/* 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-rtpL24pay

 * @see_also: rtpL24depay

 *

 * Payload raw 24-bit audio into 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 -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink

 * ]| This example pipeline will payload raw audio. Refer to

 * the rtpL24depay example to depayload and play the RTP stream.

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#  include "config.h"

#endif

#include <string.h>

#include <gst/audio/audio.h>

#include <gst/rtp/gstrtpbuffer.h>

#include "gstrtpL24pay.h"

#include "gstrtpchannels.h"

GST_DEBUG_CATEGORY_STATIC (rtpL24pay_debug);

#define GST_CAT_DEFAULT (rtpL24pay_debug)

static GstStaticPadTemplate gst_rtp_L24_pay_sink_template =

GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    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_pay_src_template =

    GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("application/x-rtp, "

        "media = (string) \"audio\", "

        "payload = (int) [ 96, 127 ], "

        "clock-rate = (int) [ 1, MAX ], "

        "encoding-name = (string) \"L24\", " "channels = (int) [ 1, MAX ];")

    );

static gboolean gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload,

    GstCaps * caps);

static GstCaps *gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload,

    GstPad * pad, GstCaps * filter);

static GstFlowReturn

gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,

    GstBuffer * buffer);

#define gst_rtp_L24_pay_parent_class parent_class

G_DEFINE_TYPE (GstRtpL24Pay, gst_rtp_L24_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);

static void

gst_rtp_L24_pay_class_init (GstRtpL24PayClass * klass)

{

  GstElementClass *gstelement_class;

  GstRTPBasePayloadClass *gstrtpbasepayload_class;

  gstelement_class = (GstElementClass *) klass;

  gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

  gstrtpbasepayload_class->set_caps = gst_rtp_L24_pay_setcaps;

  gstrtpbasepayload_class->get_caps = gst_rtp_L24_pay_getcaps;

  gstrtpbasepayload_class->handle_buffer = gst_rtp_L24_pay_handle_buffer;

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_L24_pay_src_template);

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_L24_pay_sink_template);

  gst_element_class_set_static_metadata (gstelement_class,

      "RTP audio payloader", "Codec/Payloader/Network/RTP",

      "Payload-encode Raw 24-bit audio into RTP packets (RFC 3190)",

      "Wim Taymans <wim.taymans@gmail.com>,"

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

  GST_DEBUG_CATEGORY_INIT (rtpL24pay_debug, "rtpL24pay", 0,

      "L24 RTP Payloader");

}

static void

gst_rtp_L24_pay_init (GstRtpL24Pay * rtpL24pay)

{

  GstRTPBaseAudioPayload *rtpbaseaudiopayload;

  rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL24pay);

  /* tell rtpbaseaudiopayload that this is a sample based codec */

  gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);

}

static gboolean

gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)

{

  GstRtpL24Pay *rtpL24pay;

  gboolean res;

  gchar *params;

  GstAudioInfo *info;

  const GstRTPChannelOrder *order;

  GstRTPBaseAudioPayload *rtpbaseaudiopayload;

  rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);

  rtpL24pay = GST_RTP_L24_PAY (basepayload);

  info = &rtpL24pay->info;

  gst_audio_info_init (info);

  if (!gst_audio_info_from_caps (info, caps))

    goto invalid_caps;

  order = gst_rtp_channels_get_by_pos (info->channels, info->position);

  rtpL24pay->order = order;

  gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L24",

      info->rate);

  params = g_strdup_printf ("%d", info->channels);

  if (!order && info->channels > 2) {

    GST_ELEMENT_WARNING (rtpL24pay, STREAM, DECODE,

        (NULL), ("Unknown channel order for %d channels", info->channels));

  }

  if (order && order->name) {

    res = gst_rtp_base_payload_set_outcaps (basepayload,

        "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,

        info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);

  } else {

    res = gst_rtp_base_payload_set_outcaps (basepayload,

        "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,

        info->channels, NULL);

  }

  g_free (params);

  /* octet-per-sample is 3 * channels for L24 */

  gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,

      3 * info->channels);

  return res;

  /* ERRORS */

invalid_caps:

  {

    GST_DEBUG_OBJECT (rtpL24pay, "invalid caps");

    return FALSE;

  }

}

static GstCaps *

gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,

    GstCaps * filter)

{

  GstCaps *otherpadcaps;

  GstCaps *caps;

  caps = gst_pad_get_pad_template_caps (pad);

  otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);

  if (otherpadcaps) {

    if (!gst_caps_is_empty (otherpadcaps)) {

      GstStructure *structure;

      gint channels;

      gint rate;

      structure = gst_caps_get_structure (otherpadcaps, 0);

      caps = gst_caps_make_writable (caps);

      if (gst_structure_get_int (structure, "channels", &channels)) {

        gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);

      }

      if (gst_structure_get_int (structure, "clock-rate", &rate)) {

        gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);

      }

    }

    gst_caps_unref (otherpadcaps);

  }

  if (filter) {

    GstCaps *tcaps = caps;

    caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (tcaps);

  }

  return caps;

}

static GstFlowReturn

gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,

    GstBuffer * buffer)

{

  GstRtpL24Pay *rtpL24pay;

  rtpL24pay = GST_RTP_L24_PAY (basepayload);

  buffer = gst_buffer_make_writable (buffer);

  if (rtpL24pay->order &&

      !gst_audio_buffer_reorder_channels (buffer, rtpL24pay->info.finfo->format,

          rtpL24pay->info.channels, rtpL24pay->info.position,

          rtpL24pay->order->pos)) {

    return GST_FLOW_ERROR;

  }

  return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,

      buffer);

}

gboolean

gst_rtp_L24_pay_plugin_init (GstPlugin * plugin)

{

  return gst_element_register (plugin, "rtpL24pay",

      GST_RANK_SECONDARY, GST_TYPE_RTP_L24_PAY);

}