gstreamer/gst/rtp/gstrtpsirenpay.c

/*
 * Siren Payloader Gst Element
 *
 *   @author: Youness Alaoui <kakaroto@kakaroto.homelinux.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 "gstrtpsirenpay.h"
#include <gst/rtp/gstrtpbuffer.h>

/* elementfactory information */
static GstElementDetails gst_rtpsirenpay_details = {
  "RTP Payloader for Siren Audio",
  "Codec/Payloader/Network",
  "Packetize Siren audio streams into RTP packets",
  "Youness Alaoui <kakaroto@kakaroto.homelinux.net>"
};

GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
#define GST_CAT_DEFAULT (rtpsirenpay_debug)

static GstStaticPadTemplate gst_rtpsirenpay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
    );

static GstStaticPadTemplate gst_rtpsirenpay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp, "
        "media = (string) \"audio\", "
        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
        "clock-rate = (int) 16000, "
        "encoding-name = (string) \"SIREN\", "
        "bitrate = (string) \"16000\", " "dct-length = (int) 320")
    );

static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload,
    GstCaps * caps);

GST_BOILERPLATE (GstRTPSirenPay, gst_rtpsirenpay, GstBaseRTPAudioPayload,
    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);

static void
gst_rtpsirenpay_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_rtpsirenpay_sink_template));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_rtpsirenpay_src_template));
  gst_element_class_set_details (element_class, &gst_rtpsirenpay_details);
}

static void
gst_rtpsirenpay_class_init (GstRTPSirenPayClass * klass)
{
  GstBaseRTPPayloadClass *gstbasertppayload_class;

  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;

  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);

  gstbasertppayload_class->set_caps = gst_rtpsirenpay_setcaps;

  GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
      "siren audio RTP payloader");
}

static void
gst_rtpsirenpay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass)
{
  GstBaseRTPPayload *basertppayload;
  GstBaseRTPAudioPayload *basertpaudiopayload;

  basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);

  /* we don't set the payload type, it should be set by the application using
   * the pt property or the default 96 will be used */
  basertppayload->clock_rate = 16000;

  /* tell basertpaudiopayload that this is a frame based codec */
  gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
}

static gboolean
gst_rtpsirenpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
{
  GstRTPSirenPay *rtpsirenpay;
  GstBaseRTPAudioPayload *basertpaudiopayload;
  gint dct_length;
  GstStructure *structure;
  const char *payload_name;

  rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);

  structure = gst_caps_get_structure (caps, 0);

  gst_structure_get_int (structure, "dct-length", &dct_length);
  if (dct_length != 320)
    goto wrong_dct;

  payload_name = gst_structure_get_name (structure);
  if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
    goto wrong_caps;

  gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
      16000);
  /* set options for this frame based audio codec */
  gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);

  return gst_basertppayload_set_outcaps (basertppayload, NULL);

  /* ERRORS */
wrong_dct:
  {
    GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
        dct_length);
    return FALSE;
  }
wrong_caps:
  {
    GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
        payload_name);
    return FALSE;
  }
}

gboolean
gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
{
  return gst_element_register (plugin, "rtpsirenpay",
      GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY);
}