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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtppcmupay.h"
/* elementfactory information */
static const GstElementDetails gst_rtp_pcmu_pay_details =
GST_ELEMENT_DETAILS ("RTP PCMU payloader",
"Codec/Payloader/Network",
"Payload-encodes PCMU audio into a RTP packet",
"Edgard Lima <edgard.lima@indt.org.br>");
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) 8000, " "encoding-name = (string) \"PCMU\"")
);
static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_pcmu_pay_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_pcmu_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_pcmu_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_pcmu_pay_details);
}
static void
gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
{
GstBaseRTPPayloadClass *gstbasertppayload_class;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
}
static void
gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay);
GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
/* tell basertpaudiopayload that this is a sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
/* octet-per-sample is 1 for PCM */
gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
}
static gboolean
gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gboolean res;
payload->pt = GST_RTP_PAYLOAD_PCMU;
gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
res = gst_basertppayload_set_outcaps (payload, NULL);
return res;
}
gboolean
gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtppcmupay",
GST_RANK_NONE, GST_TYPE_RTP_PCMU_PAY);
}
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