gstreamer/gst/rtp/gstrtpmp4vpay.c
Philippe Kalaf 9a6ca70be2 gst-plugins-good/gst/rtp/: Fixed payload range in payloder caps. Removed payload range completly from depayloaders as...
Original commit message from CVS:
2005-12-14  Philippe Khalaf  <burger@speedy.org>

* gst-plugins-good/gst/rtp/gstasteriskh263.c:
* gst-plugins-good/gst/rtp/gstrtpamrdepay.c:
* gst-plugins-good/gst/rtp/gstrtpamrpay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpgsmdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263pay.c:
* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263ppay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vpay.c:
* gst-plugins-good/gst/rtp/gstrtpmpadepay.c:
* gst-plugins-good/gst/rtp/gstrtpmpapay.c:
* gst-plugins-good/gst/rtp/README:
Fixed payload range in payloder caps. Removed payload range completly from
depayloaders as they don't require payload type in their caps. In effect,
there isn't any specific payload type for any given codec, only suggestions.
Fixes bug #324011.
2005-12-14 18:07:16 +00:00

449 lines
12 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmp4vpay.h"
/* elementfactory information */
static GstElementDetails gst_rtp_mp4vpay_details = {
"RTP packet parser",
"Codec/Payloader/Network",
"Payode MPEG4 video as RTP packets (RFC 3016)",
"Wim Taymans <wim@fluendo.com>"
};
static GstStaticPadTemplate gst_rtp_mp4v_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/mpeg,"
"mpegversion=(int) 4," "systemstream=(boolean)false")
);
static GstStaticPadTemplate gst_rtp_mp4v_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) [ 96, 127 ], "
"clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4V-ES\""
/* two string params
*
"profile-level-id = (string) [1,MAX]"
"config = (string) [1,MAX]"
*/
)
);
#define DEFAULT_SEND_CONFIG FALSE
enum
{
ARG_0,
ARG_SEND_CONFIG
};
static void gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass);
static void gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * klass);
static void gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay);
static void gst_rtp_mp4v_pay_finalize (GObject * object);
static void gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_mp4v_pay_get_type (void)
{
static GType rtpmp4vpay_type = 0;
if (!rtpmp4vpay_type) {
static const GTypeInfo rtpmp4vpay_info = {
sizeof (GstRtpMP4VPayClass),
(GBaseInitFunc) gst_rtp_mp4v_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_mp4v_pay_class_init,
NULL,
NULL,
sizeof (GstRtpMP4VPay),
0,
(GInstanceInitFunc) gst_rtp_mp4v_pay_init,
};
rtpmp4vpay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4VPay",
&rtpmp4vpay_info, 0);
}
return rtpmp4vpay_type;
}
static void
gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4v_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4v_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mp4vpay_details);
}
static void
gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gobject_class->set_property = gst_rtp_mp4v_pay_set_property;
gobject_class->get_property = gst_rtp_mp4v_pay_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SEND_CONFIG,
g_param_spec_boolean ("send-config", "Send Config",
"Send the config parameters in RTP packets as well",
DEFAULT_SEND_CONFIG, G_PARAM_READWRITE));
gobject_class->finalize = gst_rtp_mp4v_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_mp4v_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mp4v_pay_handle_buffer;
}
static void
gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay)
{
rtpmp4vpay->adapter = gst_adapter_new ();
rtpmp4vpay->rate = 90000;
rtpmp4vpay->profile = 1;
rtpmp4vpay->send_config = DEFAULT_SEND_CONFIG;
}
static void
gst_rtp_mp4v_pay_finalize (GObject * object)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
g_object_unref (rtpmp4vpay->adapter);
rtpmp4vpay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
{
gchar *profile, *config;
GValue v = { 0 };
profile = g_strdup_printf ("%d", rtpmp4vpay->profile);
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4vpay->config);
config = gst_value_serialize (&v);
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
"profile-level-id", G_TYPE_STRING, profile,
"config", G_TYPE_STRING, config, NULL);
g_value_unset (&v);
g_free (profile);
g_free (config);
}
static gboolean
gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
gst_basertppayload_set_options (payload, "video", TRUE, "MP4V-ES",
rtpmp4vpay->rate);
return TRUE;
}
static GstFlowReturn
gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the MP4V data
* over multiple packets. */
avail = gst_adapter_available (rtpmp4vpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* this will be the total lenght 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, GST_BASE_RTP_PAYLOAD_MTU (rtpmp4vpay));
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtpmp4vpay->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtpmp4vpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4vpay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
}
return ret;
}
#define VOS_STARTCODE 0x000001B0
#define VOS_ENDCODE 0x000001B1
#define USER_DATA_STARTCODE 0x000001B2
#define GOP_STARTCODE 0x000001B3
#define VISUAL_OBJECT_STARTCODE 0x000001B5
#define VOP_STARTCODE 0x000001B6
static gboolean
gst_rtp_mp4v_pay_depay_data (GstRtpMP4VPay * enc, guint8 * data, guint size,
gint * strip)
{
guint32 code;
gboolean result;
*strip = 0;
if (size < 5)
return FALSE;
code = GST_READ_UINT32_BE (data);
switch (code) {
case VOS_STARTCODE:
{
gint i;
guint8 profile;
gboolean newprofile = FALSE;
gboolean equal;
/* profile_and_level_indication */
profile = data[4];
if (profile != enc->profile) {
newprofile = TRUE;
enc->profile = profile;
}
/* up to the next GOP_STARTCODE or VOP_STARTCODE is
* the config information */
code = 0xffffffff;
for (i = 5; i < size - 4; i++) {
code = (code << 8) | data[i];
if (code == GOP_STARTCODE || code == VOP_STARTCODE)
break;
}
i -= 3;
/* see if config changed */
equal = FALSE;
if (enc->config) {
if (GST_BUFFER_SIZE (enc->config) == i) {
equal = memcmp (GST_BUFFER_DATA (enc->config), data, i) == 0;
}
}
/* if config string changed or new profile, make new caps */
if (!equal || newprofile) {
if (enc->config)
gst_buffer_unref (enc->config);
enc->config = gst_buffer_new_and_alloc (i);
memcpy (GST_BUFFER_DATA (enc->config), data, i);
gst_rtp_mp4v_pay_new_caps (enc);
}
*strip = i;
/* we need to flush out the current packet. */
result = TRUE;
break;
}
case VOP_STARTCODE:
/* VOP startcode, we don't have to flush the packet */
result = FALSE;
break;
default:
/* all other startcodes need a flush */
result = TRUE;
break;
}
return result;
}
/* we expect buffers starting on startcodes.
*/
static GstFlowReturn
gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4VPay *rtpmp4vpay;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
guint8 *data;
gboolean flush;
gint strip;
GstClockTime duration;
ret = GST_FLOW_OK;
rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmp4vpay->adapter);
/* empty buffer, take timestamp */
if (avail == 0) {
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmp4vpay->duration = 0;
}
/* depay incomming data and see if we need to start a new RTP
* packet */
flush = gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, data, size, &strip);
if (strip) {
/* strip off config if requested */
if (!rtpmp4vpay->send_config) {
GstBuffer *subbuf;
/* strip off header */
subbuf = gst_buffer_create_sub (buffer, strip, size - strip);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buffer);
gst_buffer_unref (buffer);
buffer = subbuf;
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
}
}
/* if we need to flush, do so now */
if (flush) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmp4vpay->duration = 0;
avail = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpmp4vpay->duration + duration)) {
ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmp4vpay->duration = 0;
}
/* push new data */
gst_adapter_push (rtpmp4vpay->adapter, buffer);
rtpmp4vpay->duration += duration;
return ret;
}
static void
gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
switch (prop_id) {
case ARG_SEND_CONFIG:
rtpmp4vpay->send_config = g_value_get_boolean (value);
break;
default:
break;
}
}
static void
gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpMP4VPay *rtpmp4vpay;
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
switch (prop_id) {
case ARG_SEND_CONFIG:
g_value_set_boolean (value, rtpmp4vpay->send_config);
break;
default:
break;
}
}
gboolean
gst_rtp_mp4v_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4vpay",
GST_RANK_NONE, GST_TYPE_RTP_MP4V_PAY);
}