gstreamer/gst/rtp/gstrtpgsmpay.c
Stefan Kost 11aaae270b gst/rtp/: Add log category.
Original commit message from CVS:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtpgsmpay.c:
Add log category.
2007-10-04 07:29:48 +00:00

181 lines
5.4 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.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 "gstrtpgsmpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
#define GST_CAT_DEFAULT (rtpgsmpay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_gsm_pay_details =
GST_ELEMENT_DETAILS ("RTP GSM audio payloader",
"Codec/Payloader/Network",
"Payload-encodes GSM audio into a RTP packet",
"Zeeshan Ali <zeenix@gmail.com>");
static GstStaticPadTemplate gst_rtp_gsm_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
);
static GstStaticPadTemplate gst_rtp_gsm_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_GSM_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
);
static gboolean gst_rtp_gsm_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
GST_BOILERPLATE (GstRTPGSMPay, gst_rtp_gsm_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_gsm_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_gsm_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_gsm_pay_details);
}
static void
gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * 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_gsm_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
"GSM Audio RTP Payloader");
}
static void
gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay, GstRTPGSMPayClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
GST_BASE_RTP_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
}
static gboolean
gst_rtp_gsm_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
const char *stname;
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
stname = gst_structure_get_name (structure);
if (0 == strcmp ("audio/x-gsm", stname)) {
gst_basertppayload_set_options (payload, "audio", FALSE, "GSM", 8000);
} else {
return FALSE;
}
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRTPGSMPay *rtpgsmpay;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* FIXME, only one GSM frame per RTP packet for now */
payload_len = size;
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay));
/* copy timestamp and duration */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
gboolean
gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpgsmpay",
GST_RANK_NONE, GST_TYPE_RTP_GSM_PAY);
}