/* 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"

GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE (
  "sink",
  GST_PAD_SINK,
  GST_PAD_ALWAYS,
  GST_STATIC_CAPS ("audio/mpeg, "
      "systemstream = (bool) FALSE, "
      "mpegversion = { (int) 2, (int) 4 }"
  )
);

GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE (
  "src",
  GST_PAD_SRC,
  GST_PAD_ALWAYS,
  GST_STATIC_CAPS ("audio/x-raw-int, "
      "endianness = (int) BYTE_ORDER, "
      "signed = (bool) TRUE, "
      "width = (int) { 16, 24, 32 }, "
      "depth = (int) { 16, 24, 32 }, "
      "rate = (int) [ 8000, 96000 ], "
      "channels = (int) [ 1, 6 ]; "
    "audio/x-raw-float, "
      "endianness = (int) BYTE_ORDER, "
      "depth = (int) { 32, 64 }, "
      "rate = (int) [ 8000, 96000 ], "
      "channels = (int) [ 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,
				       const GstCaps *caps);
static GstPadLinkReturn
                gst_faad_srcconnect   (GstPad       *pad,
				       const GstCaps *caps);
static GstCaps *gst_faad_srcgetcaps   (GstPad       *pad);
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_faad_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_static_pad_template_get (&src_template));
  gst_element_class_add_pad_template (element_class,
	gst_static_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;

  GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE);

  faad->sinkpad = gst_pad_new_from_template (
	gst_static_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_static_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);

  /* This was originally intended as a getcaps() function, but
   * in the end, we needed a srcconnect() function, so this is
   * not really useful. However, srcconnect() uses it, so it is
   * still there... */
  /*gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);*/
}

static GstPadLinkReturn
gst_faad_sinkconnect (GstPad  *pad,
		      const GstCaps *caps)
{
  /* 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)
{
  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_simple ("audio/x-raw-int",
	    "signed", G_TYPE_BOOLEAN, TRUE,
	    "width",  G_TYPE_INT,     16,
	    "depth",  G_TYPE_INT,     16,
	    NULL);
        break;
      case FAAD_FMT_24BIT:
        caps = gst_caps_new_simple ("audio/x-raw-int",
	    "signed", G_TYPE_BOOLEAN, TRUE,
	    "width",  G_TYPE_INT,     24,
	    "depth",  G_TYPE_INT,     24,
	    NULL);
        break;
      case FAAD_FMT_32BIT:
        caps = gst_caps_new_simple ("audio/x-raw-int",
	    "signed", G_TYPE_BOOLEAN, TRUE,
	    "width",  G_TYPE_INT,     32,
	    "depth",  G_TYPE_INT,     32,
	    NULL);
        break;
      case FAAD_FMT_FLOAT:
        caps = gst_caps_new_simple ("audio/x-raw-float",
	    "depth",  G_TYPE_INT,     32,
	    NULL);
        break;
      case FAAD_FMT_DOUBLE:
        caps = gst_caps_new_simple ("audio/x-raw-float",
	    "depth",  G_TYPE_INT,     64,
	    NULL);
        break;
      default:
        caps = gst_caps_new_empty ();
        break;
    }

    if (!gst_caps_is_empty (caps)) {
      GstStructure *structure = gst_caps_get_structure (caps, 0);

      if (faad->samplerate != -1) {
	gst_structure_set (structure, 
	    "rate", G_TYPE_INT, faad->samplerate, 
	    NULL);
      } else {
	gst_structure_set (structure, 
	    "rate", GST_TYPE_INT_RANGE, 8000, 96000, 
	    NULL);
      }

      if (faad->channels != -1) {
	gst_structure_set (structure, 
	    "channels", G_TYPE_INT, faad->channels, 
	    NULL);
      } else {
	gst_structure_set (structure, 
	    "channels", GST_TYPE_INT_RANGE, 1, 6, 
	    NULL);
      }

      gst_structure_set (structure, 
	  "endianness", G_TYPE_INT, G_BYTE_ORDER, 
	  NULL);
    }

    return caps;
  }

  return GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad));
}

static GstPadLinkReturn
gst_faad_srcconnect (GstPad  *pad,
		     const GstCaps *caps)
{
  GstStructure *structure;
  const gchar *mimetype;
  gint fmt = 0;
  gint depth;
  GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));

  if (!faad->handle ||
      (faad->samplerate == -1 || faad->channels == -1)) {
    return GST_PAD_LINK_DELAYED;
  }

  structure = gst_caps_get_structure (caps, 0);
  mimetype = gst_structure_get_name (structure);
  
  if (!strcmp (mimetype, "audio/x-raw-int")) {
    gint width;

    if (!gst_structure_get_int (structure, "depth", &depth) ||
	!gst_structure_get_int (structure, "width", &width))
      return GST_PAD_LINK_REFUSED;
    if (depth != width)
      return GST_PAD_LINK_REFUSED;

    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_structure_get_int (structure, "depth", &depth))
      return GST_PAD_LINK_REFUSED;

    switch (depth) {
      case 32:
	fmt = FAAD_FMT_FLOAT;
	break;
      case 64:
	fmt = FAAD_FMT_DOUBLE;
          break;
    }
  }

  if (fmt) {
    GstCaps *newcaps, *intersect;
    faacDecConfiguration *conf;

    conf = faacDecGetCurrentConfiguration (faad->handle);
    conf->outputFormat = fmt;
    faacDecSetConfiguration (faad->handle, conf);
    /* FIXME: handle return value, how? */

    newcaps = gst_faad_srcgetcaps (pad);
    g_assert (gst_caps_is_fixed (newcaps));
    intersect = gst_caps_intersect (newcaps, caps);
    gst_caps_free (newcaps);
    if (!gst_caps_is_empty (intersect)) {
      gst_caps_free (intersect);
      faad->bps = depth / 8;
      return GST_PAD_LINK_OK;
    }
    gst_caps_free (intersect);
  }

  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 (faad, CORE, NEGOTIATION, (NULL), (NULL));
      gst_buffer_unref (buf);
      return;
    }
  }

  out = faacDecDecode (faad->handle, &info,
		       GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
  if (info.error) {
    GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
		       ("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 (faad, CORE, NEGOTIATION, (NULL), (NULL));
      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_PACKAGE,
  GST_ORIGIN
)