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  1. /*
  2. * RTMP network protocol
  3. * Copyright (c) 2009 Kostya Shishkov
  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 libavformat/rtmpproto.c
  23. * RTMP protocol
  24. */
  25. #include "libavcodec/bytestream.h"
  26. #include "libavutil/avstring.h"
  27. #include "libavutil/lfg.h"
  28. #include "libavutil/sha.h"
  29. #include "avformat.h"
  30. #include "network.h"
  31. #include "flv.h"
  32. #include "rtmp.h"
  33. #include "rtmppkt.h"
  34. /* we can't use av_log() with URLContext yet... */
  35. #if LIBAVFORMAT_VERSION_MAJOR < 53
  36. #define LOG_CONTEXT NULL
  37. #else
  38. #define LOG_CONTEXT s
  39. #endif
  40. /** RTMP protocol handler state */
  41. typedef enum {
  42. STATE_START, ///< client has not done anything yet
  43. STATE_HANDSHAKED, ///< client has performed handshake
  44. STATE_RELEASING, ///< client releasing stream before publish it (for output)
  45. STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
  46. STATE_CONNECTING, ///< client connected to server successfully
  47. STATE_READY, ///< client has sent all needed commands and waits for server reply
  48. STATE_PLAYING, ///< client has started receiving multimedia data from server
  49. STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
  50. } ClientState;
  51. /** protocol handler context */
  52. typedef struct RTMPContext {
  53. URLContext* stream; ///< TCP stream used in interactions with RTMP server
  54. RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
  55. int chunk_size; ///< size of the chunks RTMP packets are divided into
  56. int is_input; ///< input/output flag
  57. char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
  58. char app[128]; ///< application
  59. ClientState state; ///< current state
  60. int main_channel_id; ///< an additional channel ID which is used for some invocations
  61. uint8_t* flv_data; ///< buffer with data for demuxer
  62. int flv_size; ///< current buffer size
  63. int flv_off; ///< number of bytes read from current buffer
  64. RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
  65. } RTMPContext;
  66. #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
  67. /** Client key used for digest signing */
  68. static const uint8_t rtmp_player_key[] = {
  69. 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
  70. 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
  71. 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
  72. 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
  73. 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
  74. };
  75. #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
  76. /** Key used for RTMP server digest signing */
  77. static const uint8_t rtmp_server_key[] = {
  78. 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
  79. 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
  80. 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
  81. 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
  82. 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
  83. 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
  84. };
  85. /**
  86. * Generates 'connect' call and sends it to the server.
  87. */
  88. static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
  89. const char *host, int port)
  90. {
  91. RTMPPacket pkt;
  92. uint8_t ver[64], *p;
  93. char tcurl[512];
  94. ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
  95. p = pkt.data;
  96. snprintf(tcurl, sizeof(tcurl), "%s://%s:%d/%s", proto, host, port, rt->app);
  97. ff_amf_write_string(&p, "connect");
  98. ff_amf_write_number(&p, 1.0);
  99. ff_amf_write_object_start(&p);
  100. ff_amf_write_field_name(&p, "app");
  101. ff_amf_write_string(&p, rt->app);
  102. if (rt->is_input) {
  103. snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
  104. RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
  105. } else {
  106. snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
  107. ff_amf_write_field_name(&p, "type");
  108. ff_amf_write_string(&p, "nonprivate");
  109. }
  110. ff_amf_write_field_name(&p, "flashVer");
  111. ff_amf_write_string(&p, ver);
  112. ff_amf_write_field_name(&p, "tcUrl");
  113. ff_amf_write_string(&p, tcurl);
  114. if (rt->is_input) {
  115. ff_amf_write_field_name(&p, "fpad");
  116. ff_amf_write_bool(&p, 0);
  117. ff_amf_write_field_name(&p, "capabilities");
  118. ff_amf_write_number(&p, 15.0);
  119. ff_amf_write_field_name(&p, "audioCodecs");
  120. ff_amf_write_number(&p, 1639.0);
  121. ff_amf_write_field_name(&p, "videoCodecs");
  122. ff_amf_write_number(&p, 252.0);
  123. ff_amf_write_field_name(&p, "videoFunction");
  124. ff_amf_write_number(&p, 1.0);
  125. }
  126. ff_amf_write_object_end(&p);
  127. pkt.data_size = p - pkt.data;
  128. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  129. }
  130. /**
  131. * Generates 'releaseStream' call and sends it to the server. It should make
  132. * the server release some channel for media streams.
  133. */
  134. static void gen_release_stream(URLContext *s, RTMPContext *rt)
  135. {
  136. RTMPPacket pkt;
  137. uint8_t *p;
  138. ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
  139. 29 + strlen(rt->playpath));
  140. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n");
  141. p = pkt.data;
  142. ff_amf_write_string(&p, "releaseStream");
  143. ff_amf_write_number(&p, 2.0);
  144. ff_amf_write_null(&p);
  145. ff_amf_write_string(&p, rt->playpath);
  146. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  147. ff_rtmp_packet_destroy(&pkt);
  148. }
  149. /**
  150. * Generates 'FCPublish' call and sends it to the server. It should make
  151. * the server preapare for receiving media streams.
  152. */
  153. static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
  154. {
  155. RTMPPacket pkt;
  156. uint8_t *p;
  157. ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
  158. 25 + strlen(rt->playpath));
  159. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n");
  160. p = pkt.data;
  161. ff_amf_write_string(&p, "FCPublish");
  162. ff_amf_write_number(&p, 3.0);
  163. ff_amf_write_null(&p);
  164. ff_amf_write_string(&p, rt->playpath);
  165. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  166. ff_rtmp_packet_destroy(&pkt);
  167. }
  168. /**
  169. * Generates 'FCUnpublish' call and sends it to the server. It should make
  170. * the server destroy stream.
  171. */
  172. static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
  173. {
  174. RTMPPacket pkt;
  175. uint8_t *p;
  176. ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
  177. 27 + strlen(rt->playpath));
  178. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n");
  179. p = pkt.data;
  180. ff_amf_write_string(&p, "FCUnpublish");
  181. ff_amf_write_number(&p, 5.0);
  182. ff_amf_write_null(&p);
  183. ff_amf_write_string(&p, rt->playpath);
  184. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  185. ff_rtmp_packet_destroy(&pkt);
  186. }
  187. /**
  188. * Generates 'createStream' call and sends it to the server. It should make
  189. * the server allocate some channel for media streams.
  190. */
  191. static void gen_create_stream(URLContext *s, RTMPContext *rt)
  192. {
  193. RTMPPacket pkt;
  194. uint8_t *p;
  195. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n");
  196. ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
  197. p = pkt.data;
  198. ff_amf_write_string(&p, "createStream");
  199. ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
  200. ff_amf_write_null(&p);
  201. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  202. ff_rtmp_packet_destroy(&pkt);
  203. }
  204. /**
  205. * Generates 'deleteStream' call and sends it to the server. It should make
  206. * the server remove some channel for media streams.
  207. */
  208. static void gen_delete_stream(URLContext *s, RTMPContext *rt)
  209. {
  210. RTMPPacket pkt;
  211. uint8_t *p;
  212. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n");
  213. ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
  214. p = pkt.data;
  215. ff_amf_write_string(&p, "deleteStream");
  216. ff_amf_write_number(&p, 0.0);
  217. ff_amf_write_null(&p);
  218. ff_amf_write_number(&p, rt->main_channel_id);
  219. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  220. ff_rtmp_packet_destroy(&pkt);
  221. }
  222. /**
  223. * Generates 'play' call and sends it to the server, then pings the server
  224. * to start actual playing.
  225. */
  226. static void gen_play(URLContext *s, RTMPContext *rt)
  227. {
  228. RTMPPacket pkt;
  229. uint8_t *p;
  230. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
  231. ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
  232. 20 + strlen(rt->playpath));
  233. pkt.extra = rt->main_channel_id;
  234. p = pkt.data;
  235. ff_amf_write_string(&p, "play");
  236. ff_amf_write_number(&p, 0.0);
  237. ff_amf_write_null(&p);
  238. ff_amf_write_string(&p, rt->playpath);
  239. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  240. ff_rtmp_packet_destroy(&pkt);
  241. // set client buffer time disguised in ping packet
  242. ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
  243. p = pkt.data;
  244. bytestream_put_be16(&p, 3);
  245. bytestream_put_be32(&p, 1);
  246. bytestream_put_be32(&p, 256); //TODO: what is a good value here?
  247. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  248. ff_rtmp_packet_destroy(&pkt);
  249. }
  250. /**
  251. * Generates 'publish' call and sends it to the server.
  252. */
  253. static void gen_publish(URLContext *s, RTMPContext *rt)
  254. {
  255. RTMPPacket pkt;
  256. uint8_t *p;
  257. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
  258. ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
  259. 30 + strlen(rt->playpath));
  260. pkt.extra = rt->main_channel_id;
  261. p = pkt.data;
  262. ff_amf_write_string(&p, "publish");
  263. ff_amf_write_number(&p, 0.0);
  264. ff_amf_write_null(&p);
  265. ff_amf_write_string(&p, rt->playpath);
  266. ff_amf_write_string(&p, "live");
  267. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  268. ff_rtmp_packet_destroy(&pkt);
  269. }
  270. /**
  271. * Generates ping reply and sends it to the server.
  272. */
  273. static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
  274. {
  275. RTMPPacket pkt;
  276. uint8_t *p;
  277. ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
  278. p = pkt.data;
  279. bytestream_put_be16(&p, 7);
  280. bytestream_put_be32(&p, AV_RB32(ppkt->data+2) + 1);
  281. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  282. ff_rtmp_packet_destroy(&pkt);
  283. }
  284. //TODO: Move HMAC code somewhere. Eventually.
  285. #define HMAC_IPAD_VAL 0x36
  286. #define HMAC_OPAD_VAL 0x5C
  287. /**
  288. * Calculates HMAC-SHA2 digest for RTMP handshake packets.
  289. *
  290. * @param src input buffer
  291. * @param len input buffer length (should be 1536)
  292. * @param gap offset in buffer where 32 bytes should not be taken into account
  293. * when calculating digest (since it will be used to store that digest)
  294. * @param key digest key
  295. * @param keylen digest key length
  296. * @param dst buffer where calculated digest will be stored (32 bytes)
  297. */
  298. static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
  299. const uint8_t *key, int keylen, uint8_t *dst)
  300. {
  301. struct AVSHA *sha;
  302. uint8_t hmac_buf[64+32] = {0};
  303. int i;
  304. sha = av_mallocz(av_sha_size);
  305. if (keylen < 64) {
  306. memcpy(hmac_buf, key, keylen);
  307. } else {
  308. av_sha_init(sha, 256);
  309. av_sha_update(sha,key, keylen);
  310. av_sha_final(sha, hmac_buf);
  311. }
  312. for (i = 0; i < 64; i++)
  313. hmac_buf[i] ^= HMAC_IPAD_VAL;
  314. av_sha_init(sha, 256);
  315. av_sha_update(sha, hmac_buf, 64);
  316. if (gap <= 0) {
  317. av_sha_update(sha, src, len);
  318. } else { //skip 32 bytes used for storing digest
  319. av_sha_update(sha, src, gap);
  320. av_sha_update(sha, src + gap + 32, len - gap - 32);
  321. }
  322. av_sha_final(sha, hmac_buf + 64);
  323. for (i = 0; i < 64; i++)
  324. hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
  325. av_sha_init(sha, 256);
  326. av_sha_update(sha, hmac_buf, 64+32);
  327. av_sha_final(sha, dst);
  328. av_free(sha);
  329. }
  330. /**
  331. * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest
  332. * will be stored) into that packet.
  333. *
  334. * @param buf handshake data (1536 bytes)
  335. * @return offset to the digest inside input data
  336. */
  337. static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
  338. {
  339. int i, digest_pos = 0;
  340. for (i = 8; i < 12; i++)
  341. digest_pos += buf[i];
  342. digest_pos = (digest_pos % 728) + 12;
  343. rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
  344. rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
  345. buf + digest_pos);
  346. return digest_pos;
  347. }
  348. /**
  349. * Verifies that the received server response has the expected digest value.
  350. *
  351. * @param buf handshake data received from the server (1536 bytes)
  352. * @param off position to search digest offset from
  353. * @return 0 if digest is valid, digest position otherwise
  354. */
  355. static int rtmp_validate_digest(uint8_t *buf, int off)
  356. {
  357. int i, digest_pos = 0;
  358. uint8_t digest[32];
  359. for (i = 0; i < 4; i++)
  360. digest_pos += buf[i + off];
  361. digest_pos = (digest_pos % 728) + off + 4;
  362. rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
  363. rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
  364. digest);
  365. if (!memcmp(digest, buf + digest_pos, 32))
  366. return digest_pos;
  367. return 0;
  368. }
  369. /**
  370. * Performs handshake with the server by means of exchanging pseudorandom data
  371. * signed with HMAC-SHA2 digest.
  372. *
  373. * @return 0 if handshake succeeds, negative value otherwise
  374. */
  375. static int rtmp_handshake(URLContext *s, RTMPContext *rt)
  376. {
  377. AVLFG rnd;
  378. uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
  379. 3, // unencrypted data
  380. 0, 0, 0, 0, // client uptime
  381. RTMP_CLIENT_VER1,
  382. RTMP_CLIENT_VER2,
  383. RTMP_CLIENT_VER3,
  384. RTMP_CLIENT_VER4,
  385. };
  386. uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
  387. uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
  388. int i;
  389. int server_pos, client_pos;
  390. uint8_t digest[32];
  391. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n");
  392. av_lfg_init(&rnd, 0xDEADC0DE);
  393. // generate handshake packet - 1536 bytes of pseudorandom data
  394. for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
  395. tosend[i] = av_lfg_get(&rnd) >> 24;
  396. client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
  397. url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
  398. i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
  399. if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
  400. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
  401. return -1;
  402. }
  403. i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
  404. if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
  405. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
  406. return -1;
  407. }
  408. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
  409. serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
  410. if (rt->is_input) {
  411. server_pos = rtmp_validate_digest(serverdata + 1, 772);
  412. if (!server_pos) {
  413. server_pos = rtmp_validate_digest(serverdata + 1, 8);
  414. if (!server_pos) {
  415. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n");
  416. return -1;
  417. }
  418. }
  419. rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
  420. rtmp_server_key, sizeof(rtmp_server_key),
  421. digest);
  422. rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
  423. digest, 32,
  424. digest);
  425. if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
  426. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n");
  427. return -1;
  428. }
  429. for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
  430. tosend[i] = av_lfg_get(&rnd) >> 24;
  431. rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
  432. rtmp_player_key, sizeof(rtmp_player_key),
  433. digest);
  434. rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
  435. digest, 32,
  436. tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
  437. // write reply back to the server
  438. url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
  439. } else {
  440. url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
  441. }
  442. return 0;
  443. }
  444. /**
  445. * Parses received packet and may perform some action depending on
  446. * the packet contents.
  447. * @return 0 for no errors, negative values for serious errors which prevent
  448. * further communications, positive values for uncritical errors
  449. */
  450. static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
  451. {
  452. int i, t;
  453. const uint8_t *data_end = pkt->data + pkt->data_size;
  454. switch (pkt->type) {
  455. case RTMP_PT_CHUNK_SIZE:
  456. if (pkt->data_size != 4) {
  457. av_log(LOG_CONTEXT, AV_LOG_ERROR,
  458. "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
  459. return -1;
  460. }
  461. if (!rt->is_input)
  462. ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
  463. rt->chunk_size = AV_RB32(pkt->data);
  464. if (rt->chunk_size <= 0) {
  465. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
  466. return -1;
  467. }
  468. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
  469. break;
  470. case RTMP_PT_PING:
  471. t = AV_RB16(pkt->data);
  472. if (t == 6)
  473. gen_pong(s, rt, pkt);
  474. break;
  475. case RTMP_PT_INVOKE:
  476. //TODO: check for the messages sent for wrong state?
  477. if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
  478. uint8_t tmpstr[256];
  479. if (!ff_amf_get_field_value(pkt->data + 9, data_end,
  480. "description", tmpstr, sizeof(tmpstr)))
  481. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
  482. return -1;
  483. } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
  484. switch (rt->state) {
  485. case STATE_HANDSHAKED:
  486. if (!rt->is_input) {
  487. gen_release_stream(s, rt);
  488. gen_fcpublish_stream(s, rt);
  489. rt->state = STATE_RELEASING;
  490. } else {
  491. rt->state = STATE_CONNECTING;
  492. }
  493. gen_create_stream(s, rt);
  494. break;
  495. case STATE_FCPUBLISH:
  496. rt->state = STATE_CONNECTING;
  497. break;
  498. case STATE_RELEASING:
  499. rt->state = STATE_FCPUBLISH;
  500. /* hack for Wowza Media Server, it does not send result for
  501. * releaseStream and FCPublish calls */
  502. if (!pkt->data[10]) {
  503. int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
  504. if (pkt_id == 4)
  505. rt->state = STATE_CONNECTING;
  506. }
  507. if (rt->state != STATE_CONNECTING)
  508. break;
  509. case STATE_CONNECTING:
  510. //extract a number from the result
  511. if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
  512. av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n");
  513. } else {
  514. rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
  515. }
  516. if (rt->is_input) {
  517. gen_play(s, rt);
  518. } else {
  519. gen_publish(s, rt);
  520. }
  521. rt->state = STATE_READY;
  522. break;
  523. }
  524. } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
  525. const uint8_t* ptr = pkt->data + 11;
  526. uint8_t tmpstr[256];
  527. for (i = 0; i < 2; i++) {
  528. t = ff_amf_tag_size(ptr, data_end);
  529. if (t < 0)
  530. return 1;
  531. ptr += t;
  532. }
  533. t = ff_amf_get_field_value(ptr, data_end,
  534. "level", tmpstr, sizeof(tmpstr));
  535. if (!t && !strcmp(tmpstr, "error")) {
  536. if (!ff_amf_get_field_value(ptr, data_end,
  537. "description", tmpstr, sizeof(tmpstr)))
  538. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
  539. return -1;
  540. }
  541. t = ff_amf_get_field_value(ptr, data_end,
  542. "code", tmpstr, sizeof(tmpstr));
  543. if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
  544. if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
  545. }
  546. break;
  547. }
  548. return 0;
  549. }
  550. /**
  551. * Interacts with the server by receiving and sending RTMP packets until
  552. * there is some significant data (media data or expected status notification).
  553. *
  554. * @param s reading context
  555. * @param for_header non-zero value tells function to work until it
  556. * gets notification from the server that playing has been started,
  557. * otherwise function will work until some media data is received (or
  558. * an error happens)
  559. * @return 0 for successful operation, negative value in case of error
  560. */
  561. static int get_packet(URLContext *s, int for_header)
  562. {
  563. RTMPContext *rt = s->priv_data;
  564. int ret;
  565. for (;;) {
  566. RTMPPacket rpkt;
  567. if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
  568. rt->chunk_size, rt->prev_pkt[0])) != 0) {
  569. if (ret > 0) {
  570. return AVERROR(EAGAIN);
  571. } else {
  572. return AVERROR(EIO);
  573. }
  574. }
  575. ret = rtmp_parse_result(s, rt, &rpkt);
  576. if (ret < 0) {//serious error in current packet
  577. ff_rtmp_packet_destroy(&rpkt);
  578. return -1;
  579. }
  580. if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
  581. ff_rtmp_packet_destroy(&rpkt);
  582. return 0;
  583. }
  584. if (!rpkt.data_size || !rt->is_input) {
  585. ff_rtmp_packet_destroy(&rpkt);
  586. continue;
  587. }
  588. if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
  589. (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
  590. uint8_t *p;
  591. uint32_t ts = rpkt.timestamp;
  592. // generate packet header and put data into buffer for FLV demuxer
  593. rt->flv_off = 0;
  594. rt->flv_size = rpkt.data_size + 15;
  595. rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
  596. bytestream_put_byte(&p, rpkt.type);
  597. bytestream_put_be24(&p, rpkt.data_size);
  598. bytestream_put_be24(&p, ts);
  599. bytestream_put_byte(&p, ts >> 24);
  600. bytestream_put_be24(&p, 0);
  601. bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
  602. bytestream_put_be32(&p, 0);
  603. ff_rtmp_packet_destroy(&rpkt);
  604. return 0;
  605. } else if (rpkt.type == RTMP_PT_METADATA) {
  606. // we got raw FLV data, make it available for FLV demuxer
  607. rt->flv_off = 0;
  608. rt->flv_size = rpkt.data_size;
  609. rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
  610. memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
  611. ff_rtmp_packet_destroy(&rpkt);
  612. return 0;
  613. }
  614. ff_rtmp_packet_destroy(&rpkt);
  615. }
  616. return 0;
  617. }
  618. static int rtmp_close(URLContext *h)
  619. {
  620. RTMPContext *rt = h->priv_data;
  621. if (!rt->is_input) {
  622. rt->flv_data = NULL;
  623. if (rt->out_pkt.data_size)
  624. ff_rtmp_packet_destroy(&rt->out_pkt);
  625. if (rt->state > STATE_FCPUBLISH)
  626. gen_fcunpublish_stream(h, rt);
  627. }
  628. if (rt->state > STATE_HANDSHAKED)
  629. gen_delete_stream(h, rt);
  630. av_freep(&rt->flv_data);
  631. url_close(rt->stream);
  632. av_free(rt);
  633. return 0;
  634. }
  635. /**
  636. * Opens RTMP connection and verifies that the stream can be played.
  637. *
  638. * URL syntax: rtmp://server[:port][/app][/playpath]
  639. * where 'app' is first one or two directories in the path
  640. * (e.g. /ondemand/, /flash/live/, etc.)
  641. * and 'playpath' is a file name (the rest of the path,
  642. * may be prefixed with "mp4:")
  643. */
  644. static int rtmp_open(URLContext *s, const char *uri, int flags)
  645. {
  646. RTMPContext *rt;
  647. char proto[8], hostname[256], path[1024], *fname;
  648. uint8_t buf[2048];
  649. int port;
  650. int ret;
  651. rt = av_mallocz(sizeof(RTMPContext));
  652. if (!rt)
  653. return AVERROR(ENOMEM);
  654. s->priv_data = rt;
  655. rt->is_input = !(flags & URL_WRONLY);
  656. url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
  657. path, sizeof(path), s->filename);
  658. if (port < 0)
  659. port = RTMP_DEFAULT_PORT;
  660. snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port);
  661. if (url_open(&rt->stream, buf, URL_RDWR) < 0) {
  662. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);
  663. goto fail;
  664. }
  665. rt->state = STATE_START;
  666. if (rtmp_handshake(s, rt))
  667. return -1;
  668. rt->chunk_size = 128;
  669. rt->state = STATE_HANDSHAKED;
  670. //extract "app" part from path
  671. if (!strncmp(path, "/ondemand/", 10)) {
  672. fname = path + 10;
  673. memcpy(rt->app, "ondemand", 9);
  674. } else {
  675. char *p = strchr(path + 1, '/');
  676. if (!p) {
  677. fname = path + 1;
  678. rt->app[0] = '\0';
  679. } else {
  680. char *c = strchr(p + 1, ':');
  681. fname = strchr(p + 1, '/');
  682. if (!fname || c < fname) {
  683. fname = p + 1;
  684. av_strlcpy(rt->app, path + 1, p - path);
  685. } else {
  686. fname++;
  687. av_strlcpy(rt->app, path + 1, fname - path - 1);
  688. }
  689. }
  690. }
  691. if (!strchr(fname, ':') &&
  692. (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
  693. !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
  694. memcpy(rt->playpath, "mp4:", 5);
  695. } else {
  696. rt->playpath[0] = 0;
  697. }
  698. strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
  699. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
  700. proto, path, rt->app, rt->playpath);
  701. gen_connect(s, rt, proto, hostname, port);
  702. do {
  703. ret = get_packet(s, 1);
  704. } while (ret == EAGAIN);
  705. if (ret < 0)
  706. goto fail;
  707. if (rt->is_input) {
  708. // generate FLV header for demuxer
  709. rt->flv_size = 13;
  710. rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
  711. rt->flv_off = 0;
  712. memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
  713. } else {
  714. rt->flv_size = 0;
  715. rt->flv_data = NULL;
  716. rt->flv_off = 0;
  717. }
  718. s->max_packet_size = url_get_max_packet_size(rt->stream);
  719. s->is_streamed = 1;
  720. return 0;
  721. fail:
  722. rtmp_close(s);
  723. return AVERROR(EIO);
  724. }
  725. static int rtmp_read(URLContext *s, uint8_t *buf, int size)
  726. {
  727. RTMPContext *rt = s->priv_data;
  728. int orig_size = size;
  729. int ret;
  730. while (size > 0) {
  731. int data_left = rt->flv_size - rt->flv_off;
  732. if (data_left >= size) {
  733. memcpy(buf, rt->flv_data + rt->flv_off, size);
  734. rt->flv_off += size;
  735. return orig_size;
  736. }
  737. if (data_left > 0) {
  738. memcpy(buf, rt->flv_data + rt->flv_off, data_left);
  739. buf += data_left;
  740. size -= data_left;
  741. rt->flv_off = rt->flv_size;
  742. }
  743. if ((ret = get_packet(s, 0)) < 0)
  744. return ret;
  745. }
  746. return orig_size;
  747. }
  748. static int rtmp_write(URLContext *h, uint8_t *buf, int size)
  749. {
  750. RTMPContext *rt = h->priv_data;
  751. int size_temp = size;
  752. int pktsize, pkttype;
  753. uint32_t ts;
  754. const uint8_t *buf_temp = buf;
  755. if (size < 11) {
  756. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
  757. return 0;
  758. }
  759. do {
  760. if (!rt->flv_off) {
  761. //skip flv header
  762. if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
  763. buf_temp += 9 + 4;
  764. size_temp -= 9 + 4;
  765. }
  766. pkttype = bytestream_get_byte(&buf_temp);
  767. pktsize = bytestream_get_be24(&buf_temp);
  768. ts = bytestream_get_be24(&buf_temp);
  769. ts |= bytestream_get_byte(&buf_temp) << 24;
  770. bytestream_get_be24(&buf_temp);
  771. size_temp -= 11;
  772. rt->flv_size = pktsize;
  773. //force 12bytes header
  774. if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
  775. pkttype == RTMP_PT_NOTIFY) {
  776. if (pkttype == RTMP_PT_NOTIFY)
  777. pktsize += 16;
  778. rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
  779. }
  780. //this can be a big packet, it's better to send it right here
  781. ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
  782. rt->out_pkt.extra = rt->main_channel_id;
  783. rt->flv_data = rt->out_pkt.data;
  784. if (pkttype == RTMP_PT_NOTIFY)
  785. ff_amf_write_string(&rt->flv_data, "@setDataFrame");
  786. }
  787. if (rt->flv_size - rt->flv_off > size_temp) {
  788. bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
  789. rt->flv_off += size_temp;
  790. } else {
  791. bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
  792. rt->flv_off += rt->flv_size - rt->flv_off;
  793. }
  794. if (rt->flv_off == rt->flv_size) {
  795. bytestream_get_be32(&buf_temp);
  796. ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
  797. ff_rtmp_packet_destroy(&rt->out_pkt);
  798. rt->flv_size = 0;
  799. rt->flv_off = 0;
  800. }
  801. } while (buf_temp - buf < size_temp);
  802. return size;
  803. }
  804. URLProtocol rtmp_protocol = {
  805. "rtmp",
  806. rtmp_open,
  807. rtmp_read,
  808. rtmp_write,
  809. NULL, /* seek */
  810. rtmp_close,
  811. };