gstreamer/gst/rtp/gstrtpgsmdepay.c
Sebastian Dröge b1089fb520 rtp: Copy metadata in the (de)payloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2015-08-11 12:47:23 +02:00

153 lines
4.7 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Zeeshan Ali <zeenix@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 <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpgsmdepay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpgsmdepay_debug);
#define GST_CAT_DEFAULT (rtpgsmdepay_debug)
/* RTPGSMDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
static GstStaticPadTemplate gst_rtp_gsm_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = 1")
);
static GstStaticPadTemplate gst_rtp_gsm_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) \"GSM\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_GSM_STRING ", "
"clock-rate = (int) 8000")
);
static GstBuffer *gst_rtp_gsm_depay_process (GstRTPBaseDepayload * _depayload,
GstRTPBuffer * rtp);
static gboolean gst_rtp_gsm_depay_setcaps (GstRTPBaseDepayload * _depayload,
GstCaps * caps);
#define gst_rtp_gsm_depay_parent_class parent_class
G_DEFINE_TYPE (GstRTPGSMDepay, gst_rtp_gsm_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbase_depayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbase_depayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_gsm_depay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_gsm_depay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP GSM depayloader", "Codec/Depayloader/Network/RTP",
"Extracts GSM audio from RTP packets", "Zeeshan Ali <zeenix@gmail.com>");
gstrtpbase_depayload_class->process_rtp_packet = gst_rtp_gsm_depay_process;
gstrtpbase_depayload_class->set_caps = gst_rtp_gsm_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpgsmdepay_debug, "rtpgsmdepay", 0,
"GSM Audio RTP Depayloader");
}
static void
gst_rtp_gsm_depay_init (GstRTPGSMDepay * rtpgsmdepay)
{
}
static gboolean
gst_rtp_gsm_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean ret;
GstStructure *structure;
gint clock_rate;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 8000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/x-gsm",
"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return ret;
}
static GstBuffer *
gst_rtp_gsm_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstBuffer *outbuf = NULL;
gboolean marker;
marker = gst_rtp_buffer_get_marker (rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer), marker,
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (marker && outbuf) {
/* mark start of talkspurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
return outbuf;
}
gboolean
gst_rtp_gsm_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpgsmdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_DEPAY);
}