gstreamer/gst/rtp/gstrtph263ppay.c
Wim Taymans 33f18b8ea4 Merge branch 'master' into 0.11
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
2011-09-06 16:06:25 +02:00

360 lines
11 KiB
C

/* GStreamer
* Copyright (C) <2005> 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., 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 "gstrtph263ppay.h"
#define DEFAULT_FRAGMENTATION_MODE GST_FRAGMENTATION_MODE_NORMAL
enum
{
PROP_0,
PROP_FRAGMENTATION_MODE
};
#define GST_TYPE_FRAGMENTATION_MODE (gst_fragmentation_mode_get_type())
static GType
gst_fragmentation_mode_get_type (void)
{
static GType fragmentation_mode_type = 0;
static const GEnumValue fragmentation_mode[] = {
{GST_FRAGMENTATION_MODE_NORMAL, "Normal", "normal"},
{GST_FRAGMENTATION_MODE_SYNC, "Fragment at sync points", "sync"},
{0, NULL, NULL},
};
if (!fragmentation_mode_type) {
fragmentation_mode_type =
g_enum_register_static ("GstFragmentationMode", fragmentation_mode);
}
return fragmentation_mode_type;
}
GST_DEBUG_CATEGORY_STATIC (rtph263ppay_debug);
#define GST_CAT_DEFAULT rtph263ppay_debug
static GstStaticPadTemplate gst_rtp_h263p_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-h263, " "variant = (string) \"itu\" ")
);
static GstStaticPadTemplate gst_rtp_h263p_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) \"H263-1998\"; "
"application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"H263-2000\"")
);
static void gst_rtp_h263p_pay_finalize (GObject * object);
static void gst_rtp_h263p_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_h263p_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_h263p_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_h263p_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
#define gst_rtp_h263p_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpH263PPay, gst_rtp_h263p_pay, GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_h263p_pay_class_init (GstRtpH263PPayClass * 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_h263p_pay_finalize;
gobject_class->set_property = gst_rtp_h263p_pay_set_property;
gobject_class->get_property = gst_rtp_h263p_pay_get_property;
gstbasertppayload_class->set_caps = gst_rtp_h263p_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_h263p_pay_handle_buffer;
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_FRAGMENTATION_MODE, g_param_spec_enum ("fragmentation-mode",
"Fragmentation Mode",
"Packet Fragmentation Mode", GST_TYPE_FRAGMENTATION_MODE,
DEFAULT_FRAGMENTATION_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_h263p_pay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_h263p_pay_sink_template));
gst_element_class_set_details_simple (gstelement_class, "RTP H263 payloader",
"Codec/Payloader/Network/RTP",
"Payload-encodes H263/+/++ video in RTP packets (RFC 4629)",
"Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (rtph263ppay_debug, "rtph263ppay",
0, "rtph263ppay (RFC 4629)");
}
static void
gst_rtp_h263p_pay_init (GstRtpH263PPay * rtph263ppay)
{
rtph263ppay->adapter = gst_adapter_new ();
rtph263ppay->fragmentation_mode = DEFAULT_FRAGMENTATION_MODE;
}
static void
gst_rtp_h263p_pay_finalize (GObject * object)
{
GstRtpH263PPay *rtph263ppay;
rtph263ppay = GST_RTP_H263P_PAY (object);
g_object_unref (rtph263ppay->adapter);
rtph263ppay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_h263p_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gboolean res;
GstCaps *peercaps;
gchar *encoding_name = NULL;
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
peercaps =
gst_pad_peer_get_caps (GST_BASE_RTP_PAYLOAD_SRCPAD (payload), NULL);
if (peercaps) {
GstCaps *intersect = gst_caps_intersect (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_RTP_PAYLOAD_SRCPAD (payload)));
gst_caps_unref (peercaps);
if (!gst_caps_is_empty (intersect)) {
GstStructure *s = gst_caps_get_structure (intersect, 0);
encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name"));
}
gst_caps_unref (intersect);
}
if (!encoding_name)
encoding_name = g_strdup ("H263-1998");
gst_basertppayload_set_options (payload, "video", TRUE,
(gchar *) encoding_name, 90000);
res = gst_basertppayload_set_outcaps (payload, NULL);
g_free (encoding_name);
return res;
}
static void
gst_rtp_h263p_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpH263PPay *rtph263ppay;
rtph263ppay = GST_RTP_H263P_PAY (object);
switch (prop_id) {
case PROP_FRAGMENTATION_MODE:
rtph263ppay->fragmentation_mode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_h263p_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpH263PPay *rtph263ppay;
rtph263ppay = GST_RTP_H263P_PAY (object);
switch (prop_id) {
case PROP_FRAGMENTATION_MODE:
g_value_set_enum (value, rtph263ppay->fragmentation_mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstFlowReturn
gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
gboolean fragmented;
avail = gst_adapter_available (rtph263ppay->adapter);
if (avail == 0)
return GST_FLOW_OK;
fragmented = FALSE;
/* This algorithm assumes the H263/+/++ encoder sends complete frames in each
* buffer */
/* With Fragmentation Mode at GST_FRAGMENTATION_MODE_NORMAL:
* This algorithm implements the Follow-on packets method for packetization.
* This assumes low packet loss network.
* With Fragmentation Mode at GST_FRAGMENTATION_MODE_SYNC:
* This algorithm separates large frames at synchronisation points (Segments)
* (See RFC 4629 section 6). It would be interesting to have a property such as network
* quality to select between both packetization methods */
/* TODO Add VRC supprt (See RFC 4629 section 5.2) */
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
gint header_len;
guint next_gop = 0;
gboolean found_gob = FALSE;
GstRTPBuffer rtp = { NULL };
if (rtph263ppay->fragmentation_mode == GST_FRAGMENTATION_MODE_SYNC) {
/* start after 1st gop possible */
guint parsed_len = 3;
const guint8 *parse_data = NULL;
parse_data = gst_adapter_map (rtph263ppay->adapter, avail);
/* Check if we have a gob or eos , eossbs */
/* FIXME EOS and EOSSBS packets should never contain any gobs and vice-versa */
if (avail >= 3 && *parse_data == 0 && *(parse_data + 1) == 0
&& *(parse_data + 2) >= 0x80) {
GST_DEBUG_OBJECT (rtph263ppay, " Found GOB header");
found_gob = TRUE;
}
/* Find next and cut the packet accordingly */
/* TODO we should get as many gobs as possible until MTU is reached, this
* code seems to just get one GOB per packet */
while (parsed_len + 2 < avail) {
if (parse_data[parsed_len] == 0 && parse_data[parsed_len + 1] == 0
&& parse_data[parsed_len + 2] >= 0x80) {
next_gop = parsed_len;
GST_DEBUG_OBJECT (rtph263ppay, " Next GOB Detected at : %d",
next_gop);
break;
}
parsed_len++;
}
gst_adapter_unmap (rtph263ppay->adapter, 0);
}
/* for picture start frames (non-fragmented), we need to remove the first
* two 0x00 bytes and set P=1 */
header_len = (fragmented && !found_gob) ? 2 : 0;
towrite = MIN (avail, gst_rtp_buffer_calc_payload_len
(GST_BASE_RTP_PAYLOAD_MTU (rtph263ppay) - header_len, 0, 0));
if (next_gop > 0)
towrite = MIN (next_gop, towrite);
payload_len = header_len + towrite;
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* last fragment gets the marker bit set */
gst_rtp_buffer_set_marker (&rtp, avail > towrite ? 0 : 1);
payload = gst_rtp_buffer_get_payload (&rtp);
gst_adapter_copy (rtph263ppay->adapter, &payload[header_len], 0, towrite);
/* 0 1
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | RR |P|V| PLEN |PEBIT|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
/* if fragmented or gop header , write p bit =1 */
payload[0] = (fragmented && !found_gob) ? 0x00 : 0x04;
payload[1] = 0;
GST_BUFFER_TIMESTAMP (outbuf) = rtph263ppay->first_timestamp;
GST_BUFFER_DURATION (outbuf) = rtph263ppay->first_duration;
gst_rtp_buffer_unmap (&rtp);
gst_adapter_flush (rtph263ppay->adapter, towrite);
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
avail -= towrite;
fragmented = TRUE;
}
return ret;
}
static GstFlowReturn
gst_rtp_h263p_pay_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer)
{
GstRtpH263PPay *rtph263ppay;
GstFlowReturn ret;
rtph263ppay = GST_RTP_H263P_PAY (payload);
rtph263ppay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
rtph263ppay->first_duration = GST_BUFFER_DURATION (buffer);
/* we always encode and flush a full picture */
gst_adapter_push (rtph263ppay->adapter, buffer);
ret = gst_rtp_h263p_pay_flush (rtph263ppay);
return ret;
}
gboolean
gst_rtp_h263p_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtph263ppay",
GST_RANK_SECONDARY, GST_TYPE_RTP_H263P_PAY);
}