mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-28 12:41:05 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/ext/srtp/gstsrtpenc.c
Jan Alexander Steffens (heftig) e9aa117200 srtp: Support libsrtp2 
For libsrtp 1, add defines that translate the new namespaced identifiers
to the old unnamespaced ones. Also move the code for setting and getting
a stream's ROC into two compat functions that match libsrtp2's API.

It seems that libsrtp2 properly supports changing the ROC without having
to touch the sequence numbers afterwards, given that srtp_set_stream_roc
sets a pending_roc field, so the entire roc_changed dance should not be
needed anymore. The compat functions for libsrtp 1 just contain our
preexisting hacks, however, so it's still needed there.

libsrtp2 has no means of discovering the streams in the session, so to
create the stats structure we need to iterate over our own set of SSRCs.
For this we also need to re-add the previously removed ssrcs_set to the
encoder.

https://bugzilla.gnome.org/show_bug.cgi?id=776901
2018-01-29 09:58:11 +02:00

1389 lines
41 KiB
C
Raw Blame History

/*
* GStreamer - GStreamer SRTP encoder
*
* Copyright 2009-2011 Collabora Ltd.
* @author: Gabriel Millaire <gabriel.millaire@collabora.com>
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* 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.
*/
/**
* SECTION:element-srtpenc
* @title: srtpenc
* @see_also: srtpdec
*
* gstrtpenc acts as an encoder that adds security to RTP and RTCP
* packets in the form of encryption and authentication. It outs SRTP
* and SRTCP.
*
* An application can request multiple RTP and RTCP pads to protect,
* but every sink pad requested must receive packets from the same
* source (identical SSRC). If a packet received contains a different
* SSRC, a warning is emited and the valid SSRC is forced on the packet.
*
* This element uses libsrtp library. When receiving the first packet,
* the library is initialized with a new stream (based on the SSRC). It
* uses the default RTP and RTCP encryption and authentication mechanisms,
* unless the user has set the relevant properties first. It also uses
* a master key that MUST be set by property (key) at the beginning. The
* master key must be of a maximum length of 46 characters (14 characters
* for the salt plus the key). The encryption and authentication mecanisms
* available are :
*
* Encryption (properties rtp-cipher and rtcp-cipher)
* - AES_ICM 256 bits (maximum security)
* - AES_ICM 128 bits (default)
* - NULL
*
* Authentication (properties rtp-auth and rtcp-auth)
* - HMAC_SHA1 80 bits (default, maximum protection)
* - HMAC_SHA1 32 bits
* - NULL
*
* Note that for SRTP protection, authentication is mandatory (non-null)
* if encryption is used (non-null).
*
* When requested to create a sink pad, a linked source pad is created.
* Each packet received is first analysed (checked for valid SSRC) then
* its buffer is protected with libsrtp, then pushed on the source pad.
* If protection failed or the stream could not be created, the buffer
* is dropped and a warning is emitted. The packets pushed on the source
* pad are of type 'application/x-srtp' or 'application/x-srtcp'.
*
* When the maximum usage of the master key is reached, a soft-limit
* signal is sent to the user. The user must then set a new master key
* by property. If the hard limit is reached, a flag is set and every
* subsequent packet is dropped, until a new key is set and the stream
* has been updated.
*
* If a stream is to be shared between multiple clients it is also
* possible to request the internal SRTP rollover counter for a given
* SSRC. The rollover counter should be then transmitted and used by the
* clients to authenticate and decrypt the packets. Failing to do that
* the clients will start with a rollover counter of 0 which will
* probably be incorrect if the stream has been transmitted for a
* while to other clients.
*/
#include "gstsrtpenc.h"
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <string.h>
#include <stdio.h>
GST_DEBUG_CATEGORY_STATIC (gst_srtp_enc_debug);
#define GST_CAT_DEFAULT gst_srtp_enc_debug
/* 128 bit key size: 14 (salt) + 16 */
#define MASTER_128_KEY_SIZE 30
/* 256 bit key size: 14 (salt) + 16 + 16 */
#define MASTER_256_KEY_SIZE 46
/* Properties default values */
#define DEFAULT_MASTER_KEY NULL
#define DEFAULT_RTP_CIPHER GST_SRTP_CIPHER_AES_128_ICM
#define DEFAULT_RTP_AUTH GST_SRTP_AUTH_HMAC_SHA1_80
#define DEFAULT_RTCP_CIPHER DEFAULT_RTP_CIPHER
#define DEFAULT_RTCP_AUTH DEFAULT_RTP_AUTH
#define DEFAULT_RANDOM_KEY FALSE
#define DEFAULT_REPLAY_WINDOW_SIZE 128
#define DEFAULT_ALLOW_REPEAT_TX FALSE
#define HAS_CRYPTO(filter) (filter->rtp_cipher != GST_SRTP_CIPHER_NULL || \
filter->rtcp_cipher != GST_SRTP_CIPHER_NULL || \
filter->rtp_auth != GST_SRTP_AUTH_NULL || \
filter->rtcp_auth != GST_SRTP_AUTH_NULL)
/* Filter signals and args */
enum
{
SIGNAL_SOFT_LIMIT,
SIGNAL_GET_ROLLOVER_COUNTER,
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MKEY,
PROP_RTP_CIPHER,
PROP_RTP_AUTH,
PROP_RTCP_CIPHER,
PROP_RTCP_AUTH,
PROP_RANDOM_KEY,
PROP_REPLAY_WINDOW_SIZE,
PROP_ALLOW_REPEAT_TX,
PROP_STATS
};
typedef struct ProcessBufferItData
{
GstSrtpEnc *filter;
GstPad *pad;
GstBufferList *out_list;
gboolean is_rtcp;
} ProcessBufferItData;
/* the capabilities of the inputs and outputs.
*
* describe the real formats here.
*/
static GstStaticPadTemplate rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtp_src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-srtp")
);
static GstStaticPadTemplate rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-srtcp")
);
G_DEFINE_TYPE (GstSrtpEnc, gst_srtp_enc, GST_TYPE_ELEMENT);
static guint gst_srtp_enc_signals[LAST_SIGNAL] = { 0 };
static void gst_srtp_enc_dispose (GObject * object);
static void gst_srtp_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_srtp_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_srtp_enc_sink_query_rtp (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_srtp_enc_sink_query_rtcp (GstPad * pad, GstObject * parent,
GstQuery * query);
static GstIterator *gst_srtp_enc_iterate_internal_links_rtp (GstPad * pad,
GstObject * parent);
static GstIterator *gst_srtp_enc_iterate_internal_links_rtcp (GstPad * pad,
GstObject * parent);
static GstFlowReturn gst_srtp_enc_chain_rtp (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static GstFlowReturn gst_srtp_enc_chain_rtcp (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static GstFlowReturn gst_srtp_enc_chain_list_rtp (GstPad * pad,
GstObject * parent, GstBufferList * buf);
static GstFlowReturn gst_srtp_enc_chain_list_rtcp (GstPad * pad,
GstObject * parent, GstBufferList * buf);
static gboolean gst_srtp_enc_sink_event_rtp (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_srtp_enc_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event);
static GstStateChangeReturn gst_srtp_enc_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_srtp_enc_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static void gst_srtp_enc_release_pad (GstElement * element, GstPad * pad);
/* initialize the srtpenc's class
*/
static void
gst_srtp_enc_class_init (GstSrtpEncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gst_element_class_add_static_pad_template (gstelement_class,
&rtp_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtp_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtcp_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtcp_sink_template);
gst_element_class_set_static_metadata (gstelement_class, "SRTP encoder",
"Filter/Network/SRTP",
"A SRTP and SRTCP encoder",
"Gabriel Millaire <millaire.gabriel@collabora.com>");
/* Install callbacks */
gobject_class->set_property = gst_srtp_enc_set_property;
gobject_class->get_property = gst_srtp_enc_get_property;
gobject_class->dispose = gst_srtp_enc_dispose;
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_srtp_enc_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_srtp_enc_release_pad);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_srtp_enc_change_state);
/* Install properties */
g_object_class_install_property (gobject_class, PROP_MKEY,
g_param_spec_boxed ("key", "Key", "Master key (minimum of "
G_STRINGIFY (MASTER_128_KEY_SIZE) " and maximum of "
G_STRINGIFY (MASTER_256_KEY_SIZE) " bytes)",
GST_TYPE_BUFFER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_PLAYING));
g_object_class_install_property (gobject_class, PROP_RTP_CIPHER,
g_param_spec_enum ("rtp-cipher", "RTP Cipher", "RTP Cipher",
GST_TYPE_SRTP_CIPHER_TYPE, DEFAULT_RTP_CIPHER,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTP_AUTH,
g_param_spec_enum ("rtp-auth", "RTP Authentication",
"RTP Authentication", GST_TYPE_SRTP_AUTH_TYPE, DEFAULT_RTP_AUTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_CIPHER,
g_param_spec_enum ("rtcp-cipher", "RTCP Cipher",
"RTCP Cipher", GST_TYPE_SRTP_CIPHER_TYPE, DEFAULT_RTCP_CIPHER,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_AUTH,
g_param_spec_enum ("rtcp-auth", "RTCP Authentication",
"RTCP Authentication", GST_TYPE_SRTP_AUTH_TYPE, DEFAULT_RTCP_AUTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RANDOM_KEY,
g_param_spec_boolean ("random-key", "Generate random key",
"Generate a random key if TRUE",
DEFAULT_RANDOM_KEY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_REPLAY_WINDOW_SIZE,
g_param_spec_uint ("replay-window-size", "Replay window size",
"Size of the replay protection window",
64, 0x8000, DEFAULT_REPLAY_WINDOW_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_ALLOW_REPEAT_TX,
g_param_spec_boolean ("allow-repeat-tx",
"Allow repeat packets transmission",
"Whether retransmissions of packets with the same sequence number are allowed"
"(Note that such repeated transmissions must have the same RTP payload, "
"or a severe security weakness is introduced!)",
DEFAULT_ALLOW_REPEAT_TX, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_STATS,
g_param_spec_boxed ("stats", "Statistics", "Various statistics",
GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstSrtpEnc::soft-limit:
* @gstsrtpenc: the element on which the signal is emitted
*
* Signal emited when the stream with @ssrc has reached the soft
* limit of utilisation of it's master encryption key. User should
* provide a new key by setting the #GstSrtpEnc:key property.
*/
gst_srtp_enc_signals[SIGNAL_SOFT_LIMIT] =
g_signal_new ("soft-limit", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
}
/* initialize the new element
*/
static void
gst_srtp_enc_init (GstSrtpEnc * filter)
{
filter->key_changed = TRUE;
filter->first_session = TRUE;
filter->key = DEFAULT_MASTER_KEY;
filter->rtp_cipher = DEFAULT_RTP_CIPHER;
filter->rtp_auth = DEFAULT_RTP_AUTH;
filter->rtcp_cipher = DEFAULT_RTCP_CIPHER;
filter->rtcp_auth = DEFAULT_RTCP_AUTH;
filter->replay_window_size = DEFAULT_REPLAY_WINDOW_SIZE;
filter->allow_repeat_tx = DEFAULT_ALLOW_REPEAT_TX;
filter->ssrcs_set = g_hash_table_new (g_direct_hash, g_direct_equal);
}
static guint
max_cipher_key_size (GstSrtpEnc * filter)
{
guint rtp_size, rtcp_size;
rtp_size = cipher_key_size (filter->rtp_cipher);
rtcp_size = cipher_key_size (filter->rtcp_cipher);
return (rtp_size > rtcp_size) ? rtp_size : rtcp_size;
}
/* Create stream
*
* Should be called with the filter locked
*/
static srtp_err_status_t
gst_srtp_enc_create_session (GstSrtpEnc * filter)
{
srtp_err_status_t ret;
srtp_policy_t policy;
GstMapInfo map;
guchar tmp[1];
memset (&policy, 0, sizeof (srtp_policy_t));
if (HAS_CRYPTO (filter)) {
guint expected;
gsize keysize;
if (filter->key == NULL) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
("Cipher is not NULL, key must be set"),
("Cipher is not NULL, key must be set"));
GST_OBJECT_LOCK (filter);
return srtp_err_status_fail;
}
expected = max_cipher_key_size (filter);
keysize = gst_buffer_get_size (filter->key);
if (expected != keysize) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
("Master key size is wrong"),
("Expected master key of %d bytes, but received %" G_GSIZE_FORMAT
" bytes", expected, keysize));
GST_OBJECT_LOCK (filter);
return srtp_err_status_fail;
}
}
GST_DEBUG_OBJECT (filter, "Setting RTP/RTCP policy to %d / %d",
filter->rtp_cipher, filter->rtcp_cipher);
set_crypto_policy_cipher_auth (filter->rtp_cipher, filter->rtp_auth,
&policy.rtp);
set_crypto_policy_cipher_auth (filter->rtcp_cipher, filter->rtcp_auth,
&policy.rtcp);
if (HAS_CRYPTO (filter)) {
gst_buffer_map (filter->key, &map, GST_MAP_READ);
policy.key = (guchar *) map.data;
} else {
policy.key = tmp;
}
policy.ssrc.value = 0;
policy.ssrc.type = ssrc_any_outbound;
policy.next = NULL;
policy.window_size = filter->replay_window_size;
policy.allow_repeat_tx = filter->allow_repeat_tx;
/* If it is the first stream, create the session
* If not, add the stream to the session
*/
ret = srtp_create (&filter->session, &policy);
filter->first_session = FALSE;
if (HAS_CRYPTO (filter))
gst_buffer_unmap (filter->key, &map);
return ret;
}
/* Release ressources and set default values
*/
static void
gst_srtp_enc_reset_no_lock (GstSrtpEnc * filter)
{
if (!filter->first_session) {
srtp_dealloc (filter->session);
filter->session = NULL;
g_hash_table_remove_all (filter->ssrcs_set);
}
filter->first_session = TRUE;
filter->key_changed = FALSE;
}
static void
gst_srtp_enc_reset (GstSrtpEnc * filter)
{
GST_OBJECT_LOCK (filter);
gst_srtp_enc_reset_no_lock (filter);
GST_OBJECT_UNLOCK (filter);
}
/* Create sinkpad to receive RTP packets from encers
* and a srcpad for the RTP packets
*/
static GstPad *
create_rtp_sink (GstSrtpEnc * filter, const gchar * name)
{
GstPad *sinkpad, *srcpad;
gchar *sinkpadname, *srcpadname;
guint nb = 0;
GST_DEBUG_OBJECT (filter, "creating RTP sink pad");
sinkpad = gst_pad_new_from_static_template (&rtp_sink_template, name);
sinkpadname = gst_pad_get_name (sinkpad);
sscanf (sinkpadname, "rtp_sink_%u", &nb);
srcpadname = g_strdup_printf ("rtp_src_%u", nb);
GST_DEBUG_OBJECT (filter, "creating RTP source pad");
srcpad = gst_pad_new_from_static_template (&rtp_src_template, srcpadname);
g_free (srcpadname);
g_free (sinkpadname);
gst_pad_set_element_private (sinkpad, srcpad);
gst_pad_set_element_private (srcpad, sinkpad);
gst_pad_set_query_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_query_rtp));
gst_pad_set_iterate_internal_links_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtp));
gst_pad_set_chain_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_chain_rtp));
gst_pad_set_chain_list_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_chain_list_rtp));
gst_pad_set_event_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_event_rtp));
gst_pad_set_active (sinkpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), sinkpad);
gst_pad_set_iterate_internal_links_function (srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtp));
gst_pad_set_active (srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), srcpad);
return sinkpad;
}
/* Create sinkpad to receive RTCP packets from encers
* and a srcpad for the RTCP packets
*/
static GstPad *
create_rtcp_sink (GstSrtpEnc * filter, const gchar * name)
{
GstPad *srcpad, *sinkpad;
gchar *sinkpadname, *srcpadname;
guint nb = 0;
GST_DEBUG_OBJECT (filter, "creating RTCP sink pad");
sinkpad = gst_pad_new_from_static_template (&rtcp_sink_template, name);
sinkpadname = gst_pad_get_name (sinkpad);
sscanf (sinkpadname, "rtcp_sink_%u", &nb);
srcpadname = g_strdup_printf ("rtcp_src_%u", nb);
GST_DEBUG_OBJECT (filter, "creating RTCP source pad");
srcpad = gst_pad_new_from_static_template (&rtcp_src_template, srcpadname);
g_free (srcpadname);
g_free (sinkpadname);
gst_pad_set_element_private (sinkpad, srcpad);
gst_pad_set_element_private (srcpad, sinkpad);
gst_pad_set_query_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_query_rtcp));
gst_pad_set_iterate_internal_links_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtcp));
gst_pad_set_chain_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_chain_rtcp));
gst_pad_set_chain_list_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_chain_list_rtcp));
gst_pad_set_event_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_event_rtcp));
gst_pad_set_active (sinkpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), sinkpad);
gst_pad_set_iterate_internal_links_function (srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtcp));
gst_pad_set_active (srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), srcpad);
return sinkpad;
}
/* Handling new pad request
*/
static GstPad *
gst_srtp_enc_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstElementClass *klass;
GstSrtpEnc *filter;
filter = GST_SRTP_ENC (element);
klass = GST_ELEMENT_GET_CLASS (element);
GST_INFO_OBJECT (element, "New pad requested");
if (templ == gst_element_class_get_pad_template (klass, "rtp_sink_%u"))
return create_rtp_sink (filter, name);
if (templ == gst_element_class_get_pad_template (klass, "rtcp_sink_%u"))
return create_rtcp_sink (filter, name);
GST_ERROR_OBJECT (element, "Could not find specified template");
return NULL;
}
/* Dispose
*/
static void
gst_srtp_enc_dispose (GObject * object)
{
GstSrtpEnc *filter = GST_SRTP_ENC (object);
GstIterator *it;
GValue val = { 0 };
GST_DEBUG_OBJECT (object, "Dispose...");
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (object));
while (gst_iterator_next (it, &val) == GST_ITERATOR_OK) {
gst_srtp_enc_release_pad (GST_ELEMENT_CAST (object),
g_value_get_object (&val));
g_value_unset (&val);
gst_iterator_resync (it);
}
gst_iterator_free (it);
if (filter->key)
gst_buffer_unref (filter->key);
filter->key = NULL;
if (filter->ssrcs_set)
g_hash_table_unref (filter->ssrcs_set);
filter->ssrcs_set = NULL;
G_OBJECT_CLASS (gst_srtp_enc_parent_class)->dispose (object);
}
static GstStructure *
gst_srtp_enc_create_stats (GstSrtpEnc * filter)
{
GstStructure *s;
GValue va = G_VALUE_INIT;
GValue v = G_VALUE_INIT;
s = gst_structure_new_empty ("application/x-srtp-encoder-stats");
g_value_init (&va, GST_TYPE_ARRAY);
g_value_init (&v, GST_TYPE_STRUCTURE);
if (filter->session) {
GHashTableIter iter;
gpointer key;
g_hash_table_iter_init (&iter, filter->ssrcs_set);
while (g_hash_table_iter_next (&iter, &key, NULL)) {
GstStructure *ss;
guint32 ssrc = GPOINTER_TO_UINT (key);
srtp_err_status_t status;
guint32 roc;
status = srtp_get_stream_roc (filter->session, ssrc, &roc);
if (status != srtp_err_status_ok) {
continue;
}
ss = gst_structure_new ("application/x-srtp-stream",
"ssrc", G_TYPE_UINT, ssrc, "roc", G_TYPE_UINT, roc, NULL);
g_value_take_boxed (&v, ss);
gst_value_array_append_value (&va, &v);
}
}
gst_structure_take_value (s, "streams", &va);
g_value_unset (&v);
return s;
}
static void
gst_srtp_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSrtpEnc *filter = GST_SRTP_ENC (object);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
case PROP_MKEY:
if (filter->key)
gst_buffer_unref (filter->key);
filter->key = g_value_dup_boxed (value);
filter->key_changed = TRUE;
GST_INFO_OBJECT (object, "Set property: key=[%p]", filter->key);
break;
case PROP_RTP_CIPHER:
filter->rtp_cipher = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtp cipher=%d",
filter->rtp_cipher);
break;
case PROP_RTP_AUTH:
filter->rtp_auth = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtp auth=%d", filter->rtp_auth);
break;
case PROP_RTCP_CIPHER:
filter->rtcp_cipher = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtcp cipher=%d",
filter->rtcp_cipher);
break;
case PROP_RTCP_AUTH:
filter->rtcp_auth = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtcp auth=%d", filter->rtcp_auth);
break;
case PROP_RANDOM_KEY:
filter->random_key = g_value_get_boolean (value);
break;
case PROP_REPLAY_WINDOW_SIZE:
filter->replay_window_size = g_value_get_uint (value);
break;
case PROP_ALLOW_REPEAT_TX:
filter->allow_repeat_tx = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (filter);
}
static void
gst_srtp_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSrtpEnc *filter = GST_SRTP_ENC (object);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
case PROP_MKEY:
if (filter->key)
g_value_set_boxed (value, filter->key);
break;
case PROP_RTP_CIPHER:
g_value_set_enum (value, filter->rtp_cipher);
break;
case PROP_RTCP_CIPHER:
g_value_set_enum (value, filter->rtcp_cipher);
break;
case PROP_RTP_AUTH:
g_value_set_enum (value, filter->rtp_auth);
break;
case PROP_RTCP_AUTH:
g_value_set_enum (value, filter->rtcp_auth);
break;
case PROP_RANDOM_KEY:
g_value_set_boolean (value, filter->random_key);
break;
case PROP_REPLAY_WINDOW_SIZE:
g_value_set_uint (value, filter->replay_window_size);
break;
case PROP_ALLOW_REPEAT_TX:
g_value_set_boolean (value, filter->allow_repeat_tx);
break;
case PROP_STATS:
g_value_take_boxed (value, gst_srtp_enc_create_stats (filter));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (filter);
}
/* Returns the source pad linked with the sink pad
*/
static GstPad *
get_rtp_other_pad (GstPad * pad)
{
return GST_PAD (gst_pad_get_element_private (pad));
}
/* Release a sink pad and it's linked source pad
*/
static void
gst_srtp_enc_release_pad (GstElement * element, GstPad * sinkpad)
{
GstPad *srcpad;
GST_INFO_OBJECT (element, "Releasing pad %s:%s",
GST_DEBUG_PAD_NAME (sinkpad));
srcpad = GST_PAD (gst_pad_get_element_private (sinkpad));
gst_pad_set_element_private (sinkpad, NULL);
gst_pad_set_element_private (srcpad, NULL);
/* deactivate from source to sink */
gst_pad_set_active (srcpad, FALSE);
gst_pad_set_active (sinkpad, FALSE);
/* remove pads */
gst_element_remove_pad (element, srcpad);
gst_element_remove_pad (element, sinkpad);
}
/* Common setcaps function
* Handles the link with other elements
*/
static gboolean
gst_srtp_enc_sink_setcaps (GstPad * pad, GstSrtpEnc * filter,
GstCaps * caps, gboolean is_rtcp)
{
GstPad *otherpad = NULL;
GstStructure *ps = NULL;
gboolean ret = FALSE;
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
caps = gst_caps_copy (caps);
ps = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (pad, "Sink caps: %" GST_PTR_FORMAT, caps);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-srtcp");
else
gst_structure_set_name (ps, "application/x-srtp");
GST_OBJECT_LOCK (filter);
if (gst_structure_has_field_typed (ps, "ssrc", G_TYPE_UINT)) {
guint ssrc;
gst_structure_get_uint (ps, "ssrc", &ssrc);
g_hash_table_add (filter->ssrcs_set, GUINT_TO_POINTER (ssrc));
}
if (HAS_CRYPTO (filter))
gst_structure_set (ps, "srtp-key", GST_TYPE_BUFFER, filter->key, NULL);
/* Add srtp-specific params to source caps */
gst_structure_set (ps,
"srtp-cipher", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_CIPHER_TYPE, filter->rtp_cipher),
"srtp-auth", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_AUTH_TYPE, filter->rtp_auth),
"srtcp-cipher", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_CIPHER_TYPE, filter->rtcp_cipher),
"srtcp-auth", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_AUTH_TYPE, filter->rtcp_auth), NULL);
GST_OBJECT_UNLOCK (filter);
GST_DEBUG_OBJECT (pad, "Source caps: %" GST_PTR_FORMAT, caps);
/* Set caps on source pad */
otherpad = get_rtp_other_pad (pad);
ret = gst_pad_set_caps (otherpad, caps);
gst_caps_unref (caps);
return ret;
}
static gboolean
gst_srtp_enc_sink_query (GstPad * pad, GstObject * parent, GstQuery * query,
gboolean is_rtcp)
{
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter = NULL;
GstCaps *other_filter = NULL;
GstPad *otherpad;
GstCaps *other_caps;
GstCaps *ret;
GstCaps *template_caps;
int i;
otherpad = get_rtp_other_pad (pad);
gst_query_parse_caps (query, &filter);
if (filter) {
other_filter = gst_caps_copy (filter);
for (i = 0; i < gst_caps_get_size (other_filter); i++) {
GstStructure *ps = gst_caps_get_structure (other_filter, i);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-srtcp");
else
gst_structure_set_name (ps, "application/x-srtp");
}
}
other_caps = gst_pad_peer_query_caps (otherpad, other_filter);
if (other_filter)
gst_caps_unref (other_filter);
if (!other_caps)
goto return_template;
template_caps = gst_pad_get_pad_template_caps (otherpad);
ret = gst_caps_intersect_full (other_caps, template_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (other_caps);
gst_caps_unref (template_caps);
ret = gst_caps_make_writable (ret);
for (i = 0; i < gst_caps_get_size (ret); i++) {
GstStructure *ps = gst_caps_get_structure (ret, i);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-rtcp");
else
gst_structure_set_name (ps, "application/x-rtp");
gst_structure_remove_fields (ps, "srtp-key", "srtp-cipher", "srtp-auth",
"srtcp-cipher", "srtcp-auth", NULL);
}
gst_query_set_caps_result (query, ret);
gst_caps_unref (ret);
return TRUE;
return_template:
ret = gst_pad_get_pad_template_caps (pad);
gst_query_set_caps_result (query, ret);
gst_caps_unref (ret);
return TRUE;
}
default:
return gst_pad_query_default (pad, parent, query);
}
}
static gboolean
gst_srtp_enc_sink_query_rtp (GstPad * pad, GstObject * parent, GstQuery * query)
{
return gst_srtp_enc_sink_query (pad, parent, query, FALSE);
}
static gboolean
gst_srtp_enc_sink_query_rtcp (GstPad * pad, GstObject * parent,
GstQuery * query)
{
return gst_srtp_enc_sink_query (pad, parent, query, TRUE);
}
static GstIterator *
gst_srtp_enc_iterate_internal_links (GstPad * pad, GstObject * parent,
gboolean is_rtcp)
{
GstSrtpEnc *filter = GST_SRTP_ENC (parent);
GstPad *otherpad = NULL;
GstIterator *it = NULL;
otherpad = get_rtp_other_pad (pad);
if (otherpad) {
GValue val = { 0 };
g_value_init (&val, GST_TYPE_PAD);
g_value_set_object (&val, otherpad);
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
g_value_unset (&val);
} else {
GST_ELEMENT_ERROR (GST_ELEMENT_CAST (filter), CORE, PAD, (NULL),
("Unable to get linked pad"));
}
return it;
}
static GstIterator *
gst_srtp_enc_iterate_internal_links_rtp (GstPad * pad, GstObject * parent)
{
return gst_srtp_enc_iterate_internal_links (pad, parent, FALSE);
}
static GstIterator *
gst_srtp_enc_iterate_internal_links_rtcp (GstPad * pad, GstObject * parent)
{
return gst_srtp_enc_iterate_internal_links (pad, parent, TRUE);
}
static void
gst_srtp_enc_replace_random_key (GstSrtpEnc * filter)
{
guint i;
guint key_size;
GstMapInfo map;
GST_DEBUG_OBJECT (filter, "Generating random key");
if (filter->key)
gst_buffer_unref (filter->key);
key_size = max_cipher_key_size (filter);
filter->key = gst_buffer_new_allocate (NULL, key_size, NULL);
gst_buffer_map (filter->key, &map, GST_MAP_WRITE);
for (i = 0; i < map.size; i += 4)
GST_WRITE_UINT32_BE (map.data + i, g_random_int ());
gst_buffer_unmap (filter->key, &map);
filter->key_changed = TRUE;
}
static GstFlowReturn
gst_srtp_enc_check_set_caps (GstSrtpEnc * filter, GstPad * pad,
gboolean is_rtcp)
{
gboolean do_setcaps = FALSE;
GST_OBJECT_LOCK (filter);
if (filter->key_changed) {
gst_srtp_enc_reset_no_lock (filter);
do_setcaps = TRUE;
}
if (filter->first_session) {
srtp_err_status_t status = gst_srtp_enc_create_session (filter);
if (status != srtp_err_status_ok) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, INIT,
("Could not initialize SRTP encoder"),
("Failed to add stream to SRTP encoder (err: %d)", status));
return GST_FLOW_ERROR;
}
}
GST_OBJECT_UNLOCK (filter);
/* Update source caps if asked */
if (do_setcaps) {
GstCaps *caps;
caps = gst_pad_get_current_caps (pad);
if (!gst_srtp_enc_sink_setcaps (pad, filter, caps, is_rtcp)) {
gst_caps_unref (caps);
return GST_FLOW_NOT_NEGOTIATED;
}
gst_caps_unref (caps);
}
return GST_FLOW_OK;
}
static GstBuffer *
gst_srtp_enc_process_buffer (GstSrtpEnc * filter, GstPad * pad,
GstBuffer * buf, gboolean is_rtcp)
{
gint size_max, size;
GstBuffer *bufout = NULL;
GstMapInfo mapout;
srtp_err_status_t err;
/* Create a bigger buffer to add protection */
size = gst_buffer_get_size (buf);
size_max = size + SRTP_MAX_TRAILER_LEN + 10;
bufout = gst_buffer_new_allocate (NULL, size_max, NULL);
gst_buffer_map (bufout, &mapout, GST_MAP_READWRITE);
gst_buffer_extract (buf, 0, mapout.data, size);
GST_OBJECT_LOCK (filter);
gst_srtp_init_event_reporter ();
if (is_rtcp)
err = srtp_protect_rtcp (filter->session, mapout.data, &size);
else
err = srtp_protect (filter->session, mapout.data, &size);
GST_OBJECT_UNLOCK (filter);
gst_buffer_unmap (bufout, &mapout);
if (err == srtp_err_status_ok) {
/* Buffer protected */
gst_buffer_set_size (bufout, size);
gst_buffer_copy_into (bufout, buf, GST_BUFFER_COPY_METADATA, 0, -1);
GST_LOG_OBJECT (pad, "Encoding %s buffer of size %d",
is_rtcp ? "RTCP" : "RTP", size);
} else if (err == srtp_err_status_key_expired) {
GST_ELEMENT_ERROR (GST_ELEMENT_CAST (filter), STREAM, ENCODE,
("Key usage limit has been reached"),
("Unable to protect buffer (hard key usage limit reached)"));
goto fail;
} else {
/* srtp_protect failed */
GST_ELEMENT_ERROR (filter, LIBRARY, FAILED, (NULL),
("Unable to protect buffer (protect failed) code %d", err));
goto fail;
}
return bufout;
fail:
gst_buffer_unref (bufout);
return NULL;
}
static GstFlowReturn
gst_srtp_enc_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
gboolean is_rtcp)
{
GstSrtpEnc *filter = GST_SRTP_ENC (parent);
GstFlowReturn ret = GST_FLOW_OK;
GstPad *otherpad;
GstBuffer *bufout = NULL;
if ((ret = gst_srtp_enc_check_set_caps (filter, pad, is_rtcp)) != GST_FLOW_OK) {
goto out;
}
GST_OBJECT_LOCK (filter);
if (!HAS_CRYPTO (filter)) {
GST_OBJECT_UNLOCK (filter);
otherpad = get_rtp_other_pad (pad);
return gst_pad_push (otherpad, buf);
}
GST_OBJECT_UNLOCK (filter);
if ((bufout = gst_srtp_enc_process_buffer (filter, pad, buf, is_rtcp))) {
/* Push buffer to source pad */
otherpad = get_rtp_other_pad (pad);
ret = gst_pad_push (otherpad, bufout);
bufout = NULL;
if (ret != GST_FLOW_OK)
goto out;
} else {
goto fail;
}
GST_OBJECT_LOCK (filter);
if (gst_srtp_get_soft_limit_reached ()) {
GST_OBJECT_UNLOCK (filter);
g_signal_emit (filter, gst_srtp_enc_signals[SIGNAL_SOFT_LIMIT], 0);
GST_OBJECT_LOCK (filter);
if (filter->random_key && !filter->key_changed)
gst_srtp_enc_replace_random_key (filter);
}
GST_OBJECT_UNLOCK (filter);
out:
gst_buffer_unref (buf);
return ret;
fail:
ret = GST_FLOW_ERROR;
goto out;
}
static gboolean
process_buffer_it (GstBuffer ** buffer, guint index, gpointer user_data)
{
ProcessBufferItData *data = user_data;
GstBuffer *bufout;
if ((bufout =
gst_srtp_enc_process_buffer (data->filter, data->pad, *buffer,
data->is_rtcp))) {
gst_buffer_list_add (data->out_list, bufout);
} else {
GST_WARNING_OBJECT (data->filter, "Error encoding buffer, dropping");
}
return TRUE;
}
static GstFlowReturn
gst_srtp_enc_chain_list (GstPad * pad, GstObject * parent,
GstBufferList * buf_list, gboolean is_rtcp)
{
GstSrtpEnc *filter = GST_SRTP_ENC (parent);
GstFlowReturn ret = GST_FLOW_OK;
GstPad *otherpad;
GstBufferList *out_list = NULL;
ProcessBufferItData process_data;
GST_LOG_OBJECT (pad, "Buffer chain with list of %d",
gst_buffer_list_length (buf_list));
if (!gst_buffer_list_length (buf_list))
goto out;
if ((ret = gst_srtp_enc_check_set_caps (filter, pad, is_rtcp)) != GST_FLOW_OK)
goto out;
GST_OBJECT_LOCK (filter);
if (!HAS_CRYPTO (filter)) {
GST_OBJECT_UNLOCK (filter);
otherpad = get_rtp_other_pad (pad);
return gst_pad_push_list (otherpad, buf_list);
}
GST_OBJECT_UNLOCK (filter);
out_list = gst_buffer_list_new ();
process_data.filter = filter;
process_data.pad = pad;
process_data.is_rtcp = is_rtcp;
process_data.out_list = out_list;
gst_buffer_list_foreach (buf_list, process_buffer_it, &process_data);
if (!gst_buffer_list_length (out_list)) {
gst_buffer_list_unref (out_list);
ret = GST_FLOW_OK;
goto out;
}
/* Push buffer to source pad */
otherpad = get_rtp_other_pad (pad);
GST_LOG_OBJECT (pad, "Pushing buffer chain of %d",
gst_buffer_list_length (buf_list));
ret = gst_pad_push_list (otherpad, out_list);
if (ret != GST_FLOW_OK) {
goto out;
}
GST_OBJECT_LOCK (filter);
if (gst_srtp_get_soft_limit_reached ()) {
GST_OBJECT_UNLOCK (filter);
g_signal_emit (filter, gst_srtp_enc_signals[SIGNAL_SOFT_LIMIT], 0);
GST_OBJECT_LOCK (filter);
if (filter->random_key && !filter->key_changed)
gst_srtp_enc_replace_random_key (filter);
}
GST_OBJECT_UNLOCK (filter);
out:
gst_buffer_list_unref (buf_list);
return ret;
}
static GstFlowReturn
gst_srtp_enc_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_enc_chain (pad, parent, buf, FALSE);
}
static GstFlowReturn
gst_srtp_enc_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_enc_chain (pad, parent, buf, TRUE);
}
static GstFlowReturn
gst_srtp_enc_chain_list_rtp (GstPad * pad, GstObject * parent,
GstBufferList * buf_list)
{
return gst_srtp_enc_chain_list (pad, parent, buf_list, FALSE);
}
static GstFlowReturn
gst_srtp_enc_chain_list_rtcp (GstPad * pad, GstObject * parent,
GstBufferList * buf_list)
{
return gst_srtp_enc_chain_list (pad, parent, buf_list, TRUE);
}
/* Change state
*/
static GstStateChangeReturn
gst_srtp_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstSrtpEnc *filter;
filter = GST_SRTP_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (filter->rtp_cipher != GST_SRTP_CIPHER_NULL ||
filter->rtcp_cipher != GST_SRTP_CIPHER_NULL ||
filter->rtp_auth != GST_SRTP_AUTH_NULL ||
filter->rtcp_auth != GST_SRTP_AUTH_NULL) {
if (!filter->key) {
if (filter->random_key) {
gst_srtp_enc_replace_random_key (filter);
} else {
GST_ERROR_OBJECT (element, "Need a key to get to READY");
return GST_STATE_CHANGE_FAILURE;
}
}
}
if ((filter->rtcp_cipher != SRTP_NULL_CIPHER)
&& (filter->rtcp_auth == SRTP_NULL_AUTH)) {
GST_ERROR_OBJECT (filter,
"RTCP authentication can't be NULL if encryption is not NULL.");
return GST_STATE_CHANGE_FAILURE;
}
GST_OBJECT_LOCK (filter);
if (!filter->first_session)
gst_srtp_enc_reset_no_lock (filter);
GST_OBJECT_UNLOCK (filter);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
res = GST_ELEMENT_CLASS (gst_srtp_enc_parent_class)->change_state (element,
transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_srtp_enc_reset (filter);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
static gboolean
gst_srtp_enc_sink_event (GstPad * pad, GstObject * parent, GstEvent * event,
gboolean is_rtcp)
{
GstSrtpEnc *filter = GST_SRTP_ENC (parent);
gboolean ret;
GstPad *otherpad;
otherpad = get_rtp_other_pad (pad);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
GST_DEBUG_OBJECT (pad, "Encing event Flush stop (%d)",
GST_EVENT_TYPE (event));
gst_srtp_enc_reset (filter);
ret = gst_pad_push_event (otherpad, event);
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_srtp_enc_sink_setcaps (pad, filter, caps, is_rtcp);
gst_event_unref (event);
break;
}
default:
GST_DEBUG_OBJECT (pad, "Encing event default (%d)",
GST_EVENT_TYPE (event));
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static gboolean
gst_srtp_enc_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event)
{
return gst_srtp_enc_sink_event (pad, parent, event, FALSE);
}
static gboolean
gst_srtp_enc_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event)
{
return gst_srtp_enc_sink_event (pad, parent, event, TRUE);
}
/* entry point to initialize the plug-in
* initialize the plug-in itself
* register the element factories and other features
*/
gboolean
gst_srtp_enc_plugin_init (GstPlugin * srtpenc)
{
GST_DEBUG_CATEGORY_INIT (gst_srtp_enc_debug, "srtpenc", 0, "SRTP Enc");
return gst_element_register (srtpenc, "srtpenc", GST_RANK_NONE,
GST_TYPE_SRTP_ENC);
}
Reference in a new issue View git blame Copy permalink