mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-30 12:10:37 +00:00
22eb34e2fe
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.
149 lines
4.4 KiB
C
149 lines
4.4 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 <string.h>
|
|
#include <gst/rtp/gstrtpbuffer.h>
|
|
#include "gstrtpspeexdepay.h"
|
|
|
|
/* elementfactory information */
|
|
static const GstElementDetails gst_rtp_speexdepay_details =
|
|
GST_ELEMENT_DETAILS ("RTP packet depayloader",
|
|
"Codec/Depayloader/Network",
|
|
"Extracts Speex audio from RTP packets",
|
|
"Edgard Lima <edgard.lima@indt.org.br>");
|
|
|
|
/* RtpSPEEXDepay signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_0
|
|
};
|
|
|
|
static GstStaticPadTemplate gst_rtp_speex_depay_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("application/x-rtp, "
|
|
"media = (string) \"audio\", "
|
|
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
|
|
"clock-rate = (int) [6000, 48000], "
|
|
"encoding-name = (string) \"SPEEX\", "
|
|
"encoding-params = (string) \"1\"")
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_speex_depay_src_template =
|
|
GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-speex")
|
|
);
|
|
|
|
static GstBuffer *gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload,
|
|
GstBuffer * buf);
|
|
static gboolean gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload,
|
|
GstCaps * caps);
|
|
|
|
GST_BOILERPLATE (GstRtpSPEEXDepay, gst_rtp_speex_depay, GstBaseRTPDepayload,
|
|
GST_TYPE_BASE_RTP_DEPAYLOAD);
|
|
|
|
static void
|
|
gst_rtp_speex_depay_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_depay_src_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&gst_rtp_speex_depay_sink_template));
|
|
gst_element_class_set_details (element_class, &gst_rtp_speexdepay_details);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
|
|
|
|
parent_class = g_type_class_peek_parent (klass);
|
|
|
|
gstbasertpdepayload_class->process = gst_rtp_speex_depay_process;
|
|
gstbasertpdepayload_class->set_caps = gst_rtp_speex_depay_setcaps;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay,
|
|
GstRtpSPEEXDepayClass * klass)
|
|
{
|
|
GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay)->clock_rate = 8000;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
|
|
{
|
|
GstCaps *srccaps;
|
|
gboolean ret;
|
|
|
|
srccaps =
|
|
gst_static_pad_template_get_caps (&gst_rtp_speex_depay_src_template);
|
|
ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
|
|
|
|
gst_caps_unref (srccaps);
|
|
return ret;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
|
|
{
|
|
GstBuffer *outbuf = NULL;
|
|
gint payload_len;
|
|
guint8 *payload;
|
|
|
|
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
|
|
GST_BUFFER_SIZE (buf),
|
|
gst_rtp_buffer_get_marker (buf),
|
|
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
|
|
|
|
payload_len = gst_rtp_buffer_get_payload_len (buf);
|
|
payload = gst_rtp_buffer_get_payload (buf);
|
|
|
|
outbuf = gst_buffer_new_and_alloc (payload_len);
|
|
memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
|
|
return outbuf;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_speex_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpspeexdepay",
|
|
GST_RANK_MARGINAL, GST_TYPE_RTP_SPEEX_DEPAY);
|
|
}
|