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authorMartin Storsjö <martin@martin.st>2012-12-12 00:22:48 +0200
committerMartin Storsjö <martin@martin.st>2013-01-15 11:55:10 +0200
commit2f3bada63e57345329c4f9b48e9b81b5cfc03d05 (patch)
tree0fca9f3fa78273b26df15fa5bdbbc8e9bdcd14d1 /libavformat/srtpproto.c
parent424da308302bef604844d3110a39f2f03bf5358e (diff)
lavf: Add a protocol for SRTP encryption/decryption
This is mostly useful for encryption together with the RTP muxer, but could also be set up as IO towards the peer with the SDP demuxer with custom IO. Signed-off-by: Martin Storsjö <martin@martin.st>
Diffstat (limited to 'libavformat/srtpproto.c')
-rw-r--r--libavformat/srtpproto.c144
1 files changed, 144 insertions, 0 deletions
diff --git a/libavformat/srtpproto.c b/libavformat/srtpproto.c
new file mode 100644
index 0000000000..1cacfdbb06
--- /dev/null
+++ b/libavformat/srtpproto.c
@@ -0,0 +1,144 @@
+/*
+ * 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 "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,
+};