/* GStreamer * Copyright (C) 2020 Collabora Ltd. * Author: Guillaume Desmottes , Collabora Ltd. * * 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. */ /** * SECTION:element-rtpisacdepay * @title: rtpisacdepay * @short_description: iSAC RTP Depayloader * * Since: 1.20 * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include "gstrtpelements.h" #include "gstrtpisacdepay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpisacdepay_debug); #define GST_CAT_DEFAULT (rtpisacdepay_debug) static GstStaticPadTemplate gst_rtp_isac_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) { 16000, 32000 }, " "encoding-name = (string) \"ISAC\"") ); static GstStaticPadTemplate gst_rtp_isac_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/isac, " "rate = (int) { 16000, 32000 }, " "channels = (int) 1") ); struct _GstRtpIsacDepay { /*< private > */ GstRTPBaseDepayload parent; guint64 packet; }; #define gst_rtp_isac_depay_parent_class parent_class G_DEFINE_TYPE (GstRtpIsacDepay, gst_rtp_isac_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpisacdepay, "rtpisacdepay", GST_RANK_SECONDARY, GST_TYPE_RTP_ISAC_DEPAY, rtp_element_init (plugin)); static gboolean gst_rtp_isac_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) { GstCaps *src_caps; GstStructure *s; gint rate; gboolean ret; GST_DEBUG_OBJECT (depayload, "sink caps: %" GST_PTR_FORMAT, caps); s = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (s, "clock-rate", &rate)) { GST_ERROR_OBJECT (depayload, "Missing 'clock-rate' in caps"); return FALSE; } src_caps = gst_caps_new_simple ("audio/isac", "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, rate, NULL); ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), src_caps); GST_DEBUG_OBJECT (depayload, "set caps on source: %" GST_PTR_FORMAT " (ret=%d)", src_caps, ret); gst_caps_unref (src_caps); return ret; } static GstBuffer * gst_rtp_isac_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp_buffer) { GstBuffer *outbuf; outbuf = gst_rtp_buffer_get_payload_buffer (rtp_buffer); gst_rtp_drop_non_audio_meta (depayload, outbuf); return outbuf; } static void gst_rtp_isac_depay_class_init (GstRtpIsacDepayClass * klass) { GstElementClass *gstelement_class = (GstElementClass *) klass; GstRTPBaseDepayloadClass *depayload_class = (GstRTPBaseDepayloadClass *) klass; depayload_class->set_caps = gst_rtp_isac_depay_setcaps; depayload_class->process_rtp_packet = gst_rtp_isac_depay_process; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_isac_depay_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_isac_depay_src_template); gst_element_class_set_static_metadata (gstelement_class, "RTP iSAC depayloader", "Codec/Depayloader/Network/RTP", "Extracts iSAC audio from RTP packets", "Guillaume Desmottes "); GST_DEBUG_CATEGORY_INIT (rtpisacdepay_debug, "rtpisacdepay", 0, "iSAC RTP Depayloader"); } static void gst_rtp_isac_depay_init (GstRtpIsacDepay * rtpisacdepay) { }