gstreamer/gst/rtp/gstrtpspeexpay.c
Wim Taymans 22eb34e2fe gst/rtp/: Fix case of encoding-name and key/value pairs to match the document.
Original commit message from CVS:
* gst/rtp/README:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix case of encoding-name and key/value pairs to match the document.
This is to make interoperation with SDP case-insensitive as required by
the relevant RFCs.
2007-01-25 14:22:53 +00:00

155 lines
4.6 KiB
C

/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* 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 "gstrtpspeexpay.h"
/* elementfactory information */
static const GstElementDetails gst_rtp_speex_pay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload-encodes Speex audio into a RTP packet",
"Edgard Lima <edgard.lima@indt.org.br>");
static GstStaticPadTemplate gst_rtp_speex_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstStaticPadTemplate gst_rtp_speex_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) \"SPEEX\", "
"encoding-params = (string) \"1\"")
);
static gboolean gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
GST_BOILERPLATE (GstRtpSPEEXPay, gst_rtp_speex_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_speex_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_speex_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_speex_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_speex_pay_details);
}
static void
gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * 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_speex_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;
}
static void
gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay,
GstRtpSPEEXPayClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
GST_BASE_RTP_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
}
static gboolean
gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpSPEEXPay *rtpspeexpay;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
GstFlowReturn ret;
rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one SPEEX 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 (rtpspeexpay));
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* 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);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
gboolean
gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexpay",
GST_RANK_NONE, GST_TYPE_RTP_SPEEX_PAY);
}