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_readwrite.h"
30 #include "libavutil/lfg.h"
31 #include "libavutil/sha.h"
44 /** RTMP protocol handler state */
46 STATE_START, ///< client has not done anything yet
47 STATE_HANDSHAKED, ///< client has performed handshake
48 STATE_RELEASING, ///< client releasing stream before publish it (for output)
49 STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
50 STATE_CONNECTING, ///< client connected to server successfully
51 STATE_READY, ///< client has sent all needed commands and waits for server reply
52 STATE_PLAYING, ///< client has started receiving multimedia data from server
53 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
54 STATE_STOPPED, ///< the broadcast has been stopped
57 /** protocol handler context */
58 typedef struct RTMPContext {
59 URLContext* stream; ///< TCP stream used in interactions with RTMP server
60 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
61 int chunk_size; ///< size of the chunks RTMP packets are divided into
62 int is_input; ///< input/output flag
63 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
64 char app[128]; ///< application
65 ClientState state; ///< current state
66 int main_channel_id; ///< an additional channel ID which is used for some invocations
67 uint8_t* flv_data; ///< buffer with data for demuxer
68 int flv_size; ///< current buffer size
69 int flv_off; ///< number of bytes read from current buffer
70 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
71 uint32_t client_report_size; ///< number of bytes after which client should report to server
72 uint32_t bytes_read; ///< number of bytes read from server
73 uint32_t last_bytes_read; ///< number of bytes read last reported to server
76 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
77 /** Client key used for digest signing */
78 static const uint8_t rtmp_player_key[] = {
79 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
80 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
82 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
83 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
84 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
87 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
88 /** Key used for RTMP server digest signing */
89 static const uint8_t rtmp_server_key[] = {
90 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
91 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
92 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
94 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
95 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
96 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
100 * Generate 'connect' call and send it to the server.
102 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
103 const char *host, int port)
109 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
112 ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
113 ff_amf_write_string(&p, "connect");
114 ff_amf_write_number(&p, 1.0);
115 ff_amf_write_object_start(&p);
116 ff_amf_write_field_name(&p, "app");
117 ff_amf_write_string(&p, rt->app);
120 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
121 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
123 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
124 ff_amf_write_field_name(&p, "type");
125 ff_amf_write_string(&p, "nonprivate");
127 ff_amf_write_field_name(&p, "flashVer");
128 ff_amf_write_string(&p, ver);
129 ff_amf_write_field_name(&p, "tcUrl");
130 ff_amf_write_string(&p, tcurl);
132 ff_amf_write_field_name(&p, "fpad");
133 ff_amf_write_bool(&p, 0);
134 ff_amf_write_field_name(&p, "capabilities");
135 ff_amf_write_number(&p, 15.0);
136 ff_amf_write_field_name(&p, "audioCodecs");
137 ff_amf_write_number(&p, 1639.0);
138 ff_amf_write_field_name(&p, "videoCodecs");
139 ff_amf_write_number(&p, 252.0);
140 ff_amf_write_field_name(&p, "videoFunction");
141 ff_amf_write_number(&p, 1.0);
143 ff_amf_write_object_end(&p);
145 pkt.data_size = p - pkt.data;
147 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
148 ff_rtmp_packet_destroy(&pkt);
152 * Generate 'releaseStream' call and send it to the server. It should make
153 * the server release some channel for media streams.
155 static void gen_release_stream(URLContext *s, RTMPContext *rt)
160 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
161 29 + strlen(rt->playpath));
163 av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
165 ff_amf_write_string(&p, "releaseStream");
166 ff_amf_write_number(&p, 2.0);
167 ff_amf_write_null(&p);
168 ff_amf_write_string(&p, rt->playpath);
170 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
171 ff_rtmp_packet_destroy(&pkt);
175 * Generate 'FCPublish' call and send it to the server. It should make
176 * the server preapare for receiving media streams.
178 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
183 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
184 25 + strlen(rt->playpath));
186 av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
188 ff_amf_write_string(&p, "FCPublish");
189 ff_amf_write_number(&p, 3.0);
190 ff_amf_write_null(&p);
191 ff_amf_write_string(&p, rt->playpath);
193 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
194 ff_rtmp_packet_destroy(&pkt);
198 * Generate 'FCUnpublish' call and send it to the server. It should make
199 * the server destroy stream.
201 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
206 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
207 27 + strlen(rt->playpath));
209 av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
211 ff_amf_write_string(&p, "FCUnpublish");
212 ff_amf_write_number(&p, 5.0);
213 ff_amf_write_null(&p);
214 ff_amf_write_string(&p, rt->playpath);
216 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
217 ff_rtmp_packet_destroy(&pkt);
221 * Generate 'createStream' call and send it to the server. It should make
222 * the server allocate some channel for media streams.
224 static void gen_create_stream(URLContext *s, RTMPContext *rt)
229 av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
230 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
233 ff_amf_write_string(&p, "createStream");
234 ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
235 ff_amf_write_null(&p);
237 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
238 ff_rtmp_packet_destroy(&pkt);
243 * Generate 'deleteStream' call and send it to the server. It should make
244 * the server remove some channel for media streams.
246 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
251 av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
252 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
255 ff_amf_write_string(&p, "deleteStream");
256 ff_amf_write_number(&p, 0.0);
257 ff_amf_write_null(&p);
258 ff_amf_write_number(&p, rt->main_channel_id);
260 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
261 ff_rtmp_packet_destroy(&pkt);
265 * Generate 'play' call and send it to the server, then ping the server
266 * to start actual playing.
268 static void gen_play(URLContext *s, RTMPContext *rt)
273 av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
274 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
275 20 + strlen(rt->playpath));
276 pkt.extra = rt->main_channel_id;
279 ff_amf_write_string(&p, "play");
280 ff_amf_write_number(&p, 0.0);
281 ff_amf_write_null(&p);
282 ff_amf_write_string(&p, rt->playpath);
284 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
285 ff_rtmp_packet_destroy(&pkt);
287 // set client buffer time disguised in ping packet
288 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
291 bytestream_put_be16(&p, 3);
292 bytestream_put_be32(&p, 1);
293 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
295 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
296 ff_rtmp_packet_destroy(&pkt);
300 * Generate 'publish' call and send it to the server.
302 static void gen_publish(URLContext *s, RTMPContext *rt)
307 av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
308 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
309 30 + strlen(rt->playpath));
310 pkt.extra = rt->main_channel_id;
313 ff_amf_write_string(&p, "publish");
314 ff_amf_write_number(&p, 0.0);
315 ff_amf_write_null(&p);
316 ff_amf_write_string(&p, rt->playpath);
317 ff_amf_write_string(&p, "live");
319 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
320 ff_rtmp_packet_destroy(&pkt);
324 * Generate ping reply and send it to the server.
326 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
331 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
333 bytestream_put_be16(&p, 7);
334 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
335 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
336 ff_rtmp_packet_destroy(&pkt);
340 * Generate report on bytes read so far and send it to the server.
342 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
347 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
349 bytestream_put_be32(&p, rt->bytes_read);
350 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
351 ff_rtmp_packet_destroy(&pkt);
354 //TODO: Move HMAC code somewhere. Eventually.
355 #define HMAC_IPAD_VAL 0x36
356 #define HMAC_OPAD_VAL 0x5C
359 * Calculate HMAC-SHA2 digest for RTMP handshake packets.
361 * @param src input buffer
362 * @param len input buffer length (should be 1536)
363 * @param gap offset in buffer where 32 bytes should not be taken into account
364 * when calculating digest (since it will be used to store that digest)
365 * @param key digest key
366 * @param keylen digest key length
367 * @param dst buffer where calculated digest will be stored (32 bytes)
369 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
370 const uint8_t *key, int keylen, uint8_t *dst)
373 uint8_t hmac_buf[64+32] = {0};
376 sha = av_mallocz(av_sha_size);
379 memcpy(hmac_buf, key, keylen);
381 av_sha_init(sha, 256);
382 av_sha_update(sha,key, keylen);
383 av_sha_final(sha, hmac_buf);
385 for (i = 0; i < 64; i++)
386 hmac_buf[i] ^= HMAC_IPAD_VAL;
388 av_sha_init(sha, 256);
389 av_sha_update(sha, hmac_buf, 64);
391 av_sha_update(sha, src, len);
392 } else { //skip 32 bytes used for storing digest
393 av_sha_update(sha, src, gap);
394 av_sha_update(sha, src + gap + 32, len - gap - 32);
396 av_sha_final(sha, hmac_buf + 64);
398 for (i = 0; i < 64; i++)
399 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
400 av_sha_init(sha, 256);
401 av_sha_update(sha, hmac_buf, 64+32);
402 av_sha_final(sha, dst);
408 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
409 * will be stored) into that packet.
411 * @param buf handshake data (1536 bytes)
412 * @return offset to the digest inside input data
414 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
416 int i, digest_pos = 0;
418 for (i = 8; i < 12; i++)
419 digest_pos += buf[i];
420 digest_pos = (digest_pos % 728) + 12;
422 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
423 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
429 * Verify that the received server response has the expected digest value.
431 * @param buf handshake data received from the server (1536 bytes)
432 * @param off position to search digest offset from
433 * @return 0 if digest is valid, digest position otherwise
435 static int rtmp_validate_digest(uint8_t *buf, int off)
437 int i, digest_pos = 0;
440 for (i = 0; i < 4; i++)
441 digest_pos += buf[i + off];
442 digest_pos = (digest_pos % 728) + off + 4;
444 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
445 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
447 if (!memcmp(digest, buf + digest_pos, 32))
453 * Perform handshake with the server by means of exchanging pseudorandom data
454 * signed with HMAC-SHA2 digest.
456 * @return 0 if handshake succeeds, negative value otherwise
458 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
461 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
462 3, // unencrypted data
463 0, 0, 0, 0, // client uptime
469 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
470 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
472 int server_pos, client_pos;
475 av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
477 av_lfg_init(&rnd, 0xDEADC0DE);
478 // generate handshake packet - 1536 bytes of pseudorandom data
479 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
480 tosend[i] = av_lfg_get(&rnd) >> 24;
481 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
483 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
484 i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
485 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
486 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
489 i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
490 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
491 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
495 av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
496 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
498 if (rt->is_input && serverdata[5] >= 3) {
499 server_pos = rtmp_validate_digest(serverdata + 1, 772);
501 server_pos = rtmp_validate_digest(serverdata + 1, 8);
503 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
508 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
509 rtmp_server_key, sizeof(rtmp_server_key),
511 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
514 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
515 av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
519 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
520 tosend[i] = av_lfg_get(&rnd) >> 24;
521 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
522 rtmp_player_key, sizeof(rtmp_player_key),
524 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
526 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
528 // write reply back to the server
529 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
531 ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
538 * Parse received packet and possibly perform some action depending on
539 * the packet contents.
540 * @return 0 for no errors, negative values for serious errors which prevent
541 * further communications, positive values for uncritical errors
543 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
546 const uint8_t *data_end = pkt->data + pkt->data_size;
549 ff_rtmp_packet_dump(s, pkt);
553 case RTMP_PT_CHUNK_SIZE:
554 if (pkt->data_size != 4) {
555 av_log(s, AV_LOG_ERROR,
556 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
560 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
561 rt->chunk_size = AV_RB32(pkt->data);
562 if (rt->chunk_size <= 0) {
563 av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
566 av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
569 t = AV_RB16(pkt->data);
571 gen_pong(s, rt, pkt);
573 case RTMP_PT_CLIENT_BW:
574 if (pkt->data_size < 4) {
575 av_log(s, AV_LOG_ERROR,
576 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
580 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
581 rt->client_report_size = AV_RB32(pkt->data) >> 1;
584 //TODO: check for the messages sent for wrong state?
585 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
588 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
589 "description", tmpstr, sizeof(tmpstr)))
590 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
592 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
594 case STATE_HANDSHAKED:
596 gen_release_stream(s, rt);
597 gen_fcpublish_stream(s, rt);
598 rt->state = STATE_RELEASING;
600 rt->state = STATE_CONNECTING;
602 gen_create_stream(s, rt);
604 case STATE_FCPUBLISH:
605 rt->state = STATE_CONNECTING;
607 case STATE_RELEASING:
608 rt->state = STATE_FCPUBLISH;
609 /* hack for Wowza Media Server, it does not send result for
610 * releaseStream and FCPublish calls */
611 if (!pkt->data[10]) {
612 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
614 rt->state = STATE_CONNECTING;
616 if (rt->state != STATE_CONNECTING)
618 case STATE_CONNECTING:
619 //extract a number from the result
620 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
621 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
623 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
630 rt->state = STATE_READY;
633 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
634 const uint8_t* ptr = pkt->data + 11;
637 for (i = 0; i < 2; i++) {
638 t = ff_amf_tag_size(ptr, data_end);
643 t = ff_amf_get_field_value(ptr, data_end,
644 "level", tmpstr, sizeof(tmpstr));
645 if (!t && !strcmp(tmpstr, "error")) {
646 if (!ff_amf_get_field_value(ptr, data_end,
647 "description", tmpstr, sizeof(tmpstr)))
648 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
651 t = ff_amf_get_field_value(ptr, data_end,
652 "code", tmpstr, sizeof(tmpstr));
653 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
654 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
655 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
656 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
664 * Interact with the server by receiving and sending RTMP packets until
665 * there is some significant data (media data or expected status notification).
667 * @param s reading context
668 * @param for_header non-zero value tells function to work until it
669 * gets notification from the server that playing has been started,
670 * otherwise function will work until some media data is received (or
672 * @return 0 for successful operation, negative value in case of error
674 static int get_packet(URLContext *s, int for_header)
676 RTMPContext *rt = s->priv_data;
681 uint32_t ts, cts, pts=0;
683 if (rt->state == STATE_STOPPED)
687 RTMPPacket rpkt = { 0 };
688 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
689 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
691 return AVERROR(EAGAIN);
696 rt->bytes_read += ret;
697 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
698 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
699 gen_bytes_read(s, rt, rpkt.timestamp + 1);
700 rt->last_bytes_read = rt->bytes_read;
703 ret = rtmp_parse_result(s, rt, &rpkt);
704 if (ret < 0) {//serious error in current packet
705 ff_rtmp_packet_destroy(&rpkt);
708 if (rt->state == STATE_STOPPED) {
709 ff_rtmp_packet_destroy(&rpkt);
712 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
713 ff_rtmp_packet_destroy(&rpkt);
716 if (!rpkt.data_size || !rt->is_input) {
717 ff_rtmp_packet_destroy(&rpkt);
720 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
721 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
724 // generate packet header and put data into buffer for FLV demuxer
726 rt->flv_size = rpkt.data_size + 15;
727 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
728 bytestream_put_byte(&p, rpkt.type);
729 bytestream_put_be24(&p, rpkt.data_size);
730 bytestream_put_be24(&p, ts);
731 bytestream_put_byte(&p, ts >> 24);
732 bytestream_put_be24(&p, 0);
733 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
734 bytestream_put_be32(&p, 0);
735 ff_rtmp_packet_destroy(&rpkt);
737 } else if (rpkt.type == RTMP_PT_METADATA) {
738 // we got raw FLV data, make it available for FLV demuxer
740 rt->flv_size = rpkt.data_size;
741 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
742 /* rewrite timestamps */
745 while (next - rpkt.data < rpkt.data_size - 11) {
747 data_size = bytestream_get_be24(&next);
749 cts = bytestream_get_be24(&next);
750 cts |= bytestream_get_byte(&next) << 24;
755 bytestream_put_be24(&p, ts);
756 bytestream_put_byte(&p, ts >> 24);
757 next += data_size + 3 + 4;
759 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
760 ff_rtmp_packet_destroy(&rpkt);
763 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,