diff --git a/omx/Makefile.am b/omx/Makefile.am
index 72a7a2666d..a67157ddd4 100644
--- a/omx/Makefile.am
+++ b/omx/Makefile.am
@@ -12,6 +12,7 @@ libgstopenmax_la_SOURCES = \
 	gstomxmpeg4videoenc.c \
 	gstomxh264enc.c \
 	gstomxh263enc.c \
+	gstomxaacenc.c \
 	gstbasevideocodec.c \
 	gstbasevideodecoder.c \
 	gstbasevideoencoder.c \
@@ -32,6 +33,7 @@ noinst_HEADERS = \
 	gstomxmpeg4videoenc.h \
 	gstomxh264enc.h \
 	gstomxh263enc.h \
+	gstomxaacenc.h \
 	gstbasevideocodec.h \
 	gstbasevideodecoder.h \
 	gstbasevideoencoder.h \
diff --git a/omx/gstomx.c b/omx/gstomx.c
index 05766f641b..772f3c48fc 100644
--- a/omx/gstomx.c
+++ b/omx/gstomx.c
@@ -33,6 +33,7 @@
 #include "gstomxmpeg4videoenc.h"
 #include "gstomxh264enc.h"
 #include "gstomxh263enc.h"
+#include "gstomxaacenc.h"
 
 GST_DEBUG_CATEGORY (gstomx_debug);
 #define GST_CAT_DEFAULT gstomx_debug
@@ -1772,7 +1773,7 @@ static GType (*types[]) (void) = {
 gst_omx_mpeg4_video_dec_get_type, gst_omx_h264_dec_get_type,
       gst_omx_h263_dec_get_type, gst_omx_wmv_dec_get_type,
       gst_omx_mpeg4_video_enc_get_type, gst_omx_h264_enc_get_type,
-      gst_omx_h263_enc_get_type};
+      gst_omx_h263_enc_get_type, gst_omx_aac_enc_get_type};
 
 static GKeyFile *config = NULL;
 GKeyFile *
diff --git a/omx/gstomx.conf b/omx/gstomx.conf
index 944fe170dc..78f8e7d23c 100644
--- a/omx/gstomx.conf
+++ b/omx/gstomx.conf
@@ -24,3 +24,12 @@ rank=0
 in-port-index=0
 out-port-index=1
 hacks=event-port-settings-changed-ndata-parameter-swap;video-framerate-integer;syncframe-flag-not-used
+
+[omxaacenc]
+type-name=GstOMXAACEnc
+core-name=/usr/local/lib/libomxil-bellagio.so.0
+component-name=OMX.st.audio_encoder.aac
+rank=0
+in-port-index=0
+out-port-index=1
+hacks=event-port-settings-changed-ndata-parameter-swap
diff --git a/omx/gstomxaacenc.c b/omx/gstomxaacenc.c
new file mode 100644
index 0000000000..703e77690a
--- /dev/null
+++ b/omx/gstomxaacenc.c
@@ -0,0 +1,432 @@
+/*
+ * Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
+ *   Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation
+ * version 2.1 of the License.
+ *
+ * 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+
+#include "gstomxaacenc.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_omx_aac_enc_debug_category);
+#define GST_CAT_DEFAULT gst_omx_aac_enc_debug_category
+
+/* prototypes */
+static void gst_omx_aac_enc_finalize (GObject * object);
+static void gst_omx_aac_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_omx_aac_enc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static gboolean gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc,
+    GstOMXPort * port, GstAudioState * state);
+static GstCaps *gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc,
+    GstOMXPort * port, GstAudioState * state);
+
+enum
+{
+  PROP_0,
+  PROP_BITRATE,
+  PROP_AAC_TOOLS,
+  PROP_AAC_ERROR_RESILIENCE_TOOLS
+};
+
+#define DEFAULT_BITRATE (128000)
+#define DEFAULT_AAC_TOOLS (OMX_AUDIO_AACToolMS | OMX_AUDIO_AACToolIS | OMX_AUDIO_AACToolTNS | OMX_AUDIO_AACToolPNS | OMX_AUDIO_AACToolLTP)
+#define DEFAULT_AAC_ER_TOOLS (OMX_AUDIO_AACERNone)
+
+#define GST_TYPE_OMX_AAC_TOOLS (gst_omx_aac_tools_get_type ())
+static GType
+gst_omx_aac_tools_get_type (void)
+{
+  static gsize id = 0;
+  static const GFlagsValue values[] = {
+    {OMX_AUDIO_AACToolMS, "Mid/side joint coding", "ms"},
+    {OMX_AUDIO_AACToolIS, "Intensity stereo", "is"},
+    {OMX_AUDIO_AACToolTNS, "Temporal noise shaping", "tns"},
+    {OMX_AUDIO_AACToolPNS, "Perceptual noise substitution", "pns"},
+    {OMX_AUDIO_AACToolLTP, "Long term prediction", "ltp"},
+    {0, NULL, NULL}
+  };
+
+  if (g_once_init_enter (&id)) {
+    GType tmp = g_flags_register_static ("GstOMXAACTools", values);
+    g_once_init_leave (&id, tmp);
+  }
+
+  return (GType) id;
+}
+
+#define GST_TYPE_OMX_AAC_ER_TOOLS (gst_omx_aac_er_tools_get_type ())
+static GType
+gst_omx_aac_er_tools_get_type (void)
+{
+  static gsize id = 0;
+  static const GFlagsValue values[] = {
+    {OMX_AUDIO_AACERVCB11, "Virtual code books", "vcb11"},
+    {OMX_AUDIO_AACERRVLC, "Reversible variable length coding", "rvlc"},
+    {OMX_AUDIO_AACERHCR, "Huffman codeword reordering", "hcr"},
+    {0, NULL, NULL}
+  };
+
+  if (g_once_init_enter (&id)) {
+    GType tmp = g_flags_register_static ("GstOMXAACERTools", values);
+    g_once_init_leave (&id, tmp);
+  }
+
+  return (GType) id;
+}
+
+/* class initialization */
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_omx_aac_enc_debug_category, "omxaacenc", 0, \
+      "debug category for gst-omx audio encoder base class");
+
+GST_BOILERPLATE_FULL (GstOMXAACEnc, gst_omx_aac_enc,
+    GstOMXAudioEnc, GST_TYPE_OMX_AUDIO_ENC, DEBUG_INIT);
+
+static void
+gst_omx_aac_enc_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  GstOMXAudioEncClass *audioenc_class = GST_OMX_AUDIO_ENC_CLASS (g_class);
+
+  gst_element_class_set_details_simple (element_class,
+      "OpenMAX AAC Audio Encoder",
+      "Codec/Encoder/Audio",
+      "Encode AAC audio streams",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
+  /* If no role was set from the config file we set the
+   * default AAC audio encoder role */
+  if (!audioenc_class->component_role)
+    audioenc_class->component_role = "audio_encoder.aac";
+}
+
+static void
+gst_omx_aac_enc_class_init (GstOMXAACEncClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstOMXAudioEncClass *audioenc_class = GST_OMX_AUDIO_ENC_CLASS (klass);
+
+  gobject_class->finalize = gst_omx_aac_enc_finalize;
+  gobject_class->set_property = gst_omx_aac_enc_set_property;
+  gobject_class->get_property = gst_omx_aac_enc_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_BITRATE,
+      g_param_spec_uint ("bitrate", "Bitrate",
+          "Bitrate",
+          0, G_MAXUINT, DEFAULT_BITRATE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+          GST_PARAM_MUTABLE_READY));
+
+  g_object_class_install_property (gobject_class, PROP_AAC_TOOLS,
+      g_param_spec_flags ("aac-tools", "AAC Tools",
+          "Allowed AAC tools",
+          GST_TYPE_OMX_AAC_TOOLS,
+          DEFAULT_AAC_TOOLS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+          GST_PARAM_MUTABLE_READY));
+
+  g_object_class_install_property (gobject_class,
+      PROP_AAC_ERROR_RESILIENCE_TOOLS,
+      g_param_spec_flags ("aac-error-resilience-tools",
+          "AAC Error Resilience Tools", "Allowed AAC error resilience tools",
+          GST_TYPE_OMX_AAC_ER_TOOLS, DEFAULT_AAC_ER_TOOLS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+          GST_PARAM_MUTABLE_READY));
+
+  audioenc_class->set_format = GST_DEBUG_FUNCPTR (gst_omx_aac_enc_set_format);
+  audioenc_class->get_caps = GST_DEBUG_FUNCPTR (gst_omx_aac_enc_get_caps);
+
+  audioenc_class->default_src_template_caps = "audio/mpeg, "
+      "mpegversion=(int){2, 4}, "
+      "stream-format=(string){raw, adts, adif, loas, latm}";
+}
+
+static void
+gst_omx_aac_enc_init (GstOMXAACEnc * self, GstOMXAACEncClass * klass)
+{
+  self->bitrate = DEFAULT_BITRATE;
+  self->aac_tools = DEFAULT_AAC_TOOLS;
+  self->aac_er_tools = DEFAULT_AAC_ER_TOOLS;
+}
+
+static void
+gst_omx_aac_enc_finalize (GObject * object)
+{
+  /* GstOMXAACEnc *self = GST_OMX_AAC_ENC (object); */
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_omx_aac_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstOMXAACEnc *self = GST_OMX_AAC_ENC (object);
+
+  switch (prop_id) {
+    case PROP_BITRATE:
+      self->bitrate = g_value_get_uint (value);
+      break;
+    case PROP_AAC_TOOLS:
+      self->aac_tools = g_value_get_flags (value);
+      break;
+    case PROP_AAC_ERROR_RESILIENCE_TOOLS:
+      self->aac_er_tools = g_value_get_flags (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_omx_aac_enc_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstOMXAACEnc *self = GST_OMX_AAC_ENC (object);
+
+  switch (prop_id) {
+    case PROP_BITRATE:
+      g_value_set_uint (value, self->bitrate);
+      break;
+    case PROP_AAC_TOOLS:
+      g_value_set_flags (value, self->aac_tools);
+      break;
+    case PROP_AAC_ERROR_RESILIENCE_TOOLS:
+      g_value_set_flags (value, self->aac_er_tools);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+gst_omx_aac_enc_set_format (GstOMXAudioEnc * enc, GstOMXPort * port,
+    GstAudioState * state)
+{
+  GstOMXAACEnc *self = GST_OMX_AAC_ENC (enc);
+  OMX_AUDIO_PARAM_AACPROFILETYPE aac_profile;
+  GstCaps *peercaps;
+  OMX_AUDIO_AACSTREAMFORMATTYPE stream_format = OMX_AUDIO_AACStreamFormatRAW;
+  OMX_AUDIO_AACPROFILETYPE profile = OMX_AUDIO_AACObjectLC;
+  OMX_ERRORTYPE err;
+
+  GST_OMX_INIT_STRUCT (&aac_profile);
+  aac_profile.nPortIndex = enc->out_port->index;
+
+  err =
+      gst_omx_component_get_parameter (enc->component, OMX_IndexParamAudioAac,
+      &aac_profile);
+  if (err != OMX_ErrorNone) {
+    GST_ERROR_OBJECT (self,
+        "Failed to get AAC parameters from component: %s (0x%08x)",
+        gst_omx_error_to_string (err), err);
+    return FALSE;
+  }
+
+  peercaps = gst_pad_peer_get_caps (GST_BASE_AUDIO_ENCODER_SRC_PAD (self));
+  if (peercaps) {
+    GstCaps *intersection;
+    GstStructure *s;
+    gint mpegversion = 0;
+    const gchar *profile_string, *stream_format_string;
+
+    intersection =
+        gst_caps_intersect (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_AUDIO_ENCODER_SRC_PAD (self)));
+    gst_caps_unref (peercaps);
+    if (gst_caps_is_empty (intersection)) {
+      gst_caps_unref (intersection);
+      GST_ERROR_OBJECT (self, "Empty caps");
+      return FALSE;
+    }
+
+    s = gst_caps_get_structure (intersection, 0);
+
+    if (gst_structure_get_int (s, "mpegversion", &mpegversion)) {
+      profile_string =
+          gst_structure_get_string (s,
+          ((mpegversion == 2) ? "profile" : "base-profile"));
+
+      if (profile_string) {
+        if (g_str_equal (profile_string, "main")) {
+          profile = OMX_AUDIO_AACObjectMain;
+        } else if (g_str_equal (profile_string, "lc")) {
+          profile = OMX_AUDIO_AACObjectLC;
+        } else if (g_str_equal (profile_string, "ssr")) {
+          profile = OMX_AUDIO_AACObjectSSR;
+        } else if (g_str_equal (profile_string, "ltp")) {
+          profile = OMX_AUDIO_AACObjectLTP;
+        } else {
+          GST_ERROR_OBJECT (self, "Unsupported profile '%s'", profile_string);
+          gst_caps_unref (intersection);
+          return FALSE;
+        }
+      }
+    }
+
+    stream_format_string = gst_structure_get_string (s, "stream-format");
+    if (stream_format_string) {
+      if (g_str_equal (stream_format_string, "raw")) {
+        stream_format = OMX_AUDIO_AACStreamFormatRAW;
+      } else if (g_str_equal (stream_format_string, "adts")) {
+        if (mpegversion == 2) {
+          stream_format = OMX_AUDIO_AACStreamFormatMP2ADTS;
+        } else {
+          stream_format = OMX_AUDIO_AACStreamFormatMP4ADTS;
+        }
+      } else if (g_str_equal (stream_format_string, "loas")) {
+        stream_format = OMX_AUDIO_AACStreamFormatMP4LOAS;
+      } else if (g_str_equal (stream_format_string, "latm")) {
+        stream_format = OMX_AUDIO_AACStreamFormatMP4LATM;
+      } else if (g_str_equal (stream_format_string, "adif")) {
+        stream_format = OMX_AUDIO_AACStreamFormatADIF;
+      } else {
+        GST_ERROR_OBJECT (self, "Unsupported stream-format '%s'",
+            stream_format_string);
+        gst_caps_unref (intersection);
+        return FALSE;
+      }
+    }
+
+    gst_caps_unref (intersection);
+  }
+
+  aac_profile.eAACProfile = profile;
+  aac_profile.eAACStreamFormat = stream_format;
+
+  aac_profile.nAACtools = self->aac_tools;
+  aac_profile.nAACERtools = self->aac_er_tools;
+
+  aac_profile.nBitRate = self->bitrate;
+
+  err =
+      gst_omx_component_set_parameter (enc->component, OMX_IndexParamAudioAac,
+      &aac_profile);
+  if (err != OMX_ErrorNone) {
+    GST_ERROR_OBJECT (self, "Error setting AAC parameters: %s (0x%08x)",
+        gst_omx_error_to_string (err), err);
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+static GstCaps *
+gst_omx_aac_enc_get_caps (GstOMXAudioEnc * enc, GstOMXPort * port,
+    GstAudioState * state)
+{
+  GstCaps *caps;
+  OMX_ERRORTYPE err;
+  OMX_AUDIO_PARAM_AACPROFILETYPE aac_profile;
+  gint mpegversion = 4;
+  const gchar *stream_format = NULL, *profile = NULL;
+
+  GST_OMX_INIT_STRUCT (&aac_profile);
+  aac_profile.nPortIndex = enc->out_port->index;
+
+  err =
+      gst_omx_component_get_parameter (enc->component, OMX_IndexParamAudioAac,
+      &aac_profile);
+  if (err != OMX_ErrorNone) {
+    GST_ERROR_OBJECT (enc,
+        "Failed to get AAC parameters from component: %s (0x%08x)",
+        gst_omx_error_to_string (err), err);
+    return NULL;
+  }
+
+  switch (aac_profile.eAACProfile) {
+    case OMX_AUDIO_AACObjectMain:
+      profile = "main";
+      break;
+    case OMX_AUDIO_AACObjectLC:
+      profile = "lc";
+      break;
+    case OMX_AUDIO_AACObjectSSR:
+      profile = "ssr";
+      break;
+    case OMX_AUDIO_AACObjectLTP:
+      profile = "ltp";
+      break;
+    case OMX_AUDIO_AACObjectHE:
+    case OMX_AUDIO_AACObjectScalable:
+    case OMX_AUDIO_AACObjectERLC:
+    case OMX_AUDIO_AACObjectLD:
+    case OMX_AUDIO_AACObjectHE_PS:
+    default:
+      GST_ERROR_OBJECT (enc, "Unsupported profile %d", aac_profile.eAACProfile);
+      break;
+  }
+
+  switch (aac_profile.eAACStreamFormat) {
+    case OMX_AUDIO_AACStreamFormatMP2ADTS:
+      mpegversion = 2;
+      stream_format = "adts";
+      break;
+    case OMX_AUDIO_AACStreamFormatMP4ADTS:
+      mpegversion = 4;
+      stream_format = "adts";
+      break;
+    case OMX_AUDIO_AACStreamFormatMP4LOAS:
+      mpegversion = 4;
+      stream_format = "loas";
+      break;
+    case OMX_AUDIO_AACStreamFormatMP4LATM:
+      mpegversion = 4;
+      stream_format = "latm";
+      break;
+    case OMX_AUDIO_AACStreamFormatADIF:
+      mpegversion = 4;
+      stream_format = "adif";
+      break;
+    case OMX_AUDIO_AACStreamFormatRAW:
+      mpegversion = 4;
+      stream_format = "raw";
+      break;
+    case OMX_AUDIO_AACStreamFormatMP4FF:
+    default:
+      GST_ERROR_OBJECT (enc, "Unsupported stream-format %u",
+          aac_profile.eAACStreamFormat);
+      break;
+  }
+
+  caps = gst_caps_new_simple ("audio/mpeg", NULL);
+
+  if (mpegversion != 0)
+    gst_caps_set_simple (caps, "mpegversion", G_TYPE_INT, mpegversion,
+        "stream-format", G_TYPE_STRING, stream_format, NULL);
+  if (profile != NULL && mpegversion == 2)
+    gst_caps_set_simple (caps, "profile", G_TYPE_STRING, profile, NULL);
+  if (profile != NULL && mpegversion == 4)
+    gst_caps_set_simple (caps, "base-profile", G_TYPE_STRING, profile, NULL);
+  if (aac_profile.nChannels != 0)
+    gst_caps_set_simple (caps, "channels", G_TYPE_INT, aac_profile.nChannels,
+        NULL);
+  if (aac_profile.nSampleRate != 0)
+    gst_caps_set_simple (caps, "rate", G_TYPE_INT, aac_profile.nSampleRate,
+        NULL);
+
+  return caps;
+}
diff --git a/omx/gstomxaacenc.h b/omx/gstomxaacenc.h
new file mode 100644
index 0000000000..b6c3daa9b7
--- /dev/null
+++ b/omx/gstomxaacenc.h
@@ -0,0 +1,65 @@
+/*
+ * Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
+ *   Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation
+ * version 2.1 of the License.
+ *
+ * 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+ *
+ */
+
+#ifndef __GST_OMX_AAC_ENC_H__
+#define __GST_OMX_AAC_ENC_H__
+
+#include <gst/gst.h>
+#include "gstomxaudioenc.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OMX_AAC_ENC \
+  (gst_omx_aac_enc_get_type())
+#define GST_OMX_AAC_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OMX_AAC_ENC,GstOMXAACEnc))
+#define GST_OMX_AAC_ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OMX_AAC_ENC,GstOMXAACEncClass))
+#define GST_OMX_AAC_ENC_GET_CLASS(obj) \
+  (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_OMX_AAC_ENC,GstOMXAACEncClass))
+#define GST_IS_OMX_AAC_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OMX_AAC_ENC))
+#define GST_IS_OMX_AAC_ENC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OMX_AAC_ENC))
+
+typedef struct _GstOMXAACEnc GstOMXAACEnc;
+typedef struct _GstOMXAACEncClass GstOMXAACEncClass;
+
+struct _GstOMXAACEnc
+{
+  GstOMXAudioEnc parent;
+
+  /* properties */
+  guint bitrate;
+  guint aac_tools;
+  guint aac_er_tools;
+};
+
+struct _GstOMXAACEncClass
+{
+  GstOMXAudioEncClass parent_class;
+};
+
+GType gst_omx_aac_enc_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_OMX_AAC_ENC_H__ */
+