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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1507 lines
44 KiB
C
1507 lines
44 KiB
C
/*
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* GStreamer - GStreamer SRTP encoder
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*
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* Copyright 2009-2011 Collabora Ltd.
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* @author: Gabriel Millaire <gabriel.millaire@collabora.com>
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* @author: Olivier Crete <olivier.crete@collabora.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*
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* Alternatively, the contents of this file may be used under the
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* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
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* which case the following provisions apply instead of the ones
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* mentioned above:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-srtpenc
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* @title: srtpenc
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* @see_also: srtpdec
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*
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* gstrtpenc acts as an encoder that adds security to RTP and RTCP
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* packets in the form of encryption and authentication. It outs SRTP
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* and SRTCP.
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*
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* An application can request multiple RTP and RTCP pads to protect,
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* but every sink pad requested must receive packets from the same
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* source (identical SSRC). If a packet received contains a different
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* SSRC, a warning is emitted and the valid SSRC is forced on the packet.
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*
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* This element uses libsrtp library. When receiving the first packet,
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* the library is initialized with a new stream (based on the SSRC). It
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* uses the default RTP and RTCP encryption and authentication mechanisms,
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* unless the user has set the relevant properties first. It also uses
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* a master key that MUST be set by property (key) at the beginning. The
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* master key must be of a maximum length of 46 characters (14 characters
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* for the salt plus the key). The encryption and authentication mechanisms
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* available are :
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*
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* Encryption (properties rtp-cipher and rtcp-cipher)
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* - AES_ICM 256 bits (maximum security)
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* - AES_ICM 128 bits (default)
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* - NULL
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*
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* Authentication (properties rtp-auth and rtcp-auth)
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* - HMAC_SHA1 80 bits (default, maximum protection)
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* - HMAC_SHA1 32 bits
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* - NULL
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*
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* Note that for SRTP protection, authentication is mandatory (non-null)
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* if encryption is used (non-null).
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*
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* When requested to create a sink pad, a linked source pad is created.
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* Each packet received is first analysed (checked for valid SSRC) then
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* its buffer is protected with libsrtp, then pushed on the source pad.
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* If protection failed or the stream could not be created, the buffer
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* is dropped and a warning is emitted. The packets pushed on the source
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* pad are of type 'application/x-srtp' or 'application/x-srtcp'.
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*
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* When the maximum usage of the master key is reached, a soft-limit
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* signal is sent to the user. The user must then set a new master key
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* by property. If the hard limit is reached, a flag is set and every
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* subsequent packet is dropped, until a new key is set and the stream
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* has been updated.
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*
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* If a stream is to be shared between multiple clients it is also
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* possible to request the internal SRTP rollover counter for a given
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* SSRC. The rollover counter should be then transmitted and used by the
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* clients to authenticate and decrypt the packets. Failing to do that
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* the clients will start with a rollover counter of 0 which will
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* probably be incorrect if the stream has been transmitted for a
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* while to other clients.
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*
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* This element supports sending with a single Master Key, it is possible to set the
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* Master Key Identifier (MKI) using the "mki" property. If this property is set, the MKI
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* will be added to every buffer.
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*/
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#include "gstsrtpenc.h"
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <string.h>
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#include <stdio.h>
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GST_DEBUG_CATEGORY_STATIC (gst_srtp_enc_debug);
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#define GST_CAT_DEFAULT gst_srtp_enc_debug
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/* 128 bit key size: 14 (salt) + 16 */
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#define MASTER_128_KEY_SIZE 30
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/* 256 bit key size: 14 (salt) + 16 + 16 */
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#define MASTER_256_KEY_SIZE 46
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/* Properties default values */
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#define DEFAULT_MASTER_KEY NULL
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#define DEFAULT_RTP_CIPHER GST_SRTP_CIPHER_AES_128_ICM
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#define DEFAULT_RTP_AUTH GST_SRTP_AUTH_HMAC_SHA1_80
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#define DEFAULT_RTCP_CIPHER DEFAULT_RTP_CIPHER
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#define DEFAULT_RTCP_AUTH DEFAULT_RTP_AUTH
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#define DEFAULT_RANDOM_KEY FALSE
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#define DEFAULT_REPLAY_WINDOW_SIZE 128
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#define DEFAULT_ALLOW_REPEAT_TX FALSE
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#define HAS_CRYPTO(filter) (filter->rtp_cipher != GST_SRTP_CIPHER_NULL || \
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filter->rtcp_cipher != GST_SRTP_CIPHER_NULL || \
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filter->rtp_auth != GST_SRTP_AUTH_NULL || \
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filter->rtcp_auth != GST_SRTP_AUTH_NULL)
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/* Filter signals and args */
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enum
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{
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SIGNAL_SOFT_LIMIT,
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SIGNAL_GET_ROLLOVER_COUNTER,
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_MKEY,
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PROP_RTP_CIPHER,
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PROP_RTP_AUTH,
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PROP_RTCP_CIPHER,
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PROP_RTCP_AUTH,
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PROP_RANDOM_KEY,
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PROP_REPLAY_WINDOW_SIZE,
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PROP_ALLOW_REPEAT_TX,
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PROP_STATS,
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PROP_MKI
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};
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typedef struct ProcessBufferItData
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{
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GstSrtpEnc *filter;
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GstPad *pad;
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GstBufferList *out_list;
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GstFlowReturn flowret;
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gboolean is_rtcp;
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} ProcessBufferItData;
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/* the capabilities of the inputs and outputs.
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*
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* describe the real formats here.
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*/
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static GstStaticPadTemplate rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtp_sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtp_src_%u",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-srtp")
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);
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static GstStaticPadTemplate rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_src_%u",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-srtcp")
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);
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G_DEFINE_TYPE (GstSrtpEnc, gst_srtp_enc, GST_TYPE_ELEMENT);
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static guint gst_srtp_enc_signals[LAST_SIGNAL] = { 0 };
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static void gst_srtp_enc_dispose (GObject * object);
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static void gst_srtp_enc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_srtp_enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_srtp_enc_sink_query_rtp (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static gboolean gst_srtp_enc_sink_query_rtcp (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static GstIterator *gst_srtp_enc_iterate_internal_links_rtp (GstPad * pad,
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GstObject * parent);
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static GstIterator *gst_srtp_enc_iterate_internal_links_rtcp (GstPad * pad,
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GstObject * parent);
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static GstFlowReturn gst_srtp_enc_chain_rtp (GstPad * pad, GstObject * parent,
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GstBuffer * buf);
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static GstFlowReturn gst_srtp_enc_chain_rtcp (GstPad * pad, GstObject * parent,
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GstBuffer * buf);
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static GstFlowReturn gst_srtp_enc_chain_list_rtp (GstPad * pad,
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GstObject * parent, GstBufferList * buf);
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static GstFlowReturn gst_srtp_enc_chain_list_rtcp (GstPad * pad,
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GstObject * parent, GstBufferList * buf);
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static gboolean gst_srtp_enc_sink_event_rtp (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_srtp_enc_sink_event_rtcp (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static GstStateChangeReturn gst_srtp_enc_change_state (GstElement * element,
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GstStateChange transition);
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static GstPad *gst_srtp_enc_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
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static void gst_srtp_enc_release_pad (GstElement * element, GstPad * pad);
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/* initialize the srtpenc's class
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*/
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static void
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gst_srtp_enc_class_init (GstSrtpEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&rtp_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&rtp_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&rtcp_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&rtcp_sink_template);
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gst_element_class_set_static_metadata (gstelement_class, "SRTP encoder",
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"Filter/Network/SRTP",
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"A SRTP and SRTCP encoder",
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"Gabriel Millaire <millaire.gabriel@collabora.com>");
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/* Install callbacks */
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gobject_class->set_property = gst_srtp_enc_set_property;
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gobject_class->get_property = gst_srtp_enc_get_property;
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gobject_class->dispose = gst_srtp_enc_dispose;
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_srtp_enc_request_new_pad);
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gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_srtp_enc_release_pad);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_srtp_enc_change_state);
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/* Install properties */
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g_object_class_install_property (gobject_class, PROP_MKEY,
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g_param_spec_boxed ("key", "Key", "Master key (minimum of "
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G_STRINGIFY (MASTER_128_KEY_SIZE) " and maximum of "
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G_STRINGIFY (MASTER_256_KEY_SIZE) " bytes)",
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GST_TYPE_BUFFER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_PLAYING));
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g_object_class_install_property (gobject_class, PROP_RTP_CIPHER,
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g_param_spec_enum ("rtp-cipher", "RTP Cipher", "RTP Cipher",
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GST_TYPE_SRTP_CIPHER_TYPE, DEFAULT_RTP_CIPHER,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RTP_AUTH,
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g_param_spec_enum ("rtp-auth", "RTP Authentication",
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"RTP Authentication", GST_TYPE_SRTP_AUTH_TYPE, DEFAULT_RTP_AUTH,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RTCP_CIPHER,
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g_param_spec_enum ("rtcp-cipher", "RTCP Cipher",
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"RTCP Cipher", GST_TYPE_SRTP_CIPHER_TYPE, DEFAULT_RTCP_CIPHER,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RTCP_AUTH,
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g_param_spec_enum ("rtcp-auth", "RTCP Authentication",
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"RTCP Authentication", GST_TYPE_SRTP_AUTH_TYPE, DEFAULT_RTCP_AUTH,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RANDOM_KEY,
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g_param_spec_boolean ("random-key", "Generate random key",
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"Generate a random key if TRUE",
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DEFAULT_RANDOM_KEY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_REPLAY_WINDOW_SIZE,
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g_param_spec_uint ("replay-window-size", "Replay window size",
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"Size of the replay protection window",
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64, 0x8000, DEFAULT_REPLAY_WINDOW_SIZE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_ALLOW_REPEAT_TX,
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g_param_spec_boolean ("allow-repeat-tx",
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"Allow repeat packets transmission",
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"Whether retransmissions of packets with the same sequence number are allowed"
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"(Note that such repeated transmissions must have the same RTP payload, "
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"or a severe security weakness is introduced!)",
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DEFAULT_ALLOW_REPEAT_TX, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_STATS,
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g_param_spec_boxed ("stats", "Statistics", "Various statistics",
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GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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#ifdef HAVE_SRTP2
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g_object_class_install_property (gobject_class, PROP_MKI,
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g_param_spec_boxed ("mki", "MKI",
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"Master key Identifier (NULL means no MKI)", GST_TYPE_BUFFER,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_PLAYING));
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#endif
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/**
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* GstSrtpEnc::soft-limit:
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* @gstsrtpenc: the element on which the signal is emitted
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*
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* Signal emitted when the stream with @ssrc has reached the soft
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* limit of utilisation of it's master encryption key. User should
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* provide a new key by setting the #GstSrtpEnc:key property.
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*/
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gst_srtp_enc_signals[SIGNAL_SOFT_LIMIT] =
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g_signal_new ("soft-limit", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
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}
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/* initialize the new element
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*/
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static void
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gst_srtp_enc_init (GstSrtpEnc * filter)
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{
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filter->key_changed = TRUE;
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filter->first_session = TRUE;
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filter->key = DEFAULT_MASTER_KEY;
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filter->rtp_cipher = DEFAULT_RTP_CIPHER;
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filter->rtp_auth = DEFAULT_RTP_AUTH;
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filter->rtcp_cipher = DEFAULT_RTCP_CIPHER;
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filter->rtcp_auth = DEFAULT_RTCP_AUTH;
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filter->replay_window_size = DEFAULT_REPLAY_WINDOW_SIZE;
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filter->allow_repeat_tx = DEFAULT_ALLOW_REPEAT_TX;
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filter->ssrcs_set = g_hash_table_new (g_direct_hash, g_direct_equal);
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}
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static guint
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max_cipher_key_size (GstSrtpEnc * filter)
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{
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guint rtp_size, rtcp_size;
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rtp_size = cipher_key_size (filter->rtp_cipher);
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rtcp_size = cipher_key_size (filter->rtcp_cipher);
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return (rtp_size > rtcp_size) ? rtp_size : rtcp_size;
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}
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/* Create stream
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*
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* Should be called with the filter locked
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*/
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static srtp_err_status_t
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gst_srtp_enc_create_session (GstSrtpEnc * filter)
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{
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srtp_err_status_t ret;
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srtp_policy_t policy;
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GstMapInfo map;
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guchar tmp[1];
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#ifdef HAVE_SRTP2
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srtp_master_key_t mkey;
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srtp_master_key_t *mkey_ptr = &mkey;
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gboolean has_mki = FALSE;
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GstMapInfo mki_map;
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#endif
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memset (&policy, 0, sizeof (srtp_policy_t));
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if (HAS_CRYPTO (filter)) {
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guint expected;
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gsize keysize;
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if (filter->key == NULL) {
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GST_OBJECT_UNLOCK (filter);
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GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
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("Cipher is not NULL, key must be set"),
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("Cipher is not NULL, key must be set"));
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GST_OBJECT_LOCK (filter);
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return srtp_err_status_fail;
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}
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expected = max_cipher_key_size (filter);
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keysize = gst_buffer_get_size (filter->key);
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if (expected != keysize) {
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GST_OBJECT_UNLOCK (filter);
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GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
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("Master key size is wrong"),
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("Expected master key of %d bytes, but received %" G_GSIZE_FORMAT
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" bytes", expected, keysize));
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GST_OBJECT_LOCK (filter);
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return srtp_err_status_fail;
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}
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}
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GST_DEBUG_OBJECT (filter, "Setting RTP/RTCP policy to %d / %d",
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filter->rtp_cipher, filter->rtcp_cipher);
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set_crypto_policy_cipher_auth (filter->rtp_cipher, filter->rtp_auth,
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&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;
|
|
}
|
|
|
|
#ifdef HAVE_SRTP2
|
|
if (filter->mki) {
|
|
if (!gst_buffer_map (filter->mki, &mki_map, GST_MAP_READ)) {
|
|
GST_OBJECT_UNLOCK (filter);
|
|
GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS, ("Could not map MKI"),
|
|
(NULL));
|
|
GST_OBJECT_LOCK (filter);
|
|
|
|
ret = srtp_err_status_fail;
|
|
goto done;
|
|
}
|
|
has_mki = TRUE;
|
|
|
|
policy.num_master_keys = 1;
|
|
policy.keys = &mkey_ptr;
|
|
mkey.key = policy.key;
|
|
policy.key = NULL;
|
|
|
|
mkey.mki_id = (guchar *) mki_map.data;
|
|
mkey.mki_size = mki_map.size;
|
|
}
|
|
#endif
|
|
|
|
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;
|
|
|
|
#ifdef HAVE_SRTP2
|
|
done:
|
|
|
|
if (has_mki)
|
|
gst_buffer_unmap (filter->mki, &mki_map);
|
|
#endif
|
|
|
|
if (HAS_CRYPTO (filter))
|
|
gst_buffer_unmap (filter->key, &map);
|
|
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* Release resources and set default values
|
|
*/
|
|
static void
|
|
gst_srtp_enc_reset_no_lock (GstSrtpEnc * filter)
|
|
{
|
|
if (!filter->first_session) {
|
|
if (filter->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);
|
|
|
|
gst_buffer_replace (&filter->key, NULL);
|
|
gst_buffer_replace (&filter->mki, 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:
|
|
gst_clear_buffer (&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;
|
|
#ifdef HAVE_SRTP2
|
|
case PROP_MKI:
|
|
gst_clear_buffer (&filter->mki);
|
|
filter->mki = g_value_dup_boxed (value);
|
|
filter->key_changed = TRUE;
|
|
GST_INFO_OBJECT (object, "Set property: mki=[%p]", filter->mki);
|
|
break;
|
|
#endif
|
|
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;
|
|
#ifdef HAVE_SRTP2
|
|
case PROP_MKI:
|
|
if (filter->mki)
|
|
g_value_set_boxed (value, filter->mki);
|
|
break;
|
|
#endif
|
|
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));
|
|
}
|
|
|
|
static void
|
|
gst_srtp_enc_add_ssrc (GstSrtpEnc * filter, guint ssrc)
|
|
{
|
|
gboolean is_added =
|
|
g_hash_table_add (filter->ssrcs_set, GUINT_TO_POINTER (ssrc));
|
|
if (is_added) {
|
|
GST_DEBUG_OBJECT (filter, "Added ssrc %u", ssrc);
|
|
}
|
|
}
|
|
|
|
/* 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);
|
|
gst_srtp_enc_add_ssrc (filter, ssrc);
|
|
}
|
|
|
|
if (HAS_CRYPTO (filter))
|
|
gst_structure_set (ps, "srtp-key", GST_TYPE_BUFFER, filter->key, NULL);
|
|
|
|
#ifdef HAVE_SRTP2
|
|
if (filter->mki)
|
|
gst_structure_set (ps, "mki", GST_TYPE_BUFFER, filter->mki, NULL);
|
|
#endif
|
|
|
|
/* 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", "mki", 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 void
|
|
gst_srtp_enc_ensure_ssrc (GstSrtpEnc * filter, GstBuffer * buf)
|
|
{
|
|
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
|
|
if (gst_rtp_buffer_map (buf,
|
|
GST_MAP_READ | GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING, &rtpbuf)) {
|
|
guint32 ssrc = gst_rtp_buffer_get_ssrc (&rtpbuf);
|
|
gst_srtp_enc_add_ssrc (filter, ssrc);
|
|
gst_rtp_buffer_unmap (&rtpbuf);
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_srtp_enc_process_buffer (GstSrtpEnc * filter, GstPad * pad,
|
|
GstBuffer * buf, gboolean is_rtcp, GstBuffer ** outbuf_ptr)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
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 (filter->session == NULL) {
|
|
/* The rtcp session disappeared (element shutting down) */
|
|
GST_OBJECT_UNLOCK (filter);
|
|
ret = GST_FLOW_FLUSHING;
|
|
goto fail;
|
|
}
|
|
|
|
gst_srtp_enc_ensure_ssrc (filter, buf);
|
|
|
|
#ifdef HAVE_SRTP2
|
|
if (is_rtcp)
|
|
err = srtp_protect_rtcp_mki (filter->session, mapout.data, &size,
|
|
(filter->mki != NULL), 0);
|
|
else
|
|
err = srtp_protect_mki (filter->session, mapout.data, &size,
|
|
(filter->mki != NULL), 0);
|
|
#else
|
|
if (is_rtcp)
|
|
err = srtp_protect_rtcp (filter->session, mapout.data, &size);
|
|
else
|
|
err = srtp_protect (filter->session, mapout.data, &size);
|
|
#endif
|
|
|
|
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)"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto fail;
|
|
|
|
} else {
|
|
/* srtp_protect failed */
|
|
GST_ELEMENT_ERROR (filter, LIBRARY, FAILED, (NULL),
|
|
("Unable to protect buffer (protect failed) code %d", err));
|
|
ret = GST_FLOW_ERROR;
|
|
goto fail;
|
|
}
|
|
|
|
*outbuf_ptr = bufout;
|
|
return ret;
|
|
|
|
fail:
|
|
gst_buffer_unref (bufout);
|
|
return ret;
|
|
}
|
|
|
|
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);
|
|
|
|
ret = gst_srtp_enc_process_buffer (filter, pad, buf, is_rtcp, &bufout);
|
|
if (ret != GST_FLOW_OK)
|
|
goto out;
|
|
|
|
/* 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;
|
|
|
|
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;
|
|
}
|
|
|
|
static gboolean
|
|
process_buffer_it (GstBuffer ** buffer, guint index, gpointer user_data)
|
|
{
|
|
ProcessBufferItData *data = user_data;
|
|
GstBuffer *bufout;
|
|
GstFlowReturn ret;
|
|
|
|
ret = gst_srtp_enc_process_buffer (data->filter, data->pad, *buffer,
|
|
data->is_rtcp, &bufout);
|
|
if (ret != GST_FLOW_OK) {
|
|
data->flowret = ret;
|
|
return FALSE;
|
|
}
|
|
|
|
gst_buffer_list_add (data->out_list, bufout);
|
|
|
|
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;
|
|
process_data.flowret = GST_FLOW_OK;
|
|
|
|
if (!gst_buffer_list_foreach (buf_list, process_buffer_it, &process_data)) {
|
|
ret = process_data.flowret;
|
|
goto out;
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* RFC 3711 says in "3. SRTP Framework" that SRTCP message authentication
|
|
* is MANDATORY. In case of GCM let the pipeline handle any errors.
|
|
*/
|
|
if (filter->rtcp_cipher != GST_SRTP_CIPHER_AES_128_GCM
|
|
&& filter->rtcp_cipher != GST_SRTP_CIPHER_AES_256_GCM
|
|
&& filter->rtcp_cipher != GST_SRTP_CIPHER_NULL
|
|
&& filter->rtcp_auth == GST_SRTP_AUTH_NULL) {
|
|
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);
|
|
}
|