gstreamer/gst/rtp/gstrtpac3depay.c
Wim Taymans 4397c8ffbf rtpdepay: remove payload type restrictions
Remove the pt restrictions for all the depayloaders that have an
encoding-name. We can use this to autoplug decoders.
Remove the encoding-name for all the payloaders with a fixed payload
type.
We now either have an encoding-name or a pt in the sinkpad caps of
a depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=639292
2013-01-28 12:41:04 +01:00

213 lines
5.8 KiB
C

/* 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.
*/
#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)
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,
GstBuffer * buf);
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_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_ac3_depay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&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 = 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;
}
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 (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpAC3Depay *rtpac3depay;
GstBuffer *outbuf;
GstRTPBuffer rtp = { NULL, };
guint8 *payload;
guint16 FT, NF;
rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
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);
gst_rtp_buffer_unmap (&rtp);
if (outbuf)
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));
gst_rtp_buffer_unmap (&rtp);
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);
}