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
23 * @file libavformat/rtmpproto.c
27 #include "libavcodec/bytestream.h"
28 #include "libavutil/avstring.h"
29 #include "libavutil/lfg.h"
30 #include "libavutil/sha.h"
39 /* we can't use av_log() with URLContext yet... */
40 #if LIBAVFORMAT_VERSION_MAJOR < 53
41 #define LOG_CONTEXT NULL
46 /** RTMP protocol handler state */
48 STATE_START, ///< client has not done anything yet
49 STATE_HANDSHAKED, ///< client has performed handshake
50 STATE_RELEASING, ///< client releasing stream before publish it (for output)
51 STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
52 STATE_CONNECTING, ///< client connected to server successfully
53 STATE_READY, ///< client has sent all needed commands and waits for server reply
54 STATE_PLAYING, ///< client has started receiving multimedia data from server
55 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
56 STATE_STOPPED, ///< the broadcast has been stopped
59 /** protocol handler context */
60 typedef struct RTMPContext {
61 URLContext* stream; ///< TCP stream used in interactions with RTMP server
62 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
63 int chunk_size; ///< size of the chunks RTMP packets are divided into
64 int is_input; ///< input/output flag
65 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
66 char app[128]; ///< application
67 ClientState state; ///< current state
68 int main_channel_id; ///< an additional channel ID which is used for some invocations
69 uint8_t* flv_data; ///< buffer with data for demuxer
70 int flv_size; ///< current buffer size
71 int flv_off; ///< number of bytes read from current buffer
72 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
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 * Generates 'connect' call and sends 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 snprintf(tcurl, sizeof(tcurl), "%s://%s:%d/%s", proto, host, port, 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 * Generates 'releaseStream' call and sends 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(LOG_CONTEXT, 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 * Generates 'FCPublish' call and sends 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(LOG_CONTEXT, 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 * Generates 'FCUnpublish' call and sends 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(LOG_CONTEXT, 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 * Generates 'createStream' call and sends 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(LOG_CONTEXT, 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 * Generates 'deleteStream' call and sends 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(LOG_CONTEXT, 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 * Generates 'play' call and sends it to the server, then pings the server
265 * to start actual playing.
267 static void gen_play(URLContext *s, RTMPContext *rt)
272 av_log(LOG_CONTEXT, 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 * Generates 'publish' call and sends it to the server.
301 static void gen_publish(URLContext *s, RTMPContext *rt)
306 av_log(LOG_CONTEXT, 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 * Generates ping reply and sends 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) + 1);
334 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
335 ff_rtmp_packet_destroy(&pkt);
338 //TODO: Move HMAC code somewhere. Eventually.
339 #define HMAC_IPAD_VAL 0x36
340 #define HMAC_OPAD_VAL 0x5C
343 * Calculates HMAC-SHA2 digest for RTMP handshake packets.
345 * @param src input buffer
346 * @param len input buffer length (should be 1536)
347 * @param gap offset in buffer where 32 bytes should not be taken into account
348 * when calculating digest (since it will be used to store that digest)
349 * @param key digest key
350 * @param keylen digest key length
351 * @param dst buffer where calculated digest will be stored (32 bytes)
353 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
354 const uint8_t *key, int keylen, uint8_t *dst)
357 uint8_t hmac_buf[64+32] = {0};
360 sha = av_mallocz(av_sha_size);
363 memcpy(hmac_buf, key, keylen);
365 av_sha_init(sha, 256);
366 av_sha_update(sha,key, keylen);
367 av_sha_final(sha, hmac_buf);
369 for (i = 0; i < 64; i++)
370 hmac_buf[i] ^= HMAC_IPAD_VAL;
372 av_sha_init(sha, 256);
373 av_sha_update(sha, hmac_buf, 64);
375 av_sha_update(sha, src, len);
376 } else { //skip 32 bytes used for storing digest
377 av_sha_update(sha, src, gap);
378 av_sha_update(sha, src + gap + 32, len - gap - 32);
380 av_sha_final(sha, hmac_buf + 64);
382 for (i = 0; i < 64; i++)
383 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
384 av_sha_init(sha, 256);
385 av_sha_update(sha, hmac_buf, 64+32);
386 av_sha_final(sha, dst);
392 * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest
393 * will be stored) into that packet.
395 * @param buf handshake data (1536 bytes)
396 * @return offset to the digest inside input data
398 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
400 int i, digest_pos = 0;
402 for (i = 8; i < 12; i++)
403 digest_pos += buf[i];
404 digest_pos = (digest_pos % 728) + 12;
406 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
407 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
413 * Verifies that the received server response has the expected digest value.
415 * @param buf handshake data received from the server (1536 bytes)
416 * @param off position to search digest offset from
417 * @return 0 if digest is valid, digest position otherwise
419 static int rtmp_validate_digest(uint8_t *buf, int off)
421 int i, digest_pos = 0;
424 for (i = 0; i < 4; i++)
425 digest_pos += buf[i + off];
426 digest_pos = (digest_pos % 728) + off + 4;
428 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
429 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
431 if (!memcmp(digest, buf + digest_pos, 32))
437 * Performs handshake with the server by means of exchanging pseudorandom data
438 * signed with HMAC-SHA2 digest.
440 * @return 0 if handshake succeeds, negative value otherwise
442 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
445 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
446 3, // unencrypted data
447 0, 0, 0, 0, // client uptime
453 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
454 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
456 int server_pos, client_pos;
459 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n");
461 av_lfg_init(&rnd, 0xDEADC0DE);
462 // generate handshake packet - 1536 bytes of pseudorandom data
463 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
464 tosend[i] = av_lfg_get(&rnd) >> 24;
465 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
467 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
468 i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
469 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
470 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
473 i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
474 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
475 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
479 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
480 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
483 server_pos = rtmp_validate_digest(serverdata + 1, 772);
485 server_pos = rtmp_validate_digest(serverdata + 1, 8);
487 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n");
492 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
493 rtmp_server_key, sizeof(rtmp_server_key),
495 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
498 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
499 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n");
503 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
504 tosend[i] = av_lfg_get(&rnd) >> 24;
505 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
506 rtmp_player_key, sizeof(rtmp_player_key),
508 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
510 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
512 // write reply back to the server
513 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
515 url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
522 * Parses received packet and may perform some action depending on
523 * the packet contents.
524 * @return 0 for no errors, negative values for serious errors which prevent
525 * further communications, positive values for uncritical errors
527 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
530 const uint8_t *data_end = pkt->data + pkt->data_size;
533 case RTMP_PT_CHUNK_SIZE:
534 if (pkt->data_size != 4) {
535 av_log(LOG_CONTEXT, AV_LOG_ERROR,
536 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
540 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
541 rt->chunk_size = AV_RB32(pkt->data);
542 if (rt->chunk_size <= 0) {
543 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
546 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
549 t = AV_RB16(pkt->data);
551 gen_pong(s, rt, pkt);
554 //TODO: check for the messages sent for wrong state?
555 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
558 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
559 "description", tmpstr, sizeof(tmpstr)))
560 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
562 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
564 case STATE_HANDSHAKED:
566 gen_release_stream(s, rt);
567 gen_fcpublish_stream(s, rt);
568 rt->state = STATE_RELEASING;
570 rt->state = STATE_CONNECTING;
572 gen_create_stream(s, rt);
574 case STATE_FCPUBLISH:
575 rt->state = STATE_CONNECTING;
577 case STATE_RELEASING:
578 rt->state = STATE_FCPUBLISH;
579 /* hack for Wowza Media Server, it does not send result for
580 * releaseStream and FCPublish calls */
581 if (!pkt->data[10]) {
582 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
584 rt->state = STATE_CONNECTING;
586 if (rt->state != STATE_CONNECTING)
588 case STATE_CONNECTING:
589 //extract a number from the result
590 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
591 av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n");
593 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
600 rt->state = STATE_READY;
603 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
604 const uint8_t* ptr = pkt->data + 11;
607 for (i = 0; i < 2; i++) {
608 t = ff_amf_tag_size(ptr, data_end);
613 t = ff_amf_get_field_value(ptr, data_end,
614 "level", tmpstr, sizeof(tmpstr));
615 if (!t && !strcmp(tmpstr, "error")) {
616 if (!ff_amf_get_field_value(ptr, data_end,
617 "description", tmpstr, sizeof(tmpstr)))
618 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
621 t = ff_amf_get_field_value(ptr, data_end,
622 "code", tmpstr, sizeof(tmpstr));
623 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
624 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
625 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
626 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
634 * Interacts with the server by receiving and sending RTMP packets until
635 * there is some significant data (media data or expected status notification).
637 * @param s reading context
638 * @param for_header non-zero value tells function to work until it
639 * gets notification from the server that playing has been started,
640 * otherwise function will work until some media data is received (or
642 * @return 0 for successful operation, negative value in case of error
644 static int get_packet(URLContext *s, int for_header)
646 RTMPContext *rt = s->priv_data;
649 if (rt->state == STATE_STOPPED)
654 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
655 rt->chunk_size, rt->prev_pkt[0])) != 0) {
657 return AVERROR(EAGAIN);
663 ret = rtmp_parse_result(s, rt, &rpkt);
664 if (ret < 0) {//serious error in current packet
665 ff_rtmp_packet_destroy(&rpkt);
668 if (rt->state == STATE_STOPPED) {
669 ff_rtmp_packet_destroy(&rpkt);
672 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
673 ff_rtmp_packet_destroy(&rpkt);
676 if (!rpkt.data_size || !rt->is_input) {
677 ff_rtmp_packet_destroy(&rpkt);
680 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
681 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
683 uint32_t ts = rpkt.timestamp;
685 // generate packet header and put data into buffer for FLV demuxer
687 rt->flv_size = rpkt.data_size + 15;
688 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
689 bytestream_put_byte(&p, rpkt.type);
690 bytestream_put_be24(&p, rpkt.data_size);
691 bytestream_put_be24(&p, ts);
692 bytestream_put_byte(&p, ts >> 24);
693 bytestream_put_be24(&p, 0);
694 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
695 bytestream_put_be32(&p, 0);
696 ff_rtmp_packet_destroy(&rpkt);
698 } else if (rpkt.type == RTMP_PT_METADATA) {
699 // we got raw FLV data, make it available for FLV demuxer
701 rt->flv_size = rpkt.data_size;
702 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
703 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
704 ff_rtmp_packet_destroy(&rpkt);
707 ff_rtmp_packet_destroy(&rpkt);
712 static int rtmp_close(URLContext *h)
714 RTMPContext *rt = h->priv_data;
718 if (rt->out_pkt.data_size)
719 ff_rtmp_packet_destroy(&rt->out_pkt);
720 if (rt->state > STATE_FCPUBLISH)
721 gen_fcunpublish_stream(h, rt);
723 if (rt->state > STATE_HANDSHAKED)
724 gen_delete_stream(h, rt);
726 av_freep(&rt->flv_data);
727 url_close(rt->stream);
733 * Opens RTMP connection and verifies that the stream can be played.
735 * URL syntax: rtmp://server[:port][/app][/playpath]
736 * where 'app' is first one or two directories in the path
737 * (e.g. /ondemand/, /flash/live/, etc.)
738 * and 'playpath' is a file name (the rest of the path,
739 * may be prefixed with "mp4:")
741 static int rtmp_open(URLContext *s, const char *uri, int flags)
744 char proto[8], hostname[256], path[1024], *fname;
749 rt = av_mallocz(sizeof(RTMPContext));
751 return AVERROR(ENOMEM);
753 rt->is_input = !(flags & URL_WRONLY);
755 url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
756 path, sizeof(path), s->filename);
759 port = RTMP_DEFAULT_PORT;
760 snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port);
762 if (url_open(&rt->stream, buf, URL_RDWR) < 0) {
763 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);
767 rt->state = STATE_START;
768 if (rtmp_handshake(s, rt))
771 rt->chunk_size = 128;
772 rt->state = STATE_HANDSHAKED;
773 //extract "app" part from path
774 if (!strncmp(path, "/ondemand/", 10)) {
776 memcpy(rt->app, "ondemand", 9);
778 char *p = strchr(path + 1, '/');
783 char *c = strchr(p + 1, ':');
784 fname = strchr(p + 1, '/');
785 if (!fname || c < fname) {
787 av_strlcpy(rt->app, path + 1, p - path);
790 av_strlcpy(rt->app, path + 1, fname - path - 1);
794 if (!strchr(fname, ':') &&
795 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
796 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
797 memcpy(rt->playpath, "mp4:", 5);
801 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
803 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
804 proto, path, rt->app, rt->playpath);
805 gen_connect(s, rt, proto, hostname, port);
808 ret = get_packet(s, 1);
809 } while (ret == EAGAIN);
814 // generate FLV header for demuxer
816 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
818 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
825 s->max_packet_size = url_get_max_packet_size(rt->stream);
834 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
836 RTMPContext *rt = s->priv_data;
837 int orig_size = size;
841 int data_left = rt->flv_size - rt->flv_off;
843 if (data_left >= size) {
844 memcpy(buf, rt->flv_data + rt->flv_off, size);
849 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
852 rt->flv_off = rt->flv_size;
854 if ((ret = get_packet(s, 0)) < 0)
860 static int rtmp_write(URLContext *h, uint8_t *buf, int size)
862 RTMPContext *rt = h->priv_data;
863 int size_temp = size;
864 int pktsize, pkttype;
866 const uint8_t *buf_temp = buf;
869 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
876 if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
881 pkttype = bytestream_get_byte(&buf_temp);
882 pktsize = bytestream_get_be24(&buf_temp);
883 ts = bytestream_get_be24(&buf_temp);
884 ts |= bytestream_get_byte(&buf_temp) << 24;
885 bytestream_get_be24(&buf_temp);
887 rt->flv_size = pktsize;
889 //force 12bytes header
890 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
891 pkttype == RTMP_PT_NOTIFY) {
892 if (pkttype == RTMP_PT_NOTIFY)
894 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
897 //this can be a big packet, it's better to send it right here
898 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
899 rt->out_pkt.extra = rt->main_channel_id;
900 rt->flv_data = rt->out_pkt.data;
902 if (pkttype == RTMP_PT_NOTIFY)
903 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
906 if (rt->flv_size - rt->flv_off > size_temp) {
907 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
908 rt->flv_off += size_temp;
910 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
911 rt->flv_off += rt->flv_size - rt->flv_off;
914 if (rt->flv_off == rt->flv_size) {
915 bytestream_get_be32(&buf_temp);
917 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
918 ff_rtmp_packet_destroy(&rt->out_pkt);
922 } while (buf_temp - buf < size_temp);
926 URLProtocol rtmp_protocol = {