/* 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, 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) /* elementfactory information */ static GstElementDetails gst_rtp_gsmdepay_details = { "RTP GSM depayloader", "Codec/Depayloader/Network", "Extracts GSM audio from RTP packets", "Zeeshan Ali <zeenix@gmail.com>" }; /* 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\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "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 (GstBaseRTPDepayload * _depayload, GstBuffer * buf); static gboolean gst_rtp_gsm_depay_setcaps (GstBaseRTPDepayload * _depayload, GstCaps * caps); GST_BOILERPLATE (GstRTPGSMDepay, gst_rtp_gsm_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static void gst_rtp_gsm_depay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_gsm_depay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_gsm_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_gsmdepay_details); } static void gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass) { GstBaseRTPDepayloadClass *gstbasertp_depayload_class; gstbasertp_depayload_class = (GstBaseRTPDepayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertp_depayload_class->process = gst_rtp_gsm_depay_process; gstbasertp_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, GstRTPGSMDepayClass * klass) { /* needed because of GST_BOILERPLATE */ } static gboolean gst_rtp_gsm_depay_setcaps (GstBaseRTPDepayload * 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_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps); gst_caps_unref (srccaps); return ret; } static GstBuffer * gst_rtp_gsm_depay_process (GstBaseRTPDepayload * _depayload, GstBuffer * buf) { GstBuffer *outbuf = NULL; gboolean marker; marker = gst_rtp_buffer_get_marker (buf); GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d", GST_BUFFER_SIZE (buf), marker, gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf)); outbuf = gst_rtp_buffer_get_payload_buffer (buf); if (marker) { /* mark start of talkspurt with DISCONT */ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); } return outbuf; } gboolean gst_rtp_gsm_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpgsmdepay", GST_RANK_MARGINAL, GST_TYPE_RTP_GSM_DEPAY); }