->cache = NULL;
-      }
-      clear_queued (a52dec);
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      if (a52dec->state) {
-        a52_free (a52dec->state);
-        a52dec->state = NULL;
-      }
-      break;
-    default:
-      break;
-  }
-
-  return ret;
-}
-
 static void
 gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
     GParamSpec * pspec)

ext/a52dec/gsta52dec.h

@@ -22,6 +22,7 @@
 #define __GST_A52DEC_H__
 
 #include <gst/gst.h>
+#include <gst/audio/gstaudiodecoder.h>
 
 G_BEGIN_DECLS
 
@@ -40,41 +41,33 @@ typedef struct _GstA52Dec GstA52Dec;
 typedef struct _GstA52DecClass GstA52DecClass;
 
 struct _GstA52Dec {
-  GstElement     element;
+  GstAudioDecoder element;
 
-  /* pads */
+  GstPadChainFunction base_chain;
-  GstPad        *sinkpad,
-                *srcpad;
-  GstSegment     segment;
 
   gboolean       dvdmode;
-  gboolean       sent_segment;
-  gboolean       discont;
-
   gboolean       flag_update;
   int            prev_flags;
 
+  /* stream properties */
   int            bit_rate;
   int            sample_rate;
   int            stream_channels;
   int            request_channels;
   int            using_channels;
 
+  /* decoding properties */
   sample_t       level;
   sample_t       bias;
   gboolean       dynamic_range_compression;
   sample_t      *samples;
   a52_state_t   *state;
 
-  GstBuffer     *cache;
+  GstTagList    *pending_tags;
-  GstClockTime   time;
-
-  /* reverse */
-  GList         *queued;
 };
 
 struct _GstA52DecClass {
-  GstElementClass parent_class;
+  GstAudioDecoderClass parent_class;
 
   guint32 a52_cpuflags;
 };