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/* GStreamer
* Copyright (C) <2007> 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.
*/
/**
* SECTION:element-rtpac3depay
* @see_also: rtpac3pay
*
* Extract AC3 audio from RTP packets according to RFC 4184.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)AC3, payload=(int)96' ! rtpac3depay ! a52dec ! pulsesink
* ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
* the rtpac3pay example to create the RTP stream.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include <string.h>
#include "gstrtpac3depay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpac3depay_debug);
#define GST_CAT_DEFAULT (rtpac3depay_debug)
static GstStaticPadTemplate gst_rtp_ac3_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/ac3")
);
static GstStaticPadTemplate gst_rtp_ac3_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) { 32000, 44100, 48000 }, "
"encoding-name = (string) \"AC3\"")
);
G_DEFINE_TYPE (GstRtpAC3Depay, gst_rtp_ac3_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
static gboolean gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static void
gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_ac3_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_ac3_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP AC3 depayloader", "Codec/Depayloader/Network/RTP",
"Extracts AC3 audio from RTP packets (RFC 4184)",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasedepayload_class->set_caps = gst_rtp_ac3_depay_setcaps;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_ac3_depay_process;
GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0,
"AC3 Audio RTP Depayloader");
}
static void
gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay)
{
/* needed because of G_DEFINE_TYPE */
}
static gboolean
gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
gint clock_rate;
GstCaps *srccaps;
gboolean res;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_empty_simple ("audio/ac3");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
}
static GstBuffer *
gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpAC3Depay *rtpac3depay;
GstBuffer *outbuf;
guint8 *payload;
guint16 FT, NF;
rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
if (gst_rtp_buffer_get_payload_len (rtp) < 2)
goto empty_packet;
payload = gst_rtp_buffer_get_payload (rtp);
/* strip off header
*
* 0 1
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ | FT| NF |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
FT = payload[0] & 0x3;
NF = payload[1];
GST_DEBUG_OBJECT (rtpac3depay, "FT: %d, NF: %d", FT, NF);
/* We don't bother with fragmented packets yet */
outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 2, -1);
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpac3depay), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
gboolean
gst_rtp_ac3_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpac3depay",
GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_DEPAY);
}
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