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  1. /*
  2. * Microsoft RTP/ASF support.
  3. * Copyright (c) 2008 Ronald S. Bultje
  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. * @file
  23. * @brief Microsoft RTP/ASF support
  24. * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
  25. */
  26. #include "libavutil/base64.h"
  27. #include "libavutil/avstring.h"
  28. #include "libavutil/intreadwrite.h"
  29. #include "rtp.h"
  30. #include "rtpdec_asf.h"
  31. #include "rtsp.h"
  32. #include "asf.h"
  33. /**
  34. * From MSDN 2.2.1.4, we learn that ASF data packets over RTP should not
  35. * contain any padding. Unfortunately, the header min/max_pktsize are not
  36. * updated (thus making min_pktsize invalid). Here, we "fix" these faulty
  37. * min_pktsize values in the ASF file header.
  38. * @return 0 on success, <0 on failure (currently -1).
  39. */
  40. static int rtp_asf_fix_header(uint8_t *buf, int len)
  41. {
  42. uint8_t *p = buf, *end = buf + len;
  43. if (len < sizeof(ff_asf_guid) * 2 + 22 ||
  44. memcmp(p, ff_asf_header, sizeof(ff_asf_guid))) {
  45. return -1;
  46. }
  47. p += sizeof(ff_asf_guid) + 14;
  48. do {
  49. uint64_t chunksize = AV_RL64(p + sizeof(ff_asf_guid));
  50. if (memcmp(p, ff_asf_file_header, sizeof(ff_asf_guid))) {
  51. if (chunksize > end - p)
  52. return -1;
  53. p += chunksize;
  54. continue;
  55. }
  56. /* skip most of the file header, to min_pktsize */
  57. p += 6 * 8 + 3 * 4 + sizeof(ff_asf_guid) * 2;
  58. if (p + 8 <= end && AV_RL32(p) == AV_RL32(p + 4)) {
  59. /* and set that to zero */
  60. AV_WL32(p, 0);
  61. return 0;
  62. }
  63. break;
  64. } while (end - p >= sizeof(ff_asf_guid) + 8);
  65. return -1;
  66. }
  67. /**
  68. * The following code is basically a buffered ByteIOContext,
  69. * with the added benefit of returning -EAGAIN (instead of 0)
  70. * on packet boundaries, such that the ASF demuxer can return
  71. * safely and resume business at the next packet.
  72. */
  73. static int packetizer_read(void *opaque, uint8_t *buf, int buf_size)
  74. {
  75. return AVERROR(EAGAIN);
  76. }
  77. static void init_packetizer(ByteIOContext *pb, uint8_t *buf, int len)
  78. {
  79. init_put_byte(pb, buf, len, 0, NULL, packetizer_read, NULL, NULL);
  80. /* this "fills" the buffer with its current content */
  81. pb->pos = len;
  82. pb->buf_end = buf + len;
  83. }
  84. int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
  85. {
  86. int ret = 0;
  87. if (av_strstart(p, "pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,", &p)) {
  88. ByteIOContext pb;
  89. RTSPState *rt = s->priv_data;
  90. int len = strlen(p) * 6 / 8;
  91. char *buf = av_mallocz(len);
  92. av_base64_decode(buf, p, len);
  93. if (rtp_asf_fix_header(buf, len) < 0)
  94. av_log(s, AV_LOG_ERROR,
  95. "Failed to fix invalid RTSP-MS/ASF min_pktsize\n");
  96. init_packetizer(&pb, buf, len);
  97. if (rt->asf_ctx) {
  98. av_close_input_stream(rt->asf_ctx);
  99. rt->asf_ctx = NULL;
  100. }
  101. ret = av_open_input_stream(&rt->asf_ctx, &pb, "", &asf_demuxer, NULL);
  102. if (ret < 0)
  103. return ret;
  104. rt->asf_pb_pos = url_ftell(&pb);
  105. av_free(buf);
  106. rt->asf_ctx->pb = NULL;
  107. }
  108. return ret;
  109. }
  110. static int asfrtp_parse_sdp_line(AVFormatContext *s, int stream_index,
  111. PayloadContext *asf, const char *line)
  112. {
  113. if (av_strstart(line, "stream:", &line)) {
  114. RTSPState *rt = s->priv_data;
  115. s->streams[stream_index]->id = strtol(line, NULL, 10);
  116. if (rt->asf_ctx) {
  117. int i;
  118. for (i = 0; i < rt->asf_ctx->nb_streams; i++) {
  119. if (s->streams[stream_index]->id == rt->asf_ctx->streams[i]->id) {
  120. *s->streams[stream_index]->codec =
  121. *rt->asf_ctx->streams[i]->codec;
  122. rt->asf_ctx->streams[i]->codec->extradata_size = 0;
  123. rt->asf_ctx->streams[i]->codec->extradata = NULL;
  124. av_set_pts_info(s->streams[stream_index], 32, 1, 1000);
  125. }
  126. }
  127. }
  128. }
  129. return 0;
  130. }
  131. struct PayloadContext {
  132. ByteIOContext *pktbuf, pb;
  133. uint8_t *buf;
  134. };
  135. /**
  136. * @return 0 when a packet was written into /p pkt, and no more data is left;
  137. * 1 when a packet was written into /p pkt, and more packets might be left;
  138. * <0 when not enough data was provided to return a full packet, or on error.
  139. */
  140. static int asfrtp_parse_packet(AVFormatContext *s, PayloadContext *asf,
  141. AVStream *st, AVPacket *pkt,
  142. uint32_t *timestamp,
  143. const uint8_t *buf, int len, int flags)
  144. {
  145. ByteIOContext *pb = &asf->pb;
  146. int res, mflags, len_off;
  147. RTSPState *rt = s->priv_data;
  148. if (!rt->asf_ctx)
  149. return -1;
  150. if (len > 0) {
  151. int off, out_len = 0;
  152. if (len < 4)
  153. return -1;
  154. av_freep(&asf->buf);
  155. init_put_byte(pb, buf, len, 0, NULL, NULL, NULL, NULL);
  156. while (url_ftell(pb) + 4 < len) {
  157. int start_off = url_ftell(pb);
  158. mflags = get_byte(pb);
  159. if (mflags & 0x80)
  160. flags |= RTP_FLAG_KEY;
  161. len_off = get_be24(pb);
  162. if (mflags & 0x20) /**< relative timestamp */
  163. url_fskip(pb, 4);
  164. if (mflags & 0x10) /**< has duration */
  165. url_fskip(pb, 4);
  166. if (mflags & 0x8) /**< has location ID */
  167. url_fskip(pb, 4);
  168. off = url_ftell(pb);
  169. if (!(mflags & 0x40)) {
  170. /**
  171. * If 0x40 is not set, the len_off field specifies an offset
  172. * of this packet's payload data in the complete (reassembled)
  173. * ASF packet. This is used to spread one ASF packet over
  174. * multiple RTP packets.
  175. */
  176. if (asf->pktbuf && len_off != url_ftell(asf->pktbuf)) {
  177. uint8_t *p;
  178. url_close_dyn_buf(asf->pktbuf, &p);
  179. asf->pktbuf = NULL;
  180. av_free(p);
  181. }
  182. if (!len_off && !asf->pktbuf &&
  183. (res = url_open_dyn_buf(&asf->pktbuf)) < 0)
  184. return res;
  185. if (!asf->pktbuf)
  186. return AVERROR(EIO);
  187. put_buffer(asf->pktbuf, buf + off, len - off);
  188. url_fskip(pb, len - off);
  189. if (!(flags & RTP_FLAG_MARKER))
  190. return -1;
  191. out_len = url_close_dyn_buf(asf->pktbuf, &asf->buf);
  192. asf->pktbuf = NULL;
  193. } else {
  194. /**
  195. * If 0x40 is set, the len_off field specifies the length of
  196. * the next ASF packet that can be read from this payload
  197. * data alone. This is commonly the same as the payload size,
  198. * but could be less in case of packet splitting (i.e.
  199. * multiple ASF packets in one RTP packet).
  200. */
  201. int cur_len = start_off + len_off - off;
  202. int prev_len = out_len;
  203. out_len += cur_len;
  204. asf->buf = av_realloc(asf->buf, out_len);
  205. memcpy(asf->buf + prev_len, buf + off, cur_len);
  206. url_fskip(pb, cur_len);
  207. }
  208. }
  209. init_packetizer(pb, asf->buf, out_len);
  210. pb->pos += rt->asf_pb_pos;
  211. pb->eof_reached = 0;
  212. rt->asf_ctx->pb = pb;
  213. }
  214. for (;;) {
  215. int i;
  216. res = av_read_packet(rt->asf_ctx, pkt);
  217. rt->asf_pb_pos = url_ftell(pb);
  218. if (res != 0)
  219. break;
  220. for (i = 0; i < s->nb_streams; i++) {
  221. if (s->streams[i]->id == rt->asf_ctx->streams[pkt->stream_index]->id) {
  222. pkt->stream_index = i;
  223. return 1; // FIXME: return 0 if last packet
  224. }
  225. }
  226. av_free_packet(pkt);
  227. }
  228. return res == 1 ? -1 : res;
  229. }
  230. static PayloadContext *asfrtp_new_context(void)
  231. {
  232. return av_mallocz(sizeof(PayloadContext));
  233. }
  234. static void asfrtp_free_context(PayloadContext *asf)
  235. {
  236. if (asf->pktbuf) {
  237. uint8_t *p = NULL;
  238. url_close_dyn_buf(asf->pktbuf, &p);
  239. asf->pktbuf = NULL;
  240. av_free(p);
  241. }
  242. av_freep(&asf->buf);
  243. av_free(asf);
  244. }
  245. #define RTP_ASF_HANDLER(n, s, t) \
  246. RTPDynamicProtocolHandler ff_ms_rtp_ ## n ## _handler = { \
  247. .enc_name = s, \
  248. .codec_type = t, \
  249. .codec_id = CODEC_ID_NONE, \
  250. .parse_sdp_a_line = asfrtp_parse_sdp_line, \
  251. .open = asfrtp_new_context, \
  252. .close = asfrtp_free_context, \
  253. .parse_packet = asfrtp_parse_packet, \
  254. };
  255. RTP_ASF_HANDLER(asf_pfv, "x-asf-pf", AVMEDIA_TYPE_VIDEO);
  256. RTP_ASF_HANDLER(asf_pfa, "x-asf-pf", AVMEDIA_TYPE_AUDIO);