/* 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <stdlib.h> #include <string.h> #include <gst/rtp/gstrtpbuffer.h> #include "gstrtpgsmpay.h" GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug); #define GST_CAT_DEFAULT (rtpgsmpay_debug) static GstStaticPadTemplate gst_rtp_gsm_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1") ); static GstStaticPadTemplate gst_rtp_gsm_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_GSM_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"; " "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"") ); static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer); #define gst_rtp_gsm_pay_parent_class parent_class G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_RTP_BASE_PAYLOAD); static void gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass) { GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0, "GSM Audio RTP Payloader"); gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_gsm_pay_sink_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template)); gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader", "Codec/Payloader/Network/RTP", "Payload-encodes GSM audio into a RTP packet", "Zeeshan Ali <zeenix@gmail.com>"); gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps; gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer; } static void gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay) { GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000; GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM; } static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) { const char *stname; GstStructure *structure; gboolean res; structure = gst_caps_get_structure (caps, 0); stname = gst_structure_get_name (structure); if (strcmp ("audio/x-gsm", stname)) goto invalid_type; gst_rtp_base_payload_set_options (payload, "audio", FALSE, "GSM", 8000); res = gst_rtp_base_payload_set_outcaps (payload, NULL); return res; /* ERRORS */ invalid_type: { GST_WARNING_OBJECT (payload, "invalid media type received"); return FALSE; } } static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstRTPGSMPay *rtpgsmpay; guint payload_len; GstBuffer *outbuf; GstMapInfo map; guint8 *payload; GstClockTime timestamp, duration; GstFlowReturn ret; GstRTPBuffer rtp = { NULL }; rtpgsmpay = GST_RTP_GSM_PAY (basepayload); gst_buffer_map (buffer, &map, GST_MAP_READ); timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); /* FIXME, only one GSM frame per RTP packet for now */ payload_len = map.size; /* FIXME, just error out for now */ if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)) goto too_big; outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); /* copy timestamp and duration */ GST_BUFFER_TIMESTAMP (outbuf) = timestamp; GST_BUFFER_DURATION (outbuf) = duration; /* get payload */ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); /* copy data in payload */ payload = gst_rtp_buffer_get_payload (&rtp); memcpy (payload, map.data, map.size); gst_rtp_buffer_unmap (&rtp); gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT, gst_buffer_get_size (outbuf)); ret = gst_rtp_base_payload_push (basepayload, outbuf); return ret; /* ERRORS */ too_big: { GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL), ("payload_len %u > mtu %u", payload_len, GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))); gst_buffer_unmap (buffer, &map); return GST_FLOW_ERROR; } } gboolean gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpgsmpay", GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_PAY); }