/* GStreamer

 * Copyright (C) <2007> Wim Taymans <wim.taymans@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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:element-rtpL24depay

 * @see_also: rtpL24pay

 *

 * Extract raw audio from RTP packets according to RFC 3190, section 4.

 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt

 *

 * <refsect2>

 * <title>Example pipeline</title>

 * |[

 * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink

 * ]| This example pipeline will depayload an RTP raw audio stream. Refer to

 * the rtpL24pay example to create the RTP stream.

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <string.h>

#include <stdlib.h>

#include <gst/audio/audio.h>

#include "gstrtpL24depay.h"

#include "gstrtpchannels.h"

#include "gstrtputils.h"

GST_DEBUG_CATEGORY_STATIC (rtpL24depay_debug);

#define GST_CAT_DEFAULT (rtpL24depay_debug)

static GstStaticPadTemplate gst_rtp_L24_depay_src_template =

GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-raw, "

        "format = (string) S24BE, "

        "layout = (string) interleaved, "

        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")

    );

static GstStaticPadTemplate gst_rtp_L24_depay_sink_template =

GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("application/x-rtp, "

        "media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], "

        "encoding-name = (string) \"L24\"")

    );

#define gst_rtp_L24_depay_parent_class parent_class

G_DEFINE_TYPE (GstRtpL24Depay, gst_rtp_L24_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);

static gboolean gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload,

    GstCaps * caps);

static GstBuffer *gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload,

    GstRTPBuffer * rtp);

static void

gst_rtp_L24_depay_class_init (GstRtpL24DepayClass * klass)

{

  GstElementClass *gstelement_class;

  GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;

  gstelement_class = (GstElementClass *) klass;

  gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;

  gstrtpbasedepayload_class->set_caps = gst_rtp_L24_depay_setcaps;

  gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L24_depay_process;

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_L24_depay_src_template);

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_L24_depay_sink_template);

  gst_element_class_set_static_metadata (gstelement_class,

      "RTP audio depayloader", "Codec/Depayloader/Network/RTP",

      "Extracts raw 24-bit audio from RTP packets",

      "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>,"

      "David Holroyd <dave@badgers-in-foil.co.uk>");

  GST_DEBUG_CATEGORY_INIT (rtpL24depay_debug, "rtpL24depay", 0,

      "Raw Audio RTP Depayloader");

}

static void

gst_rtp_L24_depay_init (GstRtpL24Depay * rtpL24depay)

{

}

static gint

gst_rtp_L24_depay_parse_int (GstStructure * structure, const gchar * field,

    gint def)

{

  const gchar *str;

  gint res;

  if ((str = gst_structure_get_string (structure, field)))

    return atoi (str);

  if (gst_structure_get_int (structure, field, &res))

    return res;

  return def;

}

static gboolean

gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)

{

  GstStructure *structure;

  GstRtpL24Depay *rtpL24depay;

  gint clock_rate, payload;

  gint channels;

  GstCaps *srccaps;

  gboolean res;

  const gchar *channel_order;

  const GstRTPChannelOrder *order;

  GstAudioInfo *info;

  rtpL24depay = GST_RTP_L24_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  payload = 96;

  gst_structure_get_int (structure, "payload", &payload);

  /* no fixed mapping, we need clock-rate */

  channels = 0;

  clock_rate = 0;

  /* caps can overwrite defaults */

  clock_rate =

      gst_rtp_L24_depay_parse_int (structure, "clock-rate", clock_rate);

  if (clock_rate == 0)

    goto no_clockrate;

  channels =

      gst_rtp_L24_depay_parse_int (structure, "encoding-params", channels);

  if (channels == 0) {

    channels = gst_rtp_L24_depay_parse_int (structure, "channels", channels);

    if (channels == 0) {

      /* channels defaults to 1 otherwise */

      channels = 1;

    }

  }

  depayload->clock_rate = clock_rate;

  info = &rtpL24depay->info;

  gst_audio_info_init (info);

  info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S24BE);

  info->rate = clock_rate;

  info->channels = channels;

  info->bpf = (info->finfo->width / 8) * channels;

  /* add channel positions */

  channel_order = gst_structure_get_string (structure, "channel-order");

  order = gst_rtp_channels_get_by_order (channels, channel_order);

  rtpL24depay->order = order;

  if (order) {

    memcpy (info->position, order->pos,

        sizeof (GstAudioChannelPosition) * channels);

    gst_audio_channel_positions_to_valid_order (info->position, info->channels);

  } else {

    GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,

        (NULL), ("Unknown channel order '%s' for %d channels",

            GST_STR_NULL (channel_order), channels));

    /* create default NONE layout */

    gst_rtp_channels_create_default (channels, info->position);

  }

  srccaps = gst_audio_info_to_caps (info);

  res = gst_pad_set_caps (depayload->srcpad, srccaps);

  gst_caps_unref (srccaps);

  return res;

  /* ERRORS */

no_clockrate:

  {

    GST_ERROR_OBJECT (depayload, "no clock-rate specified");

    return FALSE;

  }

}

static GstBuffer *

gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)

{

  GstRtpL24Depay *rtpL24depay;

  GstBuffer *outbuf;

  gint payload_len;

  gboolean marker;

  rtpL24depay = GST_RTP_L24_DEPAY (depayload);

  payload_len = gst_rtp_buffer_get_payload_len (rtp);

  if (payload_len <= 0)

    goto empty_packet;

  GST_DEBUG_OBJECT (rtpL24depay, "got payload of %d bytes", payload_len);

  outbuf = gst_rtp_buffer_get_payload_buffer (rtp);

  marker = gst_rtp_buffer_get_marker (rtp);

  if (marker) {

    /* mark talk spurt with RESYNC */

    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);

  }

  outbuf = gst_buffer_make_writable (outbuf);

  if (outbuf) {

    gst_rtp_drop_non_audio_meta (rtpL24depay, outbuf);

  }

  if (rtpL24depay->order &&

      !gst_audio_buffer_reorder_channels (outbuf,

          rtpL24depay->info.finfo->format, rtpL24depay->info.channels,

          rtpL24depay->info.position, rtpL24depay->order->pos)) {

    goto reorder_failed;

  }

  return outbuf;

  /* ERRORS */

empty_packet:

  {

    GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,

        ("Empty Payload."), (NULL));

    return NULL;

  }

reorder_failed:

  {

    GST_ELEMENT_ERROR (rtpL24depay, STREAM, DECODE,

        ("Channel reordering failed."), (NULL));

    return NULL;

  }

}

gboolean

gst_rtp_L24_depay_plugin_init (GstPlugin * plugin)

{

  return gst_element_register (plugin, "rtpL24depay",

      GST_RANK_SECONDARY, GST_TYPE_RTP_L24_DEPAY);

}