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