/* GStreamer
 * Copyright (C) <2009> Digitro Corporation
 *  @author: Tiago Katcipis <tiago.katcipis@gmail.com>
 *
 * 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 <gst/rtp/gstrtpbuffer.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtpg723depay.h"

/* references:
 * RFC 3551 (4.5.3)
 */

#define GST_G723_RTP_DEPAY_FRAME_SIZE 24
#define GST_G723_RTP_DEPAY_SID_SIZE 4

/* elementfactory information */
static const GstElementDetails gst_rtp_g723depay_details =
GST_ELEMENT_DETAILS ("RTP G.723 depayloader",
    "Codec/Depayloader/Network",
    "Extracts G.723 audio from RTP packets (RFC 3551)",
    "Tiago Katcipis <tiagokatcipis@gmail.com>");

enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  ARG_0
};

/* input is an RTP packet
 *
 */
static GstStaticPadTemplate gst_rtp_g723_depay_sink_template =
    GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp, "
        "media = (string) \"audio\", "
        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
        "clock-rate = (int) 8000, "
        "encoding-name = (string) \"G723\"; "
        "application/x-rtp, "
        "media = (string) \"audio\", "
        "payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
        "clock-rate = (int) 8000")
    );

static GstStaticPadTemplate gst_rtp_g723_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/G723, " "channels = (int) 1," "rate = (int) 8000")
    );

static gboolean gst_rtp_g723_depay_setcaps (GstBaseRTPDepayload * depayload,
    GstCaps * caps);
static GstBuffer *gst_rtp_g723_depay_process (GstBaseRTPDepayload * depayload,
    GstBuffer * buf);

GST_BOILERPLATE (GstRtpG723Depay, gst_rtp_g723_depay, GstBaseRTPDepayload,
    GST_TYPE_BASE_RTP_DEPAYLOAD);

static void
gst_rtp_g723_depay_base_init (gpointer klass)
{
  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_rtp_g723_depay_src_template));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_rtp_g723_depay_sink_template));

  gst_element_class_set_details (element_class, &gst_rtp_g723depay_details);
}

static void
gst_rtp_g723_depay_class_init (GstRtpG723DepayClass * klass)
{
  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;

  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;

  parent_class = g_type_class_peek_parent (klass);

  gstbasertpdepayload_class->process = gst_rtp_g723_depay_process;
  gstbasertpdepayload_class->set_caps = gst_rtp_g723_depay_setcaps;
}

static void
gst_rtp_g723_depay_init (GstRtpG723Depay * rtpg723depay,
    GstRtpG723DepayClass * klass)
{
  GstBaseRTPDepayload *depayload;

  depayload = GST_BASE_RTP_DEPAYLOAD (rtpg723depay);

  gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}

static gboolean
gst_rtp_g723_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
  GstStructure *structure;
  GstCaps *srccaps;
  GstRtpG723Depay *rtpg723depay;
  const gchar *params;
  gint clock_rate, channels;
  gboolean ret;

  rtpg723depay = GST_RTP_G723_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  if (!(params = gst_structure_get_string (structure, "encoding-params")))
    channels = 1;
  else {
    channels = atoi (params);
  }

  if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
    clock_rate = 8000;

  if (channels != 1)
    goto wrong_channels;

  if (clock_rate != 8000)
    goto wrong_clock_rate;

  depayload->clock_rate = clock_rate;

  srccaps = gst_caps_new_simple ("audio/G723",
      "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, clock_rate, NULL);
  ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
  gst_caps_unref (srccaps);

  return ret;

  /* ERRORS */
wrong_channels:
  {
    GST_DEBUG_OBJECT (rtpg723depay, "expected 1 channel, got %d",
        rtpg723depay->channels);
    return FALSE;
  }
wrong_clock_rate:
  {
    GST_DEBUG_OBJECT (rtpg723depay, "expected 8000 clock-rate, got %d",
        clock_rate);
    return FALSE;
  }
}


static GstBuffer *
gst_rtp_g723_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
  GstRtpG723Depay *rtpg723depay;
  GstBuffer *outbuf = NULL;
  gint payload_len;
  gboolean marker;

  rtpg723depay = GST_RTP_G723_DEPAY (depayload);

  payload_len = gst_rtp_buffer_get_payload_len (buf);

  /* At least 4 bytes (SID from G723) */
  if (payload_len < GST_G723_RTP_DEPAY_SID_SIZE) {
    GST_ELEMENT_WARNING (rtpg723depay, STREAM, DECODE,
        (NULL), ("G723 RTP payload too small (%d)", payload_len));
    goto bad_packet;
  }

  GST_DEBUG_OBJECT (rtpg723depay, "payload len %d", payload_len);

  if ((payload_len % GST_G723_RTP_DEPAY_FRAME_SIZE) == GST_G723_RTP_DEPAY_SID_SIZE) {
    GST_DEBUG_OBJECT (rtpg723depay, "G723 payload contains SID frame");
  }

  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
  marker = gst_rtp_buffer_get_marker (buf);

  if (marker) {
    /* marker bit starts talkspurt */
    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
  }

  GST_DEBUG ("gst_rtp_g723_depay_chain: pushing buffer of size %d", GST_BUFFER_SIZE (outbuf));
  return outbuf;

  /* ERRORS */
bad_packet:
  {
    /* no fatal error */
    return NULL;
  }
}

/*Plugin init functions*/
static gboolean
plugin_init (GstPlugin * plugin)
{
  return gst_element_register (plugin, "rtpg723depay", GST_RANK_NONE, gst_rtp_g723_depay_get_type());
}

GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
    GST_VERSION_MINOR,
    "rtpg723depay",
    "Extracts G.723 audio from RTP packets",
    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
