gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpg723depay.c

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/* GStreamer
*
* Copyright (C) <2010> 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.
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*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtpelements.h"
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#include "gstrtpg723depay.h"
GST_DEBUG_CATEGORY_STATIC (rtpg723depay_debug);
#define GST_CAT_DEFAULT (rtpg723depay_debug)
/* references:
*
* RFC 3551 (4.5.3)
*/
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
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};
/* input is an RTP packet
*
*/
static GstStaticPadTemplate gst_rtp_g723_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) 8000, "
"encoding-name = (string) \"G723\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
"clock-rate = (int) 8000")
);
static GstStaticPadTemplate gst_rtp_g723_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G723, " "channels = (int) 1," "rate = (int) 8000")
);
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static gboolean gst_rtp_g723_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_g723_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
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#define gst_rtp_g723_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpG723Depay, gst_rtp_g723_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpg723depay, "rtpg723depay",
GST_RANK_SECONDARY, GST_TYPE_RTP_G723_DEPAY, rtp_element_init (plugin));
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static void
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gst_rtp_g723_depay_class_init (GstRtpG723DepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpg723depay_debug, "rtpg723depay", 0,
"G.723 RTP Depayloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g723_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g723_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP G.723 depayloader", "Codec/Depayloader/Network/RTP",
"Extracts G.723 audio from RTP packets (RFC 3551)",
"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g723_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_g723_depay_setcaps;
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}
static void
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gst_rtp_g723_depay_init (GstRtpG723Depay * rtpg723depay)
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{
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GstRTPBaseDepayload *depayload;
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depayload = GST_RTP_BASE_DEPAYLOAD (rtpg723depay);
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gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
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}
static gboolean
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gst_rtp_g723_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
GstStructure *structure;
GstCaps *srccaps;
GstRtpG723Depay *rtpg723depay;
const gchar *params;
gint clock_rate, channels;
gboolean ret;
rtpg723depay = GST_RTP_G723_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!(params = gst_structure_get_string (structure, "encoding-params")))
channels = 1;
else {
channels = atoi (params);
}
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 8000;
if (channels != 1)
goto wrong_channels;
if (clock_rate != 8000)
goto wrong_clock_rate;
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/G723",
"channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, clock_rate, NULL);
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
return ret;
/* ERRORS */
wrong_channels:
{
GST_DEBUG_OBJECT (rtpg723depay, "expected 1 channel, got %d", channels);
return FALSE;
}
wrong_clock_rate:
{
GST_DEBUG_OBJECT (rtpg723depay, "expected 8000 clock-rate, got %d",
clock_rate);
return FALSE;
}
}
static GstBuffer *
gst_rtp_g723_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
GstRtpG723Depay *rtpg723depay;
GstBuffer *outbuf = NULL;
gint payload_len;
gboolean marker;
rtpg723depay = GST_RTP_G723_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
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/* At least 4 bytes */
if (payload_len < 4)
goto too_small;
GST_LOG_OBJECT (rtpg723depay, "payload len %d", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
marker = gst_rtp_buffer_get_marker (rtp);
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if (marker) {
/* marker bit starts talkspurt */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
GST_LOG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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return outbuf;
/* ERRORS */
too_small:
{
GST_ELEMENT_WARNING (rtpg723depay, STREAM, DECODE,
(NULL), ("G723 RTP payload too small (%d)", payload_len));
goto bad_packet;
}
bad_packet:
{
/* no fatal error */
return NULL;
}
}