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
74 int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
75 uint8_t flv_header[11]; ///< partial incoming flv packet header
76 int flv_header_bytes; ///< number of initialized bytes in flv_header
79 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
80 /** Client key used for digest signing */
81 static const uint8_t rtmp_player_key[] = {
82 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
83 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
85 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
86 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
87 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
90 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
91 /** Key used for RTMP server digest signing */
92 static const uint8_t rtmp_server_key[] = {
93 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
94 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
95 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
97 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
98 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
99 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
103 * Generate 'connect' call and send it to the server.
105 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
106 const char *host, int port)
112 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
115 ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
116 ff_amf_write_string(&p, "connect");
117 ff_amf_write_number(&p, 1.0);
118 ff_amf_write_object_start(&p);
119 ff_amf_write_field_name(&p, "app");
120 ff_amf_write_string(&p, rt->app);
123 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
124 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
126 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
127 ff_amf_write_field_name(&p, "type");
128 ff_amf_write_string(&p, "nonprivate");
130 ff_amf_write_field_name(&p, "flashVer");
131 ff_amf_write_string(&p, ver);
132 ff_amf_write_field_name(&p, "tcUrl");
133 ff_amf_write_string(&p, tcurl);
135 ff_amf_write_field_name(&p, "fpad");
136 ff_amf_write_bool(&p, 0);
137 ff_amf_write_field_name(&p, "capabilities");
138 ff_amf_write_number(&p, 15.0);
139 ff_amf_write_field_name(&p, "audioCodecs");
140 ff_amf_write_number(&p, 1639.0);
141 ff_amf_write_field_name(&p, "videoCodecs");
142 ff_amf_write_number(&p, 252.0);
143 ff_amf_write_field_name(&p, "videoFunction");
144 ff_amf_write_number(&p, 1.0);
146 ff_amf_write_object_end(&p);
148 pkt.data_size = p - pkt.data;
150 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
151 ff_rtmp_packet_destroy(&pkt);
155 * Generate 'releaseStream' call and send it to the server. It should make
156 * the server release some channel for media streams.
158 static void gen_release_stream(URLContext *s, RTMPContext *rt)
163 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
164 29 + strlen(rt->playpath));
166 av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
168 ff_amf_write_string(&p, "releaseStream");
169 ff_amf_write_number(&p, 2.0);
170 ff_amf_write_null(&p);
171 ff_amf_write_string(&p, rt->playpath);
173 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
174 ff_rtmp_packet_destroy(&pkt);
178 * Generate 'FCPublish' call and send it to the server. It should make
179 * the server preapare for receiving media streams.
181 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
186 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
187 25 + strlen(rt->playpath));
189 av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
191 ff_amf_write_string(&p, "FCPublish");
192 ff_amf_write_number(&p, 3.0);
193 ff_amf_write_null(&p);
194 ff_amf_write_string(&p, rt->playpath);
196 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
197 ff_rtmp_packet_destroy(&pkt);
201 * Generate 'FCUnpublish' call and send it to the server. It should make
202 * the server destroy stream.
204 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
209 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
210 27 + strlen(rt->playpath));
212 av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
214 ff_amf_write_string(&p, "FCUnpublish");
215 ff_amf_write_number(&p, 5.0);
216 ff_amf_write_null(&p);
217 ff_amf_write_string(&p, rt->playpath);
219 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
220 ff_rtmp_packet_destroy(&pkt);
224 * Generate 'createStream' call and send it to the server. It should make
225 * the server allocate some channel for media streams.
227 static void gen_create_stream(URLContext *s, RTMPContext *rt)
232 av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
233 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
236 ff_amf_write_string(&p, "createStream");
237 ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
238 ff_amf_write_null(&p);
240 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
241 ff_rtmp_packet_destroy(&pkt);
246 * Generate 'deleteStream' call and send it to the server. It should make
247 * the server remove some channel for media streams.
249 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
254 av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
255 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
258 ff_amf_write_string(&p, "deleteStream");
259 ff_amf_write_number(&p, 0.0);
260 ff_amf_write_null(&p);
261 ff_amf_write_number(&p, rt->main_channel_id);
263 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
264 ff_rtmp_packet_destroy(&pkt);
268 * Generate 'play' call and send it to the server, then ping the server
269 * to start actual playing.
271 static void gen_play(URLContext *s, RTMPContext *rt)
276 av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
277 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
278 20 + strlen(rt->playpath));
279 pkt.extra = rt->main_channel_id;
282 ff_amf_write_string(&p, "play");
283 ff_amf_write_number(&p, 0.0);
284 ff_amf_write_null(&p);
285 ff_amf_write_string(&p, rt->playpath);
287 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
288 ff_rtmp_packet_destroy(&pkt);
290 // set client buffer time disguised in ping packet
291 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
294 bytestream_put_be16(&p, 3);
295 bytestream_put_be32(&p, 1);
296 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
298 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
299 ff_rtmp_packet_destroy(&pkt);
303 * Generate 'publish' call and send it to the server.
305 static void gen_publish(URLContext *s, RTMPContext *rt)
310 av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
311 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
312 30 + strlen(rt->playpath));
313 pkt.extra = rt->main_channel_id;
316 ff_amf_write_string(&p, "publish");
317 ff_amf_write_number(&p, 0.0);
318 ff_amf_write_null(&p);
319 ff_amf_write_string(&p, rt->playpath);
320 ff_amf_write_string(&p, "live");
322 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
323 ff_rtmp_packet_destroy(&pkt);
327 * Generate ping reply and send it to the server.
329 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
334 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
336 bytestream_put_be16(&p, 7);
337 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
338 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
339 ff_rtmp_packet_destroy(&pkt);
343 * Generate report on bytes read so far and send it to the server.
345 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
350 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
352 bytestream_put_be32(&p, rt->bytes_read);
353 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
354 ff_rtmp_packet_destroy(&pkt);
357 //TODO: Move HMAC code somewhere. Eventually.
358 #define HMAC_IPAD_VAL 0x36
359 #define HMAC_OPAD_VAL 0x5C
362 * Calculate HMAC-SHA2 digest for RTMP handshake packets.
364 * @param src input buffer
365 * @param len input buffer length (should be 1536)
366 * @param gap offset in buffer where 32 bytes should not be taken into account
367 * when calculating digest (since it will be used to store that digest)
368 * @param key digest key
369 * @param keylen digest key length
370 * @param dst buffer where calculated digest will be stored (32 bytes)
372 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
373 const uint8_t *key, int keylen, uint8_t *dst)
376 uint8_t hmac_buf[64+32] = {0};
379 sha = av_mallocz(av_sha_size);
382 memcpy(hmac_buf, key, keylen);
384 av_sha_init(sha, 256);
385 av_sha_update(sha,key, keylen);
386 av_sha_final(sha, hmac_buf);
388 for (i = 0; i < 64; i++)
389 hmac_buf[i] ^= HMAC_IPAD_VAL;
391 av_sha_init(sha, 256);
392 av_sha_update(sha, hmac_buf, 64);
394 av_sha_update(sha, src, len);
395 } else { //skip 32 bytes used for storing digest
396 av_sha_update(sha, src, gap);
397 av_sha_update(sha, src + gap + 32, len - gap - 32);
399 av_sha_final(sha, hmac_buf + 64);
401 for (i = 0; i < 64; i++)
402 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
403 av_sha_init(sha, 256);
404 av_sha_update(sha, hmac_buf, 64+32);
405 av_sha_final(sha, dst);
411 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
412 * will be stored) into that packet.
414 * @param buf handshake data (1536 bytes)
415 * @return offset to the digest inside input data
417 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
419 int i, digest_pos = 0;
421 for (i = 8; i < 12; i++)
422 digest_pos += buf[i];
423 digest_pos = (digest_pos % 728) + 12;
425 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
426 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
432 * Verify that the received server response has the expected digest value.
434 * @param buf handshake data received from the server (1536 bytes)
435 * @param off position to search digest offset from
436 * @return 0 if digest is valid, digest position otherwise
438 static int rtmp_validate_digest(uint8_t *buf, int off)
440 int i, digest_pos = 0;
443 for (i = 0; i < 4; i++)
444 digest_pos += buf[i + off];
445 digest_pos = (digest_pos % 728) + off + 4;
447 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
448 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
450 if (!memcmp(digest, buf + digest_pos, 32))
456 * Perform handshake with the server by means of exchanging pseudorandom data
457 * signed with HMAC-SHA2 digest.
459 * @return 0 if handshake succeeds, negative value otherwise
461 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
464 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
465 3, // unencrypted data
466 0, 0, 0, 0, // client uptime
472 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
473 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
475 int server_pos, client_pos;
478 av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
480 av_lfg_init(&rnd, 0xDEADC0DE);
481 // generate handshake packet - 1536 bytes of pseudorandom data
482 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
483 tosend[i] = av_lfg_get(&rnd) >> 24;
484 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
486 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
487 i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
488 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
489 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
492 i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
493 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
494 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
498 av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
499 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
501 if (rt->is_input && serverdata[5] >= 3) {
502 server_pos = rtmp_validate_digest(serverdata + 1, 772);
504 server_pos = rtmp_validate_digest(serverdata + 1, 8);
506 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
511 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
512 rtmp_server_key, sizeof(rtmp_server_key),
514 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
517 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
518 av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
522 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
523 tosend[i] = av_lfg_get(&rnd) >> 24;
524 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
525 rtmp_player_key, sizeof(rtmp_player_key),
527 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
529 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
531 // write reply back to the server
532 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
534 ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
541 * Parse received packet and possibly perform some action depending on
542 * the packet contents.
543 * @return 0 for no errors, negative values for serious errors which prevent
544 * further communications, positive values for uncritical errors
546 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
549 const uint8_t *data_end = pkt->data + pkt->data_size;
552 ff_rtmp_packet_dump(s, pkt);
556 case RTMP_PT_CHUNK_SIZE:
557 if (pkt->data_size != 4) {
558 av_log(s, AV_LOG_ERROR,
559 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
563 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
564 rt->chunk_size = AV_RB32(pkt->data);
565 if (rt->chunk_size <= 0) {
566 av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
569 av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
572 t = AV_RB16(pkt->data);
574 gen_pong(s, rt, pkt);
576 case RTMP_PT_CLIENT_BW:
577 if (pkt->data_size < 4) {
578 av_log(s, AV_LOG_ERROR,
579 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
583 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
584 rt->client_report_size = AV_RB32(pkt->data) >> 1;
587 //TODO: check for the messages sent for wrong state?
588 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
591 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
592 "description", tmpstr, sizeof(tmpstr)))
593 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
595 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
597 case STATE_HANDSHAKED:
599 gen_release_stream(s, rt);
600 gen_fcpublish_stream(s, rt);
601 rt->state = STATE_RELEASING;
603 rt->state = STATE_CONNECTING;
605 gen_create_stream(s, rt);
607 case STATE_FCPUBLISH:
608 rt->state = STATE_CONNECTING;
610 case STATE_RELEASING:
611 rt->state = STATE_FCPUBLISH;
612 /* hack for Wowza Media Server, it does not send result for
613 * releaseStream and FCPublish calls */
614 if (!pkt->data[10]) {
615 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
617 rt->state = STATE_CONNECTING;
619 if (rt->state != STATE_CONNECTING)
621 case STATE_CONNECTING:
622 //extract a number from the result
623 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
624 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
626 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
633 rt->state = STATE_READY;
636 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
637 const uint8_t* ptr = pkt->data + 11;
640 for (i = 0; i < 2; i++) {
641 t = ff_amf_tag_size(ptr, data_end);
646 t = ff_amf_get_field_value(ptr, data_end,
647 "level", tmpstr, sizeof(tmpstr));
648 if (!t && !strcmp(tmpstr, "error")) {
649 if (!ff_amf_get_field_value(ptr, data_end,
650 "description", tmpstr, sizeof(tmpstr)))
651 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
654 t = ff_amf_get_field_value(ptr, data_end,
655 "code", tmpstr, sizeof(tmpstr));
656 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
657 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
658 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
659 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
667 * Interact with the server by receiving and sending RTMP packets until
668 * there is some significant data (media data or expected status notification).
670 * @param s reading context
671 * @param for_header non-zero value tells function to work until it
672 * gets notification from the server that playing has been started,
673 * otherwise function will work until some media data is received (or
675 * @return 0 for successful operation, negative value in case of error
677 static int get_packet(URLContext *s, int for_header)
679 RTMPContext *rt = s->priv_data;
684 uint32_t ts, cts, pts=0;
686 if (rt->state == STATE_STOPPED)
690 RTMPPacket rpkt = { 0 };
691 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
692 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
694 return AVERROR(EAGAIN);
699 rt->bytes_read += ret;
700 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
701 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
702 gen_bytes_read(s, rt, rpkt.timestamp + 1);
703 rt->last_bytes_read = rt->bytes_read;
706 ret = rtmp_parse_result(s, rt, &rpkt);
707 if (ret < 0) {//serious error in current packet
708 ff_rtmp_packet_destroy(&rpkt);
711 if (rt->state == STATE_STOPPED) {
712 ff_rtmp_packet_destroy(&rpkt);
715 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
716 ff_rtmp_packet_destroy(&rpkt);
719 if (!rpkt.data_size || !rt->is_input) {
720 ff_rtmp_packet_destroy(&rpkt);
723 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
724 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
727 // generate packet header and put data into buffer for FLV demuxer
729 rt->flv_size = rpkt.data_size + 15;
730 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
731 bytestream_put_byte(&p, rpkt.type);
732 bytestream_put_be24(&p, rpkt.data_size);
733 bytestream_put_be24(&p, ts);
734 bytestream_put_byte(&p, ts >> 24);
735 bytestream_put_be24(&p, 0);
736 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
737 bytestream_put_be32(&p, 0);
738 ff_rtmp_packet_destroy(&rpkt);
740 } else if (rpkt.type == RTMP_PT_METADATA) {
741 // we got raw FLV data, make it available for FLV demuxer
743 rt->flv_size = rpkt.data_size;
744 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
745 /* rewrite timestamps */
748 while (next - rpkt.data < rpkt.data_size - 11) {
750 data_size = bytestream_get_be24(&next);
752 cts = bytestream_get_be24(&next);
753 cts |= bytestream_get_byte(&next) << 24;
758 bytestream_put_be24(&p, ts);
759 bytestream_put_byte(&p, ts >> 24);
760 next += data_size + 3 + 4;
762 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
763 ff_rtmp_packet_destroy(&rpkt);
766 ff_rtmp_packet_destroy(&rpkt);
770 static int rtmp_close(URLContext *h)
772 RTMPContext *rt = h->priv_data;
776 if (rt->out_pkt.data_size)
777 ff_rtmp_packet_destroy(&rt->out_pkt);
778 if (rt->state > STATE_FCPUBLISH)
779 gen_fcunpublish_stream(h, rt);
781 if (rt->state > STATE_HANDSHAKED)
782 gen_delete_stream(h, rt);
784 av_freep(&rt->flv_data);
785 ffurl_close(rt->stream);
791 * Open RTMP connection and verify that the stream can be played.
793 * URL syntax: rtmp://server[:port][/app][/playpath]
794 * where 'app' is first one or two directories in the path
795 * (e.g. /ondemand/, /flash/live/, etc.)
796 * and 'playpath' is a file name (the rest of the path,
797 * may be prefixed with "mp4:")
799 static int rtmp_open(URLContext *s, const char *uri, int flags)
802 char proto[8], hostname[256], path[1024], *fname;
807 rt = av_mallocz(sizeof(RTMPContext));
809 return AVERROR(ENOMEM);
811 rt->is_input = !(flags & AVIO_FLAG_WRITE);
813 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
814 path, sizeof(path), s->filename);
817 port = RTMP_DEFAULT_PORT;
818 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
820 if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
821 &s->interrupt_callback, NULL) < 0) {
822 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
826 rt->state = STATE_START;
827 if (rtmp_handshake(s, rt))
830 rt->chunk_size = 128;
831 rt->state = STATE_HANDSHAKED;
832 //extract "app" part from path
833 if (!strncmp(path, "/ondemand/", 10)) {
835 memcpy(rt->app, "ondemand", 9);
837 char *p = strchr(path + 1, '/');
842 char *c = strchr(p + 1, ':');
843 fname = strchr(p + 1, '/');
844 if (!fname || c < fname) {
846 av_strlcpy(rt->app, path + 1, p - path);
849 av_strlcpy(rt->app, path + 1, fname - path - 1);
853 if (!strchr(fname, ':') &&
854 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
855 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
856 memcpy(rt->playpath, "mp4:", 5);
860 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
862 rt->client_report_size = 1048576;
864 rt->last_bytes_read = 0;
866 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
867 proto, path, rt->app, rt->playpath);
868 gen_connect(s, rt, proto, hostname, port);
871 ret = get_packet(s, 1);
872 } while (ret == EAGAIN);
877 // generate FLV header for demuxer
879 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
881 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
889 s->max_packet_size = rt->stream->max_packet_size;
898 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
900 RTMPContext *rt = s->priv_data;
901 int orig_size = size;
905 int data_left = rt->flv_size - rt->flv_off;
907 if (data_left >= size) {
908 memcpy(buf, rt->flv_data + rt->flv_off, size);
913 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
916 rt->flv_off = rt->flv_size;
919 if ((ret = get_packet(s, 0)) < 0)
925 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
927 RTMPContext *rt = s->priv_data;
928 int size_temp = size;
929 int pktsize, pkttype;
931 const uint8_t *buf_temp = buf;
934 if (rt->skip_bytes) {
935 int skip = FFMIN(rt->skip_bytes, size_temp);
938 rt->skip_bytes -= skip;
942 if (rt->flv_header_bytes < 11) {
943 const uint8_t *header = rt->flv_header;
944 int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
945 bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
946 rt->flv_header_bytes += copy;
948 if (rt->flv_header_bytes < 11)
951 pkttype = bytestream_get_byte(&header);
952 pktsize = bytestream_get_be24(&header);
953 ts = bytestream_get_be24(&header);
954 ts |= bytestream_get_byte(&header) << 24;
955 bytestream_get_be24(&header);
956 rt->flv_size = pktsize;
958 //force 12bytes header
959 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
960 pkttype == RTMP_PT_NOTIFY) {
961 if (pkttype == RTMP_PT_NOTIFY)
963 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
966 //this can be a big packet, it's better to send it right here
967 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
968 rt->out_pkt.extra = rt->main_channel_id;
969 rt->flv_data = rt->out_pkt.data;
971 if (pkttype == RTMP_PT_NOTIFY)
972 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
975 if (rt->flv_size - rt->flv_off > size_temp) {
976 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
977 rt->flv_off += size_temp;
980 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
981 size_temp -= rt->flv_size - rt->flv_off;
982 rt->flv_off += rt->flv_size - rt->flv_off;
985 if (rt->flv_off == rt->flv_size) {
988 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
989 ff_rtmp_packet_destroy(&rt->out_pkt);
992 rt->flv_header_bytes = 0;
994 } while (buf_temp - buf < size);
998 URLProtocol ff_rtmp_protocol = {
1000 .url_open = rtmp_open,
1001 .url_read = rtmp_read,
1002 .url_write = rtmp_write,
1003 .url_close = rtmp_close,