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/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include <string.h>
#include "gstrtpelements.h"
#include "gstrtpmpadepay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmpadepay_debug);
#define GST_CAT_DEFAULT (rtpmpadepay_debug)
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_MPA_STRING ", "
"clock-rate = (int) 90000 ;"
"application/x-rtp, "
"media = (string) \"audio\", "
"encoding-name = (string) \"MPA\", clock-rate = (int) [1, MAX]")
);
#define gst_rtp_mpa_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpMPADepay, gst_rtp_mpa_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmpadepay, "rtpmpadepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_DEPAY, rtp_element_init (plugin));
static gboolean gst_rtp_mpa_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_mpa_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static void
gst_rtp_mpa_depay_class_init (GstRtpMPADepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpmpadepay_debug, "rtpmpadepay", 0,
"MPEG Audio RTP Depayloader");
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mpa_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mpa_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP",
"Extracts MPEG audio from RTP packets (RFC 2038)",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasedepayload_class->set_caps = gst_rtp_mpa_depay_setcaps;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mpa_depay_process;
}
static void
gst_rtp_mpa_depay_init (GstRtpMPADepay * rtpmpadepay)
{
}
static gboolean
gst_rtp_mpa_depay_setcaps (GstRTPBaseDepayload * 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 (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpMPADepay *rtpmpadepay;
GstBuffer *outbuf;
gint payload_len;
#if 0
guint8 *payload;
guint16 frag_offset;
#endif
gboolean marker;
rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
if (payload_len <= 4)
goto empty_packet;
#if 0
payload = gst_rtp_buffer_get_payload (&rtp);
/* 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];
#endif
/* subbuffer skipping the 4 header bytes */
outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 4, -1);
marker = gst_rtp_buffer_get_marker (rtp);
if (marker) {
/* mark start of talkspurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
GST_DEBUG_OBJECT (rtpmpadepay,
"gst_rtp_mpa_depay_chain: pushing buffer of size %" G_GSIZE_FORMAT "",
gst_buffer_get_size (outbuf));
if (outbuf) {
gst_rtp_drop_non_audio_meta (rtpmpadepay, outbuf);
}
/* FIXME, we can push half mpeg frames when they are split over multiple
* RTP packets */
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
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