gstreamer/gst/rtp/gstrtph264pay.c
Laurent Glayal d94a696bcd gst/rtp/: Added H264 payloader. Fixes #423782.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
(gst_rtp_h264_pay_plugin_init):
* gst/rtp/gstrtph264pay.h:
Added H264 payloader. Fixes #423782.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Small fixes.
2007-03-29 08:08:49 +00:00

337 lines
9.2 KiB
C

/* GStreamer
*
* 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 <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtph264pay.h"
GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
#define GST_CAT_DEFAULT (rtph264pay_debug)
/* references:
*
* RFC 3984
*/
/* elementfactory information */
static const GstElementDetails gst_rtp_h264pay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload-encode H264 video into RTP packets (RFC 3984)",
"Laurent Glayal <spglegle@yahoo.fr>");
static GstStaticPadTemplate gst_rtp_h264_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-h264")
);
static GstStaticPadTemplate gst_rtp_h264_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) \"H264\"")
);
static void gst_rtp_h264_pay_finalize (GObject * object);
static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
GST_BOILERPLATE (GstRtpH264Pay, gst_rtp_h264_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_h264_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_h264_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_h264_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_h264pay_details);
}
static void
gst_rtp_h264_pay_class_init (GstRtpH264PayClass * 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_h264_pay_finalize;
gobject_class->set_property = gst_rtp_h264_pay_set_property;
gobject_class->get_property = gst_rtp_h264_pay_get_property;
gstelement_class->change_state = gst_rtp_h264_pay_change_state;
gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
"H264 RTP Payloader");
}
static void
gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay, GstRtpH264PayClass * klass)
{
}
static void
gst_rtp_h264_pay_finalize (GObject * object)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (basepayload);
gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
gst_basertppayload_set_outcaps (basepayload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint size, idxdata;
GstBuffer *outbuf;
guint8 *payload, *data, *pdata;
guint8 nalType;
GstClockTime timestamp;
guint packet_len, payload_len, mtu;
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
/* H264 stream analysis */
pdata = data;
idxdata = size;
while (idxdata > 5 &&
(pdata[0] != 0x00 || pdata[1] != 0x00 || pdata[2] != 0x1 ||
(pdata[3] & 0x1f) < 1 || (pdata[3] & 0x1f) > 23)
) {
pdata++;
idxdata--;
GST_DEBUG_OBJECT (basepayload, "idxdata=%d", idxdata);
}
if (idxdata < 5) {
GST_DEBUG_OBJECT (basepayload,
"Returning GST_FLOW_OK without creating RTP packet");
return GST_FLOW_OK;
}
pdata += 3;
idxdata -= 3;
nalType = pdata[0] & 0x1f;
GST_DEBUG_OBJECT (basepayload, "Processing Buffer with NAL TYPE=%d", nalType);
packet_len = gst_rtp_buffer_calc_packet_len (idxdata, 0, 0);
if (packet_len < mtu) {
GST_DEBUG_OBJECT (basepayload,
"NAL Unit fit in one packet datasize=%d mtu=%d", idxdata, mtu);
/* will fit in one packet */
outbuf = gst_rtp_buffer_new_allocate (idxdata, 0, 0);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
gst_rtp_buffer_set_marker (outbuf, 1);
payload = gst_rtp_buffer_get_payload (outbuf);
GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", idxdata);
memcpy (payload, pdata, idxdata);
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
} else {
GST_DEBUG_OBJECT (basepayload,
"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", idxdata, mtu);
/* Fragmentation Units FU-A */
guint8 nalHeader;
guint limitedSize;
int ii = 0, start = 1, end = 0, first = 0;
nalHeader = *pdata;
pdata++;
idxdata--;
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
idxdata);
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0); /* We keep 2 bytes for FU indicator and FU Header */
while (end == 0) {
limitedSize = idxdata < payload_len ? idxdata : payload_len;
GST_DEBUG_OBJECT (basepayload,
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
ii);
outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
gst_rtp_buffer_set_marker (outbuf, end);
payload = gst_rtp_buffer_get_payload (outbuf);
if (limitedSize == idxdata) {
GST_DEBUG_OBJECT (basepayload, "end idxdata=%d iteration=%d", idxdata,
ii);
end = 1;
}
/* FU indicator */
payload[0] = (nalHeader & 0x60) | 28;
/* FU Header */
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
memcpy (&payload[2], pdata + first, limitedSize);
GST_DEBUG_OBJECT (basepayload,
"recorded %d payload bytes into packet iteration=%d", limitedSize + 2,
ii);
ret = gst_basertppayload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
idxdata -= limitedSize;
first += limitedSize;
ii++;
start = 0;
}
gst_buffer_unref (buffer);
return ret;
}
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("Should not be there !!"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
static void
gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpH264Pay *rtph264pay;
GstStateChangeReturn ret;
rtph264pay = GST_RTP_H264_PAY (element);
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
gboolean
gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtph264pay",
GST_RANK_NONE, GST_TYPE_RTP_H264_PAY);
}