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authorMichael Niedermayer <michaelni@gmx.at>2013-01-15 16:05:00 +0100
committerMichael Niedermayer <michaelni@gmx.at>2013-01-15 16:05:34 +0100
commitb52925d2cde98b2a3ffaf451248191c8627d356a (patch)
treeb5bc13efa1c4f0151fb47fecf295935000c71dc7
parente7e0186eeb0a0aa1e2ca805b97eb60cbd14e0567 (diff)
parent2f3bada63e57345329c4f9b48e9b81b5cfc03d05 (diff)
downloadffmpeg-b52925d2cde98b2a3ffaf451248191c8627d356a.tar.gz
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>
-rw-r--r--Changelog1
-rw-r--r--libavformat/Makefile4
-rw-r--r--libavformat/allformats.c1
-rw-r--r--libavformat/rtpdec.c14
-rw-r--r--libavformat/rtpdec.h6
-rw-r--r--libavformat/rtsp.c12
-rw-r--r--libavformat/rtsp.h3
-rw-r--r--libavformat/srtpproto.c144
-rw-r--r--libavformat/version.h4
9 files changed, 185 insertions, 4 deletions
diff --git a/Changelog b/Changelog
index 62d63026d6..6a520d21d7 100644
--- a/Changelog
+++ b/Changelog
@@ -3,6 +3,7 @@ releases are sorted from youngest to oldest.
version <next>:
- VDPAU hardware acceleration through normal hwaccel
+- SRTP support
version 1.1:
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 756059795e..dc3abfd786 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -43,7 +43,8 @@ OBJS-$(CONFIG_RTPDEC) += rdt.o \
rtpdec_qt.o \
rtpdec_svq3.o \
rtpdec_vp8.o \
- rtpdec_xiph.o
+ rtpdec_xiph.o \
+ srtp.o
OBJS-$(CONFIG_RTPENC_CHAIN) += rtpenc_chain.o rtp.o
# muxers/demuxers
@@ -430,6 +431,7 @@ OBJS-$(CONFIG_RTMPTE_PROTOCOL) += rtmpproto.o rtmppkt.o
OBJS-$(CONFIG_RTMPTS_PROTOCOL) += rtmpproto.o rtmppkt.o
OBJS-$(CONFIG_RTP_PROTOCOL) += rtpproto.o
OBJS-$(CONFIG_SCTP_PROTOCOL) += sctp.o
+OBJS-$(CONFIG_SRTP_PROTOCOL) += srtpproto.o srtp.o
OBJS-$(CONFIG_TCP_PROTOCOL) += tcp.o
OBJS-$(CONFIG_TLS_PROTOCOL) += tls.o
OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 83094f95df..f49b436f50 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -327,6 +327,7 @@ void av_register_all(void)
REGISTER_PROTOCOL(RTMPTS, rtmpts);
REGISTER_PROTOCOL(RTP, rtp);
REGISTER_PROTOCOL(SCTP, sctp);
+ REGISTER_PROTOCOL(SRTP, srtp);
REGISTER_PROTOCOL(TCP, tcp);
REGISTER_PROTOCOL(TLS, tls);
REGISTER_PROTOCOL(UDP, udp);
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 597b8040a3..fb1e5b9ea7 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -26,6 +26,7 @@
#include "avformat.h"
#include "mpegts.h"
#include "network.h"
+#include "srtp.h"
#include "url.h"
#include "rtpdec.h"
#include "rtpdec_formats.h"
@@ -543,6 +544,13 @@ void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
s->handler = handler;
}
+void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
+ const char *params)
+{
+ if (!ff_srtp_set_crypto(&s->srtp, suite, params))
+ s->srtp_enabled = 1;
+}
+
/**
* This was the second switch in rtp_parse packet.
* Normalizes time, if required, sets stream_index, etc.
@@ -879,7 +887,10 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **bufptr, int len)
{
- int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
+ int rv;
+ if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
+ return -1;
+ rv = rtp_parse_one_packet(s, pkt, bufptr, len);
s->prev_ret = rv;
while (rv == AVERROR(EAGAIN) && has_next_packet(s))
rv = rtp_parse_queued_packet(s, pkt);
@@ -892,6 +903,7 @@ void ff_rtp_parse_close(RTPDemuxContext *s)
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
ff_mpegts_parse_close(s->ts);
}
+ ff_srtp_free(&s->srtp);
av_free(s);
}
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
index cf7448850f..0c5dcabc8b 100644
--- a/libavformat/rtpdec.h
+++ b/libavformat/rtpdec.h
@@ -27,6 +27,7 @@
#include "avformat.h"
#include "rtp.h"
#include "url.h"
+#include "srtp.h"
typedef struct PayloadContext PayloadContext;
typedef struct RTPDynamicProtocolHandler RTPDynamicProtocolHandler;
@@ -43,6 +44,8 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
int payload_type, int queue_size);
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
RTPDynamicProtocolHandler *handler);
+void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
+ const char *params);
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **buf, int len);
void ff_rtp_parse_close(RTPDemuxContext *s);
@@ -163,6 +166,9 @@ struct RTPDemuxContext {
/* used to send back RTCP RR */
char hostname[256];
+ int srtp_enabled;
+ struct SRTPContext srtp;
+
/** Statistics for this stream (used by RTCP receiver reports) */
RTPStatistics statistics;
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 0582a00f8c..23ce65c0bf 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -480,6 +480,14 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
s->nb_streams > 0) {
st = s->streams[s->nb_streams - 1];
st->codec->sample_rate = atoi(p);
+ } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
+ // RFC 4568
+ rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
+ get_word(buf1, sizeof(buf1), &p); // ignore tag
+ get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
+ p += strspn(p, SPACE_CHARS);
+ if (av_strstart(p, "inline:", &p))
+ get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
} else {
if (rt->server_type == RTSP_SERVER_WMS)
ff_wms_parse_sdp_a_line(s, p);
@@ -653,6 +661,10 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
}
+ if (rtsp_st->crypto_suite[0])
+ ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
+ rtsp_st->crypto_suite,
+ rtsp_st->crypto_params);
}
return 0;
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index fb4c761495..321cd7a1f8 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -440,6 +440,9 @@ typedef struct RTSPStream {
/** Enable sending RTCP feedback messages according to RFC 4585 */
int feedback;
+
+ char crypto_suite[40];
+ char crypto_params[100];
} RTSPStream;
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
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,
+};
diff --git a/libavformat/version.h b/libavformat/version.h
index e50e501462..e1fe74e034 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -30,8 +30,8 @@
#include "libavutil/avutil.h"
#define LIBAVFORMAT_VERSION_MAJOR 54
-#define LIBAVFORMAT_VERSION_MINOR 59
-#define LIBAVFORMAT_VERSION_MICRO 107
+#define LIBAVFORMAT_VERSION_MINOR 60
+#define LIBAVFORMAT_VERSION_MICRO 100
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \