blob: 5931f85e214b51196c8e61210abd7b326e23396e [file] [log] [blame]
/* 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);
}