2 * RTMP network protocol
3 * Copyright (c) 2009 Kostya Shishkov
5 * This file is part of FFmpeg.
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.
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.
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
27 #include "libavcodec/bytestream.h"
28 #include "libavutil/avstring.h"
29 #include "libavutil/lfg.h"
30 #include "libavutil/sha.h"
43 /** RTMP protocol handler state */
45 STATE_START, ///< client has not done anything yet
46 STATE_HANDSHAKED, ///< client has performed handshake
47 STATE_RELEASING, ///< client releasing stream before publish it (for output)
48 STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
49 STATE_CONNECTING, ///< client connected to server successfully
50 STATE_READY, ///< client has sent all needed commands and waits for server reply
51 STATE_PLAYING, ///< client has started receiving multimedia data from server
52 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
53 STATE_STOPPED, ///< the broadcast has been stopped
56 /** protocol handler context */
57 typedef struct RTMPContext {
58 URLContext* stream; ///< TCP stream used in interactions with RTMP server
59 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
60 int chunk_size; ///< size of the chunks RTMP packets are divided into
61 int is_input; ///< input/output flag
62 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
63 char app[128]; ///< application
64 ClientState state; ///< current state
65 int main_channel_id; ///< an additional channel ID which is used for some invocations
66 uint8_t* flv_data; ///< buffer with data for demuxer
67 int flv_size; ///< current buffer size
68 int flv_off; ///< number of bytes read from current buffer
69 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
70 uint32_t client_report_size; ///< number of bytes after which client should report to server
71 uint32_t bytes_read; ///< number of bytes read from server
72 uint32_t last_bytes_read; ///< number of bytes read last reported to server
75 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
76 /** Client key used for digest signing */
77 static const uint8_t rtmp_player_key[] = {
78 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
79 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', '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
86 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
87 /** Key used for RTMP server digest signing */
88 static const uint8_t rtmp_server_key[] = {
89 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
90 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
91 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
93 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
94 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
95 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
99 * Generate 'connect' call and send it to the server.
101 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
102 const char *host, int port)
108 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
111 ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
112 ff_amf_write_string(&p, "connect");
113 ff_amf_write_number(&p, 1.0);
114 ff_amf_write_object_start(&p);
115 ff_amf_write_field_name(&p, "app");
116 ff_amf_write_string(&p, rt->app);
119 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
120 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
122 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
123 ff_amf_write_field_name(&p, "type");
124 ff_amf_write_string(&p, "nonprivate");
126 ff_amf_write_field_name(&p, "flashVer");
127 ff_amf_write_string(&p, ver);
128 ff_amf_write_field_name(&p, "tcUrl");
129 ff_amf_write_string(&p, tcurl);
131 ff_amf_write_field_name(&p, "fpad");
132 ff_amf_write_bool(&p, 0);
133 ff_amf_write_field_name(&p, "capabilities");
134 ff_amf_write_number(&p, 15.0);
135 ff_amf_write_field_name(&p, "audioCodecs");
136 ff_amf_write_number(&p, 1639.0);
137 ff_amf_write_field_name(&p, "videoCodecs");
138 ff_amf_write_number(&p, 252.0);
139 ff_amf_write_field_name(&p, "videoFunction");
140 ff_amf_write_number(&p, 1.0);
142 ff_amf_write_object_end(&p);
144 pkt.data_size = p - pkt.data;
146 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
147 ff_rtmp_packet_destroy(&pkt);
151 * Generate 'releaseStream' call and send it to the server. It should make
152 * the server release some channel for media streams.
154 static void gen_release_stream(URLContext *s, RTMPContext *rt)
159 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
160 29 + strlen(rt->playpath));
162 av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
164 ff_amf_write_string(&p, "releaseStream");
165 ff_amf_write_number(&p, 2.0);
166 ff_amf_write_null(&p);
167 ff_amf_write_string(&p, rt->playpath);
169 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
170 ff_rtmp_packet_destroy(&pkt);
174 * Generate 'FCPublish' call and send it to the server. It should make
175 * the server preapare for receiving media streams.
177 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
182 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
183 25 + strlen(rt->playpath));
185 av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
187 ff_amf_write_string(&p, "FCPublish");
188 ff_amf_write_number(&p, 3.0);
189 ff_amf_write_null(&p);
190 ff_amf_write_string(&p, rt->playpath);
192 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
193 ff_rtmp_packet_destroy(&pkt);
197 * Generate 'FCUnpublish' call and send it to the server. It should make
198 * the server destroy stream.
200 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
205 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
206 27 + strlen(rt->playpath));
208 av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
210 ff_amf_write_string(&p, "FCUnpublish");
211 ff_amf_write_number(&p, 5.0);
212 ff_amf_write_null(&p);
213 ff_amf_write_string(&p, rt->playpath);
215 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
216 ff_rtmp_packet_destroy(&pkt);
220 * Generate 'createStream' call and send it to the server. It should make
221 * the server allocate some channel for media streams.
223 static void gen_create_stream(URLContext *s, RTMPContext *rt)
228 av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
229 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
232 ff_amf_write_string(&p, "createStream");
233 ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
234 ff_amf_write_null(&p);
236 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
237 ff_rtmp_packet_destroy(&pkt);
242 * Generate 'deleteStream' call and send it to the server. It should make
243 * the server remove some channel for media streams.
245 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
250 av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
251 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
254 ff_amf_write_string(&p, "deleteStream");
255 ff_amf_write_number(&p, 0.0);
256 ff_amf_write_null(&p);
257 ff_amf_write_number(&p, rt->main_channel_id);
259 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
260 ff_rtmp_packet_destroy(&pkt);
264 * Generate 'play' call and send it to the server, then ping the server
265 * to start actual playing.
267 static void gen_play(URLContext *s, RTMPContext *rt)
272 av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
273 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
274 20 + strlen(rt->playpath));
275 pkt.extra = rt->main_channel_id;
278 ff_amf_write_string(&p, "play");
279 ff_amf_write_number(&p, 0.0);
280 ff_amf_write_null(&p);
281 ff_amf_write_string(&p, rt->playpath);
283 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
284 ff_rtmp_packet_destroy(&pkt);
286 // set client buffer time disguised in ping packet
287 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
290 bytestream_put_be16(&p, 3);
291 bytestream_put_be32(&p, 1);
292 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
294 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
295 ff_rtmp_packet_destroy(&pkt);
299 * Generate 'publish' call and send it to the server.
301 static void gen_publish(URLContext *s, RTMPContext *rt)
306 av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
307 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
308 30 + strlen(rt->playpath));
309 pkt.extra = rt->main_channel_id;
312 ff_amf_write_string(&p, "publish");
313 ff_amf_write_number(&p, 0.0);
314 ff_amf_write_null(&p);
315 ff_amf_write_string(&p, rt->playpath);
316 ff_amf_write_string(&p, "live");
318 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
319 ff_rtmp_packet_destroy(&pkt);
323 * Generate ping reply and send it to the server.
325 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
330 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
332 bytestream_put_be16(&p, 7);
333 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
334 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
335 ff_rtmp_packet_destroy(&pkt);
339 * Generate report on bytes read so far and send it to the server.
341 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
346 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
348 bytestream_put_be32(&p, rt->bytes_read);
349 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
350 ff_rtmp_packet_destroy(&pkt);
353 //TODO: Move HMAC code somewhere. Eventually.
354 #define HMAC_IPAD_VAL 0x36
355 #define HMAC_OPAD_VAL 0x5C
358 * Calculate HMAC-SHA2 digest for RTMP handshake packets.
360 * @param src input buffer
361 * @param len input buffer length (should be 1536)
362 * @param gap offset in buffer where 32 bytes should not be taken into account
363 * when calculating digest (since it will be used to store that digest)
364 * @param key digest key
365 * @param keylen digest key length
366 * @param dst buffer where calculated digest will be stored (32 bytes)
368 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
369 const uint8_t *key, int keylen, uint8_t *dst)
372 uint8_t hmac_buf[64+32] = {0};
375 sha = av_mallocz(av_sha_size);
378 memcpy(hmac_buf, key, keylen);
380 av_sha_init(sha, 256);
381 av_sha_update(sha,key, keylen);
382 av_sha_final(sha, hmac_buf);
384 for (i = 0; i < 64; i++)
385 hmac_buf[i] ^= HMAC_IPAD_VAL;
387 av_sha_init(sha, 256);
388 av_sha_update(sha, hmac_buf, 64);
390 av_sha_update(sha, src, len);
391 } else { //skip 32 bytes used for storing digest
392 av_sha_update(sha, src, gap);
393 av_sha_update(sha, src + gap + 32, len - gap - 32);
395 av_sha_final(sha, hmac_buf + 64);
397 for (i = 0; i < 64; i++)
398 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
399 av_sha_init(sha, 256);
400 av_sha_update(sha, hmac_buf, 64+32);
401 av_sha_final(sha, dst);
407 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
408 * will be stored) into that packet.
410 * @param buf handshake data (1536 bytes)
411 * @return offset to the digest inside input data
413 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
415 int i, digest_pos = 0;
417 for (i = 8; i < 12; i++)
418 digest_pos += buf[i];
419 digest_pos = (digest_pos % 728) + 12;
421 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
422 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
428 * Verify that the received server response has the expected digest value.
430 * @param buf handshake data received from the server (1536 bytes)
431 * @param off position to search digest offset from
432 * @return 0 if digest is valid, digest position otherwise
434 static int rtmp_validate_digest(uint8_t *buf, int off)
436 int i, digest_pos = 0;
439 for (i = 0; i < 4; i++)
440 digest_pos += buf[i + off];
441 digest_pos = (digest_pos % 728) + off + 4;
443 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
444 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
446 if (!memcmp(digest, buf + digest_pos, 32))
452 * Perform handshake with the server by means of exchanging pseudorandom data
453 * signed with HMAC-SHA2 digest.
455 * @return 0 if handshake succeeds, negative value otherwise
457 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
460 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
461 3, // unencrypted data
462 0, 0, 0, 0, // client uptime
468 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
469 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
471 int server_pos, client_pos;
474 av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
476 av_lfg_init(&rnd, 0xDEADC0DE);
477 // generate handshake packet - 1536 bytes of pseudorandom data
478 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
479 tosend[i] = av_lfg_get(&rnd) >> 24;
480 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
482 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
483 i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
484 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
485 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
488 i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
489 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
490 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
494 av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
495 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
497 if (rt->is_input && serverdata[5] >= 3) {
498 server_pos = rtmp_validate_digest(serverdata + 1, 772);
500 server_pos = rtmp_validate_digest(serverdata + 1, 8);
502 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
507 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
508 rtmp_server_key, sizeof(rtmp_server_key),
510 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
513 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
514 av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
518 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
519 tosend[i] = av_lfg_get(&rnd) >> 24;
520 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
521 rtmp_player_key, sizeof(rtmp_player_key),
523 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
525 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
527 // write reply back to the server
528 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
530 ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
537 * Parse received packet and possibly perform some action depending on
538 * the packet contents.
539 * @return 0 for no errors, negative values for serious errors which prevent
540 * further communications, positive values for uncritical errors
542 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
545 const uint8_t *data_end = pkt->data + pkt->data_size;
548 ff_rtmp_packet_dump(s, pkt);
552 case RTMP_PT_CHUNK_SIZE:
553 if (pkt->data_size != 4) {
554 av_log(s, AV_LOG_ERROR,
555 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
559 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
560 rt->chunk_size = AV_RB32(pkt->data);
561 if (rt->chunk_size <= 0) {
562 av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
565 av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
568 t = AV_RB16(pkt->data);
570 gen_pong(s, rt, pkt);
572 case RTMP_PT_CLIENT_BW:
573 if (pkt->data_size < 4) {
574 av_log(s, AV_LOG_ERROR,
575 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
579 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
580 rt->client_report_size = AV_RB32(pkt->data) >> 1;
583 //TODO: check for the messages sent for wrong state?
584 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
587 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
588 "description", tmpstr, sizeof(tmpstr)))
589 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
591 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
593 case STATE_HANDSHAKED:
595 gen_release_stream(s, rt);
596 gen_fcpublish_stream(s, rt);
597 rt->state = STATE_RELEASING;
599 rt->state = STATE_CONNECTING;
601 gen_create_stream(s, rt);
603 case STATE_FCPUBLISH:
604 rt->state = STATE_CONNECTING;
606 case STATE_RELEASING:
607 rt->state = STATE_FCPUBLISH;
608 /* hack for Wowza Media Server, it does not send result for
609 * releaseStream and FCPublish calls */
610 if (!pkt->data[10]) {
611 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
613 rt->state = STATE_CONNECTING;
615 if (rt->state != STATE_CONNECTING)
617 case STATE_CONNECTING:
618 //extract a number from the result
619 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
620 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
622 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
629 rt->state = STATE_READY;
632 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
633 const uint8_t* ptr = pkt->data + 11;
636 for (i = 0; i < 2; i++) {
637 t = ff_amf_tag_size(ptr, data_end);
642 t = ff_amf_get_field_value(ptr, data_end,
643 "level", tmpstr, sizeof(tmpstr));
644 if (!t && !strcmp(tmpstr, "error")) {
645 if (!ff_amf_get_field_value(ptr, data_end,
646 "description", tmpstr, sizeof(tmpstr)))
647 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
650 t = ff_amf_get_field_value(ptr, data_end,
651 "code", tmpstr, sizeof(tmpstr));
652 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
653 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
654 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
655 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
663 * Interact with the server by receiving and sending RTMP packets until
664 * there is some significant data (media data or expected status notification).
666 * @param s reading context
667 * @param for_header non-zero value tells function to work until it
668 * gets notification from the server that playing has been started,
669 * otherwise function will work until some media data is received (or
671 * @return 0 for successful operation, negative value in case of error
673 static int get_packet(URLContext *s, int for_header)
675 RTMPContext *rt = s->priv_data;
680 uint32_t ts, cts, pts=0;
682 if (rt->state == STATE_STOPPED)
687 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
688 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
690 return AVERROR(EAGAIN);
695 rt->bytes_read += ret;
696 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
697 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
698 gen_bytes_read(s, rt, rpkt.timestamp + 1);
699 rt->last_bytes_read = rt->bytes_read;
702 ret = rtmp_parse_result(s, rt, &rpkt);
703 if (ret < 0) {//serious error in current packet
704 ff_rtmp_packet_destroy(&rpkt);
707 if (rt->state == STATE_STOPPED) {
708 ff_rtmp_packet_destroy(&rpkt);
711 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
712 ff_rtmp_packet_destroy(&rpkt);
715 if (!rpkt.data_size || !rt->is_input) {
716 ff_rtmp_packet_destroy(&rpkt);
719 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
720 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
723 // generate packet header and put data into buffer for FLV demuxer
725 rt->flv_size = rpkt.data_size + 15;
726 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
727 bytestream_put_byte(&p, rpkt.type);
728 bytestream_put_be24(&p, rpkt.data_size);
729 bytestream_put_be24(&p, ts);
730 bytestream_put_byte(&p, ts >> 24);
731 bytestream_put_be24(&p, 0);
732 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
733 bytestream_put_be32(&p, 0);
734 ff_rtmp_packet_destroy(&rpkt);
736 } else if (rpkt.type == RTMP_PT_METADATA) {
737 // we got raw FLV data, make it available for FLV demuxer
739 rt->flv_size = rpkt.data_size;
740 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
741 /* rewrite timestamps */
744 while (next - rpkt.data < rpkt.data_size - 11) {
746 data_size = bytestream_get_be24(&next);
748 cts = bytestream_get_be24(&next);
749 cts |= bytestream_get_byte(&next) << 24;
754 bytestream_put_be24(&p, ts);
755 bytestream_put_byte(&p, ts >> 24);
756 next += data_size + 3 + 4;
758 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
759 ff_rtmp_packet_destroy(&rpkt);
762 ff_rtmp_packet_destroy(&rpkt);
767 static int rtmp_close(URLContext *h)
769 RTMPContext *rt = h->priv_data;
773 if (rt->out_pkt.data_size)
774 ff_rtmp_packet_destroy(&rt->out_pkt);
775 if (rt->state > STATE_FCPUBLISH)
776 gen_fcunpublish_stream(h, rt);
778 if (rt->state > STATE_HANDSHAKED)
779 gen_delete_stream(h, rt);
781 av_freep(&rt->flv_data);
782 ffurl_close(rt->stream);
788 * Open RTMP connection and verify that the stream can be played.
790 * URL syntax: rtmp://server[:port][/app][/playpath]
791 * where 'app' is first one or two directories in the path
792 * (e.g. /ondemand/, /flash/live/, etc.)
793 * and 'playpath' is a file name (the rest of the path,
794 * may be prefixed with "mp4:")
796 static int rtmp_open(URLContext *s, const char *uri, int flags)
799 char proto[8], hostname[256], path[1024], *fname;
804 rt = av_mallocz(sizeof(RTMPContext));
806 return AVERROR(ENOMEM);
808 rt->is_input = !(flags & AVIO_FLAG_WRITE);
810 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
811 path, sizeof(path), s->filename);
814 port = RTMP_DEFAULT_PORT;
815 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
817 if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE) < 0) {
818 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
822 rt->state = STATE_START;
823 if (rtmp_handshake(s, rt))
826 rt->chunk_size = 128;
827 rt->state = STATE_HANDSHAKED;
828 //extract "app" part from path
829 if (!strncmp(path, "/ondemand/", 10)) {
831 memcpy(rt->app, "ondemand", 9);
833 char *p = strchr(path + 1, '/');
838 char *c = strchr(p + 1, ':');
839 fname = strchr(p + 1, '/');
840 if (!fname || c < fname) {
842 av_strlcpy(rt->app, path + 1, p - path);
845 av_strlcpy(rt->app, path + 1, fname - path - 1);
849 if (!strchr(fname, ':') &&
850 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
851 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
852 memcpy(rt->playpath, "mp4:", 5);
856 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
858 rt->client_report_size = 1048576;
860 rt->last_bytes_read = 0;
862 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
863 proto, path, rt->app, rt->playpath);
864 gen_connect(s, rt, proto, hostname, port);
867 ret = get_packet(s, 1);
868 } while (ret == EAGAIN);
873 // generate FLV header for demuxer
875 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
877 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
884 s->max_packet_size = rt->stream->max_packet_size;
893 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
895 RTMPContext *rt = s->priv_data;
896 int orig_size = size;
900 int data_left = rt->flv_size - rt->flv_off;
902 if (data_left >= size) {
903 memcpy(buf, rt->flv_data + rt->flv_off, size);
908 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
911 rt->flv_off = rt->flv_size;
914 if ((ret = get_packet(s, 0)) < 0)
920 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
922 RTMPContext *rt = s->priv_data;
923 int size_temp = size;
924 int pktsize, pkttype;
926 const uint8_t *buf_temp = buf;
929 av_log(s, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
936 if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
941 pkttype = bytestream_get_byte(&buf_temp);
942 pktsize = bytestream_get_be24(&buf_temp);
943 ts = bytestream_get_be24(&buf_temp);
944 ts |= bytestream_get_byte(&buf_temp) << 24;
945 bytestream_get_be24(&buf_temp);
947 rt->flv_size = pktsize;
949 //force 12bytes header
950 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
951 pkttype == RTMP_PT_NOTIFY) {
952 if (pkttype == RTMP_PT_NOTIFY)
954 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
957 //this can be a big packet, it's better to send it right here
958 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
959 rt->out_pkt.extra = rt->main_channel_id;
960 rt->flv_data = rt->out_pkt.data;
962 if (pkttype == RTMP_PT_NOTIFY)
963 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
966 if (rt->flv_size - rt->flv_off > size_temp) {
967 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
968 rt->flv_off += size_temp;
970 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
971 rt->flv_off += rt->flv_size - rt->flv_off;
974 if (rt->flv_off == rt->flv_size) {
975 bytestream_get_be32(&buf_temp);
977 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
978 ff_rtmp_packet_destroy(&rt->out_pkt);
982 } while (buf_temp - buf < size_temp);
986 URLProtocol ff_rtmp_protocol = {
988 .url_open = rtmp_open,
989 .url_read = rtmp_read,
990 .url_write = rtmp_write,
991 .url_close = rtmp_close,