/* GStreamer * Copyright (C) <1999> Erik Walthinsen * Copyright (C) <2005> Edgard Lima * Copyright (C) <2005> Nokia Corporation * * 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 #include #include #include "gstrtppcmupay.h" /* elementfactory information */ static const GstElementDetails gst_rtp_pcmu_pay_details = GST_ELEMENT_DETAILS ("RTP packet payloader", "Codec/Payloader/Network", "Payload-encodes PCMU audio into a RTP packet", "Edgard Lima "); static GstStaticPadTemplate gst_rtp_pcmu_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-mulaw, channels=(int)1, rate=(int)8000") ); static GstStaticPadTemplate gst_rtp_pcmu_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"; " "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"") ); static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload, GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); static void gst_rtp_pcmu_pay_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_pcmu_pay_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_pcmu_pay_src_template)); gst_element_class_set_details (element_class, &gst_rtp_pcmu_pay_details); } static void gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps; } static void gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass) { GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay); GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000; /* tell basertpaudiopayload that this is a sample based codec */ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload); /* octet-per-sample is 1 for PCM */ gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1); } static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { gboolean res; payload->pt = GST_RTP_PAYLOAD_PCMU; gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000); res = gst_basertppayload_set_outcaps (payload, NULL); return res; } gboolean gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtppcmupay", GST_RANK_NONE, GST_TYPE_RTP_PCMU_PAY); }