/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * 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 "gstrtpgsmenc.h" /* elementfactory information */ static GstElementDetails gst_rtpgsmenc_details = { "RTP GSM Audio Encoder", "Codec/Encoder/Network", "Encodes GSM audio into a RTP packet", "Zeeshan Ali " }; static GstStaticPadTemplate gst_rtpgsmenc_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) [ 1000, 48000 ]") ); static GstStaticPadTemplate gst_rtpgsmenc_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) [ 96, 255 ], " "clock_rate = (int) 8000, " "encoding_name = (string) \"GSM\"") ); static void gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass); static void gst_rtpgsmenc_base_init (GstRtpGSMEncClass * klass); static void gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc); static gboolean gst_rtpgsmenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtpgsmenc_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); static GstBaseRTPPayloadClass *parent_class = NULL; static GType gst_rtpgsmenc_get_type (void) { static GType rtpgsmenc_type = 0; if (!rtpgsmenc_type) { static const GTypeInfo rtpgsmenc_info = { sizeof (GstRtpGSMEncClass), (GBaseInitFunc) gst_rtpgsmenc_base_init, NULL, (GClassInitFunc) gst_rtpgsmenc_class_init, NULL, NULL, sizeof (GstRtpGSMEnc), 0, (GInstanceInitFunc) gst_rtpgsmenc_init, }; rtpgsmenc_type = g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpGSMEnc", &rtpgsmenc_info, 0); } return rtpgsmenc_type; } static void gst_rtpgsmenc_base_init (GstRtpGSMEncClass * klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtpgsmenc_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtpgsmenc_src_template)); gst_element_class_set_details (element_class, &gst_rtpgsmenc_details); } static void gst_rtpgsmenc_class_init (GstRtpGSMEncClass * 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_ref (GST_TYPE_BASE_RTP_PAYLOAD); gstbasertppayload_class->set_caps = gst_rtpgsmenc_setcaps; gstbasertppayload_class->handle_buffer = gst_rtpgsmenc_handle_buffer; } static void gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc) { rtpgsmenc->frequency = 8000; } static gboolean gst_rtpgsmenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { GstRtpGSMEnc *rtpgsmenc; GstStructure *structure; gboolean ret; GstCaps *srccaps; rtpgsmenc = GST_RTP_GSM_ENC (payload); structure = gst_caps_get_structure (caps, 0); ret = gst_structure_get_int (structure, "rate", &rtpgsmenc->frequency); if (!ret) return FALSE; srccaps = gst_caps_new_simple ("application/x-rtp", NULL); gst_pad_set_caps (GST_BASE_RTP_PAYLOAD_SRCPAD (rtpgsmenc), srccaps); gst_caps_unref (srccaps); return TRUE; } static GstFlowReturn gst_rtpgsmenc_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpGSMEnc *rtpgsmenc; guint size, payload_len; GstBuffer *outbuf; guint8 *payload, *data; GstClockTime timestamp; GstFlowReturn ret; rtpgsmenc = GST_RTP_GSM_ENC (basepayload); size = GST_BUFFER_SIZE (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); /* FIXME, only one GSM frame per RTP packet for now */ payload_len = size; outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0); /* FIXME, assert for now */ g_assert (GST_BUFFER_SIZE (outbuf) < GST_BASE_RTP_PAYLOAD_MTU (rtpgsmenc)); gst_rtpbuffer_set_timestamp (outbuf, timestamp * 8000 / GST_SECOND); /* copy timestamp */ GST_BUFFER_TIMESTAMP (outbuf) = timestamp; /* get payload */ payload = gst_rtpbuffer_get_payload (outbuf); data = GST_BUFFER_DATA (buffer); /* copy data in payload */ memcpy (&payload[0], data, size); gst_buffer_unref (buffer); GST_DEBUG ("gst_rtpgsmenc_chain: pushing buffer of size %d", GST_BUFFER_SIZE (outbuf)); ret = gst_basertppayload_push (basepayload, outbuf); return ret; } gboolean gst_rtpgsmenc_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpgsmenc", GST_RANK_NONE, GST_TYPE_RTP_GSM_ENC); }