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Merge commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05'

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* 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>
This commit is contained in:
Michael Niedermayer 2013-01-15 16:05:00 +01:00
commit b52925d2cd
9 changed files with 185 additions and 4 deletions
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1
Changelog
View File

@ -3,6 +3,7 @@ releases are sorted from youngest to oldest.
version <next>:
- VDPAU hardware acceleration through normal hwaccel
- SRTP support
version 1.1:

4
libavformat/Makefile
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@ -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

1
libavformat/allformats.c
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@ -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);

14
libavformat/rtpdec.c
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@ -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);
}

6
libavformat/rtpdec.h
View File

@ -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;

12
libavformat/rtsp.c
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@ -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;

3
libavformat/rtsp.h
View File

@ -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,

144
libavformat/srtpproto.c Normal file
View File

@ -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,
};

4
libavformat/version.h
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@ -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, \