New libfaad (MPEG-2/4 AAC decoder) plugin

Original commit message from CVS:
New libfaad (MPEG-2/4 AAC decoder) plugin

1 files changed, 473 insertions, 0 deletions

diff --git a/ext/faad/gstfaad.c b/ext/faad/gstfaad.c
new file mode 100644
index 00000000..fcb59ff5
--- /dev/null
+++ b/ext/faad/gstfaad.c
@@ -0,0 +1,473 @@
+/* GStreamer FAAD (Free AAC Decoder) plugin
+ * Copyright (C) 2003 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 "gstfaad.h"
+
+GST_PAD_TEMPLATE_FACTORY (sink_template,
+  "sink",
+  GST_PAD_SINK,
+  GST_PAD_ALWAYS,
+  GST_CAPS_NEW (
+    "faad_mpeg_templ",
+    "audio/mpeg",
+      "systemstream", GST_PROPS_BOOLEAN (FALSE),
+      "mpegversion",  GST_PROPS_LIST (
+                        GST_PROPS_INT (2),
+                        GST_PROPS_INT (4)
+                      )
+  )
+);
+
+GST_PAD_TEMPLATE_FACTORY (src_template,
+  "src",
+  GST_PAD_SRC,
+  GST_PAD_ALWAYS,
+  GST_CAPS_NEW (
+    "faad_int_templ",
+    "audio/x-raw-int",
+      "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+      "signed",     GST_PROPS_BOOLEAN (TRUE),
+      "width",      GST_PROPS_LIST (
+		      GST_PROPS_INT (16),
+		      GST_PROPS_INT (24),
+		      GST_PROPS_INT (32)
+		    ),
+      "depth",      GST_PROPS_LIST (
+		      GST_PROPS_INT (16),
+		      GST_PROPS_INT (24),
+		      GST_PROPS_INT (32)
+		    ),
+      "rate",       GST_PROPS_INT_RANGE (8000, 96000),
+      "channels",   GST_PROPS_INT_RANGE (1, 6)
+  ),
+  GST_CAPS_NEW (
+    "faad_float_templ",
+    "audio/x-raw-float",
+      "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+      "depth",      GST_PROPS_LIST (
+		      GST_PROPS_INT (32), /* float  */
+		      GST_PROPS_INT (64)  /* double */
+		    ),
+      "rate",       GST_PROPS_INT_RANGE (8000, 96000),
+      "channels",   GST_PROPS_INT_RANGE (1, 6)
+  )
+);
+
+static void     gst_faad_base_init    (GstFaadClass *klass);
+static void     gst_faad_class_init   (GstFaadClass *klass);
+static void     gst_faad_init         (GstFaad      *faad);
+
+static GstPadLinkReturn
+                gst_faad_sinkconnect  (GstPad       *pad,
+				       GstCaps      *caps);
+static GstPadLinkReturn
+                gst_faad_srcconnect   (GstPad       *pad,
+				       GstCaps      *caps);
+static GstCaps *gst_faad_srcgetcaps   (GstPad       *pad,
+				       GstCaps      *caps);
+static void     gst_faad_chain        (GstPad       *pad,
+				       GstData      *data);
+static GstElementStateReturn
+                gst_faad_change_state (GstElement   *element);
+
+static GstElementClass *parent_class = NULL;
+/* static guint gst_lame_signals[LAST_SIGNAL] = { 0 }; */
+
+GType
+gst_faad_get_type (void)
+{
+  static GType gst_faad_type = 0;
+
+  if (!gst_faad_type) {
+    static const GTypeInfo gst_faad_info = {
+      sizeof (GstFaadClass),      
+      (GBaseInitFunc) gst_faad_base_init,
+      NULL,
+      (GClassInitFunc) gst_faad_class_init,
+      NULL,
+      NULL,
+      sizeof(GstFaad),
+      0,
+      (GInstanceInitFunc) gst_faad_init,
+    };
+
+    gst_faad_type = g_type_register_static (GST_TYPE_ELEMENT,
+					    "GstFaad",
+					    &gst_faad_info, 0);
+  }
+
+  return gst_faad_type;
+}
+
+static void
+gst_faad_base_init (GstFaadClass *klass)
+{
+  GstElementDetails gst_faad_details = {
+    "Free AAC Decoder (FAAD)",
+    "Codec/Audio/Decoder",
+    "Free MPEG-2/4 AAC decoder",
+    "Ronald Bultje <rbultje@ronald.bitfreak.net>",
+  };
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+	GST_PAD_TEMPLATE_GET (src_template));
+  gst_element_class_add_pad_template (element_class,
+	GST_PAD_TEMPLATE_GET (sink_template));
+
+  gst_element_class_set_details (element_class, &gst_faad_details);
+}
+
+static void
+gst_faad_class_init (GstFaadClass *klass)
+{
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+  gstelement_class->change_state = gst_faad_change_state;
+}
+
+static void
+gst_faad_init (GstFaad *faad)
+{
+  faad->handle = NULL;
+  faad->samplerate = -1;
+  faad->channels = -1;
+
+  faad->sinkpad = gst_pad_new_from_template (
+	GST_PAD_TEMPLATE_GET (sink_template), "sink");
+  gst_element_add_pad (GST_ELEMENT (faad), faad->sinkpad);
+  gst_pad_set_chain_function (faad->sinkpad, gst_faad_chain);
+  gst_pad_set_link_function (faad->sinkpad, gst_faad_sinkconnect);
+
+  faad->srcpad = gst_pad_new_from_template (
+	GST_PAD_TEMPLATE_GET (src_template), "src");
+  gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad);
+  gst_pad_set_link_function (faad->srcpad, gst_faad_srcconnect);
+  gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);
+}
+
+static GstPadLinkReturn
+gst_faad_sinkconnect (GstPad  *pad,
+		      GstCaps *caps)
+{
+  if (!GST_CAPS_IS_FIXED (caps))
+    return GST_PAD_LINK_DELAYED;
+
+  /* oh, we really don't care what's in here. We'll
+   * get AAC audio (MPEG-2/4) anyway, so why bother? */
+  return GST_PAD_LINK_OK;
+}
+
+static GstCaps *
+gst_faad_srcgetcaps (GstPad  *pad,
+		     GstCaps *caps)
+{
+  GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
+
+  if (faad->handle != NULL &&
+      faad->channels != -1 && faad->samplerate != -1) {
+    faacDecConfiguration *conf;
+    GstCaps *caps;
+
+    conf = faacDecGetCurrentConfiguration (faad->handle);
+
+    switch (conf->outputFormat) {
+      case FAAD_FMT_16BIT:
+        caps = GST_CAPS_NEW ("faad_src_int16",
+			     "audio/x-raw-int",
+			       "signed", GST_PROPS_BOOLEAN (TRUE),
+			       "width",  GST_PROPS_INT (16),
+			       "depth",  GST_PROPS_INT (16));
+        break;
+      case FAAD_FMT_24BIT:
+        caps = GST_CAPS_NEW ("faad_src_int24",
+			     "audio/x-raw-int",
+			       "signed", GST_PROPS_BOOLEAN (TRUE),
+			       "width",  GST_PROPS_INT (24),
+			       "depth",  GST_PROPS_INT (24));
+        break;
+      case FAAD_FMT_32BIT:
+        caps = GST_CAPS_NEW ("faad_src_int32",
+			     "audio/x-raw-int",
+			       "signed", GST_PROPS_BOOLEAN (TRUE),
+			       "width",  GST_PROPS_INT (32),
+			       "depth",  GST_PROPS_INT (32));
+        break;
+      case FAAD_FMT_FLOAT:
+        caps = GST_CAPS_NEW ("faad_src_float32",
+			     "audio/x-raw-float",
+			       "depth",  GST_PROPS_INT (32));
+        break;
+      case FAAD_FMT_DOUBLE:
+        caps = GST_CAPS_NEW ("faad_src_float64",
+			     "audio/x-raw-float",
+			       "depth",  GST_PROPS_INT (64));
+        break;
+      default:
+        caps = GST_CAPS_NONE;
+        break;
+    }
+
+    if (caps) {
+      GstPropsEntry *samplerate, *channels, *endianness;
+
+      if (faad->samplerate != -1) {
+        samplerate = gst_props_entry_new ("rate",
+		GST_PROPS_INT (faad->samplerate));
+      } else {
+        samplerate = gst_props_entry_new ("rate",
+		GST_PROPS_INT_RANGE (8000, 96000));
+      }
+      gst_props_add_entry (caps->properties, samplerate);
+
+      if (faad->channels != -1) {
+        channels = gst_props_entry_new ("channels",
+		GST_PROPS_INT (faad->channels));
+      } else {
+        channels = gst_props_entry_new ("channels",
+		GST_PROPS_INT_RANGE (1, 6));
+      }
+      gst_props_add_entry (caps->properties, channels);
+
+      endianness = gst_props_entry_new ("endianness",
+		GST_PROPS_INT (G_BYTE_ORDER));
+      gst_props_add_entry (caps->properties, endianness);
+    }
+
+    return caps;
+  }
+
+  return gst_pad_template_get_caps (
+	GST_PAD_TEMPLATE_GET (src_template));
+}
+
+static GstPadLinkReturn
+gst_faad_srcconnect (GstPad  *pad,
+		     GstCaps *caps)
+{
+  GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
+  GstCaps *t;
+
+  if (!faad->handle ||
+      (faad->samplerate == -1 || faad->channels == -1)) {
+    return GST_PAD_LINK_DELAYED;
+  }
+
+  /* we do samplerate/channels ourselves */
+  for (t = caps; t != NULL; t = t->next) {
+    gst_props_remove_entry_by_name (t->properties, "rate");
+    gst_props_remove_entry_by_name (t->properties, "channels");
+  }
+
+  /* go through list */
+  caps = gst_caps_normalize (caps);
+  for ( ; caps != NULL; caps = caps->next) {
+    const gchar *mimetype = gst_caps_get_mime (caps);
+    gint depth = 0, fmt = 0;
+
+    if (!strcmp (mimetype, "audio/x-raw-int")) {
+      gint width = 0;
+
+      if (gst_caps_has_fixed_property (caps, "depth") &&
+          gst_caps_has_fixed_property (caps, "width"))
+        gst_caps_get (caps, "depth", &depth,
+			    "width", &width, NULL);
+      if (depth != width)
+        continue;
+
+      switch (depth) {
+        case 16:
+          fmt = FAAD_FMT_16BIT;
+          break;
+        case 24:
+          fmt = FAAD_FMT_24BIT;
+          break;
+        case 32:
+          fmt = FAAD_FMT_32BIT;
+          break;
+      }
+    } else {
+      if (gst_caps_has_fixed_property (caps, "depth"))
+        gst_caps_get_int (caps, "depth", &depth);
+
+      switch (depth) {
+        case 32:
+          fmt = FAAD_FMT_FLOAT;
+          break;
+        case 64:
+          fmt = FAAD_FMT_DOUBLE;
+          break;
+      }
+    }
+
+    if (fmt) {
+      GstCaps *newcaps;
+      faacDecConfiguration *conf;
+
+      conf = faacDecGetCurrentConfiguration (faad->handle);
+      conf->outputFormat = fmt;
+      conf->defObjectType = LC;
+      conf->defSampleRate = faad->samplerate;
+      faacDecSetConfiguration (faad->handle, conf);
+      /* FIXME: handle return value, how? */
+
+      newcaps = gst_faad_srcgetcaps (pad, NULL);
+      g_assert (GST_CAPS_IS_FIXED (newcaps));
+      if (gst_pad_try_set_caps (pad, newcaps) > 0) {
+        faad->bps = depth * faad->channels / 8;
+        return GST_PAD_LINK_OK;
+      }
+    }
+  }
+
+  return GST_PAD_LINK_REFUSED;
+}
+
+static void
+gst_faad_chain (GstPad  *pad,
+		GstData *data)
+{
+  GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
+  GstBuffer *buf, *outbuf;
+  faacDecFrameInfo info;
+  void *out;
+
+  if (GST_IS_EVENT (data)) {
+    GstEvent *event = GST_EVENT (data);
+
+    switch (GST_EVENT_TYPE (event)) {
+      case GST_EVENT_EOS:
+        gst_element_set_eos (GST_ELEMENT (faad));
+        gst_pad_push (faad->srcpad, data);
+        return;
+      default:
+	gst_pad_event_default (pad, event);
+        return;
+    }
+  }
+
+  buf = GST_BUFFER (data);
+
+  if (faad->samplerate == -1 || faad->channels == -1) {
+    gulong samplerate;
+    guchar channels;
+
+    faacDecInit (faad->handle,
+		 GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
+		 &samplerate, &channels);
+    faad->samplerate = samplerate;
+    faad->channels = channels;
+    if (gst_faad_srcconnect (faad->srcpad,
+			     gst_pad_get_allowed_caps (faad->srcpad)) <= 0) {
+      gst_element_error (GST_ELEMENT (faad),
+			 "Failed to negotiate output format with next element");
+      gst_buffer_unref (buf);
+      return;
+    }
+  }
+
+  out = faacDecDecode (faad->handle, &info,
+		       GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+  if (info.error) {
+    gst_element_error (GST_ELEMENT (faad),
+		       "Failed to decode buffer: %s",
+		       faacDecGetErrorMessage (info.error));
+    gst_buffer_unref (buf);
+    return;
+  }
+  if (info.samplerate != faad->samplerate ||
+      info.channels != faad->channels) {
+    faad->samplerate = info.samplerate;
+    faad->channels = info.channels;
+    if (gst_faad_srcconnect (faad->srcpad,
+			     gst_pad_get_allowed_caps (faad->srcpad)) <= 0) {
+      gst_element_error (GST_ELEMENT (faad),
+			 "Failed to negotiate format with next element");
+      gst_buffer_unref (buf);
+      return;
+    }
+  }
+
+  /* FIXME: did it handle the whole buffer? */
+  outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
+  /* ugh */
+  memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
+  GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
+  GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
+
+  gst_buffer_unref (buf);
+  gst_pad_push (faad->srcpad, GST_DATA (outbuf));
+}
+
+static GstElementStateReturn
+gst_faad_change_state (GstElement *element)
+{
+  GstFaad *faad = GST_FAAD (element);
+
+  switch (GST_STATE_TRANSITION (element)) {
+    case GST_STATE_NULL_TO_READY:
+      if (!(faad->handle = faacDecOpen ()))
+        return GST_STATE_FAILURE;
+      break;
+    case GST_STATE_PAUSED_TO_READY:
+      faad->samplerate = -1;
+      faad->channels = -1;
+      break;
+    case GST_STATE_READY_TO_NULL:
+      faacDecClose (faad->handle);
+      faad->handle = 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 gboolean
+plugin_init (GstPlugin *plugin)
+{
+  return gst_element_register (plugin, "faad",
+			       GST_RANK_PRIMARY,
+			       GST_TYPE_FAAD);
+}
+
+GST_PLUGIN_DEFINE (
+  GST_VERSION_MAJOR,
+  GST_VERSION_MINOR,
+  "faad",
+  "Free AAC Decoder (FAAD)",
+  plugin_init,
+  VERSION,
+  "GPL",
+  GST_COPYRIGHT,
+  GST_PACKAGE,
+  GST_ORIGIN
+)