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148 lines
4.6 KiB
C
148 lines
4.6 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
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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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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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 "gstrtpgsmdepay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpgsmdepay_debug);
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#define GST_CAT_DEFAULT (rtpgsmdepay_debug)
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/* RTPGSMDepay signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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static GstStaticPadTemplate gst_rtp_gsm_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/x-gsm, " "rate = (int) 8000, " "channels = 1")
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);
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static GstStaticPadTemplate gst_rtp_gsm_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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"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\";"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_GSM_STRING ", "
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"clock-rate = (int) 8000")
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);
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static GstBuffer *gst_rtp_gsm_depay_process (GstRTPBaseDepayload * _depayload,
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GstRTPBuffer * rtp);
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static gboolean gst_rtp_gsm_depay_setcaps (GstRTPBaseDepayload * _depayload,
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GstCaps * caps);
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#define gst_rtp_gsm_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPGSMDepay, gst_rtp_gsm_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpgsmdepay, "rtpgsmdepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_DEPAY, rtp_element_init (plugin));
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static void
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gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbase_depayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbase_depayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_gsm_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_gsm_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP GSM depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts GSM audio from RTP packets", "Zeeshan Ali <zeenix@gmail.com>");
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gstrtpbase_depayload_class->process_rtp_packet = gst_rtp_gsm_depay_process;
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gstrtpbase_depayload_class->set_caps = gst_rtp_gsm_depay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpgsmdepay_debug, "rtpgsmdepay", 0,
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"GSM Audio RTP Depayloader");
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}
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static void
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gst_rtp_gsm_depay_init (GstRTPGSMDepay * rtpgsmdepay)
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{
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}
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static gboolean
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gst_rtp_gsm_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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gboolean ret;
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GstStructure *structure;
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gint clock_rate;
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 8000; /* default */
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depayload->clock_rate = clock_rate;
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srccaps = gst_caps_new_simple ("audio/x-gsm",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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return ret;
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}
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static GstBuffer *
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gst_rtp_gsm_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstBuffer *outbuf = NULL;
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gboolean marker;
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marker = gst_rtp_buffer_get_marker (rtp);
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GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
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gst_buffer_get_size (rtp->buffer), marker,
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gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
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if (marker && outbuf) {
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/* mark start of talkspurt with RESYNC */
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
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if (outbuf) {
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gst_rtp_drop_non_audio_meta (depayload, outbuf);
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}
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return outbuf;
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}
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