gstreamer/gst/rtp/gstrtpac3depay.c
Stefan Kost e7f919986a gst/rtp/: Remove copy/paste unused code (property setters and getter) found by the coverage suite (yay, saves ~20k on...
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).
2008-01-09 11:11:01 +00:00

278 lines
7.2 KiB
C

/* GStreamer
* Copyright (C) <2007> 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 "gstrtpac3depay.h"
GST_DEBUG_CATEGORY_STATIC (rtpac3depay_debug);
#define GST_CAT_DEFAULT (rtpac3depay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_ac3depay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Depayloader/Network",
"Extracts AC3 audio from RTP packets (RFC 4184)",
"Wim Taymans <wim@fluendo.com>");
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\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) { 32000, 44100, 48000 }, "
"encoding-name = (string) \"AC3\"")
);
GST_BOILERPLATE (GstRtpAC3Depay, gst_rtp_ac3_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static gboolean gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_ac3_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static GstStateChangeReturn gst_rtp_ac3_depay_change_state (GstElement *
element, GstStateChange transition);
static void
gst_rtp_ac3_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_ac3_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_ac3_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_ac3depay_details);
}
static void
gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass)
{
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstelement_class->change_state = gst_rtp_ac3_depay_change_state;
gstbasertpdepayload_class->set_caps = gst_rtp_ac3_depay_setcaps;
gstbasertpdepayload_class->process = gst_rtp_ac3_depay_process;
GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0,
"MPEG Audio RTP Depayloader");
}
static void
gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay,
GstRtpAC3DepayClass * klass)
{
/* needed because of GST_BOILERPLATE */
}
static gboolean
gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpAC3Depay *rtpac3depay;
gint clock_rate = 90000; /* default */
rtpac3depay = GST_RTP_AC3_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;
}
struct frmsize_s
{
guint16 bit_rate;
guint16 frm_size[3];
};
static const struct frmsize_s frmsizecod_tbl[] = {
{32, {64, 69, 96}},
{32, {64, 70, 96}},
{40, {80, 87, 120}},
{40, {80, 88, 120}},
{48, {96, 104, 144}},
{48, {96, 105, 144}},
{56, {112, 121, 168}},
{56, {112, 122, 168}},
{64, {128, 139, 192}},
{64, {128, 140, 192}},
{80, {160, 174, 240}},
{80, {160, 175, 240}},
{96, {192, 208, 288}},
{96, {192, 209, 288}},
{112, {224, 243, 336}},
{112, {224, 244, 336}},
{128, {256, 278, 384}},
{128, {256, 279, 384}},
{160, {320, 348, 480}},
{160, {320, 349, 480}},
{192, {384, 417, 576}},
{192, {384, 418, 576}},
{224, {448, 487, 672}},
{224, {448, 488, 672}},
{256, {512, 557, 768}},
{256, {512, 558, 768}},
{320, {640, 696, 960}},
{320, {640, 697, 960}},
{384, {768, 835, 1152}},
{384, {768, 836, 1152}},
{448, {896, 975, 1344}},
{448, {896, 976, 1344}},
{512, {1024, 1114, 1536}},
{512, {1024, 1115, 1536}},
{576, {1152, 1253, 1728}},
{576, {1152, 1254, 1728}},
{640, {1280, 1393, 1920}},
{640, {1280, 1394, 1920}}
};
static GstBuffer *
gst_rtp_ac3_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpAC3Depay *rtpac3depay;
GstBuffer *outbuf;
rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
if (!gst_rtp_buffer_validate (buf))
goto bad_packet;
{
gint payload_len;
guint8 *payload;
guint16 FT, NF;
payload_len = gst_rtp_buffer_get_payload_len (buf);
payload = gst_rtp_buffer_get_payload (buf);
if (payload_len <= 2)
goto empty_packet;
/* 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);
payload_len -= 2;
payload += 2;
/* We don't bother with fragmented packets yet */
outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 2, -1);
GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
}
return NULL;
bad_packet:
{
GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
("Packet did not validate."), (NULL));
return NULL;
}
#if 0
bad_payload:
{
GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
("Unexpected payload type."), (NULL));
return NULL;
}
#endif
empty_packet:
{
GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
static GstStateChangeReturn
gst_rtp_ac3_depay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpAC3Depay *rtpac3depay;
GstStateChangeReturn ret;
rtpac3depay = GST_RTP_AC3_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_ac3_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpac3depay",
GST_RANK_MARGINAL, GST_TYPE_RTP_AC3_DEPAY);
}