mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 18:21:04 +00:00
eac9efb92e
A marker bit on an audio packet does not mean a DISCONT (in the GStreamer sense of missing data) but it means that the packet is the end of a talkspurt and thus a good opportunity to resync to the clock. Use the RESYNC buffer flag to note this. Real discontinuities are marked with DISCONT still when the seqnum has a GAP or when the input buffer has the DISCONT flag set. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=627204
151 lines
4.6 KiB
C
151 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 "gstrtpgsmdepay.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,
|
|
GstBuffer * buf);
|
|
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 = 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, GstBuffer * buf)
|
|
{
|
|
GstBuffer *outbuf = NULL;
|
|
gboolean marker;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
|
|
|
|
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 (buf), marker,
|
|
gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
|
|
|
|
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
if (marker && outbuf) {
|
|
/* mark start of talkspurt with RESYNC */
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
|
|
}
|
|
|
|
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);
|
|
}
|