mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-19 06:46:38 +00:00
4a28e649c3
Every g_quark_from_static_string() is a hash table lookup serialised on the global quark lock in GLib. Let's just look up the two quarks we need once and cache them locally for future use. While we're at it, add new utility functions for the two most commonly used tags (audio + video). Make first argument a gpointer so we don't have to cast and make the code ugly. These are used for logging purposes only anyway.
125 lines
3.8 KiB
C
125 lines
3.8 KiB
C
/*
|
|
* Siren Depayloader Gst Element
|
|
*
|
|
* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
|
|
*
|
|
* 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 <stdlib.h>
|
|
#include <gst/rtp/gstrtpbuffer.h>
|
|
#include <gst/audio/audio.h>
|
|
#include "gstrtpsirendepay.h"
|
|
#include "gstrtputils.h"
|
|
|
|
static GstStaticPadTemplate gst_rtp_siren_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) 16000, " "encoding-name = (string) \"SIREN\"")
|
|
/* This is the default, so the peer doesn't have to specify it */
|
|
/* " "dct-length = (int) 320") */
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_siren_depay_src_template =
|
|
GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
|
|
);
|
|
|
|
static GstBuffer *gst_rtp_siren_depay_process (GstRTPBaseDepayload *
|
|
depayload, GstRTPBuffer * rtp);
|
|
static gboolean gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload *
|
|
depayload, GstCaps * caps);
|
|
|
|
G_DEFINE_TYPE (GstRTPSirenDepay, gst_rtp_siren_depay,
|
|
GST_TYPE_RTP_BASE_DEPAYLOAD);
|
|
|
|
static void gst_rtp_siren_depay_class_init (GstRTPSirenDepayClass * klass)
|
|
{
|
|
GstElementClass *gstelement_class;
|
|
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
|
|
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
|
|
|
|
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_siren_depay_process;
|
|
gstrtpbasedepayload_class->set_caps = gst_rtp_siren_depay_setcaps;
|
|
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_siren_depay_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_siren_depay_sink_template);
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"RTP Siren packet depayloader", "Codec/Depayloader/Network/RTP",
|
|
"Extracts Siren audio from RTP packets",
|
|
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_siren_depay_init (GstRTPSirenDepay * rtpsirendepay)
|
|
{
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
|
|
{
|
|
GstCaps *srccaps;
|
|
gboolean ret;
|
|
|
|
srccaps = gst_caps_new_simple ("audio/x-siren",
|
|
"dct-length", G_TYPE_INT, 320, NULL);
|
|
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
|
|
|
|
GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
|
|
gst_caps_unref (srccaps);
|
|
|
|
/* always fixed clock rate of 16000 */
|
|
depayload->clock_rate = 16000;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_rtp_siren_depay_process (GstRTPBaseDepayload * depayload,
|
|
GstRTPBuffer * rtp)
|
|
{
|
|
GstBuffer *outbuf;
|
|
|
|
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
|
|
|
|
if (outbuf) {
|
|
gst_rtp_drop_non_audio_meta (depayload, outbuf);
|
|
}
|
|
|
|
return outbuf;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_siren_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpsirendepay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_DEPAY);
|
|
}
|