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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>
* Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.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 "gstrtpg726pay.h"
static const GstElementDetails gst_rtp_g726_pay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload-encodes G.726 audio into a RTP packet",
"Axis Communications <dev-gstreamer@axis.com>");
static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-adpcm, "
"channels = (int) 1, "
"rate = (int) 8000, "
"bitrate = (int) { 16000, 24000, 32000, 40000 }, "
"layout = (string) \"g726\"")
);
static GstStaticPadTemplate gst_rtp_g726_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_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\" } ")
);
static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_g726_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_g726_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_g726_pay_details);
}
static void
gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
}
static void
gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg726pay);
GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000;
/* sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
}
static gboolean
gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gchar *encoding_name;
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstBaseRTPAudioPayload *basertpaudiopayload;
gint bitrate;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
if (!gst_structure_get_int (structure, "bitrate", &bitrate))
bitrate = 32000;
switch (bitrate) {
case 16000:
encoding_name = g_strdup ("G726-16");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
2);
break;
case 24000:
encoding_name = g_strdup ("G726-24");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
3);
break;
case 32000:
encoding_name = g_strdup ("G726-32");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
4);
break;
case 40000:
encoding_name = g_strdup ("G726-40");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
5);
break;
default:
goto invalid_bitrate;
}
gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
gst_basertppayload_set_outcaps (payload, NULL);
g_free (encoding_name);
return TRUE;
/* ERRORS */
invalid_bitrate:
{
GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", bitrate);
return FALSE;
}
}
gboolean
gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg726pay",
GST_RANK_NONE, GST_TYPE_RTP_G726_PAY);
}
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