mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
f70a623418
Conflicts: docs/libs/Makefile.am ext/kate/gstkatetiger.c ext/opus/gstopusdec.c ext/xvid/gstxvidenc.c gst-libs/gst/basecamerabinsrc/Makefile.am gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h gst-libs/gst/video/gstbasevideocodec.c gst-libs/gst/video/gstbasevideocodec.h gst-libs/gst/video/gstbasevideodecoder.c gst-libs/gst/video/gstbasevideoencoder.c gst/asfmux/gstasfmux.c gst/audiovisualizers/gstwavescope.c gst/camerabin2/gstcamerabin2.c gst/debugutils/gstcompare.c gst/frei0r/gstfrei0rmixer.c gst/mpegpsmux/mpegpsmux.c gst/mpegtsmux/mpegtsmux.c gst/mxf/mxfmux.c gst/videomeasure/gstvideomeasure_ssim.c gst/videoparsers/gsth264parse.c gst/videoparsers/gstmpeg4videoparse.c
120 lines
3.7 KiB
C
120 lines
3.7 KiB
C
/*
|
|
* Opus Depayloader Gst Element
|
|
*
|
|
* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
|
|
*
|
|
* 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., 59 Temple Place - Suite 330,
|
|
* Boston, MA 02111-1307, USA.
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include "config.h"
|
|
#endif
|
|
|
|
#include <string.h>
|
|
#include <stdlib.h>
|
|
#include <gst/rtp/gstrtpbuffer.h>
|
|
#include "gstrtpopusdepay.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug);
|
|
#define GST_CAT_DEFAULT (rtpopusdepay_debug)
|
|
|
|
static GstStaticPadTemplate gst_rtp_opus_depay_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("application/x-rtp, "
|
|
"media = (string) \"audio\", "
|
|
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ","
|
|
"clock-rate = (int) 48000, "
|
|
"encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_opus_depay_src_template =
|
|
GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-opus")
|
|
);
|
|
|
|
static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
|
|
GstBuffer * buf);
|
|
static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
|
|
GstCaps * caps);
|
|
|
|
G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
|
|
GST_TYPE_RTP_BASE_DEPAYLOAD);
|
|
|
|
static void
|
|
gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
|
|
{
|
|
GstRTPBaseDepayloadClass *gstbasertpdepayload_class;
|
|
GstElementClass *element_class;
|
|
|
|
element_class = GST_ELEMENT_CLASS (klass);
|
|
gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass;
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&gst_rtp_opus_depay_src_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&gst_rtp_opus_depay_sink_template));
|
|
gst_element_class_set_details_simple (element_class,
|
|
"RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP",
|
|
"Extracts Opus audio from RTP packets",
|
|
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
|
|
|
|
gstbasertpdepayload_class->process = gst_rtp_opus_depay_process;
|
|
gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
|
|
"Opus RTP Depayloader");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay)
|
|
{
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
|
|
{
|
|
GstCaps *srccaps;
|
|
gboolean ret;
|
|
|
|
srccaps = gst_caps_new_empty_simple ("audio/x-opus");
|
|
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
|
|
|
|
GST_DEBUG_OBJECT (depayload,
|
|
"set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
|
|
gst_caps_unref (srccaps);
|
|
|
|
depayload->clock_rate = 48000;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
|
|
{
|
|
GstBuffer *outbuf;
|
|
GstRTPBuffer rtpbuf = { NULL, };
|
|
|
|
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
|
|
outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
|
|
gst_rtp_buffer_unmap (&rtpbuf);
|
|
|
|
return outbuf;
|
|
}
|