gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpmpvpay.c

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/* 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 <gst/video/video.h>
#include "gstrtpelements.h"
#include "gstrtpmpvpay.h"
#include "gstrtputils.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\"; "
"application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_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);
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static gboolean gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
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static GstFlowReturn gst_rtp_mpv_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
static gboolean gst_rtp_mpv_pay_sink_event (GstRTPBasePayload * payload,
GstEvent * event);
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#define gst_rtp_mpv_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPMPVPay, gst_rtp_mpv_pay, GST_TYPE_RTP_BASE_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmpvpay, "rtpmpvpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MPV_PAY, rtp_element_init (plugin));
static void
gst_rtp_mpv_pay_class_init (GstRTPMPVPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
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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_static_pad_template (gstelement_class,
&gst_rtp_mpv_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mpv_pay_src_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG2 ES video payloader", "Codec/Payloader/Network/RTP",
"Payload-encodes MPEG2 ES into RTP packets (RFC 2250)",
"Thijs Vermeir <thijsvermeir@gmail.com>");
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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
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gst_rtp_mpv_pay_init (GstRTPMPVPay * rtpmpvpay)
{
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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
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gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gst_rtp_base_payload_set_options (payload, "video",
payload->pt != GST_RTP_PAYLOAD_MPV, "MPV", 90000);
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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;
}
#define RTP_HEADER_LEN 12
static GstFlowReturn
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
{
GstFlowReturn ret;
guint avail;
GstBufferList *list;
GstBuffer *outbuf;
guint8 *payload;
avail = gst_adapter_available (rtpmpvpay->adapter);
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (rtpmpvpay, "available %u", avail);
if (avail == 0)
return GST_FLOW_OK;
list =
gst_buffer_list_new_sized (avail / (GST_RTP_BASE_PAYLOAD_MTU (rtpmpvpay) -
RTP_HEADER_LEN) + 1);
while (avail > 0) {
guint towrite;
guint packet_len;
guint payload_len;
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GstRTPBuffer rtp = { NULL };
GstBuffer *paybuf;
packet_len = gst_rtp_buffer_calc_packet_len (avail + 4, 0, 0);
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towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpvpay));
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
outbuf =
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
(rtpmpvpay), 4, 0, 0);
payload_len -= 4;
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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;
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gst_rtp_buffer_set_marker (&rtp, avail == 0);
if (avail == 0)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
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gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (rtpmpvpay->adapter, payload_len);
gst_rtp_copy_video_meta (rtpmpvpay, outbuf, paybuf);
outbuf = gst_buffer_append (outbuf, paybuf);
GST_DEBUG_OBJECT (rtpmpvpay, "Adding buffer");
GST_BUFFER_PTS (outbuf) = rtpmpvpay->first_ts;
gst_buffer_list_add (list, outbuf);
}
ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmpvpay), list);
return ret;
}
static GstFlowReturn
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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_PTS (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);
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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;
}