From 95343e3ad9af7eecb2be61b1eb84c0fcb54a48db Mon Sep 17 00:00:00 2001
From: benjamin gaignard <benjamin.gaignard@linaro.org>
Date: Mon, 18 Apr 2011 17:19:00 +0200
Subject: [PATCH] [voaacenc] add new plugin for audio AAC encoder based on vo-aacenc lib
 add plugin and unit

---
 configure.ac                    |    8 +
 ext/Makefile.am                 |    7 +
 ext/voaacenc/Makefile.am        |   18 +
 ext/voaacenc/gstvoaac.c         |   38 +++
 ext/voaacenc/gstvoaacenc.c      |  665 +++++++++++++++++++++++++++++++++++++++
 ext/voaacenc/gstvoaacenc.h      |   81 +++++
 tests/check/Makefile.am         |   11 +
 tests/check/elements/voaacenc.c |  287 +++++++++++++++++
 8 files changed, 1115 insertions(+), 0 deletions(-)
 create mode 100644 ext/voaacenc/Makefile.am
 create mode 100644 ext/voaacenc/gstvoaac.c
 create mode 100644 ext/voaacenc/gstvoaacenc.c
 create mode 100644 ext/voaacenc/gstvoaacenc.h
 create mode 100644 tests/check/elements/voaacenc.c

diff --git a/configure.ac b/configure.ac
index 6b4bf57..7742eaa 100644
--- a/configure.ac
+++ b/configure.ac
@@ -569,6 +569,12 @@ AG_GST_CHECK_FEATURE(AMRWB, [amrwb library], amrwbenc, [
                         AC_SUBST(AMRWB_LIBS))
 ])
 
+dnl *** aac-enc ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_VOAACENC, true)
+AG_GST_CHECK_FEATURE(VOAACENC, [vo-aacenc library], vo-aacenc, [
+  AG_GST_PKG_CHECK_MODULES(VOAACENC, vo-aacenc >= 0.1.0)
+])
+
 dnl *** apexsink ***
 translit(dnm, m, l) AM_CONDITIONAL(USE_APEXSINK, true)
 AG_GST_CHECK_FEATURE(APEXSINK, [AirPort Express Wireless sink], apexsink, [
@@ -1585,6 +1591,7 @@ dnl but we still need to set the conditionals
 
 AM_CONDITIONAL(USE_ASSRENDER, false)
 AM_CONDITIONAL(USE_AMRWB, false)
+AM_CONDITIONAL(USE_VOAACENC, false)
 AM_CONDITIONAL(USE_APEXSINK, false)
 AM_CONDITIONAL(USE_BZ2, false)
 AM_CONDITIONAL(USE_CDAUDIO, false)
@@ -1820,6 +1827,7 @@ tests/examples/mxf/Makefile
 tests/examples/scaletempo/Makefile
 tests/icles/Makefile
 ext/amrwbenc/Makefile
+ext/voaacenc/Makefile
 ext/assrender/Makefile
 ext/apexsink/Makefile
 ext/bz2/Makefile
diff --git a/ext/Makefile.am b/ext/Makefile.am
index 9b670d4..3e4beb2 100644
--- a/ext/Makefile.am
+++ b/ext/Makefile.am
@@ -112,6 +112,12 @@ else
  FAAD_DIR=
 endif
 
+if USE_VOAACENC
+ VOAACENC_DIR=voaacenc
+else
+ VOAACENC_DIR=
+endif
+
 if USE_FLITE
 FLITE_DIR=flite
 else
@@ -368,6 +374,7 @@ endif
 
 
 SUBDIRS=\
+	$(VOAACENC_DIR) \
 	$(ASSRENDER_DIR) \
 	$(AMRWB_DIR) \
 	$(APEXSINK_DIR) \
diff --git a/ext/voaacenc/Makefile.am b/ext/voaacenc/Makefile.am
new file mode 100644
index 0000000..7506490
--- /dev/null
+++ b/ext/voaacenc/Makefile.am
@@ -0,0 +1,18 @@
+plugin_LTLIBRARIES = libgstvoaacenc.la
+
+libgstvoaacenc_la_SOURCES = \
+	gstvoaac.c \
+	gstvoaacenc.c 
+
+libgstvoaacenc_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(VOAACENC_CFLAGS)
+libgstvoaacenc_la_LIBADD = -lgstaudio-$(GST_MAJORMINOR) \
+													 $(GST_BASE_LIBS) $(GST_LIBS) $(VOAACENC_LIBS)
+libgstvoaacenc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstvoaacenc_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = \
+	gstvoaacenc.h
+
+presetdir = $(datadir)/gstreamer-$(GST_MAJORMINOR)/presets
+
+EXTRA_DIST = $(preset_DATA)
diff --git a/ext/voaacenc/gstvoaac.c b/ext/voaacenc/gstvoaac.c
new file mode 100644
index 0000000..30b5df9
--- /dev/null
+++ b/ext/voaacenc/gstvoaac.c
@@ -0,0 +1,38 @@
+/* GStreamer AAC encoder plugin
+ * Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
+ *
+ * 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 "gstvoaacenc.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "voaacenc",
+      GST_RANK_SECONDARY, GST_TYPE_VOAACENC);
+}
+
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "voaacenc",
+    "AAC audio encoder",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
diff --git a/ext/voaacenc/gstvoaacenc.c b/ext/voaacenc/gstvoaacenc.c
new file mode 100644
index 0000000..ffad419
--- /dev/null
+++ b/ext/voaacenc/gstvoaacenc.c
@@ -0,0 +1,665 @@
+/* GStreamer AAC encoder plugin
+ * Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
+ *
+ * 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.
+ */
+
+/**
+ * SECTION:element-voaacenc
+ *
+ * AAC audio encoder based on vo-aacenc library 
+ * <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voaacenc ! filesink location=abc.aac
+ * ]|
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/audio/multichannel.h>
+
+#include "gstvoaacenc.h"
+
+#define VOAAC_ENC_DEFAULT_BITRATE (128000)
+#define VOAAC_ENC_DEFAULT_CHANNELS (2)
+#define VOAAC_ENC_DEFAULT_RATE (44100)
+#define VOAAC_ENC_DEFAULT_OUTPUTFORMAT (0)      /* RAW */
+#define VOAAC_ENC_MPEGVERSION (4)
+#define VOAAC_ENC_CODECDATA_LEN (2)
+#define VOAAC_ENC_BITS_PER_SAMPLE (16)
+
+enum
+{
+  PROP_0,
+  PROP_BITRATE
+};
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "signed = (boolean) TRUE, "
+        "endianness = (int) BYTE_ORDER, "
+        "rate = (int) [8000, 96000], " "channels = (int) [1, 6]")
+    );
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg, "
+        "mpegversion = (int) 4, "
+        "rate = (int) [8000, 96000], "
+        "channels = (int) [1, 6], " "stream-format = (string) { adts, raw } ")
+    );
+
+GST_DEBUG_CATEGORY_STATIC (gst_voaacenc_debug);
+#define GST_CAT_DEFAULT gst_voaacenc_debug
+
+static void gst_voaacenc_finalize (GObject * object);
+
+static GstFlowReturn gst_voaacenc_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean gst_voaacenc_setcaps (GstPad * pad, GstCaps * caps);
+static GstStateChangeReturn gst_voaacenc_state_change (GstElement * element,
+    GstStateChange transition);
+static gboolean voaacenc_core_init (GstVoAacEnc * voaacenc);
+static gboolean voaacenc_core_set_parameter (GstVoAacEnc * voaacenc);
+static void voaacenc_core_uninit (GstVoAacEnc * voaacenc);
+static GstCaps *gst_voaacenc_getcaps (GstPad * pad);
+static GstCaps *gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc);
+static gint voaacenc_get_rate_index (gint rate);
+
+#define VOAAC_ENC_MAX_CHANNELS 6
+
+/* describe the channels position */
+const GstAudioChannelPosition
+    gst_voaacenc_channel_position[][VOAAC_ENC_MAX_CHANNELS] = {
+  {                             /* 1 ch: Mono */
+      GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
+  {                             /* 2 ch: front left + front right (front stereo) */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+  {                             /* 3 ch: front center + front stereo */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+  {                             /* 4 ch: front center + front stereo + back center */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+      GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
+  {                             /* 5 ch: front center + front stereo + back stereo */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+      GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
+  {                             /* 6ch: front center + front stereo + back stereo + LFE */
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+      GST_AUDIO_CHANNEL_POSITION_LFE}
+};
+
+static void
+_do_init (GType object_type)
+{
+  const GInterfaceInfo preset_interface_info = {
+    NULL,                       /* interface init */
+    NULL,                       /* interface finalize */
+    NULL                        /* interface_data */
+  };
+
+  g_type_add_interface_static (object_type, GST_TYPE_PRESET,
+      &preset_interface_info);
+
+  GST_DEBUG_CATEGORY_INIT (gst_voaacenc_debug, "voaacenc", 0,
+      "AAC audio encoder");
+}
+
+GST_BOILERPLATE_FULL (GstVoAacEnc, gst_voaacenc, GstElement, GST_TYPE_ELEMENT,
+    _do_init);
+
+static void
+gst_voaacenc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstVoAacEnc *self = GST_VOAACENC (object);
+
+  switch (prop_id) {
+    case PROP_BITRATE:
+      self->bitrate = g_value_get_int (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+  return;
+}
+
+static void
+gst_voaacenc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstVoAacEnc *self = GST_VOAACENC (object);
+
+  switch (prop_id) {
+    case PROP_BITRATE:
+      g_value_set_int (value, self->bitrate);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+  return;
+}
+
+static void
+gst_voaacenc_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+
+  gst_element_class_set_details_simple (element_class, "AAC audio encoder",
+      "Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
+}
+
+static void
+gst_voaacenc_class_init (GstVoAacEncClass * klass)
+{
+  GObjectClass *object_class = G_OBJECT_CLASS (klass);
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  object_class->set_property = GST_DEBUG_FUNCPTR (gst_voaacenc_set_property);
+  object_class->get_property = GST_DEBUG_FUNCPTR (gst_voaacenc_get_property);
+  object_class->finalize = GST_DEBUG_FUNCPTR (gst_voaacenc_finalize);
+
+  g_object_class_install_property (object_class, PROP_BITRATE,
+      g_param_spec_int ("bitrate",
+          "Bitrate",
+          "Target Audio Bitrate",
+          0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  element_class->change_state = GST_DEBUG_FUNCPTR (gst_voaacenc_state_change);
+}
+
+static void
+gst_voaacenc_init (GstVoAacEnc * voaacenc, GstVoAacEncClass * klass)
+{
+  /* create the sink pad */
+  voaacenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
+  gst_pad_set_setcaps_function (voaacenc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_voaacenc_setcaps));
+  gst_pad_set_getcaps_function (voaacenc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps));
+  gst_pad_set_chain_function (voaacenc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_voaacenc_chain));
+  gst_element_add_pad (GST_ELEMENT (voaacenc), voaacenc->sinkpad);
+
+  /* create the src pad */
+  voaacenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
+  gst_pad_use_fixed_caps (voaacenc->srcpad);
+  gst_element_add_pad (GST_ELEMENT (voaacenc), voaacenc->srcpad);
+
+  voaacenc->adapter = gst_adapter_new ();
+
+  voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
+  voaacenc->rate = VOAAC_ENC_DEFAULT_RATE;
+  voaacenc->channels = VOAAC_ENC_DEFAULT_CHANNELS;
+  voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
+
+  /* init rest */
+  voaacenc->handle = NULL;
+  voaacenc->sinkcaps = NULL;
+}
+
+static void
+gst_voaacenc_finalize (GObject * object)
+{
+  GstVoAacEnc *voaacenc;
+
+  voaacenc = GST_VOAACENC (object);
+
+  if (voaacenc->sinkcaps) {
+    gst_caps_unref (voaacenc->sinkcaps);
+    voaacenc->sinkcaps = NULL;
+  }
+
+  g_object_unref (G_OBJECT (voaacenc->adapter));
+  voaacenc->adapter = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* check downstream caps to configure format */
+static void
+gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
+{
+  GstCaps *caps;
+
+  caps = gst_pad_get_allowed_caps (voaacenc->srcpad);
+
+  GST_DEBUG_OBJECT (voaacenc, "allowed caps: %" GST_PTR_FORMAT, caps);
+
+  if (caps && gst_caps_get_size (caps) > 0) {
+    GstStructure *s = gst_caps_get_structure (caps, 0);
+    const gchar *str = NULL;
+
+    if ((str = gst_structure_get_string (s, "stream-format"))) {
+      if (strcmp (str, "adts") == 0) {
+        GST_DEBUG_OBJECT (voaacenc, "use ADTS format for output");
+        voaacenc->output_format = 1;
+      } else if (strcmp (str, "raw") == 0) {
+        GST_DEBUG_OBJECT (voaacenc, "use RAW format for output");
+        voaacenc->output_format = 0;
+      } else {
+        GST_DEBUG_OBJECT (voaacenc, "unknown stream-format: %s", str);
+        voaacenc->output_format = 0;
+      }
+    }
+  }
+
+  if (caps)
+    gst_caps_unref (caps);
+
+}
+
+
+static GstCaps *
+gst_voaacenc_generate_sink_caps (void)
+{
+  GstCaps *caps = gst_caps_new_empty ();
+  gint i, c;
+
+  for (i = 0; i < VOAAC_ENC_MAX_CHANNELS; i++) {
+    GValue chanpos = { 0 };
+    GValue pos = { 0 };
+    GstStructure *structure;
+
+    g_value_init (&chanpos, GST_TYPE_ARRAY);
+    g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
+
+    for (c = 0; c <= i; c++) {
+      g_value_set_enum (&pos, gst_voaacenc_channel_position[i][c]);
+      gst_value_array_append_value (&chanpos, &pos);
+    }
+
+    g_value_unset (&pos);
+
+    structure = gst_structure_new ("audio/x-raw-int",
+        "width", G_TYPE_INT, 16,
+        "depth", G_TYPE_INT, 16,
+        "signed", G_TYPE_BOOLEAN, TRUE,
+        "endianness", G_TYPE_INT, G_BYTE_ORDER,
+        "rate", GST_TYPE_INT_RANGE, 8000, 96000, "channels", G_TYPE_INT, i + 1);
+
+    gst_structure_set_value (structure, "channel-positions", &chanpos);
+    g_value_unset (&chanpos);
+
+    gst_caps_append_structure (caps, structure);
+  }
+
+  return caps;
+}
+
+
+static GstCaps *
+gst_voaacenc_getcaps (GstPad * pad)
+{
+  GstVoAacEnc *voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
+
+  if (voaacenc->sinkcaps == NULL) {
+    voaacenc->sinkcaps = gst_voaacenc_generate_sink_caps ();
+  }
+
+  GST_DEBUG_OBJECT (voaacenc, "generated sink caps: %" GST_PTR_FORMAT,
+      voaacenc->sinkcaps);
+
+  return gst_caps_ref (voaacenc->sinkcaps);
+}
+
+
+static gboolean
+gst_voaacenc_setcaps (GstPad * pad, GstCaps * caps)
+{
+  gboolean ret = FALSE;
+  GstStructure *structure;
+  GstVoAacEnc *voaacenc;
+  GstCaps *src_caps;
+
+  voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  /* get channel count */
+  gst_structure_get_int (structure, "channels", &voaacenc->channels);
+  gst_structure_get_int (structure, "rate", &voaacenc->rate);
+
+  /* precalc duration as it's constant now */
+  voaacenc->duration =
+      gst_util_uint64_scale_int (1024, GST_SECOND, voaacenc->rate);
+  voaacenc->inbuf_size = voaacenc->channels * 2 * 1024;
+
+  gst_voaacenc_negotiate (voaacenc);
+
+  /* create reverse caps */
+  src_caps = gst_voaacenc_create_source_pad_caps (voaacenc);
+
+  if (src_caps) {
+    gst_pad_set_caps (voaacenc->srcpad, src_caps);
+    gst_caps_unref (src_caps);
+    ret = voaacenc_core_set_parameter (voaacenc);
+  }
+  return ret;
+}
+
+static GstFlowReturn
+gst_voaacenc_chain (GstPad * pad, GstBuffer * buffer)
+{
+  GstVoAacEnc *voaacenc;
+  GstFlowReturn ret;
+  guint64 timestamp, distance = 0;
+
+  voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
+
+  g_return_val_if_fail (voaacenc->handle, GST_FLOW_WRONG_STATE);
+
+  if (voaacenc->rate == 0 || voaacenc->channels == 0)
+    goto not_negotiated;
+
+  /* discontinuity clears adapter, FIXME, maybe we can set some
+   * encoder flag to mask the discont. */
+  if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+    gst_adapter_clear (voaacenc->adapter);
+    voaacenc->ts = 0;
+    voaacenc->discont = TRUE;
+  }
+
+  ret = GST_FLOW_OK;
+  gst_adapter_push (voaacenc->adapter, buffer);
+
+  /* Collect samples until we have enough for an output frame */
+  while (gst_adapter_available (voaacenc->adapter) >= voaacenc->inbuf_size) {
+    GstBuffer *out;
+    guint8 *data;
+    VO_CODECBUFFER input = { 0 }
+    , output = {
+    0};
+    VO_AUDIO_OUTPUTINFO output_info = { {0}
+    };
+
+
+    /* max size */
+    if ((ret =
+            gst_pad_alloc_buffer_and_set_caps (voaacenc->srcpad, 0,
+                voaacenc->inbuf_size, GST_PAD_CAPS (voaacenc->srcpad),
+                &out)) != GST_FLOW_OK) {
+      return ret;
+    }
+
+    output.Buffer = GST_BUFFER_DATA (out);
+    output.Length = voaacenc->inbuf_size;
+
+    if (voaacenc->discont) {
+      GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
+      voaacenc->discont = FALSE;
+    }
+
+    data =
+        (guint8 *) gst_adapter_peek (voaacenc->adapter, voaacenc->inbuf_size);
+    input.Buffer = data;
+    input.Length = voaacenc->inbuf_size;
+    voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
+
+    /* encode */
+    if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
+            &output_info) != VO_ERR_NONE) {
+      gst_buffer_unref (out);
+      return GST_FLOW_ERROR;
+    }
+
+    /* get timestamp from adapter */
+    timestamp = gst_adapter_prev_timestamp (voaacenc->adapter, &distance);
+
+    if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (timestamp))) {
+      GST_BUFFER_TIMESTAMP (out) =
+          timestamp +
+          GST_FRAMES_TO_CLOCK_TIME (distance / voaacenc->channels /
+          VOAAC_ENC_BITS_PER_SAMPLE, voaacenc->rate);
+    }
+
+    GST_BUFFER_DURATION (out) =
+        GST_FRAMES_TO_CLOCK_TIME (voaacenc->inbuf_size / voaacenc->channels /
+        VOAAC_ENC_BITS_PER_SAMPLE, voaacenc->rate);
+
+    voaacenc->ts = GST_BUFFER_TIMESTAMP (out) + GST_BUFFER_DURATION (out);
+
+    GST_LOG_OBJECT (voaacenc, "Pushing out buffer time: %" GST_TIME_FORMAT
+        " duration: %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)),
+        GST_TIME_ARGS (GST_BUFFER_DURATION (out)));
+
+    GST_BUFFER_SIZE (out) = output.Length;
+
+    /* flush the among of data we have peek */
+    gst_adapter_flush (voaacenc->adapter, voaacenc->inbuf_size);
+
+    /* play */
+    if ((ret = gst_pad_push (voaacenc->srcpad, out)) != GST_FLOW_OK)
+      break;
+  }
+  return ret;
+
+  /* ERRORS */
+not_negotiated:
+  {
+    GST_ELEMENT_ERROR (voaacenc, STREAM, TYPE_NOT_FOUND,
+        (NULL), ("unknown type"));
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+}
+
+static GstStateChangeReturn
+gst_voaacenc_state_change (GstElement * element, GstStateChange transition)
+{
+  GstVoAacEnc *voaacenc;
+  GstStateChangeReturn ret;
+
+  voaacenc = GST_VOAACENC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (voaacenc_core_init (voaacenc) == FALSE)
+        return GST_STATE_CHANGE_FAILURE;
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      voaacenc->rate = 0;
+      voaacenc->channels = 0;
+      voaacenc->ts = 0;
+      voaacenc->discont = FALSE;
+      gst_adapter_clear (voaacenc->adapter);
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      voaacenc_core_uninit (voaacenc);
+      gst_adapter_clear (voaacenc->adapter);
+      break;
+    default:
+      break;
+  }
+  return ret;
+}
+
+static GstCaps *
+gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
+{
+  GstCaps *caps = NULL;
+  GstBuffer *codec_data;
+  gint index;
+
+  if ((index = voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
+
+    caps = gst_caps_new_simple ("audio/mpeg",
+        "mpegversion", G_TYPE_INT, VOAAC_ENC_MPEGVERSION,
+        "channels", G_TYPE_INT, voaacenc->channels,
+        "rate", G_TYPE_INT, voaacenc->rate,
+        "stream-format", G_TYPE_STRING,
+        (voaacenc->output_format ? "adts" : "raw")
+        , NULL);
+
+    if (!voaacenc->output_format) {
+      codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
+
+      GST_BUFFER_DATA (codec_data)[0] = ((0x02 << 3) | (index >> 1));
+      GST_BUFFER_DATA (codec_data)[1] =
+          ((index & 0x01) << 7) | (voaacenc->channels << 3);
+
+      gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
+          NULL);
+
+      gst_buffer_unref (codec_data);
+    }
+  }
+
+  return caps;
+}
+
+
+
+static VO_U32
+voaacenc_core_mem_alloc (VO_S32 uID, VO_MEM_INFO * pMemInfo)
+{
+  if (!pMemInfo)
+    return VO_ERR_INVALID_ARG;
+
+  pMemInfo->VBuffer = g_malloc (pMemInfo->Size);
+  return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_free (VO_S32 uID, VO_PTR pMem)
+{
+  g_free (pMem);
+  return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_set (VO_S32 uID, VO_PTR pBuff, VO_U8 uValue, VO_U32 uSize)
+{
+  memset (pBuff, uValue, uSize);
+  return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_copy (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
+{
+  memcpy (pDest, pSource, uSize);
+  return 0;
+}
+
+static VO_U32
+voaacenc_core_mem_check (VO_S32 uID, VO_PTR pBuffer, VO_U32 uSize)
+{
+  return 0;
+}
+
+static gboolean
+voaacenc_core_init (GstVoAacEnc * voaacenc)
+{
+  VO_CODEC_INIT_USERDATA user_data = { 0 };
+  voGetAACEncAPI (&voaacenc->codec_api);
+
+  voaacenc->mem_operator.Alloc = voaacenc_core_mem_alloc;
+  voaacenc->mem_operator.Copy = voaacenc_core_mem_copy;
+  voaacenc->mem_operator.Free = voaacenc_core_mem_free;
+  voaacenc->mem_operator.Set = voaacenc_core_mem_set;
+  voaacenc->mem_operator.Check = voaacenc_core_mem_check;
+  user_data.memflag = VO_IMF_USERMEMOPERATOR;
+  user_data.memData = &voaacenc->mem_operator;
+  voaacenc->codec_api.Init (&voaacenc->handle, VO_AUDIO_CodingAAC, &user_data);
+
+  if (voaacenc->handle == NULL) {
+    return FALSE;
+  }
+  return TRUE;
+
+}
+
+static gboolean
+voaacenc_core_set_parameter (GstVoAacEnc * voaacenc)
+{
+  AACENC_PARAM params = { 0 };
+  params.sampleRate = voaacenc->rate;
+  params.bitRate = voaacenc->bitrate;
+  params.nChannels = voaacenc->channels;
+  if (voaacenc->output_format) {
+    params.adtsUsed = 1;
+  } else {
+    params.adtsUsed = 0;
+  }
+  if (voaacenc->codec_api.SetParam (voaacenc->handle, VO_PID_AAC_ENCPARAM,
+          &params) != VO_ERR_NONE) {
+    return FALSE;
+  }
+  return TRUE;
+}
+
+static void
+voaacenc_core_uninit (GstVoAacEnc * voaacenc)
+{
+  if (voaacenc->handle) {
+    voaacenc->codec_api.Uninit (voaacenc->handle);
+    voaacenc->handle = NULL;
+  }
+}
+
+static gint
+voaacenc_get_rate_index (gint rate)
+{
+  static const gint rate_table[] = {
+    96000, 88200, 64000, 48000, 44100, 32000,
+    24000, 22050, 16000, 12000, 11025, 8000
+  };
+  gint i;
+  for (i = 0; i < G_N_ELEMENTS (rate_table); ++i) {
+    if (rate == rate_table[i]) {
+      return i;
+    }
+  }
+  return -1;
+}
diff --git a/ext/voaacenc/gstvoaacenc.h b/ext/voaacenc/gstvoaacenc.h
new file mode 100644
index 0000000..0a336cd
--- /dev/null
+++ b/ext/voaacenc/gstvoaacenc.h
@@ -0,0 +1,81 @@
+/* GStreamer AAC encoder plugin
+ * Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
+ *
+ * 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_VOAACENC_H__
+#define __GST_VOAACENC_H__
+
+#include <gst/gst.h>
+#include <vo-aacenc/voAAC.h>
+#include <vo-aacenc/cmnMemory.h>
+
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_VOAACENC \
+  (gst_voaacenc_get_type())
+#define GST_VOAACENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_VOAACENC, GstVoAacEnc))
+#define GST_VOAACENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_VOAACENC, GstVoAacEncClass))
+#define GST_IS_VOAACENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_VOAACENC))
+#define GST_IS_VOAACENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_VOAACENC))
+
+typedef struct _GstVoAacEnc GstVoAacEnc;
+typedef struct _GstVoAacEncClass GstVoAacEncClass;
+
+struct _GstVoAacEnc {
+  GstElement element;
+
+  /* pads */
+  GstPad *sinkpad, *srcpad;
+	GstCaps *sinkcaps;
+  guint64 ts;
+  gboolean discont;
+
+  GstAdapter *adapter;
+
+
+  /* desired bitrate */
+  gint bitrate;
+  gint channels;
+  gint rate;
+  gint output_format;
+  gint duration;
+
+  gint inbuf_size;
+
+  /* library handle */
+  VO_AUDIO_CODECAPI codec_api;
+  VO_HANDLE handle;
+  VO_MEM_OPERATOR mem_operator;
+
+};
+
+struct _GstVoAacEncClass {
+  GstElementClass parent_class;
+};
+
+GType gst_voaacenc_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_VOAACENC_H__ */
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 9de5de2..89b0c1c 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -46,6 +46,12 @@ else
 check_faad =
 endif
 
+if USE_VOAACENC
+check_voaacenc = elements/voaacenc
+else
+check_voaacenc =
+endif
+
 if USE_EXIF
 check_jifmux = elements/jifmux
 else
@@ -143,6 +149,7 @@ check_PROGRAMS = \
 	$(check_assrender) \
 	$(check_faac)  \
 	$(check_faad)  \
+	$(check_voaacenc) \
 	$(check_mpeg2enc)  \
 	$(check_mplex)     \
 	$(check_ofa)        \
@@ -184,6 +191,10 @@ AM_CFLAGS = $(GST_CHECK_CFLAGS) $(GST_OPTION_CFLAGS) \
 		-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
 LDADD = $(GST_CHECK_LIBS)
 
+elements_voaacenc_LDADD = \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
+	-lgstaudio-@GST_MAJORMINOR@ 
+
 elements_camerabin_CFLAGS = \
 	$(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) \
 	$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS) -DGST_USE_UNSTABLE_API
diff --git a/tests/check/elements/voaacenc.c b/tests/check/elements/voaacenc.c
new file mode 100644
index 0000000..22b42fb
--- /dev/null
+++ b/tests/check/elements/voaacenc.c
@@ -0,0 +1,287 @@
+/* GStreamer
+ *
+ * unit test for voaacenc
+ *
+ * Copyright (C) <2009> Mark Nauwelaerts <mnauw@users.sf.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.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+#include <gst/audio/multichannel.h>
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+static GstPad *mysrcpad, *mysinkpad;
+
+#define AUDIO_CAPS_STRING "audio/x-raw-int, " \
+                           "rate = (int) 48000, " \
+                           "channels = (int) 2, " \
+                           "width = (int) 16, " \
+                           "depth = (int) 16, " \
+                           "signed = (boolean) true, " \
+                           "endianness = (int) BYTE_ORDER "
+
+#define AAC_RAW_CAPS_STRING "audio/mpeg, " \
+                          "mpegversion = (int) 4, " \
+                          "rate = (int) 48000, " \
+                          "channels = (int) 2, " \
+													"stream-format = \"raw\""
+
+#define AAC_ADTS_CAPS_STRING "audio/mpeg, " \
+                          "mpegversion = (int) 4, " \
+                          "rate = (int) 48000, " \
+                          "channels = (int) 2, " \
+													"stream-format = \"adts\""
+
+
+static GstStaticPadTemplate sinktemplate_adts = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (AAC_ADTS_CAPS_STRING));
+
+static GstStaticPadTemplate sinktemplate_raw = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (AAC_RAW_CAPS_STRING));
+
+
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (AUDIO_CAPS_STRING));
+
+
+static GstElement *
+setup_voaacenc (gboolean adts)
+{
+  GstElement *voaacenc;
+
+  GST_DEBUG ("setup_voaacenc");
+  voaacenc = gst_check_setup_element ("voaacenc");
+  mysrcpad = gst_check_setup_src_pad (voaacenc, &srctemplate, NULL);
+
+  if (adts)
+    mysinkpad = gst_check_setup_sink_pad (voaacenc, &sinktemplate_adts, NULL);
+  else
+    mysinkpad = gst_check_setup_sink_pad (voaacenc, &sinktemplate_raw, NULL);
+
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  return voaacenc;
+}
+
+static void
+cleanup_voaacenc (GstElement * voaacenc)
+{
+  GST_DEBUG ("cleanup_aacenc");
+  gst_element_set_state (voaacenc, GST_STATE_NULL);
+
+  gst_pad_set_active (mysrcpad, FALSE);
+  gst_pad_set_active (mysinkpad, FALSE);
+  gst_check_teardown_src_pad (voaacenc);
+  gst_check_teardown_sink_pad (voaacenc);
+  gst_check_teardown_element (voaacenc);
+}
+
+static void
+set_channel_positions (GstCaps * caps, int channels,
+    GstAudioChannelPosition * channelpositions)
+{
+  GValue chanpos = { 0 };
+  GValue pos = { 0 };
+  GstStructure *structure = gst_caps_get_structure (caps, 0);
+  int c;
+
+  g_value_init (&chanpos, GST_TYPE_ARRAY);
+  g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
+
+  for (c = 0; c < channels; c++) {
+    g_value_set_enum (&pos, channelpositions[c]);
+    gst_value_array_append_value (&chanpos, &pos);
+  }
+  g_value_unset (&pos);
+
+  gst_structure_set_value (structure, "channel-positions", &chanpos);
+  g_value_unset (&chanpos);
+}
+
+static void
+do_test (gboolean adts)
+{
+  GstElement *voaacenc;
+  GstBuffer *inbuffer, *outbuffer;
+  GstCaps *caps;
+  gint i, num_buffers;
+  const gint nbuffers = 10;
+  GstAudioChannelPosition channel_position_layout[2] =
+      { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+    GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
+  };
+
+  voaacenc = setup_voaacenc (adts);
+  fail_unless (gst_element_set_state (voaacenc,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  /* corresponds to audio buffer mentioned in the caps */
+  inbuffer = gst_buffer_new_and_alloc (1024 * nbuffers * 2 * 2);
+  /* makes valgrind's memcheck happier */
+  memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
+  caps = gst_caps_from_string (AUDIO_CAPS_STRING);
+
+  set_channel_positions (caps, 2, channel_position_layout);
+
+  gst_buffer_set_caps (inbuffer, caps);
+  gst_caps_unref (caps);
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+
+  /* send eos to have all flushed if needed */
+  fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()) == TRUE);
+
+  num_buffers = g_list_length (buffers);
+  fail_unless_equals_int (num_buffers, nbuffers);
+
+  /* clean up buffers */
+  for (i = 0; i < num_buffers; ++i) {
+    gint size, header = 0, id;
+    guint8 *data;
+
+    outbuffer = GST_BUFFER (buffers->data);
+    fail_if (outbuffer == NULL);
+
+    data = GST_BUFFER_DATA (outbuffer);
+    size = GST_BUFFER_SIZE (outbuffer);
+
+    if (adts) {
+      gboolean protection;
+      gint k;
+
+      fail_if (size < 7);
+      protection = !(data[1] & 0x1);
+      /* expect only 1 raw data block */
+      k = (data[6] & 0x3) + 1;
+      fail_if (k != 1);
+
+      header = 7;
+      if (protection)
+        header += (k - 1) * 2 + 2;
+
+      /* check header */
+      k = GST_READ_UINT16_BE (data) & 0xFFF6;
+      /* sync */
+      fail_unless (k == 0xFFF0);
+      k = data[2];
+      /* profile */
+      fail_unless ((k >> 6) == 0x1);
+      /* rate */
+      fail_unless (((k >> 2) & 0xF) == 0x3);
+      /* channels */
+      fail_unless ((k & 0x1) == 0);
+      k = data[3];
+      fail_unless ((k >> 6) == 0x2);
+
+    } else {
+      GstCaps *caps;
+      GstStructure *s;
+      const GValue *value;
+      GstBuffer *buf;
+      gint k;
+
+      caps = gst_buffer_get_caps (outbuffer);
+      fail_if (caps == NULL);
+      s = gst_caps_get_structure (caps, 0);
+      fail_if (s == NULL);
+      value = gst_structure_get_value (s, "codec_data");
+      fail_if (value == NULL);
+      buf = gst_value_get_buffer (value);
+      fail_if (buf == NULL);
+      data = GST_BUFFER_DATA (buf);
+      size = GST_BUFFER_SIZE (buf);
+      fail_if (size < 2);
+      k = GST_READ_UINT16_BE (data);
+      /* profile, rate, channels */
+      fail_unless ((k & 0xFFF8) == ((0x02 << 11) | (0x3 << 7) | (0x02 << 3)));
+      gst_caps_unref (caps);
+
+    }
+
+    fail_if (size <= header);
+    id = data[header] & (0x7 << 5);
+    /* allow all but ID_END or ID_LFE */
+    fail_if (id == 7 || id == 3);
+
+    buffers = g_list_remove (buffers, outbuffer);
+
+    ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
+    gst_buffer_unref (outbuffer);
+    outbuffer = NULL;
+  }
+
+  cleanup_voaacenc (voaacenc);
+  g_list_free (buffers);
+  buffers = NULL;
+}
+
+GST_START_TEST (test_adts)
+{
+  do_test (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_raw)
+{
+  do_test (FALSE);
+}
+
+GST_END_TEST;
+
+static Suite *
+voaacenc_suite (void)
+{
+  Suite *s = suite_create ("voaacenc");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_adts);
+  tcase_add_test (tc_chain, test_raw);
+
+  return s;
+}
+
+int
+main (int argc, char **argv)
+{
+  int nf;
+
+  Suite *s = voaacenc_suite ();
+  SRunner *sr = srunner_create (s);
+
+  gst_check_init (&argc, &argv);
+
+  srunner_run_all (sr, CK_NORMAL);
+  nf = srunner_ntests_failed (sr);
+  srunner_free (sr);
+
+  return nf;
+}
-- 
1.7.0.4
