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mirror of https://git.videolan.org/git/ffmpeg.git synced 2024-08-03 09:49:58 +02:00
ffmpeg/libavformat/srtpproto.c
Martin Storsjö fab8156b2f avio: Copy URLContext generic options into child URLContexts
Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.

This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).

Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.

Signed-off-by: Martin Storsjö <martin@martin.st>
2016-03-24 10:34:19 +02:00

147 lines
4.8 KiB
C

/*
* SRTP network protocol
* Copyright (c) 2012 Martin Storsjo
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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 "rtpdec.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[RTP_MAX_PACKET_LENGTH];
} 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, D },
{ "srtp_in_params", "", offsetof(SRTPProtoContext, in_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D },
{ 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,
h->protocols, h)) < 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);
}
const 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,
};