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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_CONNECTING, ///< client connected to server successfully
  45. STATE_READY, ///< client has sent all needed commands and waits for server reply
  46. STATE_PLAYING, ///< client has started receiving multimedia data from server
  47. } ClientState;
  48. /** protocol handler context */
  49. typedef struct RTMPContext {
  50. URLContext* stream; ///< TCP stream used in interactions with RTMP server
  51. RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
  52. int chunk_size; ///< size of the chunks RTMP packets are divided into
  53. int is_input; ///< input/output flag
  54. char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
  55. ClientState state; ///< current state
  56. int main_channel_id; ///< an additional channel ID which is used for some invocations
  57. uint8_t* flv_data; ///< buffer with data for demuxer
  58. int flv_size; ///< current buffer size
  59. int flv_off; ///< number of bytes read from current buffer
  60. } RTMPContext;
  61. #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
  62. /** Client key used for digest signing */
  63. static const uint8_t rtmp_player_key[] = {
  64. 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
  65. 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
  66. 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
  67. 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
  68. 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
  69. };
  70. #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
  71. /** Key used for RTMP server digest signing */
  72. static const uint8_t rtmp_server_key[] = {
  73. 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
  74. 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
  75. 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
  76. 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
  77. 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
  78. 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
  79. };
  80. /**
  81. * Generates 'connect' call and sends it to the server.
  82. */
  83. static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
  84. const char *host, int port, const char *app)
  85. {
  86. RTMPPacket pkt;
  87. uint8_t ver[32], *p;
  88. char tcurl[512];
  89. ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
  90. p = pkt.data;
  91. snprintf(tcurl, sizeof(tcurl), "%s://%s:%d/%s", proto, host, port, app);
  92. ff_amf_write_string(&p, "connect");
  93. ff_amf_write_number(&p, 1.0);
  94. ff_amf_write_object_start(&p);
  95. ff_amf_write_field_name(&p, "app");
  96. ff_amf_write_string(&p, app);
  97. snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
  98. RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
  99. ff_amf_write_field_name(&p, "flashVer");
  100. ff_amf_write_string(&p, ver);
  101. ff_amf_write_field_name(&p, "tcUrl");
  102. ff_amf_write_string(&p, tcurl);
  103. ff_amf_write_field_name(&p, "fpad");
  104. ff_amf_write_bool(&p, 0);
  105. ff_amf_write_field_name(&p, "capabilities");
  106. ff_amf_write_number(&p, 15.0);
  107. ff_amf_write_field_name(&p, "audioCodecs");
  108. ff_amf_write_number(&p, 1639.0);
  109. ff_amf_write_field_name(&p, "videoCodecs");
  110. ff_amf_write_number(&p, 252.0);
  111. ff_amf_write_field_name(&p, "videoFunction");
  112. ff_amf_write_number(&p, 1.0);
  113. ff_amf_write_object_end(&p);
  114. pkt.data_size = p - pkt.data;
  115. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  116. }
  117. /**
  118. * Generates 'createStream' call and sends it to the server. It should make
  119. * the server allocate some channel for media streams.
  120. */
  121. static void gen_create_stream(URLContext *s, RTMPContext *rt)
  122. {
  123. RTMPPacket pkt;
  124. uint8_t *p;
  125. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n");
  126. ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 25);
  127. p = pkt.data;
  128. ff_amf_write_string(&p, "createStream");
  129. ff_amf_write_number(&p, 3.0);
  130. ff_amf_write_null(&p);
  131. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  132. ff_rtmp_packet_destroy(&pkt);
  133. }
  134. /**
  135. * Generates 'play' call and sends it to the server, then pings the server
  136. * to start actual playing.
  137. */
  138. static void gen_play(URLContext *s, RTMPContext *rt)
  139. {
  140. RTMPPacket pkt;
  141. uint8_t *p;
  142. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
  143. ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
  144. 20 + strlen(rt->playpath));
  145. pkt.extra = rt->main_channel_id;
  146. p = pkt.data;
  147. ff_amf_write_string(&p, "play");
  148. ff_amf_write_number(&p, 0.0);
  149. ff_amf_write_null(&p);
  150. ff_amf_write_string(&p, rt->playpath);
  151. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  152. ff_rtmp_packet_destroy(&pkt);
  153. // set client buffer time disguised in ping packet
  154. ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
  155. p = pkt.data;
  156. bytestream_put_be16(&p, 3);
  157. bytestream_put_be32(&p, 1);
  158. bytestream_put_be32(&p, 256); //TODO: what is a good value here?
  159. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  160. ff_rtmp_packet_destroy(&pkt);
  161. }
  162. /**
  163. * Generates ping reply and sends it to the server.
  164. */
  165. static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
  166. {
  167. RTMPPacket pkt;
  168. uint8_t *p;
  169. ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
  170. p = pkt.data;
  171. bytestream_put_be16(&p, 7);
  172. bytestream_put_be32(&p, AV_RB32(ppkt->data+2) + 1);
  173. ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
  174. ff_rtmp_packet_destroy(&pkt);
  175. }
  176. //TODO: Move HMAC code somewhere. Eventually.
  177. #define HMAC_IPAD_VAL 0x36
  178. #define HMAC_OPAD_VAL 0x5C
  179. /**
  180. * Calculates HMAC-SHA2 digest for RTMP handshake packets.
  181. *
  182. * @param src input buffer
  183. * @param len input buffer length (should be 1536)
  184. * @param gap offset in buffer where 32 bytes should not be taken into account
  185. * when calculating digest (since it will be used to store that digest)
  186. * @param key digest key
  187. * @param keylen digest key length
  188. * @param dst buffer where calculated digest will be stored (32 bytes)
  189. */
  190. static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
  191. const uint8_t *key, int keylen, uint8_t *dst)
  192. {
  193. struct AVSHA *sha;
  194. uint8_t hmac_buf[64+32] = {0};
  195. int i;
  196. sha = av_mallocz(av_sha_size);
  197. if (keylen < 64) {
  198. memcpy(hmac_buf, key, keylen);
  199. } else {
  200. av_sha_init(sha, 256);
  201. av_sha_update(sha,key, keylen);
  202. av_sha_final(sha, hmac_buf);
  203. }
  204. for (i = 0; i < 64; i++)
  205. hmac_buf[i] ^= HMAC_IPAD_VAL;
  206. av_sha_init(sha, 256);
  207. av_sha_update(sha, hmac_buf, 64);
  208. if (gap <= 0) {
  209. av_sha_update(sha, src, len);
  210. } else { //skip 32 bytes used for storing digest
  211. av_sha_update(sha, src, gap);
  212. av_sha_update(sha, src + gap + 32, len - gap - 32);
  213. }
  214. av_sha_final(sha, hmac_buf + 64);
  215. for (i = 0; i < 64; i++)
  216. hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
  217. av_sha_init(sha, 256);
  218. av_sha_update(sha, hmac_buf, 64+32);
  219. av_sha_final(sha, dst);
  220. av_free(sha);
  221. }
  222. /**
  223. * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest
  224. * will be stored) into that packet.
  225. *
  226. * @param buf handshake data (1536 bytes)
  227. * @return offset to the digest inside input data
  228. */
  229. static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
  230. {
  231. int i, digest_pos = 0;
  232. for (i = 8; i < 12; i++)
  233. digest_pos += buf[i];
  234. digest_pos = (digest_pos % 728) + 12;
  235. rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
  236. rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
  237. buf + digest_pos);
  238. return digest_pos;
  239. }
  240. /**
  241. * Verifies that the received server response has the expected digest value.
  242. *
  243. * @param buf handshake data received from the server (1536 bytes)
  244. * @param off position to search digest offset from
  245. * @return 0 if digest is valid, digest position otherwise
  246. */
  247. static int rtmp_validate_digest(uint8_t *buf, int off)
  248. {
  249. int i, digest_pos = 0;
  250. uint8_t digest[32];
  251. for (i = 0; i < 4; i++)
  252. digest_pos += buf[i + off];
  253. digest_pos = (digest_pos % 728) + off + 4;
  254. rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
  255. rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
  256. digest);
  257. if (!memcmp(digest, buf + digest_pos, 32))
  258. return digest_pos;
  259. return 0;
  260. }
  261. /**
  262. * Performs handshake with the server by means of exchanging pseudorandom data
  263. * signed with HMAC-SHA2 digest.
  264. *
  265. * @return 0 if handshake succeeds, negative value otherwise
  266. */
  267. static int rtmp_handshake(URLContext *s, RTMPContext *rt)
  268. {
  269. AVLFG rnd;
  270. uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
  271. 3, // unencrypted data
  272. 0, 0, 0, 0, // client uptime
  273. RTMP_CLIENT_VER1,
  274. RTMP_CLIENT_VER2,
  275. RTMP_CLIENT_VER3,
  276. RTMP_CLIENT_VER4,
  277. };
  278. uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
  279. uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
  280. int i;
  281. int server_pos, client_pos;
  282. uint8_t digest[32];
  283. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n");
  284. av_lfg_init(&rnd, 0xDEADC0DE);
  285. // generate handshake packet - 1536 bytes of pseudorandom data
  286. for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
  287. tosend[i] = av_lfg_get(&rnd) >> 24;
  288. client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
  289. url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
  290. i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
  291. if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
  292. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
  293. return -1;
  294. }
  295. i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
  296. if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
  297. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
  298. return -1;
  299. }
  300. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
  301. serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
  302. server_pos = rtmp_validate_digest(serverdata + 1, 772);
  303. if (!server_pos) {
  304. server_pos = rtmp_validate_digest(serverdata + 1, 8);
  305. if (!server_pos) {
  306. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n");
  307. return -1;
  308. }
  309. }
  310. rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
  311. rtmp_server_key, sizeof(rtmp_server_key),
  312. digest);
  313. rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
  314. digest, 32,
  315. digest);
  316. if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
  317. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n");
  318. return -1;
  319. }
  320. for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
  321. tosend[i] = av_lfg_get(&rnd) >> 24;
  322. rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
  323. rtmp_player_key, sizeof(rtmp_player_key),
  324. digest);
  325. rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
  326. digest, 32,
  327. tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
  328. // write reply back to the server
  329. url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
  330. return 0;
  331. }
  332. /**
  333. * Parses received packet and may perform some action depending on
  334. * the packet contents.
  335. * @return 0 for no errors, negative values for serious errors which prevent
  336. * further communications, positive values for uncritical errors
  337. */
  338. static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
  339. {
  340. int i, t;
  341. const uint8_t *data_end = pkt->data + pkt->data_size;
  342. switch (pkt->type) {
  343. case RTMP_PT_CHUNK_SIZE:
  344. if (pkt->data_size != 4) {
  345. av_log(LOG_CONTEXT, AV_LOG_ERROR,
  346. "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
  347. return -1;
  348. }
  349. rt->chunk_size = AV_RB32(pkt->data);
  350. if (rt->chunk_size <= 0) {
  351. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
  352. return -1;
  353. }
  354. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
  355. break;
  356. case RTMP_PT_PING:
  357. t = AV_RB16(pkt->data);
  358. if (t == 6)
  359. gen_pong(s, rt, pkt);
  360. break;
  361. case RTMP_PT_INVOKE:
  362. //TODO: check for the messages sent for wrong state?
  363. if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
  364. uint8_t tmpstr[256];
  365. if (!ff_amf_get_field_value(pkt->data + 9, data_end,
  366. "description", tmpstr, sizeof(tmpstr)))
  367. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
  368. return -1;
  369. } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
  370. switch (rt->state) {
  371. case STATE_HANDSHAKED:
  372. gen_create_stream(s, rt);
  373. rt->state = STATE_CONNECTING;
  374. break;
  375. case STATE_CONNECTING:
  376. //extract a number from the result
  377. if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
  378. av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n");
  379. } else {
  380. rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
  381. }
  382. gen_play(s, rt);
  383. rt->state = STATE_READY;
  384. break;
  385. }
  386. } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
  387. const uint8_t* ptr = pkt->data + 11;
  388. uint8_t tmpstr[256];
  389. int t;
  390. for (i = 0; i < 2; i++) {
  391. t = ff_amf_tag_size(ptr, data_end);
  392. if (t < 0)
  393. return 1;
  394. ptr += t;
  395. }
  396. t = ff_amf_get_field_value(ptr, data_end,
  397. "level", tmpstr, sizeof(tmpstr));
  398. if (!t && !strcmp(tmpstr, "error")) {
  399. if (!ff_amf_get_field_value(ptr, data_end,
  400. "description", tmpstr, sizeof(tmpstr)))
  401. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
  402. return -1;
  403. }
  404. t = ff_amf_get_field_value(ptr, data_end,
  405. "code", tmpstr, sizeof(tmpstr));
  406. if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) {
  407. rt->state = STATE_PLAYING;
  408. return 0;
  409. }
  410. }
  411. break;
  412. }
  413. return 0;
  414. }
  415. /**
  416. * Interacts with the server by receiving and sending RTMP packets until
  417. * there is some significant data (media data or expected status notification).
  418. *
  419. * @param s reading context
  420. * @param for_header non-zero value tells function to work until it
  421. * gets notification from the server that playing has been started,
  422. * otherwise function will work until some media data is received (or
  423. * an error happens)
  424. * @return 0 for successful operation, negative value in case of error
  425. */
  426. static int get_packet(URLContext *s, int for_header)
  427. {
  428. RTMPContext *rt = s->priv_data;
  429. int ret;
  430. for(;;) {
  431. RTMPPacket rpkt;
  432. if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
  433. rt->chunk_size, rt->prev_pkt[0])) != 0) {
  434. if (ret > 0) {
  435. return AVERROR(EAGAIN);
  436. } else {
  437. return AVERROR(EIO);
  438. }
  439. }
  440. ret = rtmp_parse_result(s, rt, &rpkt);
  441. if (ret < 0) {//serious error in current packet
  442. ff_rtmp_packet_destroy(&rpkt);
  443. return -1;
  444. }
  445. if (for_header && rt->state == STATE_PLAYING) {
  446. ff_rtmp_packet_destroy(&rpkt);
  447. return 0;
  448. }
  449. if (!rpkt.data_size) {
  450. ff_rtmp_packet_destroy(&rpkt);
  451. continue;
  452. }
  453. if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
  454. (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
  455. uint8_t *p;
  456. uint32_t ts = rpkt.timestamp;
  457. // generate packet header and put data into buffer for FLV demuxer
  458. rt->flv_off = 0;
  459. rt->flv_size = rpkt.data_size + 15;
  460. rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
  461. bytestream_put_byte(&p, rpkt.type);
  462. bytestream_put_be24(&p, rpkt.data_size);
  463. bytestream_put_be24(&p, ts);
  464. bytestream_put_byte(&p, ts >> 24);
  465. bytestream_put_be24(&p, 0);
  466. bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
  467. bytestream_put_be32(&p, 0);
  468. ff_rtmp_packet_destroy(&rpkt);
  469. return 0;
  470. } else if (rpkt.type == RTMP_PT_METADATA) {
  471. // we got raw FLV data, make it available for FLV demuxer
  472. rt->flv_off = 0;
  473. rt->flv_size = rpkt.data_size;
  474. rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
  475. memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
  476. ff_rtmp_packet_destroy(&rpkt);
  477. return 0;
  478. }
  479. ff_rtmp_packet_destroy(&rpkt);
  480. }
  481. return 0;
  482. }
  483. static int rtmp_close(URLContext *h)
  484. {
  485. RTMPContext *rt = h->priv_data;
  486. av_freep(&rt->flv_data);
  487. url_close(rt->stream);
  488. av_free(rt);
  489. return 0;
  490. }
  491. /**
  492. * Opens RTMP connection and verifies that the stream can be played.
  493. *
  494. * URL syntax: rtmp://server[:port][/app][/playpath]
  495. * where 'app' is first one or two directories in the path
  496. * (e.g. /ondemand/, /flash/live/, etc.)
  497. * and 'playpath' is a file name (the rest of the path,
  498. * may be prefixed with "mp4:")
  499. */
  500. static int rtmp_open(URLContext *s, const char *uri, int flags)
  501. {
  502. RTMPContext *rt;
  503. char proto[8], hostname[256], path[1024], app[128], *fname;
  504. uint8_t buf[2048];
  505. int port;
  506. int ret;
  507. rt = av_mallocz(sizeof(RTMPContext));
  508. if (!rt)
  509. return AVERROR(ENOMEM);
  510. s->priv_data = rt;
  511. rt->is_input = !(flags & URL_WRONLY);
  512. url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
  513. path, sizeof(path), s->filename);
  514. if (port < 0)
  515. port = RTMP_DEFAULT_PORT;
  516. snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port);
  517. if (url_open(&rt->stream, buf, URL_RDWR) < 0) {
  518. av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);
  519. goto fail;
  520. }
  521. if (!rt->is_input) {
  522. av_log(LOG_CONTEXT, AV_LOG_ERROR, "RTMP output is not supported yet.\n");
  523. goto fail;
  524. } else {
  525. rt->state = STATE_START;
  526. if (rtmp_handshake(s, rt))
  527. return -1;
  528. rt->chunk_size = 128;
  529. rt->state = STATE_HANDSHAKED;
  530. //extract "app" part from path
  531. if (!strncmp(path, "/ondemand/", 10)) {
  532. fname = path + 10;
  533. memcpy(app, "ondemand", 9);
  534. } else {
  535. char *p = strchr(path + 1, '/');
  536. if (!p) {
  537. fname = path + 1;
  538. app[0] = '\0';
  539. } else {
  540. char *c = strchr(p + 1, ':');
  541. fname = strchr(p + 1, '/');
  542. if (!fname || c < fname) {
  543. fname = p + 1;
  544. av_strlcpy(app, path + 1, p - path);
  545. } else {
  546. fname++;
  547. av_strlcpy(app, path + 1, fname - path - 1);
  548. }
  549. }
  550. }
  551. if (!strchr(fname, ':') &&
  552. (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
  553. !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
  554. memcpy(rt->playpath, "mp4:", 5);
  555. } else {
  556. rt->playpath[0] = 0;
  557. }
  558. strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
  559. av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
  560. proto, path, app, rt->playpath);
  561. gen_connect(s, rt, proto, hostname, port, app);
  562. do {
  563. ret = get_packet(s, 1);
  564. } while (ret == EAGAIN);
  565. if (ret < 0)
  566. goto fail;
  567. // generate FLV header for demuxer
  568. rt->flv_size = 13;
  569. rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
  570. rt->flv_off = 0;
  571. memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
  572. }
  573. s->max_packet_size = url_get_max_packet_size(rt->stream);
  574. s->is_streamed = 1;
  575. return 0;
  576. fail:
  577. rtmp_close(s);
  578. return AVERROR(EIO);
  579. }
  580. static int rtmp_read(URLContext *s, uint8_t *buf, int size)
  581. {
  582. RTMPContext *rt = s->priv_data;
  583. int orig_size = size;
  584. int ret;
  585. while (size > 0) {
  586. int data_left = rt->flv_size - rt->flv_off;
  587. if (data_left >= size) {
  588. memcpy(buf, rt->flv_data + rt->flv_off, size);
  589. rt->flv_off += size;
  590. return orig_size;
  591. }
  592. if (data_left > 0) {
  593. memcpy(buf, rt->flv_data + rt->flv_off, data_left);
  594. buf += data_left;
  595. size -= data_left;
  596. rt->flv_off = rt->flv_size;
  597. }
  598. if ((ret = get_packet(s, 0)) < 0)
  599. return ret;
  600. }
  601. return orig_size;
  602. }
  603. static int rtmp_write(URLContext *h, uint8_t *buf, int size)
  604. {
  605. return 0;
  606. }
  607. URLProtocol rtmp_protocol = {
  608. "rtmp",
  609. rtmp_open,
  610. rtmp_read,
  611. rtmp_write,
  612. NULL, /* seek */
  613. rtmp_close,
  614. };