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80f8780e92
Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
147 lines
4.3 KiB
C
147 lines
4.3 KiB
C
/* GStreamer
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* Copyright (C) 2020 Collabora Ltd.
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* Author: Guillaume Desmottes <guillaume.desmottes@collabora.com>, Collabora Ltd.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpisacdepay
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* @title: rtpisacdepay
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* @short_description: iSAC RTP Depayloader
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*
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* Since: 1.20
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "gstrtpelements.h"
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#include "gstrtpisacdepay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpisacdepay_debug);
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#define GST_CAT_DEFAULT (rtpisacdepay_debug)
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static GstStaticPadTemplate gst_rtp_isac_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) { 16000, 32000 }, "
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"encoding-name = (string) \"ISAC\"")
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);
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static GstStaticPadTemplate gst_rtp_isac_depay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/isac, "
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"rate = (int) { 16000, 32000 }, " "channels = (int) 1")
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);
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struct _GstRtpIsacDepay
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{
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/*< private > */
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GstRTPBaseDepayload parent;
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guint64 packet;
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};
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#define gst_rtp_isac_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpIsacDepay, gst_rtp_isac_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpisacdepay, "rtpisacdepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_ISAC_DEPAY, rtp_element_init (plugin));
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static gboolean
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gst_rtp_isac_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstCaps *src_caps;
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GstStructure *s;
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gint rate;
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gboolean ret;
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GST_DEBUG_OBJECT (depayload, "sink caps: %" GST_PTR_FORMAT, caps);
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s = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (s, "clock-rate", &rate)) {
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GST_ERROR_OBJECT (depayload, "Missing 'clock-rate' in caps");
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return FALSE;
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}
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src_caps = gst_caps_new_simple ("audio/isac",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, rate, NULL);
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), src_caps);
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GST_DEBUG_OBJECT (depayload,
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"set caps on source: %" GST_PTR_FORMAT " (ret=%d)", src_caps, ret);
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gst_caps_unref (src_caps);
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return ret;
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}
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static GstBuffer *
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gst_rtp_isac_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp_buffer)
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{
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GstBuffer *outbuf;
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp_buffer);
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gst_rtp_drop_non_audio_meta (depayload, outbuf);
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return outbuf;
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}
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static void
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gst_rtp_isac_depay_class_init (GstRtpIsacDepayClass * klass)
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{
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstRTPBaseDepayloadClass *depayload_class =
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(GstRTPBaseDepayloadClass *) klass;
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depayload_class->set_caps = gst_rtp_isac_depay_setcaps;
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depayload_class->process_rtp_packet = gst_rtp_isac_depay_process;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_isac_depay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_isac_depay_src_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP iSAC depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts iSAC audio from RTP packets",
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"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
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GST_DEBUG_CATEGORY_INIT (rtpisacdepay_debug, "rtpisacdepay", 0,
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"iSAC RTP Depayloader");
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}
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static void
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gst_rtp_isac_depay_init (GstRtpIsacDepay * rtpisacdepay)
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{
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}
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