mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-29 09:45:07 +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.
153 lines
4.6 KiB
C
153 lines
4.6 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_static_pad_template (gstelement_class,
|
|
&gst_rtp_gsm_depay_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&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_non_audio_meta (depayload, outbuf);
|
|
}
|
|
|
|
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);
|
|
}
|