Merge commit '2f3bada6'

* commit '2f3bada6':
  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>

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:

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

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);

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);
 }
 

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;
 

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;

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,

libavformat/srtpproto.c

+/*
+ * 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,
+};

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, \