gstreamer/gst/rtp/gstrtpmp2tpay.c
Sebastian Dröge dc059efa60 rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
The mix between all these in the RTP code is confusing, let's try to be
consistent.
2015-06-10 14:34:47 +02:00

234 lines
6.9 KiB
C

/* 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., 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 "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_MP2T_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T\" ; "
"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 (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload *
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_RTP_BASE_PAYLOAD);
static void
gst_rtp_mp2t_pay_class_init (GstRTPMP2TPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp2t_pay_finalize;
gstrtpbasepayload_class->set_caps = gst_rtp_mp2t_pay_setcaps;
gstrtpbasepayload_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_static_metadata (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_RTP_BASE_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
GST_RTP_BASE_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 (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "video", TRUE, "MP2T", 90000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
guint avail, mtu;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
avail = gst_adapter_available (rtpmp2tpay->adapter);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp2tpay);
while (avail > 0 && (ret == GST_FLOW_OK)) {
guint towrite;
guint payload_len;
guint packet_len;
GstBuffer *paybuf;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, mtu);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
payload_len -= payload_len % 188;
/* need whole packets */
if (!payload_len)
break;
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
/* get payload */
paybuf = gst_adapter_take_buffer_fast (rtpmp2tpay->adapter, payload_len);
outbuf = gst_buffer_append (outbuf, paybuf);
avail -= payload_len;
GST_BUFFER_PTS (outbuf) = rtpmp2tpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %u",
(guint) gst_buffer_get_size (outbuf));
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp2tpay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload * 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_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
again:
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 */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we have,
* or if upstream is handing us several packets, to keep latency low */
if (!size || gst_rtp_base_payload_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 */
if (buffer) {
gst_adapter_push (rtpmp2tpay->adapter, buffer);
buffer = NULL;
}
if (size >= (188 * 2)) {
size = 0;
goto again;
}
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);
}