--- a/gst_nokia_speech/gstaacenc.c	Fri Apr 16 15:15:52 2010 +0300
+++ /dev/null	Thu Jan 01 00:00:00 1970 +0000
@@ -1,603 +0,0 @@
-/* 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;
-   }
-