--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_nokia_speech/gstaacenc.c	Wed Mar 24 18:04:17 2010 -0500
@@ -0,0 +1,603 @@
+/* GStreamer AAC encoder
+ * Copyright 2009 Collabora Multimedia,
+ * Copyright 2009 Nokia Corporation
+ *  @author: Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ *
+ * 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.
+ */
+
+/* TODO non-GPL license */
+
+/**
+ * SECTION:element-nokiaaacenc
+ * @seealso: nokiaaacdec
+ *
+ * nokiaaacenc encodes raw audio to AAC streams.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+#include <string.h>
+
+#include "gstaacenc.h"
+
+GST_DEBUG_CATEGORY_STATIC (aac_enc);
+#define GST_CAT_DEFAULT aac_enc
+
+enum
+{
+  AAC_PROFILE_AUTO = 0,
+  AAC_PROFILE_LC = 2,
+  AAC_PROFILE_HE = 5
+};
+
+#define GST_TYPE_AAC_ENC_PROFILE (gst_aac_enc_profile_get_type ())
+static GType
+gst_aac_enc_profile_get_type (void)
+{
+  static GType gst_aac_enc_profile_type = 0;
+
+  if (!gst_aac_enc_profile_type) {
+    static GEnumValue gst_aac_enc_profile[] = {
+      {AAC_PROFILE_AUTO, "Codec selects LC or HE", "AUTO"},
+      {AAC_PROFILE_LC, "Low complexity profile", "LC"},
+      {AAC_PROFILE_HE, "High Efficiency", "HE"},
+      {0, NULL, NULL},
+    };
+
+    gst_aac_enc_profile_type = g_enum_register_static ("GstNokiaAacEncProfile",
+        gst_aac_enc_profile);
+  }
+
+  return gst_aac_enc_profile_type;
+}
+
+#define GST_TYPE_AAC_ENC_OUTPUTFORMAT (gst_aac_enc_outputformat_get_type ())
+static GType
+gst_aac_enc_outputformat_get_type (void)
+{
+  static GType gst_aac_enc_outputformat_type = 0;
+
+  if (!gst_aac_enc_outputformat_type) {
+    static GEnumValue gst_aac_enc_outputformat[] = {
+      {RAW, "AAC Raw format", "RAW"},
+      {USE_ADTS, "Audio Data Transport Stream format", "ADTS"},
+      {USE_ADIF, "Audio Data Interchange Format", "ADIF"},
+      {0, NULL, NULL},
+    };
+
+    gst_aac_enc_outputformat_type =
+        g_enum_register_static ("GstNokiaAacEncOutputFormat",
+        gst_aac_enc_outputformat);
+  }
+
+  return gst_aac_enc_outputformat_type;
+}
+
+enum
+{
+  PROP_0,
+  PROP_BITRATE,
+  PROP_PROFILE,
+  PROP_FORMAT
+};
+
+static GstStaticPadTemplate gst_aac_enc_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "endianness = (int) BYTE_ORDER, "
+        "signed = (bool) TRUE, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "rate = (int) [ 8000, 96000 ],  channels = (int) [ 1, 2 ] ")
+    );
+
+static GstStaticPadTemplate gst_aac_enc_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, 2 ] ")
+    );
+
+static void gst_aac_enc_base_init (gpointer g_class);
+static void gst_aac_enc_class_init (GstAACEncClass * klass);
+static void gst_aac_enc_init (GstAACEnc * filter, GstAACEncClass * klass);
+
+static void gst_aac_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_aac_enc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static void gst_aac_enc_finalize (GObject * object);
+static void gst_aac_enc_reset (GstAACEnc * enc);
+static GstStateChangeReturn gst_aac_enc_change_state (GstElement * element,
+    GstStateChange transition);
+static gboolean gst_aac_enc_sink_setcaps (GstPad * pad, GstCaps * caps);
+static GstFlowReturn gst_aac_enc_chain (GstPad * pad, GstBuffer * buffer);
+
+GST_BOILERPLATE (GstNokiaAACEnc, gst_aac_enc, GstElement, GST_TYPE_ELEMENT);
+
+static void
+gst_aac_enc_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_set_details_simple (element_class,
+      "Nokia AAC encoder", "Codec/Encoder/Audio",
+      "Nokia AAC encoder",
+      "MCC, Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_aac_enc_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_aac_enc_sink_template));
+}
+
+/* initialize the plugin's class */
+static void
+gst_aac_enc_class_init (GstAACEncClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  GST_DEBUG_CATEGORY_INIT (aac_enc, "nokiaaacenc", 0, "Nokia AAC encoder");
+
+  gobject_class->set_property = gst_aac_enc_set_property;
+  gobject_class->get_property = gst_aac_enc_get_property;
+  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_aac_enc_finalize);
+
+  /* properties */
+  g_object_class_install_property (gobject_class, PROP_BITRATE,
+      g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
+          8 * 1000, 320 * 1000, 128 * 1000,
+          (GParamFlags) (G_PARAM_READWRITE | G_PARAM_CONSTRUCT)));
+  g_object_class_install_property (gobject_class, PROP_PROFILE,
+      g_param_spec_enum ("profile", "Profile",
+          "MPEG/AAC encoding profile",
+          GST_TYPE_AAC_ENC_PROFILE, AAC_PROFILE_LC,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+  g_object_class_install_property (gobject_class, PROP_FORMAT,
+      g_param_spec_enum ("output-format", "Output format",
+          "Format of output frames",
+          GST_TYPE_AAC_ENC_OUTPUTFORMAT, RAW,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+
+  gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_aac_enc_change_state);
+}
+
+static void
+gst_aac_enc_init (GstAACEnc * enc, GstAACEncClass * klass)
+{
+  enc->sinkpad =
+      gst_pad_new_from_static_template (&gst_aac_enc_sink_template, "sink");
+  gst_pad_set_setcaps_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_aac_enc_sink_setcaps));
+  gst_pad_set_chain_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_aac_enc_chain));
+  gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
+
+  enc->srcpad =
+      gst_pad_new_from_static_template (&gst_aac_enc_src_template, "src");
+  gst_pad_use_fixed_caps (enc->srcpad);
+  gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
+
+#ifndef GST_DISABLE_GST_DEBUG
+  gst_framed_audio_enc_init (&enc->enc, GST_ELEMENT (enc), aac_enc);
+#else
+  gst_framed_audio_enc_init (&enc->enc, GST_ELEMENT (enc), NULL);
+#endif
+
+  gst_aac_enc_reset (enc);
+}
+
+static void
+gst_aac_enc_reset (GstAACEnc * enc)
+{
+  gst_framed_audio_enc_reset (&enc->enc);
+  if (enc->encoder)
+    EnAACPlus_Enc_Delete (enc->encoder);
+  enc->encoder = NULL;
+  g_free (enc->buffer);
+  enc->buffer = NULL;
+}
+
+static void
+gst_aac_enc_finalize (GObject * object)
+{
+  GstAACEnc *enc = (GstAACEnc *) object;
+
+  gst_framed_audio_enc_finalize (&enc->enc);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_aac_enc_setup_encoder (GstAACEnc * enc)
+{
+  AACPLUS_ENC_CONFIG enc_params;
+  AACPLUS_ENC_MODE mode;
+  gint rate, channels;
+  guint maxbitrate;
+
+  rate = enc->rate;
+  channels = enc->channels;
+
+  /* only up to 2 channels supported */
+  enc_params.sampleRate = rate;
+  enc_params.bitRate = enc->bitrate;
+  enc_params.nChannels = channels;
+  enc_params.aac_tools = USE_ALL;
+  enc_params.pcm_mode = 16;
+  enc_params.format = enc->format;
+
+  /* check, warn and correct if the max bitrate for the given samplerate is
+   * exceeded. Maximum of 6144 bit for a channel */
+  maxbitrate =
+      (guint) (6144.0 * (gdouble) rate / (gdouble) 1024.0 + .5) * channels;
+  if (enc_params.bitRate > maxbitrate) {
+    GST_ELEMENT_INFO (enc, RESOURCE, SETTINGS, (NULL),
+        ("bitrate %d exceeds maximum allowed bitrate of %d for samplerate %d "
+            "and %d channels.  Setting bitrate to %d",
+            enc_params.bitRate, maxbitrate, rate, channels, maxbitrate));
+    enc_params.bitRate = maxbitrate;
+  }
+
+  /* set up encoder */
+  if (enc->encoder)
+    EnAACPlus_Enc_Delete (enc->encoder);
+
+  /* only these profiles are really known to and supported by codec */
+  switch (enc->profile) {
+    case AAC_PROFILE_LC:
+      mode = MODE_AACLC;
+      break;
+    case AAC_PROFILE_HE:
+      mode = MODE_EAACPLUS;
+      break;
+    case AAC_PROFILE_AUTO:
+      mode = MODE_AUTO;
+      break;
+    default:
+      mode = MODE_AACLC;
+      g_assert_not_reached ();
+      break;
+  }
+  enc->encoder = EnAACPlus_Enc_Create (&enc_params, mode);
+
+  if (!enc->encoder)
+    goto setup_failed;
+
+  /* query and setup params,
+   * also set up some buffers for fancy HE */
+  EnAACPlus_Enc_GetSetParam (enc->encoder, &enc->info);
+
+#define DUMP_FIELD(f)  \
+  GST_DEBUG_OBJECT (enc, "encoder info: " G_STRINGIFY (f) " = %d", enc->info.f);
+
+  DUMP_FIELD (InBufSize);
+  DUMP_FIELD (OutBufSize);
+  DUMP_FIELD (Frame_Size);
+  DUMP_FIELD (writeOffset);
+  DUMP_FIELD (InBufSize);
+
+  enc->raw_frame_size = enc->info.Frame_Size;
+  enc->codec_frame_size = enc->info.OutBufSize;
+  enc->frame_duration =
+      GST_FRAMES_TO_CLOCK_TIME (enc->raw_frame_size / enc->channels / 2,
+      enc->rate);
+
+  g_free (enc->buffer);
+  /* safety margin */
+  enc->buffer = g_malloc (enc->info.InBufSize * 2);
+
+  return TRUE;
+
+  /* ERRORS */
+setup_failed:
+  {
+    GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL), (NULL));
+    return FALSE;
+  }
+}
+
+static gint
+gst_aac_enc_rate_idx (gint rate)
+{
+  static int rates[] = {
+    96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
+    8000, 7350
+  };
+  guint i;
+
+  for (i = 0; i < G_N_ELEMENTS (rates); ++i)
+    if (rates[i] == rate)
+      return i;
+
+  return 0xF;
+}
+
+static gboolean
+gst_aac_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+  GstAACEnc *enc;
+  gboolean ret = TRUE;
+  GstStructure *s;
+  GstBuffer *buf = NULL;
+  gint rate, channels;
+
+  enc = GST_AAC_ENC (GST_PAD_PARENT (pad));
+
+  /* extract stream properties */
+  s = gst_caps_get_structure (caps, 0);
+
+  if (!s)
+    goto refuse_caps;
+
+  ret = gst_structure_get_int (s, "rate", &rate);
+  ret &= gst_structure_get_int (s, "channels", &channels);
+
+  if (!ret)
+    goto refuse_caps;
+
+  enc->rate = rate;
+  enc->channels = channels;
+
+  /* NOTE:
+   * - codec only supports LC or HE (= LC + SBR etc)
+   * - HE has (more) restrictive samplerate/channels/bitrate combination
+   * - AUTO makes codec select between LC or HE (depending on settings)
+   */
+
+  gst_aac_enc_setup_encoder (enc);
+  if (!enc->encoder)
+    return FALSE;
+
+  /* HE iff writeOffset <> 0 iff Frame_Size <> 1024 * 2 * channels */
+  if (enc->info.writeOffset)
+    rate /= 2;
+
+  /* create codec_data if raw output */
+  if (enc->format == RAW) {
+    gint rate_idx;
+    guint8 *data;
+
+    buf = gst_buffer_new_and_alloc (5);
+    data = GST_BUFFER_DATA (buf);
+    rate_idx = gst_aac_enc_rate_idx (rate);
+
+    GST_DEBUG_OBJECT (enc, "codec_data: profile=%d, sri=%d, channels=%d",
+        enc->profile, rate_idx, enc->channels);
+
+    /* always write LC profile, and use implicit signaling for HE SBR */
+    data[0] = ((2 & 0x1F) << 3) | ((rate_idx & 0xE) >> 1);
+    data[1] = ((rate_idx & 0x1) << 7);
+    if (rate_idx != 0x0F) {
+      data[1] |= ((channels & 0xF) << 3);
+      GST_BUFFER_SIZE (buf) = 2;
+    } else {
+      gint srate;
+
+      srate = rate << 7;
+      data[1] |= ((srate >> 24) & 0xFF);
+      data[2] = ((srate >> 16) & 0xFF);
+      data[3] = ((srate >> 8) & 0xFF);
+      data[4] = (srate & 0xFF);
+      data[4] |= ((channels & 0xF) << 3);
+      GST_BUFFER_SIZE (buf) = 5;
+    }
+  }
+
+  /* fix some in src template */
+  caps = gst_caps_copy (gst_pad_get_pad_template_caps (enc->srcpad));
+  gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate,
+      "channels", G_TYPE_INT, channels, NULL);
+  if (buf) {
+    gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, buf, NULL);
+    gst_buffer_unref (buf);
+  }
+  ret = gst_pad_set_caps (enc->srcpad, caps);
+  gst_caps_unref (caps);
+
+  return ret;
+
+  /* ERRORS */
+refuse_caps:
+  {
+    GST_WARNING_OBJECT (enc, "refused caps %" GST_PTR_FORMAT, caps);
+    return FALSE;
+  }
+}
+
+static gint
+gst_aac_enc_get_data (GstElement * element, const guint8 * in, guint8 * out,
+    GstDtxDecision * dtx)
+{
+  GstAACEnc *enc;
+  gint res;
+  gint offset;
+  UWord32 used, encoded;
+  Word8 *inbuffer;
+
+  enc = GST_AAC_ENC_CAST (element);
+
+  offset = enc->info.writeOffset;
+  if (offset) {
+    memcpy (enc->buffer + offset, in, enc->raw_frame_size);
+    inbuffer = (Word8 *) enc->buffer;
+  } else {
+    inbuffer = (Word8 *) in;
+  }
+
+  res = EnAACPlus_Enc_Encode (enc->encoder, &enc->info, inbuffer, &used,
+      (UWord8 *) out, &encoded);
+
+  if (offset) {
+    memcpy (enc->buffer, enc->buffer + used, offset);
+  }
+
+  return res == 0 ? encoded : -1;
+}
+
+/* set parameters */
+#define AUDIO_SAMPLE_RATE   ((GST_AAC_ENC (enc->element))->rate)
+#define RAW_FRAME_SIZE      ((GST_AAC_ENC (enc->element))->raw_frame_size)
+/* safe maximum frame size */
+#define CODEC_FRAME_SIZE    ((GST_AAC_ENC (enc->element))->codec_frame_size)
+/* do not set variable frame;
+ * this will make every frame act as a silence frame and force output */
+/* #define CODEC_FRAME_VARIABLE 1 */
+#define FRAME_DURATION      ((GST_AAC_ENC (enc->element))->frame_duration)
+#define codec_get_data(enc, in, out, dtx)  \
+    gst_aac_enc_get_data (enc, in, out, dtx)
+
+/* and include code */
+#include "gstframedaudioenc.c"
+
+static GstFlowReturn
+gst_aac_enc_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstAACEnc *enc;
+
+  enc = GST_AAC_ENC (GST_PAD_PARENT (pad));
+
+  if (G_UNLIKELY (enc->encoder == NULL))
+    goto not_negotiated;
+
+  return gst_framed_audio_enc_chain (&enc->enc, buf, enc->srcpad, &enc->cnpad);
+
+  /* ERRORS */
+not_negotiated:
+  {
+    GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
+        ("format wasn't negotiated before chain function"));
+    gst_buffer_unref (buf);
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+}
+
+static void
+gst_aac_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAACEnc *enc;
+
+  enc = GST_AAC_ENC (object);
+
+  switch (prop_id) {
+    case PROP_BITRATE:
+      enc->bitrate = g_value_get_int (value);
+      break;
+    case PROP_PROFILE:
+      enc->profile = g_value_get_enum (value);
+      break;
+    case PROP_FORMAT:
+      enc->format = g_value_get_enum (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_aac_enc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAACEnc *enc;
+
+  enc = GST_AAC_ENC (object);
+
+  switch (prop_id) {
+    case PROP_BITRATE:
+      g_value_set_int (value, enc->bitrate);
+      break;
+    case PROP_PROFILE:
+      g_value_set_enum (value, enc->profile);
+      break;
+    case PROP_FORMAT:
+      g_value_set_enum (value, enc->format);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_aac_enc_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+  GstAACEnc *enc = GST_AAC_ENC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+  if (ret == GST_STATE_CHANGE_FAILURE)
+    return ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_aac_enc_reset (enc);
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+
+  if (!gst_element_register (plugin, "nokiaaacenc", GST_RANK_SECONDARY,
+          GST_TYPE_AAC_ENC))
+    return FALSE;
+
+  return TRUE;
+}
+
+/* this is the structure that gst-register looks for
+ * so keep the name plugin_desc, or you cannot get your plug-in registered */
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "nokiaaacenc",
+    "Nokia AAC MCC codec",
+    plugin_init, VERSION, "Proprietary", "gst-nokia-speech", "")
+
+EXPORT_C GstPluginDesc* _GST_PLUGIN_DESC()
+   {
+      return &gst_plugin_desc;
+   }
+