mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
RTP buffer allocation should not be done with padding for the specific MPEG2 header as the padding is done at the end of the buffer and the last byte is the size of the padding. https://bugzilla.gnome.org/show_bug.cgi?id=706970
313 lines
8.8 KiB
C
313 lines
8.8 KiB
C
/* GStreamer
|
|
* Copyright (C) <2007> Thijs Vermeir <thijsvermeir@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 "gstrtpmpvpay.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (rtpmpvpay_debug);
|
|
#define GST_CAT_DEFAULT (rtpmpvpay_debug)
|
|
|
|
static GstStaticPadTemplate gst_rtp_mpv_pay_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("video/mpeg, "
|
|
"mpegversion = (int) 2, systemstream = (boolean) FALSE")
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_mpv_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_MPV_STRING ", "
|
|
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPV\"")
|
|
);
|
|
|
|
static GstStateChangeReturn gst_rtp_mpv_pay_change_state (GstElement * element,
|
|
GstStateChange transition);
|
|
|
|
static void gst_rtp_mpv_pay_finalize (GObject * object);
|
|
|
|
static GstFlowReturn gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay);
|
|
static gboolean gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload,
|
|
GstCaps * caps);
|
|
static GstFlowReturn gst_rtp_mpv_pay_handle_buffer (GstRTPBasePayload *
|
|
payload, GstBuffer * buffer);
|
|
static gboolean gst_rtp_mpv_pay_sink_event (GstRTPBasePayload * payload,
|
|
GstEvent * event);
|
|
|
|
#define gst_rtp_mpv_pay_parent_class parent_class
|
|
G_DEFINE_TYPE (GstRTPMPVPay, gst_rtp_mpv_pay, GST_TYPE_RTP_BASE_PAYLOAD);
|
|
|
|
static void
|
|
gst_rtp_mpv_pay_class_init (GstRTPMPVPayClass * 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_mpv_pay_finalize;
|
|
|
|
gstelement_class->change_state = gst_rtp_mpv_pay_change_state;
|
|
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_rtp_mpv_pay_sink_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_rtp_mpv_pay_src_template));
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"RTP MPEG2 ES video payloader", "Codec/Payloader/Network/RTP",
|
|
"Payload-encodes MPEG2 ES into RTP packets (RFC 2250)",
|
|
"Thijs Vermeir <thijsvermeir@gmail.com>");
|
|
|
|
gstrtpbasepayload_class->set_caps = gst_rtp_mpv_pay_setcaps;
|
|
gstrtpbasepayload_class->handle_buffer = gst_rtp_mpv_pay_handle_buffer;
|
|
gstrtpbasepayload_class->sink_event = gst_rtp_mpv_pay_sink_event;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtpmpvpay_debug, "rtpmpvpay", 0,
|
|
"MPEG2 ES Video RTP Payloader");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mpv_pay_init (GstRTPMPVPay * rtpmpvpay)
|
|
{
|
|
GST_RTP_BASE_PAYLOAD (rtpmpvpay)->clock_rate = 90000;
|
|
GST_RTP_BASE_PAYLOAD_PT (rtpmpvpay) = GST_RTP_PAYLOAD_MPV;
|
|
|
|
rtpmpvpay->adapter = gst_adapter_new ();
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mpv_pay_finalize (GObject * object)
|
|
{
|
|
GstRTPMPVPay *rtpmpvpay;
|
|
|
|
rtpmpvpay = GST_RTP_MPV_PAY (object);
|
|
|
|
g_object_unref (rtpmpvpay->adapter);
|
|
rtpmpvpay->adapter = NULL;
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mpv_pay_reset (GstRTPMPVPay * pay)
|
|
{
|
|
pay->first_ts = -1;
|
|
pay->duration = 0;
|
|
gst_adapter_clear (pay->adapter);
|
|
GST_DEBUG_OBJECT (pay, "reset depayloader");
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
|
|
{
|
|
gst_rtp_base_payload_set_options (payload, "video", FALSE, "MPV", 90000);
|
|
return gst_rtp_base_payload_set_outcaps (payload, NULL);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_mpv_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
|
|
{
|
|
gboolean ret;
|
|
GstRTPMPVPay *rtpmpvpay;
|
|
|
|
rtpmpvpay = GST_RTP_MPV_PAY (payload);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
/* make sure we push the last packets in the adapter on EOS */
|
|
gst_rtp_mpv_pay_flush (rtpmpvpay);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_rtp_mpv_pay_reset (rtpmpvpay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
|
|
{
|
|
GstBuffer *outbuf;
|
|
GstFlowReturn ret;
|
|
guint avail;
|
|
|
|
guint8 *payload;
|
|
|
|
avail = gst_adapter_available (rtpmpvpay->adapter);
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
while (avail > 0) {
|
|
guint towrite;
|
|
guint packet_len;
|
|
guint payload_len;
|
|
GstRTPBuffer rtp = { NULL };
|
|
GstBuffer *paybuf;
|
|
|
|
packet_len = gst_rtp_buffer_calc_packet_len (avail + 4, 0, 0);
|
|
|
|
towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpvpay));
|
|
|
|
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
|
|
|
|
outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
|
|
|
|
payload_len -= 4;
|
|
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
|
|
|
|
payload = gst_rtp_buffer_get_payload (&rtp);
|
|
/* enable MPEG Video-specific header
|
|
*
|
|
* 0 1 2 3
|
|
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
* | MBZ |T| TR | |N|S|B|E| P | | BFC | | FFC |
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
* AN FBV FFV
|
|
*/
|
|
|
|
/* fill in the MPEG Video-specific header
|
|
* data is set to 0x0 here
|
|
*/
|
|
memset (payload, 0x0, 4);
|
|
|
|
avail -= payload_len;
|
|
|
|
gst_rtp_buffer_set_marker (&rtp, avail == 0);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
paybuf = gst_adapter_take_buffer_fast (rtpmpvpay->adapter, payload_len);
|
|
outbuf = gst_buffer_append (outbuf, paybuf);
|
|
|
|
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
|
|
|
|
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmpvpay), outbuf);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_mpv_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRTPMPVPay *rtpmpvpay;
|
|
guint avail, packet_len;
|
|
GstClockTime timestamp, duration;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
rtpmpvpay = GST_RTP_MPV_PAY (basepayload);
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
|
|
if (GST_BUFFER_IS_DISCONT (buffer)) {
|
|
GST_DEBUG_OBJECT (rtpmpvpay, "DISCONT");
|
|
gst_rtp_mpv_pay_reset (rtpmpvpay);
|
|
}
|
|
|
|
avail = gst_adapter_available (rtpmpvpay->adapter);
|
|
|
|
if (duration == -1)
|
|
duration = 0;
|
|
|
|
if (rtpmpvpay->first_ts == GST_CLOCK_TIME_NONE || avail == 0)
|
|
rtpmpvpay->first_ts = timestamp;
|
|
|
|
if (avail == 0) {
|
|
rtpmpvpay->duration = duration;
|
|
} else {
|
|
rtpmpvpay->duration += duration;
|
|
}
|
|
|
|
gst_adapter_push (rtpmpvpay->adapter, buffer);
|
|
avail = gst_adapter_available (rtpmpvpay->adapter);
|
|
|
|
/* get packet length of previous data and this new data,
|
|
* payload length includes a 4 byte MPEG video-specific header */
|
|
packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
|
|
GST_LOG_OBJECT (rtpmpvpay, "available %d, rtp packet length %d", avail,
|
|
packet_len);
|
|
|
|
if (gst_rtp_base_payload_is_filled (basepayload,
|
|
packet_len, rtpmpvpay->duration)) {
|
|
ret = gst_rtp_mpv_pay_flush (rtpmpvpay);
|
|
} else {
|
|
rtpmpvpay->first_ts = timestamp;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_mpv_pay_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstRTPMPVPay *rtpmpvpay;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpmpvpay = GST_RTP_MPV_PAY (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_rtp_mpv_pay_reset (rtpmpvpay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtp_mpv_pay_reset (rtpmpvpay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
|
|
gboolean
|
|
gst_rtp_mpv_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmpvpay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_MPV_PAY);
|
|
}
|