mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-11 02:46:33 +00:00
e7f919986a
Original commit message from CVS: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Remove copy/paste unused code (property setters and getter) found by the coverage suite (yay, saves ~20k on disk).
229 lines
6.4 KiB
C
229 lines
6.4 KiB
C
/* GStreamer
|
|
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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 <gst/rtp/gstrtpbuffer.h>
|
|
|
|
#include <string.h>
|
|
#include "gstrtpmpadepay.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (rtpmpadepay_debug);
|
|
#define GST_CAT_DEFAULT (rtpmpadepay_debug)
|
|
|
|
/* elementfactory information */
|
|
static const GstElementDetails gst_rtp_mpadepay_details =
|
|
GST_ELEMENT_DETAILS ("RTP packet depayloader",
|
|
"Codec/Depayloader/Network",
|
|
"Extracts MPEG audio from RTP packets (RFC 2038)",
|
|
"Wim Taymans <wim@fluendo.com>");
|
|
|
|
static GstStaticPadTemplate gst_rtp_mpa_depay_src_template =
|
|
GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/mpeg")
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_mpa_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) 90000, " "encoding-name = (string) \"MPA\";"
|
|
"application/x-rtp, "
|
|
"media = (string) \"audio\", "
|
|
"payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", "
|
|
"clock-rate = (int) 90000")
|
|
);
|
|
|
|
GST_BOILERPLATE (GstRtpMPADepay, gst_rtp_mpa_depay, GstBaseRTPDepayload,
|
|
GST_TYPE_BASE_RTP_DEPAYLOAD);
|
|
|
|
static gboolean gst_rtp_mpa_depay_setcaps (GstBaseRTPDepayload * depayload,
|
|
GstCaps * caps);
|
|
static GstBuffer *gst_rtp_mpa_depay_process (GstBaseRTPDepayload * depayload,
|
|
GstBuffer * buf);
|
|
|
|
static GstStateChangeReturn gst_rtp_mpa_depay_change_state (GstElement *
|
|
element, GstStateChange transition);
|
|
|
|
static void
|
|
gst_rtp_mpa_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_mpa_depay_src_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&gst_rtp_mpa_depay_sink_template));
|
|
|
|
gst_element_class_set_details (element_class, &gst_rtp_mpadepay_details);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mpa_depay_class_init (GstRtpMPADepayClass * 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);
|
|
|
|
gstelement_class->change_state = gst_rtp_mpa_depay_change_state;
|
|
|
|
gstbasertpdepayload_class->set_caps = gst_rtp_mpa_depay_setcaps;
|
|
gstbasertpdepayload_class->process = gst_rtp_mpa_depay_process;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtpmpadepay_debug, "rtpmpadepay", 0,
|
|
"MPEG Audio RTP Depayloader");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mpa_depay_init (GstRtpMPADepay * rtpmpadepay,
|
|
GstRtpMPADepayClass * klass)
|
|
{
|
|
/* needed because of GST_BOILERPLATE */
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_mpa_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
|
|
{
|
|
GstStructure *structure;
|
|
GstRtpMPADepay *rtpmpadepay;
|
|
gint clock_rate = 90000; /* default */
|
|
|
|
rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_get_int (structure, "clock-rate", &clock_rate);
|
|
depayload->clock_rate = clock_rate;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_rtp_mpa_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
|
|
{
|
|
GstRtpMPADepay *rtpmpadepay;
|
|
GstBuffer *outbuf;
|
|
|
|
rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
|
|
|
|
if (!gst_rtp_buffer_validate (buf))
|
|
goto bad_packet;
|
|
|
|
{
|
|
gint payload_len;
|
|
guint8 *payload;
|
|
guint16 frag_offset;
|
|
|
|
payload_len = gst_rtp_buffer_get_payload_len (buf);
|
|
payload = gst_rtp_buffer_get_payload (buf);
|
|
|
|
if (payload_len <= 4)
|
|
goto empty_packet;
|
|
|
|
/* strip off header
|
|
*
|
|
* 0 1 2 3
|
|
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
* | MBZ | Frag_offset |
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
*/
|
|
frag_offset = (payload[2] << 8) | payload[3];
|
|
|
|
/* subbuffer skipping the 4 header bytes */
|
|
outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 4, -1);
|
|
|
|
GST_DEBUG_OBJECT (rtpmpadepay,
|
|
"gst_rtp_mpa_depay_chain: pushing buffer of size %d",
|
|
GST_BUFFER_SIZE (outbuf));
|
|
|
|
/* FIXME, we can push half mpeg frames when they are split over multiple
|
|
* RTP packets */
|
|
return outbuf;
|
|
}
|
|
|
|
return NULL;
|
|
|
|
bad_packet:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
|
|
("Packet did not validate."), (NULL));
|
|
return NULL;
|
|
}
|
|
#if 0
|
|
bad_payload:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
|
|
("Unexpected payload type."), (NULL));
|
|
return NULL;
|
|
}
|
|
#endif
|
|
empty_packet:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
|
|
("Empty Payload."), (NULL));
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_mpa_depay_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstRtpMPADepay *rtpmpadepay;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpmpadepay = GST_RTP_MPA_DEPAY (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_mpa_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmpadepay",
|
|
GST_RANK_MARGINAL, GST_TYPE_RTP_MPA_DEPAY);
|
|
}
|