/* GStreamer

 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>

 * Copyright (C) <2005> Edgard Lima <edgard.lima@gmail.com>

 * Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.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.

 */

#ifdef HAVE_CONFIG_H

#  include "config.h"

#endif

#include <stdlib.h>

#include <string.h>

#include <gst/rtp/gstrtpbuffer.h>

#include "gstrtppcmupay.h"

static GstStaticPadTemplate gst_rtp_pcmu_pay_sink_template =

GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-mulaw, channels=(int)1, rate=(int)8000")

    );

static GstStaticPadTemplate gst_rtp_pcmu_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) " GST_RTP_PAYLOAD_PCMU_STRING ", "

        "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"; "

        "application/x-rtp, "

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

        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "

        "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"PCMU\"")

    );

static gboolean gst_rtp_pcmu_pay_setcaps (GstRTPBasePayload * payload,

    GstCaps * caps);

#define gst_rtp_pcmu_pay_parent_class parent_class

G_DEFINE_TYPE (GstRtpPcmuPay, gst_rtp_pcmu_pay,

    GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);

static void

gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)

{

  GstElementClass *gstelement_class;

  GstRTPBasePayloadClass *gstrtpbasepayload_class;

  gstelement_class = (GstElementClass *) klass;

  gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_pcmu_pay_sink_template);

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_pcmu_pay_src_template);

  gst_element_class_set_static_metadata (gstelement_class, "RTP PCMU payloader",

      "Codec/Payloader/Network/RTP",

      "Payload-encodes PCMU audio into a RTP packet",

      "Edgard Lima <edgard.lima@gmail.com>");

  gstrtpbasepayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;

}

static void

gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay)

{

  GstRTPBaseAudioPayload *rtpbaseaudiopayload;

  rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmupay);

  GST_RTP_BASE_PAYLOAD (rtppcmupay)->pt = GST_RTP_PAYLOAD_PCMU;

  GST_RTP_BASE_PAYLOAD (rtppcmupay)->clock_rate = 8000;

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

  gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);

  /* octet-per-sample is 1 for PCM */

  gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, 1);

}

static gboolean

gst_rtp_pcmu_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)

{

  gboolean res;

  gst_rtp_base_payload_set_options (payload, "audio",

      payload->pt != GST_RTP_PAYLOAD_PCMU, "PCMU", 8000);

  res = gst_rtp_base_payload_set_outcaps (payload, NULL);

  return res;

}

gboolean

gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)

{

  return gst_element_register (plugin, "rtppcmupay",

      GST_RANK_SECONDARY, GST_TYPE_RTP_PCMU_PAY);

}