diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2013-01-15 19:05:00 +0400 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2013-01-15 19:05:34 +0400 |
commit | b52925d2cde98b2a3ffaf451248191c8627d356a (patch) | |
tree | b5bc13efa1c4f0151fb47fecf295935000c71dc7 /libavformat/srtpproto.c | |
parent | e7e0186eeb0a0aa1e2ca805b97eb60cbd14e0567 (diff) | |
parent | 2f3bada63e57345329c4f9b48e9b81b5cfc03d05 (diff) |
Merge commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05'
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
lavf: Add a protocol for SRTP encryption/decryption
rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/srtpproto.c')
-rw-r--r-- | libavformat/srtpproto.c | 144 |
1 files changed, 144 insertions, 0 deletions
diff --git a/libavformat/srtpproto.c b/libavformat/srtpproto.c new file mode 100644 index 0000000000..f5e3d14090 --- /dev/null +++ b/libavformat/srtpproto.c @@ -0,0 +1,144 @@ +/* + * SRTP network protocol + * Copyright (c) 2012 Martin Storsjo + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavutil/opt.h" +#include "avformat.h" +#include "avio_internal.h" +#include "url.h" + +#include "internal.h" +#include "srtp.h" + +typedef struct SRTPProtoContext { + const AVClass *class; + URLContext *rtp_hd; + const char *out_suite, *out_params; + const char *in_suite, *in_params; + struct SRTPContext srtp_out, srtp_in; + uint8_t encryptbuf[1500]; +} SRTPProtoContext; + +#define D AV_OPT_FLAG_DECODING_PARAM +#define E AV_OPT_FLAG_ENCODING_PARAM +static const AVOption options[] = { + { "srtp_out_suite", "", offsetof(SRTPProtoContext, out_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E }, + { "srtp_out_params", "", offsetof(SRTPProtoContext, out_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E }, + { "srtp_in_suite", "", offsetof(SRTPProtoContext, in_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E }, + { "srtp_in_params", "", offsetof(SRTPProtoContext, in_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E }, + { NULL } +}; + +static const AVClass srtp_context_class = { + .class_name = "srtp", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +static int srtp_close(URLContext *h) +{ + SRTPProtoContext *s = h->priv_data; + ff_srtp_free(&s->srtp_out); + ff_srtp_free(&s->srtp_in); + ffurl_close(s->rtp_hd); + s->rtp_hd = NULL; + return 0; +} + +static int srtp_open(URLContext *h, const char *uri, int flags) +{ + SRTPProtoContext *s = h->priv_data; + char hostname[256], buf[1024], path[1024]; + int rtp_port, ret; + + if (s->out_suite && s->out_params) + if ((ret = ff_srtp_set_crypto(&s->srtp_out, s->out_suite, s->out_params)) < 0) + goto fail; + if (s->in_suite && s->in_params) + if ((ret = ff_srtp_set_crypto(&s->srtp_in, s->in_suite, s->in_params)) < 0) + goto fail; + + av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &rtp_port, + path, sizeof(path), uri); + ff_url_join(buf, sizeof(buf), "rtp", NULL, hostname, rtp_port, "%s", path); + if ((ret = ffurl_open(&s->rtp_hd, buf, flags, &h->interrupt_callback, NULL)) < 0) + goto fail; + + h->max_packet_size = FFMIN(s->rtp_hd->max_packet_size, + sizeof(s->encryptbuf)) - 14; + h->is_streamed = 1; + return 0; + +fail: + srtp_close(h); + return ret; +} + +static int srtp_read(URLContext *h, uint8_t *buf, int size) +{ + SRTPProtoContext *s = h->priv_data; + int ret; +start: + ret = ffurl_read(s->rtp_hd, buf, size); + if (ret > 0 && s->srtp_in.aes) { + if (ff_srtp_decrypt(&s->srtp_in, buf, &ret) < 0) + goto start; + } + return ret; +} + +static int srtp_write(URLContext *h, const uint8_t *buf, int size) +{ + SRTPProtoContext *s = h->priv_data; + if (!s->srtp_out.aes) + return ffurl_write(s->rtp_hd, buf, size); + size = ff_srtp_encrypt(&s->srtp_out, buf, size, s->encryptbuf, + sizeof(s->encryptbuf)); + if (size < 0) + return size; + return ffurl_write(s->rtp_hd, s->encryptbuf, size); +} + +static int srtp_get_file_handle(URLContext *h) +{ + SRTPProtoContext *s = h->priv_data; + return ffurl_get_file_handle(s->rtp_hd); +} + +static int srtp_get_multi_file_handle(URLContext *h, int **handles, + int *numhandles) +{ + SRTPProtoContext *s = h->priv_data; + return ffurl_get_multi_file_handle(s->rtp_hd, handles, numhandles); +} + +URLProtocol ff_srtp_protocol = { + .name = "srtp", + .url_open = srtp_open, + .url_read = srtp_read, + .url_write = srtp_write, + .url_close = srtp_close, + .url_get_file_handle = srtp_get_file_handle, + .url_get_multi_file_handle = srtp_get_multi_file_handle, + .priv_data_size = sizeof(SRTPProtoContext), + .priv_data_class = &srtp_context_class, + .flags = URL_PROTOCOL_FLAG_NETWORK, +}; |