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/intfloat.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
75 uint8_t flv_header[11]; ///< partial incoming flv packet header
76 int flv_header_bytes; ///< number of initialized bytes in flv_header
77 int nb_invokes; ///< keeps track of invoke messages
78 int create_stream_invoke; ///< invoke id for the create stream command
81 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
82 /** Client key used for digest signing */
83 static const uint8_t rtmp_player_key[] = {
84 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
85 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
87 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
88 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
89 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
92 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
93 /** Key used for RTMP server digest signing */
94 static const uint8_t rtmp_server_key[] = {
95 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
96 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
97 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
99 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
100 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
101 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
105 * Generate 'connect' call and send it to the server.
107 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
108 const char *host, int port)
114 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
117 ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
118 ff_amf_write_string(&p, "connect");
119 ff_amf_write_number(&p, ++rt->nb_invokes);
120 ff_amf_write_object_start(&p);
121 ff_amf_write_field_name(&p, "app");
122 ff_amf_write_string(&p, rt->app);
125 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
126 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
128 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
129 ff_amf_write_field_name(&p, "type");
130 ff_amf_write_string(&p, "nonprivate");
132 ff_amf_write_field_name(&p, "flashVer");
133 ff_amf_write_string(&p, ver);
134 ff_amf_write_field_name(&p, "tcUrl");
135 ff_amf_write_string(&p, tcurl);
137 ff_amf_write_field_name(&p, "fpad");
138 ff_amf_write_bool(&p, 0);
139 ff_amf_write_field_name(&p, "capabilities");
140 ff_amf_write_number(&p, 15.0);
141 ff_amf_write_field_name(&p, "audioCodecs");
142 ff_amf_write_number(&p, 1639.0);
143 ff_amf_write_field_name(&p, "videoCodecs");
144 ff_amf_write_number(&p, 252.0);
145 ff_amf_write_field_name(&p, "videoFunction");
146 ff_amf_write_number(&p, 1.0);
148 ff_amf_write_object_end(&p);
150 pkt.data_size = p - pkt.data;
152 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
153 ff_rtmp_packet_destroy(&pkt);
157 * Generate 'releaseStream' call and send it to the server. It should make
158 * the server release some channel for media streams.
160 static void gen_release_stream(URLContext *s, RTMPContext *rt)
165 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
166 29 + strlen(rt->playpath));
168 av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
170 ff_amf_write_string(&p, "releaseStream");
171 ff_amf_write_number(&p, ++rt->nb_invokes);
172 ff_amf_write_null(&p);
173 ff_amf_write_string(&p, rt->playpath);
175 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
176 ff_rtmp_packet_destroy(&pkt);
180 * Generate 'FCPublish' call and send it to the server. It should make
181 * the server preapare for receiving media streams.
183 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
188 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
189 25 + strlen(rt->playpath));
191 av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
193 ff_amf_write_string(&p, "FCPublish");
194 ff_amf_write_number(&p, ++rt->nb_invokes);
195 ff_amf_write_null(&p);
196 ff_amf_write_string(&p, rt->playpath);
198 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
199 ff_rtmp_packet_destroy(&pkt);
203 * Generate 'FCUnpublish' call and send it to the server. It should make
204 * the server destroy stream.
206 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
211 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
212 27 + strlen(rt->playpath));
214 av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
216 ff_amf_write_string(&p, "FCUnpublish");
217 ff_amf_write_number(&p, ++rt->nb_invokes);
218 ff_amf_write_null(&p);
219 ff_amf_write_string(&p, rt->playpath);
221 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
222 ff_rtmp_packet_destroy(&pkt);
226 * Generate 'createStream' call and send it to the server. It should make
227 * the server allocate some channel for media streams.
229 static void gen_create_stream(URLContext *s, RTMPContext *rt)
234 av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
235 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
238 ff_amf_write_string(&p, "createStream");
239 ff_amf_write_number(&p, ++rt->nb_invokes);
240 ff_amf_write_null(&p);
241 rt->create_stream_invoke = rt->nb_invokes;
243 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
244 ff_rtmp_packet_destroy(&pkt);
249 * Generate 'deleteStream' call and send it to the server. It should make
250 * the server remove some channel for media streams.
252 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
257 av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
258 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
261 ff_amf_write_string(&p, "deleteStream");
262 ff_amf_write_number(&p, ++rt->nb_invokes);
263 ff_amf_write_null(&p);
264 ff_amf_write_number(&p, rt->main_channel_id);
266 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
267 ff_rtmp_packet_destroy(&pkt);
271 * Generate 'play' call and send it to the server, then ping the server
272 * to start actual playing.
274 static void gen_play(URLContext *s, RTMPContext *rt)
279 av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
280 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
281 20 + strlen(rt->playpath));
282 pkt.extra = rt->main_channel_id;
285 ff_amf_write_string(&p, "play");
286 ff_amf_write_number(&p, ++rt->nb_invokes);
287 ff_amf_write_null(&p);
288 ff_amf_write_string(&p, rt->playpath);
290 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
291 ff_rtmp_packet_destroy(&pkt);
293 // set client buffer time disguised in ping packet
294 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
297 bytestream_put_be16(&p, 3);
298 bytestream_put_be32(&p, 1);
299 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
301 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
302 ff_rtmp_packet_destroy(&pkt);
306 * Generate 'publish' call and send it to the server.
308 static void gen_publish(URLContext *s, RTMPContext *rt)
313 av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
314 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
315 30 + strlen(rt->playpath));
316 pkt.extra = rt->main_channel_id;
319 ff_amf_write_string(&p, "publish");
320 ff_amf_write_number(&p, ++rt->nb_invokes);
321 ff_amf_write_null(&p);
322 ff_amf_write_string(&p, rt->playpath);
323 ff_amf_write_string(&p, "live");
325 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
326 ff_rtmp_packet_destroy(&pkt);
330 * Generate ping reply and send it to the server.
332 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
337 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
339 bytestream_put_be16(&p, 7);
340 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
341 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
342 ff_rtmp_packet_destroy(&pkt);
346 * Generate report on bytes read so far and send it to the server.
348 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
353 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
355 bytestream_put_be32(&p, rt->bytes_read);
356 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
357 ff_rtmp_packet_destroy(&pkt);
360 //TODO: Move HMAC code somewhere. Eventually.
361 #define HMAC_IPAD_VAL 0x36
362 #define HMAC_OPAD_VAL 0x5C
365 * Calculate HMAC-SHA2 digest for RTMP handshake packets.
367 * @param src input buffer
368 * @param len input buffer length (should be 1536)
369 * @param gap offset in buffer where 32 bytes should not be taken into account
370 * when calculating digest (since it will be used to store that digest)
371 * @param key digest key
372 * @param keylen digest key length
373 * @param dst buffer where calculated digest will be stored (32 bytes)
375 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
376 const uint8_t *key, int keylen, uint8_t *dst)
379 uint8_t hmac_buf[64+32] = {0};
382 sha = av_mallocz(av_sha_size);
385 memcpy(hmac_buf, key, keylen);
387 av_sha_init(sha, 256);
388 av_sha_update(sha,key, keylen);
389 av_sha_final(sha, hmac_buf);
391 for (i = 0; i < 64; i++)
392 hmac_buf[i] ^= HMAC_IPAD_VAL;
394 av_sha_init(sha, 256);
395 av_sha_update(sha, hmac_buf, 64);
397 av_sha_update(sha, src, len);
398 } else { //skip 32 bytes used for storing digest
399 av_sha_update(sha, src, gap);
400 av_sha_update(sha, src + gap + 32, len - gap - 32);
402 av_sha_final(sha, hmac_buf + 64);
404 for (i = 0; i < 64; i++)
405 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
406 av_sha_init(sha, 256);
407 av_sha_update(sha, hmac_buf, 64+32);
408 av_sha_final(sha, dst);
414 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
415 * will be stored) into that packet.
417 * @param buf handshake data (1536 bytes)
418 * @return offset to the digest inside input data
420 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
422 int i, digest_pos = 0;
424 for (i = 8; i < 12; i++)
425 digest_pos += buf[i];
426 digest_pos = (digest_pos % 728) + 12;
428 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
429 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
435 * Verify that the received server response has the expected digest value.
437 * @param buf handshake data received from the server (1536 bytes)
438 * @param off position to search digest offset from
439 * @return 0 if digest is valid, digest position otherwise
441 static int rtmp_validate_digest(uint8_t *buf, int off)
443 int i, digest_pos = 0;
446 for (i = 0; i < 4; i++)
447 digest_pos += buf[i + off];
448 digest_pos = (digest_pos % 728) + off + 4;
450 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
451 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
453 if (!memcmp(digest, buf + digest_pos, 32))
459 * Perform handshake with the server by means of exchanging pseudorandom data
460 * signed with HMAC-SHA2 digest.
462 * @return 0 if handshake succeeds, negative value otherwise
464 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
467 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
468 3, // unencrypted data
469 0, 0, 0, 0, // client uptime
475 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
476 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
478 int server_pos, client_pos;
481 av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
483 av_lfg_init(&rnd, 0xDEADC0DE);
484 // generate handshake packet - 1536 bytes of pseudorandom data
485 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
486 tosend[i] = av_lfg_get(&rnd) >> 24;
487 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
489 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
490 i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
491 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
492 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
495 i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
496 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
497 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
501 av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
502 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
504 if (rt->is_input && serverdata[5] >= 3) {
505 server_pos = rtmp_validate_digest(serverdata + 1, 772);
507 server_pos = rtmp_validate_digest(serverdata + 1, 8);
509 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
514 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
515 rtmp_server_key, sizeof(rtmp_server_key),
517 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
520 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
521 av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
525 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
526 tosend[i] = av_lfg_get(&rnd) >> 24;
527 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
528 rtmp_player_key, sizeof(rtmp_player_key),
530 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
532 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
534 // write reply back to the server
535 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
537 ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
544 * Parse received packet and possibly perform some action depending on
545 * the packet contents.
546 * @return 0 for no errors, negative values for serious errors which prevent
547 * further communications, positive values for uncritical errors
549 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
552 const uint8_t *data_end = pkt->data + pkt->data_size;
555 ff_rtmp_packet_dump(s, pkt);
559 case RTMP_PT_CHUNK_SIZE:
560 if (pkt->data_size != 4) {
561 av_log(s, AV_LOG_ERROR,
562 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
566 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
567 rt->chunk_size = AV_RB32(pkt->data);
568 if (rt->chunk_size <= 0) {
569 av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
572 av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
575 t = AV_RB16(pkt->data);
577 gen_pong(s, rt, pkt);
579 case RTMP_PT_CLIENT_BW:
580 if (pkt->data_size < 4) {
581 av_log(s, AV_LOG_ERROR,
582 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
586 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
587 rt->client_report_size = AV_RB32(pkt->data) >> 1;
590 //TODO: check for the messages sent for wrong state?
591 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
594 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
595 "description", tmpstr, sizeof(tmpstr)))
596 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
598 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
600 case STATE_HANDSHAKED:
602 gen_release_stream(s, rt);
603 gen_fcpublish_stream(s, rt);
604 rt->state = STATE_RELEASING;
606 rt->state = STATE_CONNECTING;
608 gen_create_stream(s, rt);
610 case STATE_FCPUBLISH:
611 rt->state = STATE_CONNECTING;
613 case STATE_RELEASING:
614 rt->state = STATE_FCPUBLISH;
615 /* hack for Wowza Media Server, it does not send result for
616 * releaseStream and FCPublish calls */
617 if (!pkt->data[10]) {
618 int pkt_id = av_int2double(AV_RB64(pkt->data + 11));
619 if (pkt_id == rt->create_stream_invoke)
620 rt->state = STATE_CONNECTING;
622 if (rt->state != STATE_CONNECTING)
624 case STATE_CONNECTING:
625 //extract a number from the result
626 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
627 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
629 rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
636 rt->state = STATE_READY;
639 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
640 const uint8_t* ptr = pkt->data + 11;
643 for (i = 0; i < 2; i++) {
644 t = ff_amf_tag_size(ptr, data_end);
649 t = ff_amf_get_field_value(ptr, data_end,
650 "level", tmpstr, sizeof(tmpstr));
651 if (!t && !strcmp(tmpstr, "error")) {
652 if (!ff_amf_get_field_value(ptr, data_end,
653 "description", tmpstr, sizeof(tmpstr)))
654 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
657 t = ff_amf_get_field_value(ptr, data_end,
658 "code", tmpstr, sizeof(tmpstr));
659 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
660 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
661 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
662 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
670 * Interact with the server by receiving and sending RTMP packets until
671 * there is some significant data (media data or expected status notification).
673 * @param s reading context
674 * @param for_header non-zero value tells function to work until it
675 * gets notification from the server that playing has been started,
676 * otherwise function will work until some media data is received (or
678 * @return 0 for successful operation, negative value in case of error
680 static int get_packet(URLContext *s, int for_header)
682 RTMPContext *rt = s->priv_data;
687 uint32_t ts, cts, pts=0;
689 if (rt->state == STATE_STOPPED)
693 RTMPPacket rpkt = { 0 };
694 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
695 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
697 return AVERROR(EAGAIN);
702 rt->bytes_read += ret;
703 if (rt->bytes_read - rt->last_bytes_read > rt->client_report_size) {
704 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
705 gen_bytes_read(s, rt, rpkt.timestamp + 1);
706 rt->last_bytes_read = rt->bytes_read;
709 ret = rtmp_parse_result(s, rt, &rpkt);
710 if (ret < 0) {//serious error in current packet
711 ff_rtmp_packet_destroy(&rpkt);
714 if (rt->state == STATE_STOPPED) {
715 ff_rtmp_packet_destroy(&rpkt);
718 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
719 ff_rtmp_packet_destroy(&rpkt);
722 if (!rpkt.data_size || !rt->is_input) {
723 ff_rtmp_packet_destroy(&rpkt);
726 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
727 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
730 // generate packet header and put data into buffer for FLV demuxer
732 rt->flv_size = rpkt.data_size + 15;
733 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
734 bytestream_put_byte(&p, rpkt.type);
735 bytestream_put_be24(&p, rpkt.data_size);
736 bytestream_put_be24(&p, ts);
737 bytestream_put_byte(&p, ts >> 24);
738 bytestream_put_be24(&p, 0);
739 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
740 bytestream_put_be32(&p, 0);
741 ff_rtmp_packet_destroy(&rpkt);
743 } else if (rpkt.type == RTMP_PT_METADATA) {
744 // we got raw FLV data, make it available for FLV demuxer
746 rt->flv_size = rpkt.data_size;
747 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
748 /* rewrite timestamps */
751 while (next - rpkt.data < rpkt.data_size - 11) {
753 data_size = bytestream_get_be24(&next);
755 cts = bytestream_get_be24(&next);
756 cts |= bytestream_get_byte(&next) << 24;
761 bytestream_put_be24(&p, ts);
762 bytestream_put_byte(&p, ts >> 24);
763 next += data_size + 3 + 4;
765 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
766 ff_rtmp_packet_destroy(&rpkt);
769 ff_rtmp_packet_destroy(&rpkt);
773 static int rtmp_close(URLContext *h)
775 RTMPContext *rt = h->priv_data;
779 if (rt->out_pkt.data_size)
780 ff_rtmp_packet_destroy(&rt->out_pkt);
781 if (rt->state > STATE_FCPUBLISH)
782 gen_fcunpublish_stream(h, rt);
784 if (rt->state > STATE_HANDSHAKED)
785 gen_delete_stream(h, rt);
787 av_freep(&rt->flv_data);
788 ffurl_close(rt->stream);
793 * Open RTMP connection and verify that the stream can be played.
795 * URL syntax: rtmp://server[:port][/app][/playpath]
796 * where 'app' is first one or two directories in the path
797 * (e.g. /ondemand/, /flash/live/, etc.)
798 * and 'playpath' is a file name (the rest of the path,
799 * may be prefixed with "mp4:")
801 static int rtmp_open(URLContext *s, const char *uri, int flags)
803 RTMPContext *rt = s->priv_data;
804 char proto[8], hostname[256], path[1024], *fname;
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,
819 &s->interrupt_callback, NULL) < 0) {
820 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
824 rt->state = STATE_START;
825 if (rtmp_handshake(s, rt))
828 rt->chunk_size = 128;
829 rt->state = STATE_HANDSHAKED;
830 //extract "app" part from path
831 if (!strncmp(path, "/ondemand/", 10)) {
833 memcpy(rt->app, "ondemand", 9);
835 char *p = strchr(path + 1, '/');
840 char *c = strchr(p + 1, ':');
841 fname = strchr(p + 1, '/');
842 if (!fname || c < fname) {
844 av_strlcpy(rt->app, path + 1, p - path);
847 av_strlcpy(rt->app, path + 1, fname - path - 1);
851 if (!strchr(fname, ':') &&
852 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
853 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
854 memcpy(rt->playpath, "mp4:", 5);
858 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
860 rt->client_report_size = 1048576;
862 rt->last_bytes_read = 0;
864 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
865 proto, path, rt->app, rt->playpath);
866 gen_connect(s, rt, proto, hostname, port);
869 ret = get_packet(s, 1);
870 } while (ret == EAGAIN);
875 // generate FLV header for demuxer
877 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
879 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
887 s->max_packet_size = rt->stream->max_packet_size;
896 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
898 RTMPContext *rt = s->priv_data;
899 int orig_size = size;
903 int data_left = rt->flv_size - rt->flv_off;
905 if (data_left >= size) {
906 memcpy(buf, rt->flv_data + rt->flv_off, size);
911 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
914 rt->flv_off = rt->flv_size;
917 if ((ret = get_packet(s, 0)) < 0)
923 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
925 RTMPContext *rt = s->priv_data;
926 int size_temp = size;
927 int pktsize, pkttype;
929 const uint8_t *buf_temp = buf;
932 if (rt->skip_bytes) {
933 int skip = FFMIN(rt->skip_bytes, size_temp);
936 rt->skip_bytes -= skip;
940 if (rt->flv_header_bytes < 11) {
941 const uint8_t *header = rt->flv_header;
942 int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
943 bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
944 rt->flv_header_bytes += copy;
946 if (rt->flv_header_bytes < 11)
949 pkttype = bytestream_get_byte(&header);
950 pktsize = bytestream_get_be24(&header);
951 ts = bytestream_get_be24(&header);
952 ts |= bytestream_get_byte(&header) << 24;
953 bytestream_get_be24(&header);
954 rt->flv_size = pktsize;
956 //force 12bytes header
957 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
958 pkttype == RTMP_PT_NOTIFY) {
959 if (pkttype == RTMP_PT_NOTIFY)
961 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
964 //this can be a big packet, it's better to send it right here
965 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
966 rt->out_pkt.extra = rt->main_channel_id;
967 rt->flv_data = rt->out_pkt.data;
969 if (pkttype == RTMP_PT_NOTIFY)
970 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
973 if (rt->flv_size - rt->flv_off > size_temp) {
974 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
975 rt->flv_off += size_temp;
978 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
979 size_temp -= rt->flv_size - rt->flv_off;
980 rt->flv_off += rt->flv_size - rt->flv_off;
983 if (rt->flv_off == rt->flv_size) {
986 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
987 ff_rtmp_packet_destroy(&rt->out_pkt);
990 rt->flv_header_bytes = 0;
992 } while (buf_temp - buf < size);
996 URLProtocol ff_rtmp_protocol = {
998 .url_open = rtmp_open,
999 .url_read = rtmp_read,
1000 .url_write = rtmp_write,
1001 .url_close = rtmp_close,
1002 .priv_data_size = sizeof(RTMPContext),
1003 .flags = URL_PROTOCOL_FLAG_NETWORK,