| /* GStreamer |
| * Copyright (C) <2007> Wim Taymans <wim.taymans@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 <stdlib.h> |
| #include <string.h> |
| #include <gst/rtp/gstrtpbuffer.h> |
| |
| #include "gstrtpmp2tpay.h" |
| |
| static GstStaticPadTemplate gst_rtp_mp2t_pay_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("video/mpegts," |
| "packetsize=(int)188," "systemstream=(boolean)true") |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_mp2t_pay_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("application/x-rtp, " |
| "media = (string) \"video\", " |
| "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " |
| "clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T\"") |
| ); |
| |
| static gboolean gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload, |
| GstCaps * caps); |
| static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload * |
| payload, GstBuffer * buffer); |
| static GstFlowReturn gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay); |
| static void gst_rtp_mp2t_pay_finalize (GObject * object); |
| |
| #define gst_rtp_mp2t_pay_parent_class parent_class |
| G_DEFINE_TYPE (GstRTPMP2TPay, gst_rtp_mp2t_pay, GST_TYPE_BASE_RTP_PAYLOAD); |
| |
| static void |
| gst_rtp_mp2t_pay_class_init (GstRTPMP2TPayClass * klass) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| GstBaseRTPPayloadClass *gstbasertppayload_class; |
| |
| gobject_class = (GObjectClass *) klass; |
| gstelement_class = (GstElementClass *) klass; |
| gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; |
| |
| gobject_class->finalize = gst_rtp_mp2t_pay_finalize; |
| |
| gstbasertppayload_class->set_caps = gst_rtp_mp2t_pay_setcaps; |
| gstbasertppayload_class->handle_buffer = gst_rtp_mp2t_pay_handle_buffer; |
| |
| gst_element_class_add_pad_template (gstelement_class, |
| gst_static_pad_template_get (&gst_rtp_mp2t_pay_sink_template)); |
| gst_element_class_add_pad_template (gstelement_class, |
| gst_static_pad_template_get (&gst_rtp_mp2t_pay_src_template)); |
| gst_element_class_set_details_simple (gstelement_class, |
| "RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network/RTP", |
| "Payload-encodes MPEG2 TS into RTP packets (RFC 2250)", |
| "Wim Taymans <wim.taymans@gmail.com>"); |
| } |
| |
| static void |
| gst_rtp_mp2t_pay_init (GstRTPMP2TPay * rtpmp2tpay) |
| { |
| GST_BASE_RTP_PAYLOAD (rtpmp2tpay)->clock_rate = 90000; |
| GST_BASE_RTP_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T; |
| |
| rtpmp2tpay->adapter = gst_adapter_new (); |
| } |
| |
| static void |
| gst_rtp_mp2t_pay_finalize (GObject * object) |
| { |
| GstRTPMP2TPay *rtpmp2tpay; |
| |
| rtpmp2tpay = GST_RTP_MP2T_PAY (object); |
| |
| g_object_unref (rtpmp2tpay->adapter); |
| rtpmp2tpay->adapter = NULL; |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| static gboolean |
| gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) |
| { |
| gboolean res; |
| |
| gst_basertppayload_set_options (payload, "video", TRUE, "MP2T", 90000); |
| res = gst_basertppayload_set_outcaps (payload, NULL); |
| |
| return res; |
| } |
| |
| static GstFlowReturn |
| gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay) |
| { |
| guint avail; |
| guint8 *payload; |
| GstFlowReturn ret; |
| GstBuffer *outbuf; |
| GstRTPBuffer rtp; |
| |
| avail = gst_adapter_available (rtpmp2tpay->adapter); |
| outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0); |
| |
| /* get payload */ |
| gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); |
| payload = gst_rtp_buffer_get_payload (&rtp); |
| |
| /* copy stuff from adapter to payload */ |
| gst_adapter_copy (rtpmp2tpay->adapter, payload, 0, avail); |
| gst_rtp_buffer_unmap (&rtp); |
| |
| GST_BUFFER_TIMESTAMP (outbuf) = rtpmp2tpay->first_ts; |
| GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration; |
| |
| GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d", |
| gst_buffer_get_size (outbuf)); |
| |
| ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf); |
| |
| /* flush the adapter content */ |
| gst_adapter_flush (rtpmp2tpay->adapter, avail); |
| |
| return ret; |
| } |
| |
| static GstFlowReturn |
| gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload * basepayload, |
| GstBuffer * buffer) |
| { |
| GstRTPMP2TPay *rtpmp2tpay; |
| guint size, avail, packet_len; |
| GstClockTime timestamp, duration; |
| GstFlowReturn ret; |
| |
| rtpmp2tpay = GST_RTP_MP2T_PAY (basepayload); |
| |
| size = gst_buffer_get_size (buffer); |
| timestamp = GST_BUFFER_TIMESTAMP (buffer); |
| duration = GST_BUFFER_DURATION (buffer); |
| |
| ret = GST_FLOW_OK; |
| avail = gst_adapter_available (rtpmp2tpay->adapter); |
| |
| /* Initialize new RTP payload */ |
| if (avail == 0) { |
| rtpmp2tpay->first_ts = timestamp; |
| rtpmp2tpay->duration = duration; |
| } |
| |
| /* get packet length of previous data and this new data, |
| * payload length includes a 4 byte header */ |
| packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0); |
| |
| /* if this buffer is going to overflow the packet, flush what we |
| * have. */ |
| if (gst_basertppayload_is_filled (basepayload, |
| packet_len, rtpmp2tpay->duration + duration)) { |
| ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay); |
| rtpmp2tpay->first_ts = timestamp; |
| rtpmp2tpay->duration = duration; |
| |
| /* keep filling the payload */ |
| } else { |
| if (GST_CLOCK_TIME_IS_VALID (duration)) |
| rtpmp2tpay->duration += duration; |
| } |
| |
| /* copy buffer to adapter */ |
| gst_adapter_push (rtpmp2tpay->adapter, buffer); |
| |
| return ret; |
| |
| } |
| |
| gboolean |
| gst_rtp_mp2t_pay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpmp2tpay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_MP2T_PAY); |
| } |