gstreamer/gst/rtp/gstrtpgsmenc.c
Thomas Vander Stichele 67abe2f343 don't use underscores
Original commit message from CVS:
don't use underscores
2005-09-20 10:51:51 +00:00

201 lines
5.5 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 "gstrtpgsmenc.h"
/* elementfactory information */
static GstElementDetails gst_rtpgsmenc_details = {
"RTP GSM Audio Encoder",
"Codec/Encoder/Network",
"Encodes GSM audio into a RTP packet",
"Zeeshan Ali <zak147@yahoo.com>"
};
static GstStaticPadTemplate gst_rtpgsmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) [ 1000, 48000 ]")
);
static GstStaticPadTemplate gst_rtpgsmenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 255 ], "
"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
);
static void gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass);
static void gst_rtpgsmenc_base_init (GstRtpGSMEncClass * klass);
static void gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc);
static gboolean gst_rtpgsmenc_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtpgsmenc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtpgsmenc_get_type (void)
{
static GType rtpgsmenc_type = 0;
if (!rtpgsmenc_type) {
static const GTypeInfo rtpgsmenc_info = {
sizeof (GstRtpGSMEncClass),
(GBaseInitFunc) gst_rtpgsmenc_base_init,
NULL,
(GClassInitFunc) gst_rtpgsmenc_class_init,
NULL,
NULL,
sizeof (GstRtpGSMEnc),
0,
(GInstanceInitFunc) gst_rtpgsmenc_init,
};
rtpgsmenc_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpGSMEnc",
&rtpgsmenc_info, 0);
}
return rtpgsmenc_type;
}
static void
gst_rtpgsmenc_base_init (GstRtpGSMEncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpgsmenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpgsmenc_src_template));
gst_element_class_set_details (element_class, &gst_rtpgsmenc_details);
}
static void
gst_rtpgsmenc_class_init (GstRtpGSMEncClass * 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_rtpgsmenc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpgsmenc_handle_buffer;
}
static void
gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc)
{
rtpgsmenc->frequency = 8000;
}
static gboolean
gst_rtpgsmenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
GstRtpGSMEnc *rtpgsmenc;
GstStructure *structure;
gboolean ret;
GstCaps *srccaps;
rtpgsmenc = GST_RTP_GSM_ENC (payload);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &rtpgsmenc->frequency);
if (!ret)
return FALSE;
srccaps = gst_caps_new_simple ("application/x-rtp", NULL);
gst_pad_set_caps (GST_BASE_RTP_PAYLOAD_SRCPAD (rtpgsmenc), srccaps);
gst_caps_unref (srccaps);
return TRUE;
}
static GstFlowReturn
gst_rtpgsmenc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpGSMEnc *rtpgsmenc;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
GstFlowReturn ret;
rtpgsmenc = GST_RTP_GSM_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one GSM frame per RTP packet for now */
payload_len = size;
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (GST_BUFFER_SIZE (outbuf) < GST_BASE_RTP_PAYLOAD_MTU (rtpgsmenc));
gst_rtpbuffer_set_timestamp (outbuf, timestamp * 8000 / GST_SECOND);
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtpbuffer_get_payload (outbuf);
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
GST_DEBUG ("gst_rtpgsmenc_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
gboolean
gst_rtpgsmenc_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpgsmenc",
GST_RANK_NONE, GST_TYPE_RTP_GSM_ENC);
}