mirror of
https://github.com/pineappleEA/pineapple-src.git
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146 lines
4.8 KiB
C
Executable File
146 lines
4.8 KiB
C
Executable File
/*
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* SRTP network protocol
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* Copyright (c) 2012 Martin Storsjo
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/opt.h"
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#include "avformat.h"
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#include "avio_internal.h"
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#include "url.h"
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#include "internal.h"
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#include "rtpdec.h"
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#include "srtp.h"
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typedef struct SRTPProtoContext {
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const AVClass *class;
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URLContext *rtp_hd;
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const char *out_suite, *out_params;
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const char *in_suite, *in_params;
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struct SRTPContext srtp_out, srtp_in;
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uint8_t encryptbuf[RTP_MAX_PACKET_LENGTH];
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} SRTPProtoContext;
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#define D AV_OPT_FLAG_DECODING_PARAM
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#define E AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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{ "srtp_out_suite", "", offsetof(SRTPProtoContext, out_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
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{ "srtp_out_params", "", offsetof(SRTPProtoContext, out_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
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{ "srtp_in_suite", "", offsetof(SRTPProtoContext, in_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D },
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{ "srtp_in_params", "", offsetof(SRTPProtoContext, in_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D },
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{ NULL }
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};
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static const AVClass srtp_context_class = {
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.class_name = "srtp",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static int srtp_close(URLContext *h)
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{
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SRTPProtoContext *s = h->priv_data;
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ff_srtp_free(&s->srtp_out);
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ff_srtp_free(&s->srtp_in);
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ffurl_closep(&s->rtp_hd);
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return 0;
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}
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static int srtp_open(URLContext *h, const char *uri, int flags)
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{
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SRTPProtoContext *s = h->priv_data;
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char hostname[256], buf[1024], path[1024];
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int rtp_port, ret;
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if (s->out_suite && s->out_params)
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if ((ret = ff_srtp_set_crypto(&s->srtp_out, s->out_suite, s->out_params)) < 0)
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goto fail;
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if (s->in_suite && s->in_params)
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if ((ret = ff_srtp_set_crypto(&s->srtp_in, s->in_suite, s->in_params)) < 0)
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goto fail;
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av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &rtp_port,
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path, sizeof(path), uri);
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ff_url_join(buf, sizeof(buf), "rtp", NULL, hostname, rtp_port, "%s", path);
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if ((ret = ffurl_open_whitelist(&s->rtp_hd, buf, flags, &h->interrupt_callback,
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NULL, h->protocol_whitelist, h->protocol_blacklist, h)) < 0)
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goto fail;
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h->max_packet_size = FFMIN(s->rtp_hd->max_packet_size,
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sizeof(s->encryptbuf)) - 14;
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h->is_streamed = 1;
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return 0;
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fail:
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srtp_close(h);
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return ret;
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}
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static int srtp_read(URLContext *h, uint8_t *buf, int size)
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{
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SRTPProtoContext *s = h->priv_data;
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int ret;
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start:
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ret = ffurl_read(s->rtp_hd, buf, size);
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if (ret > 0 && s->srtp_in.aes) {
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if (ff_srtp_decrypt(&s->srtp_in, buf, &ret) < 0)
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goto start;
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}
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return ret;
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}
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static int srtp_write(URLContext *h, const uint8_t *buf, int size)
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{
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SRTPProtoContext *s = h->priv_data;
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if (!s->srtp_out.aes)
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return ffurl_write(s->rtp_hd, buf, size);
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size = ff_srtp_encrypt(&s->srtp_out, buf, size, s->encryptbuf,
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sizeof(s->encryptbuf));
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if (size < 0)
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return size;
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return ffurl_write(s->rtp_hd, s->encryptbuf, size);
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}
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static int srtp_get_file_handle(URLContext *h)
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{
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SRTPProtoContext *s = h->priv_data;
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return ffurl_get_file_handle(s->rtp_hd);
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}
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static int srtp_get_multi_file_handle(URLContext *h, int **handles,
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int *numhandles)
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{
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SRTPProtoContext *s = h->priv_data;
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return ffurl_get_multi_file_handle(s->rtp_hd, handles, numhandles);
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}
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const URLProtocol ff_srtp_protocol = {
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.name = "srtp",
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.url_open = srtp_open,
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.url_read = srtp_read,
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.url_write = srtp_write,
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.url_close = srtp_close,
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.url_get_file_handle = srtp_get_file_handle,
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.url_get_multi_file_handle = srtp_get_multi_file_handle,
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.priv_data_size = sizeof(SRTPProtoContext),
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.priv_data_class = &srtp_context_class,
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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