2 * RTMP network protocol
3 * Copyright (c) 2009 Kostya Shishkov
5 * This file is part of Libav.
7 * Libav 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.
12 * Libav 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.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27 #include "libavcodec/bytestream.h"
28 #include "libavutil/avstring.h"
29 #include "libavutil/intfloat_readwrite.h"
30 #include "libavutil/lfg.h"
31 #include "libavutil/sha.h"
44 /** RTMP protocol handler state */
46 STATE_START, ///< client has not done anything yet
47 STATE_HANDSHAKED, ///< client has performed handshake
48 STATE_RELEASING, ///< client releasing stream before publish it (for output)
49 STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
50 STATE_CONNECTING, ///< client connected to server successfully
51 STATE_READY, ///< client has sent all needed commands and waits for server reply
52 STATE_PLAYING, ///< client has started receiving multimedia data from server
53 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
54 STATE_STOPPED, ///< the broadcast has been stopped
57 /** protocol handler context */
58 typedef struct RTMPContext {
59 URLContext* stream; ///< TCP stream used in interactions with RTMP server
60 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
61 int chunk_size; ///< size of the chunks RTMP packets are divided into
62 int is_input; ///< input/output flag
63 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
64 char app[128]; ///< application
65 ClientState state; ///< current state
66 int main_channel_id; ///< an additional channel ID which is used for some invocations
67 uint8_t* flv_data; ///< buffer with data for demuxer
68 int flv_size; ///< current buffer size
69 int flv_off; ///< number of bytes read from current buffer
70 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
71 uint32_t client_report_size; ///< number of bytes after which client should report to server
72 uint32_t bytes_read; ///< number of bytes read from server
73 uint32_t last_bytes_read; ///< number of bytes read last reported to server
74 int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
77 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
78 /** Client key used for digest signing */
79 static const uint8_t rtmp_player_key[] = {
80 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
81 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
83 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
84 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
85 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
88 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
89 /** Key used for RTMP server digest signing */
90 static const uint8_t rtmp_server_key[] = {
91 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
92 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
93 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
95 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
96 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
97 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
101 * Generate 'connect' call and send it to the server.
103 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
104 const char *host, int port)
110 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
113 ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
114 ff_amf_write_string(&p, "connect");
115 ff_amf_write_number(&p, 1.0);
116 ff_amf_write_object_start(&p);
117 ff_amf_write_field_name(&p, "app");
118 ff_amf_write_string(&p, rt->app);
121 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
122 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
124 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
125 ff_amf_write_field_name(&p, "type");
126 ff_amf_write_string(&p, "nonprivate");
128 ff_amf_write_field_name(&p, "flashVer");
129 ff_amf_write_string(&p, ver);
130 ff_amf_write_field_name(&p, "tcUrl");
131 ff_amf_write_string(&p, tcurl);
133 ff_amf_write_field_name(&p, "fpad");
134 ff_amf_write_bool(&p, 0);
135 ff_amf_write_field_name(&p, "capabilities");
136 ff_amf_write_number(&p, 15.0);
137 ff_amf_write_field_name(&p, "audioCodecs");
138 ff_amf_write_number(&p, 1639.0);
139 ff_amf_write_field_name(&p, "videoCodecs");
140 ff_amf_write_number(&p, 252.0);
141 ff_amf_write_field_name(&p, "videoFunction");
142 ff_amf_write_number(&p, 1.0);
144 ff_amf_write_object_end(&p);
146 pkt.data_size = p - pkt.data;
148 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
149 ff_rtmp_packet_destroy(&pkt);
153 * Generate 'releaseStream' call and send it to the server. It should make
154 * the server release some channel for media streams.
156 static void gen_release_stream(URLContext *s, RTMPContext *rt)
161 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
162 29 + strlen(rt->playpath));
164 av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
166 ff_amf_write_string(&p, "releaseStream");
167 ff_amf_write_number(&p, 2.0);
168 ff_amf_write_null(&p);
169 ff_amf_write_string(&p, rt->playpath);
171 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
172 ff_rtmp_packet_destroy(&pkt);
176 * Generate 'FCPublish' call and send it to the server. It should make
177 * the server preapare for receiving media streams.
179 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
184 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
185 25 + strlen(rt->playpath));
187 av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
189 ff_amf_write_string(&p, "FCPublish");
190 ff_amf_write_number(&p, 3.0);
191 ff_amf_write_null(&p);
192 ff_amf_write_string(&p, rt->playpath);
194 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
195 ff_rtmp_packet_destroy(&pkt);
199 * Generate 'FCUnpublish' call and send it to the server. It should make
200 * the server destroy stream.
202 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
207 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
208 27 + strlen(rt->playpath));
210 av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
212 ff_amf_write_string(&p, "FCUnpublish");
213 ff_amf_write_number(&p, 5.0);
214 ff_amf_write_null(&p);
215 ff_amf_write_string(&p, rt->playpath);
217 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
218 ff_rtmp_packet_destroy(&pkt);
222 * Generate 'createStream' call and send it to the server. It should make
223 * the server allocate some channel for media streams.
225 static void gen_create_stream(URLContext *s, RTMPContext *rt)
230 av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
231 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
234 ff_amf_write_string(&p, "createStream");
235 ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
236 ff_amf_write_null(&p);
238 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
239 ff_rtmp_packet_destroy(&pkt);
244 * Generate 'deleteStream' call and send it to the server. It should make
245 * the server remove some channel for media streams.
247 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
252 av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
253 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
256 ff_amf_write_string(&p, "deleteStream");
257 ff_amf_write_number(&p, 0.0);
258 ff_amf_write_null(&p);
259 ff_amf_write_number(&p, rt->main_channel_id);
261 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
262 ff_rtmp_packet_destroy(&pkt);
266 * Generate 'play' call and send it to the server, then ping the server
267 * to start actual playing.
269 static void gen_play(URLContext *s, RTMPContext *rt)
274 av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
275 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
276 20 + strlen(rt->playpath));
277 pkt.extra = rt->main_channel_id;
280 ff_amf_write_string(&p, "play");
281 ff_amf_write_number(&p, 0.0);
282 ff_amf_write_null(&p);
283 ff_amf_write_string(&p, rt->playpath);
285 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
286 ff_rtmp_packet_destroy(&pkt);
288 // set client buffer time disguised in ping packet
289 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
292 bytestream_put_be16(&p, 3);
293 bytestream_put_be32(&p, 1);
294 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
296 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
297 ff_rtmp_packet_destroy(&pkt);
301 * Generate 'publish' call and send it to the server.
303 static void gen_publish(URLContext *s, RTMPContext *rt)
308 av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
309 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
310 30 + strlen(rt->playpath));
311 pkt.extra = rt->main_channel_id;
314 ff_amf_write_string(&p, "publish");
315 ff_amf_write_number(&p, 0.0);
316 ff_amf_write_null(&p);
317 ff_amf_write_string(&p, rt->playpath);
318 ff_amf_write_string(&p, "live");
320 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
321 ff_rtmp_packet_destroy(&pkt);
325 * Generate ping reply and send it to the server.
327 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
332 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
334 bytestream_put_be16(&p, 7);
335 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
336 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
337 ff_rtmp_packet_destroy(&pkt);
341 * Generate report on bytes read so far and send it to the server.
343 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
348 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
350 bytestream_put_be32(&p, rt->bytes_read);
351 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
352 ff_rtmp_packet_destroy(&pkt);
355 //TODO: Move HMAC code somewhere. Eventually.
356 #define HMAC_IPAD_VAL 0x36
357 #define HMAC_OPAD_VAL 0x5C
360 * Calculate HMAC-SHA2 digest for RTMP handshake packets.
362 * @param src input buffer
363 * @param len input buffer length (should be 1536)
364 * @param gap offset in buffer where 32 bytes should not be taken into account
365 * when calculating digest (since it will be used to store that digest)
366 * @param key digest key
367 * @param keylen digest key length
368 * @param dst buffer where calculated digest will be stored (32 bytes)
370 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
371 const uint8_t *key, int keylen, uint8_t *dst)
374 uint8_t hmac_buf[64+32] = {0};
377 sha = av_mallocz(av_sha_size);
380 memcpy(hmac_buf, key, keylen);
382 av_sha_init(sha, 256);
383 av_sha_update(sha,key, keylen);
384 av_sha_final(sha, hmac_buf);
386 for (i = 0; i < 64; i++)
387 hmac_buf[i] ^= HMAC_IPAD_VAL;
389 av_sha_init(sha, 256);
390 av_sha_update(sha, hmac_buf, 64);
392 av_sha_update(sha, src, len);
393 } else { //skip 32 bytes used for storing digest
394 av_sha_update(sha, src, gap);
395 av_sha_update(sha, src + gap + 32, len - gap - 32);
397 av_sha_final(sha, hmac_buf + 64);
399 for (i = 0; i < 64; i++)
400 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
401 av_sha_init(sha, 256);
402 av_sha_update(sha, hmac_buf, 64+32);
403 av_sha_final(sha, dst);
409 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
410 * will be stored) into that packet.
412 * @param buf handshake data (1536 bytes)
413 * @return offset to the digest inside input data
415 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
417 int i, digest_pos = 0;
419 for (i = 8; i < 12; i++)
420 digest_pos += buf[i];
421 digest_pos = (digest_pos % 728) + 12;
423 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
424 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
430 * Verify that the received server response has the expected digest value.
432 * @param buf handshake data received from the server (1536 bytes)
433 * @param off position to search digest offset from
434 * @return 0 if digest is valid, digest position otherwise
436 static int rtmp_validate_digest(uint8_t *buf, int off)
438 int i, digest_pos = 0;
441 for (i = 0; i < 4; i++)
442 digest_pos += buf[i + off];
443 digest_pos = (digest_pos % 728) + off + 4;
445 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
446 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
448 if (!memcmp(digest, buf + digest_pos, 32))
454 * Perform handshake with the server by means of exchanging pseudorandom data
455 * signed with HMAC-SHA2 digest.
457 * @return 0 if handshake succeeds, negative value otherwise
459 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
462 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
463 3, // unencrypted data
464 0, 0, 0, 0, // client uptime
470 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
471 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
473 int server_pos, client_pos;
476 av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
478 av_lfg_init(&rnd, 0xDEADC0DE);
479 // generate handshake packet - 1536 bytes of pseudorandom data
480 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
481 tosend[i] = av_lfg_get(&rnd) >> 24;
482 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
484 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
485 i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
486 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
487 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
490 i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
491 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
492 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
496 av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
497 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
499 if (rt->is_input && serverdata[5] >= 3) {
500 server_pos = rtmp_validate_digest(serverdata + 1, 772);
502 server_pos = rtmp_validate_digest(serverdata + 1, 8);
504 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
509 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
510 rtmp_server_key, sizeof(rtmp_server_key),
512 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
515 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
516 av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
520 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
521 tosend[i] = av_lfg_get(&rnd) >> 24;
522 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
523 rtmp_player_key, sizeof(rtmp_player_key),
525 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
527 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
529 // write reply back to the server
530 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
532 ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
539 * Parse received packet and possibly perform some action depending on
540 * the packet contents.
541 * @return 0 for no errors, negative values for serious errors which prevent
542 * further communications, positive values for uncritical errors
544 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
547 const uint8_t *data_end = pkt->data + pkt->data_size;
550 ff_rtmp_packet_dump(s, pkt);
554 case RTMP_PT_CHUNK_SIZE:
555 if (pkt->data_size != 4) {
556 av_log(s, AV_LOG_ERROR,
557 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
561 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
562 rt->chunk_size = AV_RB32(pkt->data);
563 if (rt->chunk_size <= 0) {
564 av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
567 av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
570 t = AV_RB16(pkt->data);
572 gen_pong(s, rt, pkt);
574 case RTMP_PT_CLIENT_BW:
575 if (pkt->data_size < 4) {
576 av_log(s, AV_LOG_ERROR,
577 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
581 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
582 rt->client_report_size = AV_RB32(pkt->data) >> 1;
585 //TODO: check for the messages sent for wrong state?
586 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
589 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
590 "description", tmpstr, sizeof(tmpstr)))
591 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
593 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
595 case STATE_HANDSHAKED:
597 gen_release_stream(s, rt);
598 gen_fcpublish_stream(s, rt);
599 rt->state = STATE_RELEASING;
601 rt->state = STATE_CONNECTING;
603 gen_create_stream(s, rt);
605 case STATE_FCPUBLISH:
606 rt->state = STATE_CONNECTING;
608 case STATE_RELEASING:
609 rt->state = STATE_FCPUBLISH;
610 /* hack for Wowza Media Server, it does not send result for
611 * releaseStream and FCPublish calls */
612 if (!pkt->data[10]) {
613 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
615 rt->state = STATE_CONNECTING;
617 if (rt->state != STATE_CONNECTING)
619 case STATE_CONNECTING:
620 //extract a number from the result
621 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
622 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
624 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
631 rt->state = STATE_READY;
634 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
635 const uint8_t* ptr = pkt->data + 11;
638 for (i = 0; i < 2; i++) {
639 t = ff_amf_tag_size(ptr, data_end);
644 t = ff_amf_get_field_value(ptr, data_end,
645 "level", tmpstr, sizeof(tmpstr));
646 if (!t && !strcmp(tmpstr, "error")) {
647 if (!ff_amf_get_field_value(ptr, data_end,
648 "description", tmpstr, sizeof(tmpstr)))
649 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
652 t = ff_amf_get_field_value(ptr, data_end,
653 "code", tmpstr, sizeof(tmpstr));
654 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
655 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
656 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
657 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
665 * Interact with the server by receiving and sending RTMP packets until
666 * there is some significant data (media data or expected status notification).
668 * @param s reading context
669 * @param for_header non-zero value tells function to work until it
670 * gets notification from the server that playing has been started,
671 * otherwise function will work until some media data is received (or
673 * @return 0 for successful operation, negative value in case of error
675 static int get_packet(URLContext *s, int for_header)
677 RTMPContext *rt = s->priv_data;
682 uint32_t ts, cts, pts=0;
684 if (rt->state == STATE_STOPPED)
688 RTMPPacket rpkt = { 0 };
689 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
690 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
692 return AVERROR(EAGAIN);
697 rt->bytes_read += ret;
698 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
699 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
700 gen_bytes_read(s, rt, rpkt.timestamp + 1);
701 rt->last_bytes_read = rt->bytes_read;
704 ret = rtmp_parse_result(s, rt, &rpkt);
705 if (ret < 0) {//serious error in current packet
706 ff_rtmp_packet_destroy(&rpkt);
709 if (rt->state == STATE_STOPPED) {
710 ff_rtmp_packet_destroy(&rpkt);
713 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
714 ff_rtmp_packet_destroy(&rpkt);
717 if (!rpkt.data_size || !rt->is_input) {
718 ff_rtmp_packet_destroy(&rpkt);
721 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
722 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
725 // generate packet header and put data into buffer for FLV demuxer
727 rt->flv_size = rpkt.data_size + 15;
728 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
729 bytestream_put_byte(&p, rpkt.type);
730 bytestream_put_be24(&p, rpkt.data_size);
731 bytestream_put_be24(&p, ts);
732 bytestream_put_byte(&p, ts >> 24);
733 bytestream_put_be24(&p, 0);
734 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
735 bytestream_put_be32(&p, 0);
736 ff_rtmp_packet_destroy(&rpkt);
738 } else if (rpkt.type == RTMP_PT_METADATA) {
739 // we got raw FLV data, make it available for FLV demuxer
741 rt->flv_size = rpkt.data_size;
742 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
743 /* rewrite timestamps */
746 while (next - rpkt.data < rpkt.data_size - 11) {
748 data_size = bytestream_get_be24(&next);
750 cts = bytestream_get_be24(&next);
751 cts |= bytestream_get_byte(&next) << 24;
756 bytestream_put_be24(&p, ts);
757 bytestream_put_byte(&p, ts >> 24);
758 next += data_size + 3 + 4;
760 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
761 ff_rtmp_packet_destroy(&rpkt);
764 ff_rtmp_packet_destroy(&rpkt);
768 static int rtmp_close(URLContext *h)
770 RTMPContext *rt = h->priv_data;
774 if (rt->out_pkt.data_size)
775 ff_rtmp_packet_destroy(&rt->out_pkt);
776 if (rt->state > STATE_FCPUBLISH)
777 gen_fcunpublish_stream(h, rt);
779 if (rt->state > STATE_HANDSHAKED)
780 gen_delete_stream(h, rt);
782 av_freep(&rt->flv_data);
783 ffurl_close(rt->stream);
789 * Open RTMP connection and verify that the stream can be played.
791 * URL syntax: rtmp://server[:port][/app][/playpath]
792 * where 'app' is first one or two directories in the path
793 * (e.g. /ondemand/, /flash/live/, etc.)
794 * and 'playpath' is a file name (the rest of the path,
795 * may be prefixed with "mp4:")
797 static int rtmp_open(URLContext *s, const char *uri, int flags)
800 char proto[8], hostname[256], path[1024], *fname;
805 rt = av_mallocz(sizeof(RTMPContext));
807 return AVERROR(ENOMEM);
809 rt->is_input = !(flags & AVIO_FLAG_WRITE);
811 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
812 path, sizeof(path), s->filename);
815 port = RTMP_DEFAULT_PORT;
816 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
818 if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE) < 0) {
819 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
823 rt->state = STATE_START;
824 if (rtmp_handshake(s, rt))
827 rt->chunk_size = 128;
828 rt->state = STATE_HANDSHAKED;
829 //extract "app" part from path
830 if (!strncmp(path, "/ondemand/", 10)) {
832 memcpy(rt->app, "ondemand", 9);
834 char *p = strchr(path + 1, '/');
839 char *c = strchr(p + 1, ':');
840 fname = strchr(p + 1, '/');
841 if (!fname || c < fname) {
843 av_strlcpy(rt->app, path + 1, p - path);
846 av_strlcpy(rt->app, path + 1, fname - path - 1);
850 if (!strchr(fname, ':') &&
851 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
852 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
853 memcpy(rt->playpath, "mp4:", 5);
857 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
859 rt->client_report_size = 1048576;
861 rt->last_bytes_read = 0;
863 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
864 proto, path, rt->app, rt->playpath);
865 gen_connect(s, rt, proto, hostname, port);
868 ret = get_packet(s, 1);
869 } while (ret == EAGAIN);
874 // generate FLV header for demuxer
876 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
878 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
885 s->max_packet_size = rt->stream->max_packet_size;
894 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
896 RTMPContext *rt = s->priv_data;
897 int orig_size = size;
901 int data_left = rt->flv_size - rt->flv_off;
903 if (data_left >= size) {
904 memcpy(buf, rt->flv_data + rt->flv_off, size);
909 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
912 rt->flv_off = rt->flv_size;
915 if ((ret = get_packet(s, 0)) < 0)
921 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
923 RTMPContext *rt = s->priv_data;
924 int size_temp = size;
925 int pktsize, pkttype;
927 const uint8_t *buf_temp = buf;
929 if (rt->skip_bytes) {
930 int skip = FFMIN(rt->skip_bytes, size);
933 rt->skip_bytes -= skip;
938 if (!rt->flv_off && size_temp < 11) {
939 av_log(s, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
946 if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
951 pkttype = bytestream_get_byte(&buf_temp);
952 pktsize = bytestream_get_be24(&buf_temp);
953 ts = bytestream_get_be24(&buf_temp);
954 ts |= bytestream_get_byte(&buf_temp) << 24;
955 bytestream_get_be24(&buf_temp);
957 rt->flv_size = pktsize;
959 //force 12bytes header
960 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
961 pkttype == RTMP_PT_NOTIFY) {
962 if (pkttype == RTMP_PT_NOTIFY)
964 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
967 //this can be a big packet, it's better to send it right here
968 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
969 rt->out_pkt.extra = rt->main_channel_id;
970 rt->flv_data = rt->out_pkt.data;
972 if (pkttype == RTMP_PT_NOTIFY)
973 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
976 if (rt->flv_size - rt->flv_off > size_temp) {
977 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
978 rt->flv_off += size_temp;
981 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
982 size_temp -= rt->flv_size - rt->flv_off;
983 rt->flv_off += rt->flv_size - rt->flv_off;
986 if (rt->flv_off == rt->flv_size) {
988 rt->skip_bytes = 4 - size_temp;
989 buf_temp += size_temp;
992 bytestream_get_be32(&buf_temp);
995 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
996 ff_rtmp_packet_destroy(&rt->out_pkt);
1000 } while (buf_temp - buf < size);
1004 URLProtocol ff_rtmp_protocol = {
1006 .url_open = rtmp_open,
1007 .url_read = rtmp_read,
1008 .url_write = rtmp_write,
1009 .url_close = rtmp_close,