gstreamer/gst/rtp/gstrtpmp2tpay.c
Tim-Philipp Müller 92c1d289b8 rtpmp2tpay: fix up template caps so we can output the default pt 33
Add fixed payload type for mp2t to template caps as well, so
our output caps match the advertised default pt. Fixes a
regression from 1.2.

There's still something wrong with caps negotiation though,
rtpmp2tpay payload=96 ! fakesink will not output caps with
payload=96.
2014-11-01 12:40:07 +00:00

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_TIMESTAMP (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_TIMESTAMP (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);
}