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  1. /*
  2.  * SRTP network protocol
  3.  * Copyright (c) 2012 Martin Storsjo
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. #include "libavutil/opt.h"
  23. #include "avformat.h"
  24. #include "avio_internal.h"
  25. #include "url.h"
  26.  
  27. #include "internal.h"
  28. #include "rtpdec.h"
  29. #include "srtp.h"
  30.  
  31. typedef struct SRTPProtoContext {
  32.     const AVClass *class;
  33.     URLContext *rtp_hd;
  34.     const char *out_suite, *out_params;
  35.     const char *in_suite, *in_params;
  36.     struct SRTPContext srtp_out, srtp_in;
  37.     uint8_t encryptbuf[RTP_MAX_PACKET_LENGTH];
  38. } SRTPProtoContext;
  39.  
  40. #define D AV_OPT_FLAG_DECODING_PARAM
  41. #define E AV_OPT_FLAG_ENCODING_PARAM
  42. static const AVOption options[] = {
  43.     { "srtp_out_suite", "", offsetof(SRTPProtoContext, out_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
  44.     { "srtp_out_params", "", offsetof(SRTPProtoContext, out_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
  45.     { "srtp_in_suite", "", offsetof(SRTPProtoContext, in_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
  46.     { "srtp_in_params", "", offsetof(SRTPProtoContext, in_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
  47.     { NULL }
  48. };
  49.  
  50. static const AVClass srtp_context_class = {
  51.     .class_name     = "srtp",
  52.     .item_name      = av_default_item_name,
  53.     .option         = options,
  54.     .version        = LIBAVUTIL_VERSION_INT,
  55. };
  56.  
  57. static int srtp_close(URLContext *h)
  58. {
  59.     SRTPProtoContext *s = h->priv_data;
  60.     ff_srtp_free(&s->srtp_out);
  61.     ff_srtp_free(&s->srtp_in);
  62.     ffurl_close(s->rtp_hd);
  63.     s->rtp_hd = NULL;
  64.     return 0;
  65. }
  66.  
  67. static int srtp_open(URLContext *h, const char *uri, int flags)
  68. {
  69.     SRTPProtoContext *s = h->priv_data;
  70.     char hostname[256], buf[1024], path[1024];
  71.     int rtp_port, ret;
  72.  
  73.     if (s->out_suite && s->out_params)
  74.         if ((ret = ff_srtp_set_crypto(&s->srtp_out, s->out_suite, s->out_params)) < 0)
  75.             goto fail;
  76.     if (s->in_suite && s->in_params)
  77.         if ((ret = ff_srtp_set_crypto(&s->srtp_in, s->in_suite, s->in_params)) < 0)
  78.             goto fail;
  79.  
  80.     av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &rtp_port,
  81.                  path, sizeof(path), uri);
  82.     ff_url_join(buf, sizeof(buf), "rtp", NULL, hostname, rtp_port, "%s", path);
  83.     if ((ret = ffurl_open(&s->rtp_hd, buf, flags, &h->interrupt_callback, NULL)) < 0)
  84.         goto fail;
  85.  
  86.     h->max_packet_size = FFMIN(s->rtp_hd->max_packet_size,
  87.                                sizeof(s->encryptbuf)) - 14;
  88.     h->is_streamed = 1;
  89.     return 0;
  90.  
  91. fail:
  92.     srtp_close(h);
  93.     return ret;
  94. }
  95.  
  96. static int srtp_read(URLContext *h, uint8_t *buf, int size)
  97. {
  98.     SRTPProtoContext *s = h->priv_data;
  99.     int ret;
  100. start:
  101.     ret = ffurl_read(s->rtp_hd, buf, size);
  102.     if (ret > 0 && s->srtp_in.aes) {
  103.         if (ff_srtp_decrypt(&s->srtp_in, buf, &ret) < 0)
  104.             goto start;
  105.     }
  106.     return ret;
  107. }
  108.  
  109. static int srtp_write(URLContext *h, const uint8_t *buf, int size)
  110. {
  111.     SRTPProtoContext *s = h->priv_data;
  112.     if (!s->srtp_out.aes)
  113.         return ffurl_write(s->rtp_hd, buf, size);
  114.     size = ff_srtp_encrypt(&s->srtp_out, buf, size, s->encryptbuf,
  115.                            sizeof(s->encryptbuf));
  116.     if (size < 0)
  117.         return size;
  118.     return ffurl_write(s->rtp_hd, s->encryptbuf, size);
  119. }
  120.  
  121. static int srtp_get_file_handle(URLContext *h)
  122. {
  123.     SRTPProtoContext *s = h->priv_data;
  124.     return ffurl_get_file_handle(s->rtp_hd);
  125. }
  126.  
  127. static int srtp_get_multi_file_handle(URLContext *h, int **handles,
  128.                                       int *numhandles)
  129. {
  130.     SRTPProtoContext *s = h->priv_data;
  131.     return ffurl_get_multi_file_handle(s->rtp_hd, handles, numhandles);
  132. }
  133.  
  134. URLProtocol ff_srtp_protocol = {
  135.     .name                      = "srtp",
  136.     .url_open                  = srtp_open,
  137.     .url_read                  = srtp_read,
  138.     .url_write                 = srtp_write,
  139.     .url_close                 = srtp_close,
  140.     .url_get_file_handle       = srtp_get_file_handle,
  141.     .url_get_multi_file_handle = srtp_get_multi_file_handle,
  142.     .priv_data_size            = sizeof(SRTPProtoContext),
  143.     .priv_data_class           = &srtp_context_class,
  144.     .flags                     = URL_PROTOCOL_FLAG_NETWORK,
  145. };
  146.