2 * RTMP network protocol
3 * Copyright (c) 2009 Kostya Shishkov
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27 #include "libavcodec/bytestream.h"
28 #include "libavutil/avstring.h"
29 #include "libavutil/lfg.h"
30 #include "libavutil/sha.h"
40 /* we can't use av_log() with URLContext yet... */
41 #if LIBAVFORMAT_VERSION_MAJOR < 53
42 #define LOG_CONTEXT NULL
49 /** RTMP protocol handler state */
51 STATE_START, ///< client has not done anything yet
52 STATE_HANDSHAKED, ///< client has performed handshake
53 STATE_RELEASING, ///< client releasing stream before publish it (for output)
54 STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
55 STATE_CONNECTING, ///< client connected to server successfully
56 STATE_READY, ///< client has sent all needed commands and waits for server reply
57 STATE_PLAYING, ///< client has started receiving multimedia data from server
58 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
59 STATE_STOPPED, ///< the broadcast has been stopped
62 /** protocol handler context */
63 typedef struct RTMPContext {
64 URLContext* stream; ///< TCP stream used in interactions with RTMP server
65 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
66 int chunk_size; ///< size of the chunks RTMP packets are divided into
67 int is_input; ///< input/output flag
68 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
69 char app[128]; ///< application
70 ClientState state; ///< current state
71 int main_channel_id; ///< an additional channel ID which is used for some invocations
72 uint8_t* flv_data; ///< buffer with data for demuxer
73 int flv_size; ///< current buffer size
74 int flv_off; ///< number of bytes read from current buffer
75 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
76 uint32_t client_report_size; ///< number of bytes after which client should report to server
77 uint32_t bytes_read; ///< number of bytes read from server
78 uint32_t last_bytes_read; ///< number of bytes read last reported to server
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, 1.0);
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(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n");
170 ff_amf_write_string(&p, "releaseStream");
171 ff_amf_write_number(&p, 2.0);
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(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n");
193 ff_amf_write_string(&p, "FCPublish");
194 ff_amf_write_number(&p, 3.0);
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(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n");
216 ff_amf_write_string(&p, "FCUnpublish");
217 ff_amf_write_number(&p, 5.0);
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(LOG_CONTEXT, 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->is_input ? 3.0 : 4.0);
240 ff_amf_write_null(&p);
242 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
243 ff_rtmp_packet_destroy(&pkt);
248 * Generate 'deleteStream' call and send it to the server. It should make
249 * the server remove some channel for media streams.
251 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
256 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n");
257 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
260 ff_amf_write_string(&p, "deleteStream");
261 ff_amf_write_number(&p, 0.0);
262 ff_amf_write_null(&p);
263 ff_amf_write_number(&p, rt->main_channel_id);
265 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
266 ff_rtmp_packet_destroy(&pkt);
270 * Generate 'play' call and send it to the server, then ping the server
271 * to start actual playing.
273 static void gen_play(URLContext *s, RTMPContext *rt)
278 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
279 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
280 20 + strlen(rt->playpath));
281 pkt.extra = rt->main_channel_id;
284 ff_amf_write_string(&p, "play");
285 ff_amf_write_number(&p, 0.0);
286 ff_amf_write_null(&p);
287 ff_amf_write_string(&p, rt->playpath);
289 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
290 ff_rtmp_packet_destroy(&pkt);
292 // set client buffer time disguised in ping packet
293 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
296 bytestream_put_be16(&p, 3);
297 bytestream_put_be32(&p, 1);
298 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
300 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
301 ff_rtmp_packet_destroy(&pkt);
305 * Generate 'publish' call and send it to the server.
307 static void gen_publish(URLContext *s, RTMPContext *rt)
312 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
313 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
314 30 + strlen(rt->playpath));
315 pkt.extra = rt->main_channel_id;
318 ff_amf_write_string(&p, "publish");
319 ff_amf_write_number(&p, 0.0);
320 ff_amf_write_null(&p);
321 ff_amf_write_string(&p, rt->playpath);
322 ff_amf_write_string(&p, "live");
324 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
325 ff_rtmp_packet_destroy(&pkt);
329 * Generate ping reply and send it to the server.
331 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
336 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
338 bytestream_put_be16(&p, 7);
339 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
340 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
341 ff_rtmp_packet_destroy(&pkt);
345 * Generate report on bytes read so far and send it to the server.
347 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
352 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
354 bytestream_put_be32(&p, rt->bytes_read);
355 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
356 ff_rtmp_packet_destroy(&pkt);
359 //TODO: Move HMAC code somewhere. Eventually.
360 #define HMAC_IPAD_VAL 0x36
361 #define HMAC_OPAD_VAL 0x5C
364 * Calculate HMAC-SHA2 digest for RTMP handshake packets.
366 * @param src input buffer
367 * @param len input buffer length (should be 1536)
368 * @param gap offset in buffer where 32 bytes should not be taken into account
369 * when calculating digest (since it will be used to store that digest)
370 * @param key digest key
371 * @param keylen digest key length
372 * @param dst buffer where calculated digest will be stored (32 bytes)
374 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
375 const uint8_t *key, int keylen, uint8_t *dst)
378 uint8_t hmac_buf[64+32] = {0};
381 sha = av_mallocz(av_sha_size);
384 memcpy(hmac_buf, key, keylen);
386 av_sha_init(sha, 256);
387 av_sha_update(sha,key, keylen);
388 av_sha_final(sha, hmac_buf);
390 for (i = 0; i < 64; i++)
391 hmac_buf[i] ^= HMAC_IPAD_VAL;
393 av_sha_init(sha, 256);
394 av_sha_update(sha, hmac_buf, 64);
396 av_sha_update(sha, src, len);
397 } else { //skip 32 bytes used for storing digest
398 av_sha_update(sha, src, gap);
399 av_sha_update(sha, src + gap + 32, len - gap - 32);
401 av_sha_final(sha, hmac_buf + 64);
403 for (i = 0; i < 64; i++)
404 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
405 av_sha_init(sha, 256);
406 av_sha_update(sha, hmac_buf, 64+32);
407 av_sha_final(sha, dst);
413 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
414 * will be stored) into that packet.
416 * @param buf handshake data (1536 bytes)
417 * @return offset to the digest inside input data
419 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
421 int i, digest_pos = 0;
423 for (i = 8; i < 12; i++)
424 digest_pos += buf[i];
425 digest_pos = (digest_pos % 728) + 12;
427 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
428 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
434 * Verify that the received server response has the expected digest value.
436 * @param buf handshake data received from the server (1536 bytes)
437 * @param off position to search digest offset from
438 * @return 0 if digest is valid, digest position otherwise
440 static int rtmp_validate_digest(uint8_t *buf, int off)
442 int i, digest_pos = 0;
445 for (i = 0; i < 4; i++)
446 digest_pos += buf[i + off];
447 digest_pos = (digest_pos % 728) + off + 4;
449 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
450 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
452 if (!memcmp(digest, buf + digest_pos, 32))
458 * Perform handshake with the server by means of exchanging pseudorandom data
459 * signed with HMAC-SHA2 digest.
461 * @return 0 if handshake succeeds, negative value otherwise
463 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
466 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
467 3, // unencrypted data
468 0, 0, 0, 0, // client uptime
474 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
475 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
477 int server_pos, client_pos;
480 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n");
482 av_lfg_init(&rnd, 0xDEADC0DE);
483 // generate handshake packet - 1536 bytes of pseudorandom data
484 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
485 tosend[i] = av_lfg_get(&rnd) >> 24;
486 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
488 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
489 i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
490 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
491 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
494 i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
495 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
496 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
500 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
501 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
503 if (rt->is_input && serverdata[5] >= 3) {
504 server_pos = rtmp_validate_digest(serverdata + 1, 772);
506 server_pos = rtmp_validate_digest(serverdata + 1, 8);
508 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n");
513 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
514 rtmp_server_key, sizeof(rtmp_server_key),
516 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
519 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
520 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n");
524 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
525 tosend[i] = av_lfg_get(&rnd) >> 24;
526 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
527 rtmp_player_key, sizeof(rtmp_player_key),
529 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
531 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
533 // write reply back to the server
534 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
536 url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
543 * Parse received packet and possibly perform some action depending on
544 * the packet contents.
545 * @return 0 for no errors, negative values for serious errors which prevent
546 * further communications, positive values for uncritical errors
548 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
551 const uint8_t *data_end = pkt->data + pkt->data_size;
554 ff_rtmp_packet_dump(LOG_CONTEXT, pkt);
558 case RTMP_PT_CHUNK_SIZE:
559 if (pkt->data_size != 4) {
560 av_log(LOG_CONTEXT, AV_LOG_ERROR,
561 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
565 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
566 rt->chunk_size = AV_RB32(pkt->data);
567 if (rt->chunk_size <= 0) {
568 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
571 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
574 t = AV_RB16(pkt->data);
576 gen_pong(s, rt, pkt);
578 case RTMP_PT_CLIENT_BW:
579 if (pkt->data_size < 4) {
580 av_log(LOG_CONTEXT, AV_LOG_ERROR,
581 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
585 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
586 rt->client_report_size = AV_RB32(pkt->data) >> 1;
589 //TODO: check for the messages sent for wrong state?
590 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
593 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
594 "description", tmpstr, sizeof(tmpstr)))
595 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
597 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
599 case STATE_HANDSHAKED:
601 gen_release_stream(s, rt);
602 gen_fcpublish_stream(s, rt);
603 rt->state = STATE_RELEASING;
605 rt->state = STATE_CONNECTING;
607 gen_create_stream(s, rt);
609 case STATE_FCPUBLISH:
610 rt->state = STATE_CONNECTING;
612 case STATE_RELEASING:
613 rt->state = STATE_FCPUBLISH;
614 /* hack for Wowza Media Server, it does not send result for
615 * releaseStream and FCPublish calls */
616 if (!pkt->data[10]) {
617 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
619 rt->state = STATE_CONNECTING;
621 if (rt->state != STATE_CONNECTING)
623 case STATE_CONNECTING:
624 //extract a number from the result
625 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
626 av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n");
628 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
635 rt->state = STATE_READY;
638 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
639 const uint8_t* ptr = pkt->data + 11;
642 for (i = 0; i < 2; i++) {
643 t = ff_amf_tag_size(ptr, data_end);
648 t = ff_amf_get_field_value(ptr, data_end,
649 "level", tmpstr, sizeof(tmpstr));
650 if (!t && !strcmp(tmpstr, "error")) {
651 if (!ff_amf_get_field_value(ptr, data_end,
652 "description", tmpstr, sizeof(tmpstr)))
653 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
656 t = ff_amf_get_field_value(ptr, data_end,
657 "code", tmpstr, sizeof(tmpstr));
658 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
659 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
660 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
661 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
669 * Interact with the server by receiving and sending RTMP packets until
670 * there is some significant data (media data or expected status notification).
672 * @param s reading context
673 * @param for_header non-zero value tells function to work until it
674 * gets notification from the server that playing has been started,
675 * otherwise function will work until some media data is received (or
677 * @return 0 for successful operation, negative value in case of error
679 static int get_packet(URLContext *s, int for_header)
681 RTMPContext *rt = s->priv_data;
686 uint32_t ts, cts, pts=0;
688 if (rt->state == STATE_STOPPED)
693 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
694 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
696 return AVERROR(EAGAIN);
701 rt->bytes_read += ret;
702 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
703 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending bytes read report\n");
704 gen_bytes_read(s, rt, rpkt.timestamp + 1);
705 rt->last_bytes_read = rt->bytes_read;
708 ret = rtmp_parse_result(s, rt, &rpkt);
709 if (ret < 0) {//serious error in current packet
710 ff_rtmp_packet_destroy(&rpkt);
713 if (rt->state == STATE_STOPPED) {
714 ff_rtmp_packet_destroy(&rpkt);
717 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
718 ff_rtmp_packet_destroy(&rpkt);
721 if (!rpkt.data_size || !rt->is_input) {
722 ff_rtmp_packet_destroy(&rpkt);
725 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
726 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
729 // generate packet header and put data into buffer for FLV demuxer
731 rt->flv_size = rpkt.data_size + 15;
732 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
733 bytestream_put_byte(&p, rpkt.type);
734 bytestream_put_be24(&p, rpkt.data_size);
735 bytestream_put_be24(&p, ts);
736 bytestream_put_byte(&p, ts >> 24);
737 bytestream_put_be24(&p, 0);
738 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
739 bytestream_put_be32(&p, 0);
740 ff_rtmp_packet_destroy(&rpkt);
742 } else if (rpkt.type == RTMP_PT_METADATA) {
743 // we got raw FLV data, make it available for FLV demuxer
745 rt->flv_size = rpkt.data_size;
746 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
747 /* rewrite timestamps */
750 while (next - rpkt.data < rpkt.data_size - 11) {
752 data_size = bytestream_get_be24(&next);
754 cts = bytestream_get_be24(&next);
755 cts |= bytestream_get_byte(&next) << 24;
760 bytestream_put_be24(&p, ts);
761 bytestream_put_byte(&p, ts >> 24);
762 next += data_size + 3 + 4;
764 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
765 ff_rtmp_packet_destroy(&rpkt);
768 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 url_close(rt->stream);
794 * Open RTMP connection and verify that the stream can be played.
796 * URL syntax: rtmp://server[:port][/app][/playpath]
797 * where 'app' is first one or two directories in the path
798 * (e.g. /ondemand/, /flash/live/, etc.)
799 * and 'playpath' is a file name (the rest of the path,
800 * may be prefixed with "mp4:")
802 static int rtmp_open(URLContext *s, const char *uri, int flags)
805 char proto[8], hostname[256], path[1024], *fname;
810 rt = av_mallocz(sizeof(RTMPContext));
812 return AVERROR(ENOMEM);
814 rt->is_input = !(flags & URL_WRONLY);
816 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
817 path, sizeof(path), s->filename);
820 port = RTMP_DEFAULT_PORT;
821 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
823 if (url_open(&rt->stream, buf, URL_RDWR) < 0) {
824 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);
828 rt->state = STATE_START;
829 if (rtmp_handshake(s, rt))
832 rt->chunk_size = 128;
833 rt->state = STATE_HANDSHAKED;
834 //extract "app" part from path
835 if (!strncmp(path, "/ondemand/", 10)) {
837 memcpy(rt->app, "ondemand", 9);
839 char *p = strchr(path + 1, '/');
844 char *c = strchr(p + 1, ':');
845 fname = strchr(p + 1, '/');
846 if (!fname || c < fname) {
848 av_strlcpy(rt->app, path + 1, p - path);
851 av_strlcpy(rt->app, path + 1, fname - path - 1);
855 if (!strchr(fname, ':') &&
856 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
857 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
858 memcpy(rt->playpath, "mp4:", 5);
862 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
864 rt->client_report_size = 1048576;
866 rt->last_bytes_read = 0;
868 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
869 proto, path, rt->app, rt->playpath);
870 gen_connect(s, rt, proto, hostname, port);
873 ret = get_packet(s, 1);
874 } while (ret == EAGAIN);
879 // generate FLV header for demuxer
881 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
883 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
890 s->max_packet_size = url_get_max_packet_size(rt->stream);
899 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
901 RTMPContext *rt = s->priv_data;
902 int orig_size = size;
906 int data_left = rt->flv_size - rt->flv_off;
908 if (data_left >= size) {
909 memcpy(buf, rt->flv_data + rt->flv_off, size);
914 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
917 rt->flv_off = rt->flv_size;
920 if ((ret = get_packet(s, 0)) < 0)
926 static int rtmp_write(URLContext *h, const uint8_t *buf, int size)
928 RTMPContext *rt = h->priv_data;
929 int size_temp = size;
930 int pktsize, pkttype;
932 const uint8_t *buf_temp = buf;
935 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
942 if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
947 pkttype = bytestream_get_byte(&buf_temp);
948 pktsize = bytestream_get_be24(&buf_temp);
949 ts = bytestream_get_be24(&buf_temp);
950 ts |= bytestream_get_byte(&buf_temp) << 24;
951 bytestream_get_be24(&buf_temp);
953 rt->flv_size = pktsize;
955 //force 12bytes header
956 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
957 pkttype == RTMP_PT_NOTIFY) {
958 if (pkttype == RTMP_PT_NOTIFY)
960 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
963 //this can be a big packet, it's better to send it right here
964 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
965 rt->out_pkt.extra = rt->main_channel_id;
966 rt->flv_data = rt->out_pkt.data;
968 if (pkttype == RTMP_PT_NOTIFY)
969 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
972 if (rt->flv_size - rt->flv_off > size_temp) {
973 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
974 rt->flv_off += size_temp;
976 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
977 rt->flv_off += rt->flv_size - rt->flv_off;
980 if (rt->flv_off == rt->flv_size) {
981 bytestream_get_be32(&buf_temp);
983 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
984 ff_rtmp_packet_destroy(&rt->out_pkt);
988 } while (buf_temp - buf < size_temp);
992 URLProtocol rtmp_protocol = {