gstreamer/gst/rtp/gstrtpg729pay.c
Olivier Crete d6f37dadb5 gst/rtp/: Added G729 pay and depayloaders. Fixes #532409.
Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_base_init),
(gst_rtp_g729_depay_class_init), (gst_rtp_g729_depay_init),
(gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process),
(gst_rtp_g729_depay_plugin_init):
* gst/rtp/gstrtpg729depay.h:
* gst/rtp/gstrtpg729pay.c: (gst_rtpg729pay_base_init),
(gst_rtpg729pay_class_init), (gst_rtpg729pay_init),
(gst_rtpg729pay_setcaps), (gst_rtp_g729_pay_plugin_init):
* gst/rtp/gstrtpg729pay.h:
Added G729 pay and depayloaders. Fixes #532409.
2008-05-13 08:35:55 +00:00

150 lines
4.7 KiB
C

/* GStreamer
*
* 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 "gstrtpg729pay.h"
#include <gst/rtp/gstrtpbuffer.h>
/* elementfactory information */
static GstElementDetails gst_rtpg729pay_details = {
"RTP Payloader for G729 Audio",
"Codec/Payloader/Network",
"Packetize G729 audio streams into RTP packets",
"Laurent Glayal <spglegle@yahoo.fr>"
};
GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
#define GST_CAT_DEFAULT (rtpg729pay_debug)
static GstStaticPadTemplate gst_rtpg729pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G729, channels=(int)1, rate=(int)8000")
);
static GstStaticPadTemplate gst_rtpg729pay_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) \"G729\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
"clock-rate = (int) 8000")
);
static gboolean gst_rtpg729pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpG729Pay, gst_rtpg729pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtpg729pay_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_rtpg729pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpg729pay_src_template));
gst_element_class_set_details (element_class, &gst_rtpg729pay_details);
}
static void
gst_rtpg729pay_class_init (GstRtpG729PayClass * 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_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class->set_caps = gst_rtpg729pay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
"G729 audio RTP payloader");
}
static void
gst_rtpg729pay_init (GstRtpG729Pay * rtpg729pay, GstRtpG729PayClass * klass)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertppayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg729pay);
/* 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 = 8000;
/* tell basertpaudiopayload that this is a frame based codec */
gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
gst_basertppayload_set_options (basertppayload, "audio", FALSE, "G729", 8000);
gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 10, 10);
}
static gboolean
gst_rtpg729pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
{
GstRtpG729Pay *rtpg729pay;
GstBaseRTPAudioPayload *basertpaudiopayload;
gboolean ret;
GstStructure *structure;
const char *payload_name;
rtpg729pay = GST_RTP_G729_PAY (basertppayload);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
structure = gst_caps_get_structure (caps, 0);
payload_name = gst_structure_get_name (structure);
if (g_strcasecmp ("audio/G729", payload_name) != 0)
goto wrong_name;
ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
return ret;
/* ERRORS */
wrong_name:
{
GST_ERROR_OBJECT (rtpg729pay, "wrong name, expected 'audio/G729', got '%s'",
payload_name);
return FALSE;
}
}
gboolean
gst_rtp_g729_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg729pay",
GST_RANK_NONE, GST_TYPE_RTP_G729_PAY);
}