gstreamer/gst/rtp/gstrtpmpadepay.c

189 lines
5.6 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim.taymans@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 <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 MPEG audio depayloader",
"Codec/Depayloader/Network",
"Extracts MPEG audio from RTP packets (RFC 2038)",
"Wim Taymans <wim.taymans@gmail.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, " "mpegversion = (int) 1")
);
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 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)
{
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
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;
GstCaps *outcaps;
gint clock_rate;
gboolean res;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 90000;
depayload->clock_rate = clock_rate;
outcaps =
gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1, NULL);
res = gst_pad_set_caps (depayload->srcpad, outcaps);
gst_caps_unref (outcaps);
return res;
}
static GstBuffer *
gst_rtp_mpa_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpMPADepay *rtpmpadepay;
GstBuffer *outbuf;
rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
{
gint payload_len;
guint8 *payload;
guint16 frag_offset;
gboolean marker;
payload_len = gst_rtp_buffer_get_payload_len (buf);
if (payload_len <= 4)
goto empty_packet;
payload = gst_rtp_buffer_get_payload (buf);
/* 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);
marker = gst_rtp_buffer_get_marker (buf);
if (marker) {
/* mark start of talkspurt with discont */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
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;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
gboolean
gst_rtp_mpa_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmpadepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_MPA_DEPAY);
}