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)
58 /** protocol handler context */
59 typedef struct RTMPContext {
60 URLContext* stream; ///< TCP stream used in interactions with RTMP server
61 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
62 int chunk_size; ///< size of the chunks RTMP packets are divided into
63 int is_input; ///< input/output flag
64 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
65 char app[128]; ///< application
66 ClientState state; ///< current state
67 int main_channel_id; ///< an additional channel ID which is used for some invocations
68 uint8_t* flv_data; ///< buffer with data for demuxer
69 int flv_size; ///< current buffer size
70 int flv_off; ///< number of bytes read from current buffer
71 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
74 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
75 /** Client key used for digest signing */
76 static const uint8_t rtmp_player_key[] = {
77 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
78 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
80 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
81 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
82 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
85 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
86 /** Key used for RTMP server digest signing */
87 static const uint8_t rtmp_server_key[] = {
88 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
89 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
90 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
92 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
93 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
94 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
98 * Generates 'connect' call and sends it to the server.
100 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
101 const char *host, int port)
107 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
110 snprintf(tcurl, sizeof(tcurl), "%s://%s:%d/%s", proto, host, port, rt->app);
111 ff_amf_write_string(&p, "connect");
112 ff_amf_write_number(&p, 1.0);
113 ff_amf_write_object_start(&p);
114 ff_amf_write_field_name(&p, "app");
115 ff_amf_write_string(&p, rt->app);
118 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
119 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
121 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
122 ff_amf_write_field_name(&p, "type");
123 ff_amf_write_string(&p, "nonprivate");
125 ff_amf_write_field_name(&p, "flashVer");
126 ff_amf_write_string(&p, ver);
127 ff_amf_write_field_name(&p, "tcUrl");
128 ff_amf_write_string(&p, tcurl);
130 ff_amf_write_field_name(&p, "fpad");
131 ff_amf_write_bool(&p, 0);
132 ff_amf_write_field_name(&p, "capabilities");
133 ff_amf_write_number(&p, 15.0);
134 ff_amf_write_field_name(&p, "audioCodecs");
135 ff_amf_write_number(&p, 1639.0);
136 ff_amf_write_field_name(&p, "videoCodecs");
137 ff_amf_write_number(&p, 252.0);
138 ff_amf_write_field_name(&p, "videoFunction");
139 ff_amf_write_number(&p, 1.0);
141 ff_amf_write_object_end(&p);
143 pkt.data_size = p - pkt.data;
145 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
149 * Generates 'releaseStream' call and sends it to the server. It should make
150 * the server release some channel for media streams.
152 static void gen_release_stream(URLContext *s, RTMPContext *rt)
157 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
158 29 + strlen(rt->playpath));
160 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n");
162 ff_amf_write_string(&p, "releaseStream");
163 ff_amf_write_number(&p, 2.0);
164 ff_amf_write_null(&p);
165 ff_amf_write_string(&p, rt->playpath);
167 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
168 ff_rtmp_packet_destroy(&pkt);
172 * Generates 'FCPublish' call and sends it to the server. It should make
173 * the server preapare for receiving media streams.
175 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
180 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
181 25 + strlen(rt->playpath));
183 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n");
185 ff_amf_write_string(&p, "FCPublish");
186 ff_amf_write_number(&p, 3.0);
187 ff_amf_write_null(&p);
188 ff_amf_write_string(&p, rt->playpath);
190 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
191 ff_rtmp_packet_destroy(&pkt);
195 * Generates 'FCUnpublish' call and sends it to the server. It should make
196 * the server destroy stream.
198 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
203 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
204 27 + strlen(rt->playpath));
206 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n");
208 ff_amf_write_string(&p, "FCUnpublish");
209 ff_amf_write_number(&p, 5.0);
210 ff_amf_write_null(&p);
211 ff_amf_write_string(&p, rt->playpath);
213 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
214 ff_rtmp_packet_destroy(&pkt);
218 * Generates 'createStream' call and sends it to the server. It should make
219 * the server allocate some channel for media streams.
221 static void gen_create_stream(URLContext *s, RTMPContext *rt)
226 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n");
227 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
230 ff_amf_write_string(&p, "createStream");
231 ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
232 ff_amf_write_null(&p);
234 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
235 ff_rtmp_packet_destroy(&pkt);
240 * Generates 'deleteStream' call and sends it to the server. It should make
241 * the server remove some channel for media streams.
243 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
248 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n");
249 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
252 ff_amf_write_string(&p, "deleteStream");
253 ff_amf_write_number(&p, 0.0);
254 ff_amf_write_null(&p);
255 ff_amf_write_number(&p, rt->main_channel_id);
257 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
258 ff_rtmp_packet_destroy(&pkt);
262 * Generates 'play' call and sends it to the server, then pings the server
263 * to start actual playing.
265 static void gen_play(URLContext *s, RTMPContext *rt)
270 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
271 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
272 20 + strlen(rt->playpath));
273 pkt.extra = rt->main_channel_id;
276 ff_amf_write_string(&p, "play");
277 ff_amf_write_number(&p, 0.0);
278 ff_amf_write_null(&p);
279 ff_amf_write_string(&p, rt->playpath);
281 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
282 ff_rtmp_packet_destroy(&pkt);
284 // set client buffer time disguised in ping packet
285 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
288 bytestream_put_be16(&p, 3);
289 bytestream_put_be32(&p, 1);
290 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
292 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
293 ff_rtmp_packet_destroy(&pkt);
297 * Generates 'publish' call and sends it to the server.
299 static void gen_publish(URLContext *s, RTMPContext *rt)
304 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
305 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
306 30 + strlen(rt->playpath));
307 pkt.extra = rt->main_channel_id;
310 ff_amf_write_string(&p, "publish");
311 ff_amf_write_number(&p, 0.0);
312 ff_amf_write_null(&p);
313 ff_amf_write_string(&p, rt->playpath);
314 ff_amf_write_string(&p, "live");
316 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
317 ff_rtmp_packet_destroy(&pkt);
321 * Generates ping reply and sends it to the server.
323 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
328 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
330 bytestream_put_be16(&p, 7);
331 bytestream_put_be32(&p, AV_RB32(ppkt->data+2) + 1);
332 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
333 ff_rtmp_packet_destroy(&pkt);
336 //TODO: Move HMAC code somewhere. Eventually.
337 #define HMAC_IPAD_VAL 0x36
338 #define HMAC_OPAD_VAL 0x5C
341 * Calculates HMAC-SHA2 digest for RTMP handshake packets.
343 * @param src input buffer
344 * @param len input buffer length (should be 1536)
345 * @param gap offset in buffer where 32 bytes should not be taken into account
346 * when calculating digest (since it will be used to store that digest)
347 * @param key digest key
348 * @param keylen digest key length
349 * @param dst buffer where calculated digest will be stored (32 bytes)
351 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
352 const uint8_t *key, int keylen, uint8_t *dst)
355 uint8_t hmac_buf[64+32] = {0};
358 sha = av_mallocz(av_sha_size);
361 memcpy(hmac_buf, key, keylen);
363 av_sha_init(sha, 256);
364 av_sha_update(sha,key, keylen);
365 av_sha_final(sha, hmac_buf);
367 for (i = 0; i < 64; i++)
368 hmac_buf[i] ^= HMAC_IPAD_VAL;
370 av_sha_init(sha, 256);
371 av_sha_update(sha, hmac_buf, 64);
373 av_sha_update(sha, src, len);
374 } else { //skip 32 bytes used for storing digest
375 av_sha_update(sha, src, gap);
376 av_sha_update(sha, src + gap + 32, len - gap - 32);
378 av_sha_final(sha, hmac_buf + 64);
380 for (i = 0; i < 64; i++)
381 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
382 av_sha_init(sha, 256);
383 av_sha_update(sha, hmac_buf, 64+32);
384 av_sha_final(sha, dst);
390 * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest
391 * will be stored) into that packet.
393 * @param buf handshake data (1536 bytes)
394 * @return offset to the digest inside input data
396 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
398 int i, digest_pos = 0;
400 for (i = 8; i < 12; i++)
401 digest_pos += buf[i];
402 digest_pos = (digest_pos % 728) + 12;
404 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
405 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
411 * Verifies that the received server response has the expected digest value.
413 * @param buf handshake data received from the server (1536 bytes)
414 * @param off position to search digest offset from
415 * @return 0 if digest is valid, digest position otherwise
417 static int rtmp_validate_digest(uint8_t *buf, int off)
419 int i, digest_pos = 0;
422 for (i = 0; i < 4; i++)
423 digest_pos += buf[i + off];
424 digest_pos = (digest_pos % 728) + off + 4;
426 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
427 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
429 if (!memcmp(digest, buf + digest_pos, 32))
435 * Performs handshake with the server by means of exchanging pseudorandom data
436 * signed with HMAC-SHA2 digest.
438 * @return 0 if handshake succeeds, negative value otherwise
440 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
443 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
444 3, // unencrypted data
445 0, 0, 0, 0, // client uptime
451 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
452 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
454 int server_pos, client_pos;
457 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n");
459 av_lfg_init(&rnd, 0xDEADC0DE);
460 // generate handshake packet - 1536 bytes of pseudorandom data
461 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
462 tosend[i] = av_lfg_get(&rnd) >> 24;
463 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
465 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
466 i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
467 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
468 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
471 i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
472 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
473 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
477 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
478 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
481 server_pos = rtmp_validate_digest(serverdata + 1, 772);
483 server_pos = rtmp_validate_digest(serverdata + 1, 8);
485 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n");
490 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
491 rtmp_server_key, sizeof(rtmp_server_key),
493 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
496 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
497 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n");
501 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
502 tosend[i] = av_lfg_get(&rnd) >> 24;
503 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
504 rtmp_player_key, sizeof(rtmp_player_key),
506 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
508 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
510 // write reply back to the server
511 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
513 url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
520 * Parses received packet and may perform some action depending on
521 * the packet contents.
522 * @return 0 for no errors, negative values for serious errors which prevent
523 * further communications, positive values for uncritical errors
525 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
528 const uint8_t *data_end = pkt->data + pkt->data_size;
531 case RTMP_PT_CHUNK_SIZE:
532 if (pkt->data_size != 4) {
533 av_log(LOG_CONTEXT, AV_LOG_ERROR,
534 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
538 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
539 rt->chunk_size = AV_RB32(pkt->data);
540 if (rt->chunk_size <= 0) {
541 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
544 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
547 t = AV_RB16(pkt->data);
549 gen_pong(s, rt, pkt);
552 //TODO: check for the messages sent for wrong state?
553 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
556 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
557 "description", tmpstr, sizeof(tmpstr)))
558 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
560 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
562 case STATE_HANDSHAKED:
564 gen_release_stream(s, rt);
565 gen_fcpublish_stream(s, rt);
566 rt->state = STATE_RELEASING;
568 rt->state = STATE_CONNECTING;
570 gen_create_stream(s, rt);
572 case STATE_FCPUBLISH:
573 rt->state = STATE_CONNECTING;
575 case STATE_RELEASING:
576 rt->state = STATE_FCPUBLISH;
577 /* hack for Wowza Media Server, it does not send result for
578 * releaseStream and FCPublish calls */
579 if (!pkt->data[10]) {
580 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
582 rt->state = STATE_CONNECTING;
584 if (rt->state != STATE_CONNECTING)
586 case STATE_CONNECTING:
587 //extract a number from the result
588 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
589 av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n");
591 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
598 rt->state = STATE_READY;
601 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
602 const uint8_t* ptr = pkt->data + 11;
605 for (i = 0; i < 2; i++) {
606 t = ff_amf_tag_size(ptr, data_end);
611 t = ff_amf_get_field_value(ptr, data_end,
612 "level", tmpstr, sizeof(tmpstr));
613 if (!t && !strcmp(tmpstr, "error")) {
614 if (!ff_amf_get_field_value(ptr, data_end,
615 "description", tmpstr, sizeof(tmpstr)))
616 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
619 t = ff_amf_get_field_value(ptr, data_end,
620 "code", tmpstr, sizeof(tmpstr));
621 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
622 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
630 * Interacts with the server by receiving and sending RTMP packets until
631 * there is some significant data (media data or expected status notification).
633 * @param s reading context
634 * @param for_header non-zero value tells function to work until it
635 * gets notification from the server that playing has been started,
636 * otherwise function will work until some media data is received (or
638 * @return 0 for successful operation, negative value in case of error
640 static int get_packet(URLContext *s, int for_header)
642 RTMPContext *rt = s->priv_data;
647 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
648 rt->chunk_size, rt->prev_pkt[0])) != 0) {
650 return AVERROR(EAGAIN);
656 ret = rtmp_parse_result(s, rt, &rpkt);
657 if (ret < 0) {//serious error in current packet
658 ff_rtmp_packet_destroy(&rpkt);
661 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
662 ff_rtmp_packet_destroy(&rpkt);
665 if (!rpkt.data_size || !rt->is_input) {
666 ff_rtmp_packet_destroy(&rpkt);
669 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
670 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
672 uint32_t ts = rpkt.timestamp;
674 // generate packet header and put data into buffer for FLV demuxer
676 rt->flv_size = rpkt.data_size + 15;
677 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
678 bytestream_put_byte(&p, rpkt.type);
679 bytestream_put_be24(&p, rpkt.data_size);
680 bytestream_put_be24(&p, ts);
681 bytestream_put_byte(&p, ts >> 24);
682 bytestream_put_be24(&p, 0);
683 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
684 bytestream_put_be32(&p, 0);
685 ff_rtmp_packet_destroy(&rpkt);
687 } else if (rpkt.type == RTMP_PT_METADATA) {
688 // we got raw FLV data, make it available for FLV demuxer
690 rt->flv_size = rpkt.data_size;
691 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
692 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
693 ff_rtmp_packet_destroy(&rpkt);
696 ff_rtmp_packet_destroy(&rpkt);
701 static int rtmp_close(URLContext *h)
703 RTMPContext *rt = h->priv_data;
707 if (rt->out_pkt.data_size)
708 ff_rtmp_packet_destroy(&rt->out_pkt);
709 if (rt->state > STATE_FCPUBLISH)
710 gen_fcunpublish_stream(h, rt);
712 if (rt->state > STATE_HANDSHAKED)
713 gen_delete_stream(h, rt);
715 av_freep(&rt->flv_data);
716 url_close(rt->stream);
722 * Opens RTMP connection and verifies that the stream can be played.
724 * URL syntax: rtmp://server[:port][/app][/playpath]
725 * where 'app' is first one or two directories in the path
726 * (e.g. /ondemand/, /flash/live/, etc.)
727 * and 'playpath' is a file name (the rest of the path,
728 * may be prefixed with "mp4:")
730 static int rtmp_open(URLContext *s, const char *uri, int flags)
733 char proto[8], hostname[256], path[1024], *fname;
738 rt = av_mallocz(sizeof(RTMPContext));
740 return AVERROR(ENOMEM);
742 rt->is_input = !(flags & URL_WRONLY);
744 url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
745 path, sizeof(path), s->filename);
748 port = RTMP_DEFAULT_PORT;
749 snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port);
751 if (url_open(&rt->stream, buf, URL_RDWR) < 0) {
752 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);
756 rt->state = STATE_START;
757 if (rtmp_handshake(s, rt))
760 rt->chunk_size = 128;
761 rt->state = STATE_HANDSHAKED;
762 //extract "app" part from path
763 if (!strncmp(path, "/ondemand/", 10)) {
765 memcpy(rt->app, "ondemand", 9);
767 char *p = strchr(path + 1, '/');
772 char *c = strchr(p + 1, ':');
773 fname = strchr(p + 1, '/');
774 if (!fname || c < fname) {
776 av_strlcpy(rt->app, path + 1, p - path);
779 av_strlcpy(rt->app, path + 1, fname - path - 1);
783 if (!strchr(fname, ':') &&
784 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
785 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
786 memcpy(rt->playpath, "mp4:", 5);
790 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
792 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
793 proto, path, rt->app, rt->playpath);
794 gen_connect(s, rt, proto, hostname, port);
797 ret = get_packet(s, 1);
798 } while (ret == EAGAIN);
803 // generate FLV header for demuxer
805 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
807 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
814 s->max_packet_size = url_get_max_packet_size(rt->stream);
823 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
825 RTMPContext *rt = s->priv_data;
826 int orig_size = size;
830 int data_left = rt->flv_size - rt->flv_off;
832 if (data_left >= size) {
833 memcpy(buf, rt->flv_data + rt->flv_off, size);
838 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
841 rt->flv_off = rt->flv_size;
843 if ((ret = get_packet(s, 0)) < 0)
849 static int rtmp_write(URLContext *h, uint8_t *buf, int size)
851 RTMPContext *rt = h->priv_data;
852 int size_temp = size;
853 int pktsize, pkttype;
855 const uint8_t *buf_temp = buf;
858 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
865 if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
870 pkttype = bytestream_get_byte(&buf_temp);
871 pktsize = bytestream_get_be24(&buf_temp);
872 ts = bytestream_get_be24(&buf_temp);
873 ts |= bytestream_get_byte(&buf_temp) << 24;
874 bytestream_get_be24(&buf_temp);
876 rt->flv_size = pktsize;
878 //force 12bytes header
879 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
880 pkttype == RTMP_PT_NOTIFY) {
881 if (pkttype == RTMP_PT_NOTIFY)
883 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
886 //this can be a big packet, it's better to send it right here
887 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
888 rt->out_pkt.extra = rt->main_channel_id;
889 rt->flv_data = rt->out_pkt.data;
891 if (pkttype == RTMP_PT_NOTIFY)
892 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
895 if (rt->flv_size - rt->flv_off > size_temp) {
896 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
897 rt->flv_off += size_temp;
899 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
900 rt->flv_off += rt->flv_size - rt->flv_off;
903 if (rt->flv_off == rt->flv_size) {
904 bytestream_get_be32(&buf_temp);
906 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
907 ff_rtmp_packet_destroy(&rt->out_pkt);
911 } while (buf_temp - buf < size_temp);
915 URLProtocol rtmp_protocol = {