gstreamer/gst/rtp/gstrtpspeexdepay.c
Wim Taymans af6e4da92e gst/rtp/: Fix klass typos.
Original commit message from CVS:
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
(gst_rtp_mp2t_depay_plugin_init):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
Fix klass typos.
Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.
2006-09-23 15:30:40 +00:00

148 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 parser",
"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\", "
"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);
}