diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2013-01-15 16:05:00 +0100 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2013-01-15 16:05:34 +0100 |
commit | b52925d2cde98b2a3ffaf451248191c8627d356a (patch) | |
tree | b5bc13efa1c4f0151fb47fecf295935000c71dc7 /libavformat | |
parent | e7e0186eeb0a0aa1e2ca805b97eb60cbd14e0567 (diff) | |
parent | 2f3bada63e57345329c4f9b48e9b81b5cfc03d05 (diff) | |
download | ffmpeg-streaming-b52925d2cde98b2a3ffaf451248191c8627d356a.zip ffmpeg-streaming-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>
Diffstat (limited to 'libavformat')
-rw-r--r-- | libavformat/Makefile | 4 | ||||
-rw-r--r-- | libavformat/allformats.c | 1 | ||||
-rw-r--r-- | libavformat/rtpdec.c | 14 | ||||
-rw-r--r-- | libavformat/rtpdec.h | 6 | ||||
-rw-r--r-- | libavformat/rtsp.c | 12 | ||||
-rw-r--r-- | libavformat/rtsp.h | 3 | ||||
-rw-r--r-- | libavformat/srtpproto.c | 144 | ||||
-rw-r--r-- | libavformat/version.h | 4 |
8 files changed, 184 insertions, 4 deletions
diff --git a/libavformat/Makefile b/libavformat/Makefile index 7560597..dc3abfd 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 83094f9..f49b436 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 597b804..fb1e5b9 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 cf74488..0c5dcab 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 0582a00..23ce65c 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 fb4c7614..321cd7a 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 0000000..f5e3d14 --- /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 e50e501..e1fe74e 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, \ |