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gstreamer/ext/srtp/gstsrtpdec.c

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/*
* GStreamer - GStreamer SRTP decoder
*
* Copyright 2009-2011 Collabora Ltd.
* @author: Gabriel Millaire <gabriel.millaire@collabora.co.uk>
* @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-srtpdec
* @see_also: srtpenc
*
* gstrtpdec acts as a decoder that removes security from SRTP and SRTCP
* packets (encryption and authentication) and out RTP and RTCP. It
* receives packet of type 'application/x-srtp' or 'application/x-srtcp'
* on its sink pad, and outs packets of type 'application/x-rtp' or
* 'application/x-rtcp' on its source pad.
*
* For each packet received, it checks if the internal SSRC is in the list
* of streams already in use. If this is not the case, it sends a signal to
* the user to get the needed parameters to create a new stream : master
* key, encryption and authentication mecanisms for both RTP and RTCP. If
* the user can't provide those parameters, the buffer is dropped and a
* warning is emitted.
*
* This element uses libsrtp library. The encryption and authentication
* mecanisms available are :
*
* Encryption
* - AES_ICM 256 bits (maximum security)
* - AES_ICM 128 bits (default)
* - NULL
*
* Authentication
* - 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).
*
* Each packet received is first analysed (checked for valid SSRC) then
* its buffer is unprotected 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.
*
* When the maximum usage of the master key is reached, a soft-limit
* signal is sent to the user, and new parameters (master key) are needed
* in return. 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 the SRTP
* rollover counter for a given SSRC must be set in the caps "roc" field
* when the request-key signal is emitted by the decoder. The rollover
* counters should have been transmitted by a signaling protocol by some
* other means. If no rollover counter is provided by the user, 0 is
* used by default.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 udpsrc port=5004 caps='application/x-srtp, payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80, srtcp-cipher=(string)aes-128-icm, srtcp-auth=(string)hmac-sha1-80' ! srtpdec ! rtppcmadepay ! alawdec ! pulsesink
* ]| Receive PCMA SRTP packets through UDP using caps to specify
* master key and protection.
* |[
* gst-launch-1.0 audiotestsrc ! alawenc ! rtppcmapay ! 'application/x-rtp, payload=(int)8, ssrc=(uint)1356955624' ! srtpenc key="012345678901234567890123456789012345678901234567890123456789" ! udpsink port=5004
* ]| Send PCMA SRTP packets through UDP, nothing how the SSRC is forced so
* that the receiver will recognize it.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/gst.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include "gstsrtp.h"
#include "gstsrtp-enumtypes.h"
#include "gstsrtpdec.h"
#include <srtp/srtp_priv.h>
GST_DEBUG_CATEGORY_STATIC (gst_srtp_dec_debug);
#define GST_CAT_DEFAULT gst_srtp_dec_debug
#define DEFAULT_REPLAY_WINDOW_SIZE 128
/* Filter signals and args */
enum
{
SIGNAL_REQUEST_KEY = 1,
SIGNAL_CLEAR_KEYS,
SIGNAL_SOFT_LIMIT,
SIGNAL_HARD_LIMIT,
SIGNAL_REMOVE_KEY,
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_REPLAY_WINDOW_SIZE
};
typedef struct _ValidateBufferItData
{
GstSrtpDecSsrcStream **stream;
GstSrtpDec *filter;
guint32 *ssrc;
gboolean *is_rtcp;
} ValidateBufferItData;
typedef struct _DecodeBufferItData
{
GstSrtpDec *filter;
GstPad *pad;
guint32 ssrc;
gboolean is_rtcp;
} DecodeBufferItData;
/* the capabilities of the inputs and outputs.
*
* describe the real formats here.
*/
static GstStaticPadTemplate rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtp_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-srtp")
);
static GstStaticPadTemplate rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtp_src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-srtcp")
);
static GstStaticPadTemplate rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtcp")
);
static guint gst_srtp_dec_signals[LAST_SIGNAL] = { 0 };
G_DEFINE_TYPE (GstSrtpDec, gst_srtp_dec, GST_TYPE_ELEMENT);
static void gst_srtp_dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_srtp_dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_srtp_dec_clear_streams (GstSrtpDec * filter);
static void gst_srtp_dec_remove_stream (GstSrtpDec * filter, guint ssrc);
static gboolean gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_srtp_dec_sink_query_rtp (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_srtp_dec_sink_query_rtcp (GstPad * pad,
GstObject * parent, GstQuery * query);
static GstIterator *gst_srtp_dec_iterate_internal_links_rtp (GstPad * pad,
GstObject * parent);
static GstIterator *gst_srtp_dec_iterate_internal_links_rtcp (GstPad * pad,
GstObject * parent);
static GstFlowReturn gst_srtp_dec_chain_rtp (GstPad * pad,
GstObject * parent, GstBuffer * buf);
static GstFlowReturn gst_srtp_dec_chain_rtcp (GstPad * pad,
GstObject * parent, GstBuffer * buf);
static GstFlowReturn gst_srtp_dec_chain_list_rtp (GstPad * pad,
GstObject * parent, GstBufferList * buf_list);
static GstFlowReturn gst_srtp_dec_chain_list_rtcp (GstPad * pad,
GstObject * parent, GstBufferList * buf_list);
static GstStateChangeReturn gst_srtp_dec_change_state (GstElement * element,
GstStateChange transition);
static GstSrtpDecSsrcStream *request_key_with_signal (GstSrtpDec * filter,
guint32 ssrc, gint signal);
struct _GstSrtpDecSsrcStream
{
guint32 ssrc;
guint32 roc;
GstBuffer *key;
GstSrtpCipherType rtp_cipher;
GstSrtpAuthType rtp_auth;
GstSrtpCipherType rtcp_cipher;
GstSrtpAuthType rtcp_auth;
};
#define STREAM_HAS_CRYPTO(stream) \
(stream->rtp_cipher != GST_SRTP_CIPHER_NULL || \
stream->rtcp_cipher != GST_SRTP_CIPHER_NULL || \
stream->rtp_auth != GST_SRTP_AUTH_NULL || \
stream->rtcp_auth != GST_SRTP_AUTH_NULL)
/* initialize the srtpdec's class */
static void
gst_srtp_dec_class_init (GstSrtpDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_srtp_dec_set_property;
gobject_class->get_property = gst_srtp_dec_get_property;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtp_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtcp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtcp_sink_template));
gst_element_class_set_static_metadata (gstelement_class, "SRTP decoder",
"Filter/Network/SRTP",
"A SRTP and SRTCP decoder",
"Gabriel Millaire <millaire.gabriel@collabora.com>");
/* Install callbacks */
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_srtp_dec_change_state);
klass->clear_streams = GST_DEBUG_FUNCPTR (gst_srtp_dec_clear_streams);
klass->remove_stream = GST_DEBUG_FUNCPTR (gst_srtp_dec_remove_stream);
/* Install properties */
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));
/* Install signals */
/**
* GstSrtpDec::request-key:
* @gstsrtpdec: the element on which the signal is emitted
* @ssrc: The unique SSRC of the stream
*
* Signal emited to get the parameters relevant to stream
* with @ssrc. User should provide the key and the RTP and
* RTCP encryption ciphers and authentication, and return
* them wrapped in a GstCaps.
*/
gst_srtp_dec_signals[SIGNAL_REQUEST_KEY] =
g_signal_new ("request-key", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstSrtpDec::clear-keys:
* @gstsrtpdec: the element on which the signal is emitted
*
* Clear the internal list of streams
*/
gst_srtp_dec_signals[SIGNAL_CLEAR_KEYS] =
g_signal_new ("clear-keys", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (GstSrtpDecClass, clear_streams), NULL, NULL, NULL,
G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstSrtpDec::soft-limit:
* @gstsrtpdec: the element on which the signal is emitted
* @ssrc: The unique SSRC of the stream
*
* 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 and new RTP and RTCP encryption
* ciphers and authentication, and return them wrapped in a
* GstCaps.
*/
gst_srtp_dec_signals[SIGNAL_SOFT_LIMIT] =
g_signal_new ("soft-limit", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstSrtpDec::hard-limit:
* @gstsrtpdec: the element on which the signal is emitted
* @ssrc: The unique SSRC of the stream
*
* Signal emited when the stream with @ssrc has reached the
* hard limit of utilisation of it's master encryption key.
* User should provide a new key and new RTP and RTCP encryption
* ciphers and authentication, and return them wrapped in a
* GstCaps. If user could not provide those parameters or signal
* is not answered, the buffers of this stream will be dropped.
*/
gst_srtp_dec_signals[SIGNAL_HARD_LIMIT] =
g_signal_new ("hard-limit", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstSrtpDec::remove-key:
* @gstsrtpdec: the element on which the signal is emitted
* @ssrc: The SSRC for which to remove the key.
*
* Removes keys for a specific SSRC
*/
gst_srtp_dec_signals[SIGNAL_REMOVE_KEY] =
g_signal_new ("remove-key", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (GstSrtpDecClass, remove_stream), NULL, NULL, NULL,
G_TYPE_NONE, 1, G_TYPE_UINT);
}
/* initialize the new element
* instantiate pads and add them to element
* set pad calback functions
* initialize instance structure
*/
static void
gst_srtp_dec_init (GstSrtpDec * filter)
{
filter->replay_window_size = DEFAULT_REPLAY_WINDOW_SIZE;
filter->rtp_sinkpad =
gst_pad_new_from_static_template (&rtp_sink_template, "rtp_sink");
gst_pad_set_event_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_event_rtp));
gst_pad_set_query_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_query_rtp));
gst_pad_set_iterate_internal_links_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtp));
gst_pad_set_chain_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtp));
gst_pad_set_chain_list_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_list_rtp));
filter->rtp_srcpad =
gst_pad_new_from_static_template (&rtp_src_template, "rtp_src");
gst_pad_set_iterate_internal_links_function (filter->rtp_srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtp));
gst_pad_set_element_private (filter->rtp_sinkpad, filter->rtp_srcpad);
gst_pad_set_element_private (filter->rtp_srcpad, filter->rtp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtp_srcpad);
filter->rtcp_sinkpad =
gst_pad_new_from_static_template (&rtcp_sink_template, "rtcp_sink");
gst_pad_set_event_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_event_rtcp));
gst_pad_set_query_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_query_rtcp));
gst_pad_set_iterate_internal_links_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtcp));
gst_pad_set_chain_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtcp));
gst_pad_set_chain_list_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_list_rtcp));
filter->rtcp_srcpad =
gst_pad_new_from_static_template (&rtcp_src_template, "rtcp_src");
gst_pad_set_iterate_internal_links_function (filter->rtcp_srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtcp));
gst_pad_set_element_private (filter->rtcp_sinkpad, filter->rtcp_srcpad);
gst_pad_set_element_private (filter->rtcp_srcpad, filter->rtcp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtcp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtcp_srcpad);
filter->first_session = TRUE;
filter->roc_changed = FALSE;
}
static void
gst_srtp_dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSrtpDec *filter = GST_SRTP_DEC (object);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
case PROP_REPLAY_WINDOW_SIZE:
filter->replay_window_size = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (filter);
}
static void
gst_srtp_dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSrtpDec *filter = GST_SRTP_DEC (object);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
case PROP_REPLAY_WINDOW_SIZE:
g_value_set_uint (value, filter->replay_window_size);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (filter);
}
static void
gst_srtp_dec_remove_stream (GstSrtpDec * filter, guint ssrc)
{
GstSrtpDecSsrcStream *stream = NULL;
if (filter->streams == NULL)
return;
stream = g_hash_table_lookup (filter->streams, GUINT_TO_POINTER (ssrc));
if (stream) {
srtp_remove_stream (filter->session, ssrc);
g_hash_table_remove (filter->streams, GUINT_TO_POINTER (ssrc));
}
}
static GstSrtpDecSsrcStream *
find_stream_by_ssrc (GstSrtpDec * filter, guint32 ssrc)
{
return g_hash_table_lookup (filter->streams, GUINT_TO_POINTER (ssrc));
}
/* get info from buffer caps
*/
static GstSrtpDecSsrcStream *
get_stream_from_caps (GstSrtpDec * filter, GstCaps * caps, guint32 ssrc)
{
GstSrtpDecSsrcStream *stream;
GstStructure *s;
GstBuffer *buf;
const gchar *rtp_cipher, *rtp_auth, *rtcp_cipher, *rtcp_auth;
/* Create new stream structure and set default values */
stream = g_slice_new0 (GstSrtpDecSsrcStream);
stream->ssrc = ssrc;
stream->key = NULL;
/* Get info from caps */
s = gst_caps_get_structure (caps, 0);
if (!s)
goto error;
rtp_cipher = gst_structure_get_string (s, "srtp-cipher");
rtp_auth = gst_structure_get_string (s, "srtp-auth");
rtcp_cipher = gst_structure_get_string (s, "srtcp-cipher");
rtcp_auth = gst_structure_get_string (s, "srtcp-auth");
if (!rtp_cipher || !rtp_auth || !rtcp_cipher || !rtcp_auth)
goto error;
gst_structure_get_uint (s, "roc", &stream->roc);
stream->rtp_cipher = enum_value_from_nick (GST_TYPE_SRTP_CIPHER_TYPE,
rtp_cipher);
stream->rtp_auth = enum_value_from_nick (GST_TYPE_SRTP_AUTH_TYPE, rtp_auth);
stream->rtcp_cipher = enum_value_from_nick (GST_TYPE_SRTP_CIPHER_TYPE,
rtcp_cipher);
stream->rtcp_auth = enum_value_from_nick (GST_TYPE_SRTP_AUTH_TYPE, rtcp_auth);
if ((gint) stream->rtp_cipher == -1 || (gint) stream->rtp_auth == -1 ||
(gint) stream->rtcp_cipher == -1 || (gint) stream->rtcp_auth == -1) {
GST_WARNING_OBJECT (filter, "Invalid caps for stream,"
" unknown cipher or auth type");
goto error;
}
if (stream->rtcp_cipher != NULL_CIPHER && stream->rtcp_auth == NULL_AUTH) {
GST_WARNING_OBJECT (filter,
"Cannot have SRTP NULL authentication with a not-NULL encryption"
" cipher.");
goto error;
}
if (gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {
GST_DEBUG_OBJECT (filter, "Got key [%p] for SSRC %u", buf, ssrc);
stream->key = buf;
} else if (STREAM_HAS_CRYPTO (stream)) {
goto error;
}
return stream;
error:
g_slice_free (GstSrtpDecSsrcStream, stream);
return NULL;
}
/* Get SRTP params by signal
*/
static GstCaps *
signal_get_srtp_params (GstSrtpDec * filter, guint32 ssrc, gint signal)
{
GstCaps *caps = NULL;
g_signal_emit (filter, gst_srtp_dec_signals[signal], 0, ssrc, &caps);
if (caps != NULL)
GST_DEBUG_OBJECT (filter, "Caps received");
return caps;
}
/* Create a stream in the session
*/
static err_status_t
init_session_stream (GstSrtpDec * filter, guint32 ssrc,
GstSrtpDecSsrcStream * stream)
{
err_status_t ret;
srtp_policy_t policy;
GstMapInfo map;
guchar tmp[1];
memset (&policy, 0, sizeof (srtp_policy_t));
if (!stream)
return err_status_bad_param;
GST_INFO_OBJECT (filter, "Setting RTP policy...");
set_crypto_policy_cipher_auth (stream->rtp_cipher, stream->rtp_auth,
&policy.rtp);
GST_INFO_OBJECT (filter, "Setting RTCP policy...");
set_crypto_policy_cipher_auth (stream->rtcp_cipher, stream->rtcp_auth,
&policy.rtcp);
if (stream->key) {
gst_buffer_map (stream->key, &map, GST_MAP_READ);
policy.key = (guchar *) map.data;
} else {
policy.key = tmp;
}
policy.ssrc.value = ssrc;
policy.ssrc.type = ssrc_specific;
policy.window_size = filter->replay_window_size;
policy.next = NULL;
/* If it is the first stream, create the session
* If not, add the stream policy to the session
*/
if (filter->first_session)
ret = srtp_create (&filter->session, &policy);
else
ret = srtp_add_stream (filter->session, &policy);
if (stream->key)
gst_buffer_unmap (stream->key, &map);
if (ret == err_status_ok) {
srtp_stream_t srtp_stream;
srtp_stream = srtp_get_stream (filter->session, htonl (ssrc));
if (srtp_stream) {
/* Here, we just set the ROC, but we also need to set the initial
* RTP sequence number later, otherwise libsrtp will not be able
* to get the right packet index. */
rdbx_set_roc (&srtp_stream->rtp_rdbx, stream->roc);
filter->roc_changed = TRUE;
}
filter->first_session = FALSE;
g_hash_table_insert (filter->streams, GUINT_TO_POINTER (stream->ssrc),
stream);
}
return ret;
}
/* Return a stream structure for a given buffer
*/
static GstSrtpDecSsrcStream *
validate_buffer (GstSrtpDec * filter, GstBuffer * buf, guint32 * ssrc,
gboolean * is_rtcp)
{
GstSrtpDecSsrcStream *stream = NULL;
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
if (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf)) {
if (gst_rtp_buffer_get_payload_type (&rtpbuf) < 64
|| gst_rtp_buffer_get_payload_type (&rtpbuf) > 80) {
*ssrc = gst_rtp_buffer_get_ssrc (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
*is_rtcp = FALSE;
goto have_ssrc;
}
gst_rtp_buffer_unmap (&rtpbuf);
}
if (rtcp_buffer_get_ssrc (buf, ssrc)) {
*is_rtcp = TRUE;
} else {
GST_WARNING_OBJECT (filter, "No SSRC found in buffer");
return NULL;
}
have_ssrc:
stream = find_stream_by_ssrc (filter, *ssrc);
if (stream)
return stream;
return request_key_with_signal (filter, *ssrc, SIGNAL_REQUEST_KEY);
}
static void
free_stream (GstSrtpDecSsrcStream * stream)
{
if (stream->key)
gst_buffer_unref (stream->key);
g_slice_free (GstSrtpDecSsrcStream, stream);
}
/* Create new stream from params in caps
*/
static GstSrtpDecSsrcStream *
update_session_stream_from_caps (GstSrtpDec * filter, guint32 ssrc,
GstCaps * caps)
{
GstSrtpDecSsrcStream *stream = NULL;
GstSrtpDecSsrcStream *old_stream = NULL;
err_status_t err;
g_return_val_if_fail (GST_IS_SRTP_DEC (filter), NULL);
g_return_val_if_fail (GST_IS_CAPS (caps), NULL);
stream = get_stream_from_caps (filter, caps, ssrc);
old_stream = find_stream_by_ssrc (filter, ssrc);
if (stream && old_stream &&
stream->rtp_cipher == old_stream->rtp_cipher &&
stream->rtcp_cipher == old_stream->rtcp_cipher &&
stream->rtp_auth == old_stream->rtp_auth &&
stream->rtcp_auth == old_stream->rtcp_auth &&
stream->key && old_stream->key &&
gst_buffer_get_size (stream->key) ==
gst_buffer_get_size (old_stream->key)) {
GstMapInfo info;
if (gst_buffer_map (old_stream->key, &info, GST_MAP_READ)) {
gboolean equal;
equal = (gst_buffer_memcmp (stream->key, 0, info.data, info.size) == 0);
gst_buffer_unmap (old_stream->key, &info);
if (equal) {
free_stream (stream);
return old_stream;
}
}
}
/* Remove existing stream, if any */
gst_srtp_dec_remove_stream (filter, ssrc);
if (stream) {
/* Create new session stream */
err = init_session_stream (filter, ssrc, stream);
if (err != err_status_ok) {
if (stream->key)
gst_buffer_unref (stream->key);
g_slice_free (GstSrtpDecSsrcStream, stream);
stream = NULL;
}
}
return stream;
}
static gboolean
remove_yes (gpointer key, gpointer value, gpointer user_data)
{
return TRUE;
}
/* Clear the policy list
*/
static void
gst_srtp_dec_clear_streams (GstSrtpDec * filter)
{
guint nb = 0;
GST_OBJECT_LOCK (filter);
if (!filter->first_session)
srtp_dealloc (filter->session);
if (filter->streams)
nb = g_hash_table_foreach_remove (filter->streams, remove_yes, NULL);
filter->first_session = TRUE;
GST_OBJECT_UNLOCK (filter);
GST_DEBUG_OBJECT (filter, "Cleared %d streams", nb);
}
/* Send a signal
*/
static GstSrtpDecSsrcStream *
request_key_with_signal (GstSrtpDec * filter, guint32 ssrc, gint signal)
{
GstCaps *caps;
GstSrtpDecSsrcStream *stream = NULL;
caps = signal_get_srtp_params (filter, ssrc, signal);
if (caps) {
stream = update_session_stream_from_caps (filter, ssrc, caps);
if (stream)
GST_DEBUG_OBJECT (filter, "New stream set with SSRC %u", ssrc);
else
GST_WARNING_OBJECT (filter, "Could not set stream with SSRC %u", ssrc);
gst_caps_unref (caps);
} else {
GST_WARNING_OBJECT (filter, "Could not get caps for stream with SSRC %u",
ssrc);
}
return stream;
}
static gboolean
gst_srtp_dec_sink_setcaps (GstPad * pad, GstObject * parent,
GstCaps * caps, gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad;
GstStructure *ps;
gboolean ret = FALSE;
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
ps = gst_caps_get_structure (caps, 0);
if (gst_structure_has_field_typed (ps, "ssrc", G_TYPE_UINT) &&
gst_structure_has_field_typed (ps, "srtp-cipher", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtp-auth", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtcp-cipher", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtcp-auth", G_TYPE_STRING)) {
guint ssrc;
gst_structure_get_uint (ps, "ssrc", &ssrc);
if (!update_session_stream_from_caps (filter, ssrc, caps)) {
GST_WARNING_OBJECT (pad, "Could not create session from pad caps: %"
GST_PTR_FORMAT, caps);
return FALSE;
}
}
caps = gst_caps_copy (caps);
ps = gst_caps_get_structure (caps, 0);
gst_structure_remove_fields (ps, "srtp-key", "srtp-cipher", "srtp-auth",
"srtcp-cipher", "srtcp-auth", NULL);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-rtcp");
else
gst_structure_set_name (ps, "application/x-rtp");
otherpad = gst_pad_get_element_private (pad);
ret = gst_pad_set_caps (otherpad, caps);
gst_caps_unref (caps);
return ret;
}
static gboolean
gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event)
{
gboolean ret;
GstCaps *caps;
GstSrtpDec *filter = GST_SRTP_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
ret = gst_srtp_dec_sink_setcaps (pad, parent, caps, FALSE);
gst_event_unref (event);
return ret;
case GST_EVENT_SEGMENT:
/* Make sure to send a caps event downstream before the segment event,
* even if upstream didn't */
if (!gst_pad_has_current_caps (filter->rtp_srcpad)) {
GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtp");
gst_pad_set_caps (filter->rtp_srcpad, caps);
gst_caps_unref (caps);
}
filter->rtp_has_segment = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
filter->rtp_has_segment = FALSE;
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean ret;
GstCaps *caps;
GstSrtpDec *filter = GST_SRTP_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
ret = gst_srtp_dec_sink_setcaps (pad, parent, caps, TRUE);
gst_event_unref (event);
return ret;
case GST_EVENT_SEGMENT:
/* Make sure to send a caps event downstream before the segment event,
* even if upstream didn't */
if (!gst_pad_has_current_caps (filter->rtcp_srcpad)) {
GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
gst_pad_set_caps (filter->rtcp_srcpad, caps);
gst_caps_unref (caps);
}
filter->rtcp_has_segment = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
filter->rtcp_has_segment = FALSE;
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_srtp_dec_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;
GstCaps *template_caps;
GstPad *otherpad;
GstCaps *other_caps;
GstCaps *ret;
int i;
gst_query_parse_caps (query, &filter);
otherpad = (GstPad *) gst_pad_get_element_private (pad);
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-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);
}
}
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-srtcp");
else
gst_structure_set_name (ps, "application/x-srtp");
}
if (filter) {
GstCaps *tmp;
tmp = gst_caps_intersect (ret, filter);
gst_caps_unref (ret);
ret = tmp;
}
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_dec_sink_query_rtp (GstPad * pad, GstObject * parent, GstQuery * query)
{
return gst_srtp_dec_sink_query (pad, parent, query, FALSE);
}
static gboolean
gst_srtp_dec_sink_query_rtcp (GstPad * pad, GstObject * parent,
GstQuery * query)
{
return gst_srtp_dec_sink_query (pad, parent, query, TRUE);
}
static GstIterator *
gst_srtp_dec_iterate_internal_links (GstPad * pad, GstObject * parent,
gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad = NULL;
GstIterator *it = NULL;
otherpad = (GstPad *) gst_pad_get_element_private (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_dec_iterate_internal_links_rtp (GstPad * pad, GstObject * parent)
{
return gst_srtp_dec_iterate_internal_links (pad, parent, FALSE);
}
static GstIterator *
gst_srtp_dec_iterate_internal_links_rtcp (GstPad * pad, GstObject * parent)
{
return gst_srtp_dec_iterate_internal_links (pad, parent, TRUE);
}
static void
gst_srtp_dec_push_early_events (GstSrtpDec * filter, GstPad * pad,
GstPad * otherpad, gboolean is_rtcp)
{
GstEvent *otherev, *ev;
ev = gst_pad_get_sticky_event (pad, GST_EVENT_STREAM_START, 0);
if (ev) {
gst_event_unref (ev);
} else {
gchar *new_stream_id;
otherev = gst_pad_get_sticky_event (otherpad, GST_EVENT_STREAM_START, 0);
if (otherev) {
const gchar *other_stream_id;
gst_event_parse_stream_start (otherev, &other_stream_id);
new_stream_id = g_strdup_printf ("%s/%s", other_stream_id,
is_rtcp ? "rtcp" : "rtp");
gst_event_unref (otherev);
} else {
new_stream_id = gst_pad_create_stream_id (pad, GST_ELEMENT (filter),
is_rtcp ? "rtcp" : "rtp");
}
ev = gst_event_new_stream_start (new_stream_id);
g_free (new_stream_id);
gst_pad_push_event (pad, ev);
}
ev = gst_pad_get_sticky_event (pad, GST_EVENT_CAPS, 0);
if (ev) {
gst_event_unref (ev);
} else {
GstCaps *caps;
if (is_rtcp)
caps = gst_caps_new_empty_simple ("application/x-rtcp");
else
caps = gst_caps_new_empty_simple ("application/x-rtp");
gst_pad_set_caps (pad, caps);
gst_caps_unref (caps);
}
ev = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
if (ev) {
gst_event_unref (ev);
} else {
ev = gst_pad_get_sticky_event (otherpad, GST_EVENT_SEGMENT, 0);
if (ev)
gst_pad_push_event (pad, ev);
}
if (is_rtcp)
filter->rtcp_has_segment = TRUE;
else
filter->rtp_has_segment = TRUE;
}
/*
* This function should be called while holding the filter lock
*/
static gboolean
gst_srtp_dec_decode_buffer (GstSrtpDec * filter, GstPad * pad, GstBuffer * buf,
gboolean is_rtcp, guint32 ssrc)
{
GstMapInfo map;
err_status_t err;
gint size;
GST_LOG_OBJECT (pad, "Received %s buffer of size %" G_GSIZE_FORMAT
" with SSRC = %u", is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf),
ssrc);
/* Change buffer to remove protection */
buf = gst_buffer_make_writable (buf);
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
size = map.size;
unprotect:
gst_srtp_init_event_reporter ();
if (is_rtcp)
err = srtp_unprotect_rtcp (filter->session, map.data, &size);
else {
/* If ROC has changed, we know we need to set the initial RTP
* sequence number too. */
if (filter->roc_changed) {
srtp_stream_t stream;
stream = srtp_get_stream (filter->session, htonl (ssrc));
if (stream) {
guint16 seqnum = 0;
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
seqnum = gst_rtp_buffer_get_seq (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
/* We finally add the RTP sequence number to the current
* rollover counter. */
stream->rtp_rdbx.index &= ~0xFFFF;
stream->rtp_rdbx.index |= seqnum;
}
filter->roc_changed = FALSE;
}
err = srtp_unprotect (filter->session, map.data, &size);
}
GST_OBJECT_UNLOCK (filter);
if (err != err_status_ok) {
GST_WARNING_OBJECT (pad,
"Unable to unprotect buffer (unprotect failed code %d)", err);
/* Signal user depending on type of error */
switch (err) {
case err_status_key_expired:
GST_OBJECT_LOCK (filter);
/* Update stream */
if (find_stream_by_ssrc (filter, ssrc)) {
GST_OBJECT_UNLOCK (filter);
if (request_key_with_signal (filter, ssrc, SIGNAL_HARD_LIMIT)) {
GST_OBJECT_LOCK (filter);
goto unprotect;
} else {
GST_WARNING_OBJECT (filter, "Hard limit reached, no new key, "
"dropping");
}
} else {
GST_WARNING_OBJECT (filter, "Could not find matching stream, "
"dropping");
}
break;
case err_status_auth_fail:
GST_WARNING_OBJECT (filter, "Error authentication packet, dropping");
break;
case err_status_cipher_fail:
GST_WARNING_OBJECT (filter, "Error while decrypting packet, dropping");
break;
default:
GST_WARNING_OBJECT (filter, "Other error, dropping");
break;
}
gst_buffer_unmap (buf, &map);
GST_OBJECT_LOCK (filter);
return FALSE;
}
gst_buffer_unmap (buf, &map);
gst_buffer_set_size (buf, size);
GST_OBJECT_LOCK (filter);
return TRUE;
}
static GstFlowReturn
gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad;
GstSrtpDecSsrcStream *stream = NULL;
GstFlowReturn ret = GST_FLOW_OK;
guint32 ssrc = 0;
GST_OBJECT_LOCK (filter);
/* Check if this stream exists, if not create a new stream */
if (!(stream = validate_buffer (filter, buf, &ssrc, &is_rtcp))) {
GST_OBJECT_UNLOCK (filter);
GST_WARNING_OBJECT (filter, "Invalid buffer, dropping");
goto drop_buffer;
}
if (!STREAM_HAS_CRYPTO (stream)) {
GST_OBJECT_UNLOCK (filter);
goto push_out;
}
if (!gst_srtp_dec_decode_buffer (filter, pad, buf, is_rtcp, ssrc)) {
GST_OBJECT_UNLOCK (filter);
goto drop_buffer;
}
GST_OBJECT_UNLOCK (filter);
/* If all is well, we may have reached soft limit */
if (gst_srtp_get_soft_limit_reached ())
request_key_with_signal (filter, ssrc, SIGNAL_SOFT_LIMIT);
push_out:
/* Push buffer to source pad */
if (is_rtcp) {
otherpad = filter->rtcp_srcpad;
if (!filter->rtcp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtcp_srcpad,
filter->rtp_srcpad, TRUE);
} else {
otherpad = filter->rtp_srcpad;
if (!filter->rtp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtp_srcpad,
filter->rtcp_srcpad, FALSE);
}
ret = gst_pad_push (otherpad, buf);
return ret;
drop_buffer:
/* Drop buffer, except if gst_pad_push returned OK or an error */
gst_buffer_unref (buf);
return ret;
}
static gboolean
validate_buffer_it (GstBuffer ** buffer, guint index, gpointer user_data)
{
ValidateBufferItData *data = user_data;
if (!(*data->stream =
validate_buffer (data->filter, *buffer, data->ssrc, data->is_rtcp))) {
GST_WARNING_OBJECT (data->filter, "Invalid buffer, dropping");
gst_buffer_replace (buffer, NULL);
}
return TRUE;
}
static gboolean
decode_buffer_it (GstBuffer ** buffer, guint index, gpointer user_data)
{
DecodeBufferItData *data = user_data;
if (!gst_srtp_dec_decode_buffer (data->filter, data->pad, *buffer,
data->is_rtcp, data->ssrc)) {
GST_WARNING_OBJECT (data->filter, "Error decoding buffer, dropping");
gst_buffer_replace (buffer, NULL);
}
return TRUE;
}
static GstFlowReturn
gst_srtp_dec_chain_list (GstPad * pad, GstObject * parent,
GstBufferList * buf_list, gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad;
GstSrtpDecSsrcStream *stream = NULL;
GstFlowReturn ret = GST_FLOW_OK;
guint32 ssrc = 0;
ValidateBufferItData validate_data;
DecodeBufferItData decode_data;
validate_data.stream = &stream;
validate_data.filter = filter;
validate_data.ssrc = &ssrc;
validate_data.is_rtcp = &is_rtcp;
decode_data.filter = filter;
decode_data.pad = pad;
GST_OBJECT_LOCK (filter);
/* Check if this stream exists, if not create a new stream */
gst_buffer_list_foreach (buf_list, validate_buffer_it, &validate_data);
if (!gst_buffer_list_length (buf_list)) {
GST_OBJECT_LOCK (filter);
gst_buffer_list_unref (buf_list);
return GST_FLOW_OK;
}
if (!STREAM_HAS_CRYPTO (stream)) {
GST_OBJECT_UNLOCK (filter);
goto push_out;
}
decode_data.ssrc = ssrc;
decode_data.is_rtcp = is_rtcp;
gst_buffer_list_foreach (buf_list, decode_buffer_it, &decode_data);
GST_OBJECT_UNLOCK (filter);
if (!gst_buffer_list_length (buf_list)) {
gst_buffer_list_unref (buf_list);
return GST_FLOW_OK;
}
/* If all is well, we may have reached soft limit */
if (gst_srtp_get_soft_limit_reached ())
request_key_with_signal (filter, ssrc, SIGNAL_SOFT_LIMIT);
push_out:
/* Push buffer list to source pad */
if (is_rtcp) {
otherpad = filter->rtcp_srcpad;
if (!filter->rtcp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtcp_srcpad,
filter->rtp_srcpad, TRUE);
} else {
otherpad = filter->rtp_srcpad;
if (!filter->rtp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtp_srcpad,
filter->rtcp_srcpad, FALSE);
}
GST_LOG_OBJECT (pad, "Pushing buffer chain of %d",
gst_buffer_list_length (buf_list));
ret = gst_pad_push_list (otherpad, buf_list);
return ret;
}
static GstFlowReturn
gst_srtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_dec_chain (pad, parent, buf, FALSE);
}
static GstFlowReturn
gst_srtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_dec_chain (pad, parent, buf, TRUE);
}
static GstFlowReturn
gst_srtp_dec_chain_list_rtp (GstPad * pad, GstObject * parent,
GstBufferList * buf_list)
{
return gst_srtp_dec_chain_list (pad, parent, buf_list, FALSE);
}
static GstFlowReturn
gst_srtp_dec_chain_list_rtcp (GstPad * pad, GstObject * parent,
GstBufferList * buf_list)
{
return gst_srtp_dec_chain_list (pad, parent, buf_list, TRUE);
}
static GstStateChangeReturn
gst_srtp_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstSrtpDec *filter;
filter = GST_SRTP_DEC (element);
GST_OBJECT_LOCK (filter);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
filter->streams = g_hash_table_new_full (g_direct_hash, g_direct_equal,
NULL, (GDestroyNotify) free_stream);
filter->rtp_has_segment = FALSE;
filter->rtcp_has_segment = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
GST_OBJECT_UNLOCK (filter);
res = GST_ELEMENT_CLASS (gst_srtp_dec_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_dec_clear_streams (filter);
g_hash_table_unref (filter->streams);
filter->streams = NULL;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
/* entry point to initialize the plug-in
* initialize the plug-in itself
* register the element factories and other features
*/
gboolean
gst_srtp_dec_plugin_init (GstPlugin * srtpdec)
{
GST_DEBUG_CATEGORY_INIT (gst_srtp_dec_debug, "srtpdec", 0, "SRTP dec");
return gst_element_register (srtpdec, "srtpdec", GST_RANK_NONE,
GST_TYPE_SRTP_DEC);
}
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