2 * RTMP network protocol
3 * Copyright (c) 2009 Kostya Shishkov
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; 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.h"
30 #include "libavutil/lfg.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/sha.h"
40 #include "rtmpcrypt.h"
46 #define APP_MAX_LENGTH 128
47 #define PLAYPATH_MAX_LENGTH 256
48 #define TCURL_MAX_LENGTH 512
49 #define FLASHVER_MAX_LENGTH 64
51 /** RTMP protocol handler state */
53 STATE_START, ///< client has not done anything yet
54 STATE_HANDSHAKED, ///< client has performed handshake
55 STATE_RELEASING, ///< client releasing stream before publish it (for output)
56 STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
57 STATE_CONNECTING, ///< client connected to server successfully
58 STATE_READY, ///< client has sent all needed commands and waits for server reply
59 STATE_PLAYING, ///< client has started receiving multimedia data from server
60 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
61 STATE_STOPPED, ///< the broadcast has been stopped
64 /** protocol handler context */
65 typedef struct RTMPContext {
67 URLContext* stream; ///< TCP stream used in interactions with RTMP server
68 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
69 int chunk_size; ///< size of the chunks RTMP packets are divided into
70 int is_input; ///< input/output flag
71 char *playpath; ///< stream identifier to play (with possible "mp4:" prefix)
72 int live; ///< 0: recorded, -1: live, -2: both
73 char *app; ///< name of application
74 char *conn; ///< append arbitrary AMF data to the Connect message
75 ClientState state; ///< current state
76 int main_channel_id; ///< an additional channel ID which is used for some invocations
77 uint8_t* flv_data; ///< buffer with data for demuxer
78 int flv_size; ///< current buffer size
79 int flv_off; ///< number of bytes read from current buffer
80 int flv_nb_packets; ///< number of flv packets published
81 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
82 uint32_t client_report_size; ///< number of bytes after which client should report to server
83 uint32_t bytes_read; ///< number of bytes read from server
84 uint32_t last_bytes_read; ///< number of bytes read last reported to server
85 int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
86 uint8_t flv_header[11]; ///< partial incoming flv packet header
87 int flv_header_bytes; ///< number of initialized bytes in flv_header
88 int nb_invokes; ///< keeps track of invoke messages
89 int create_stream_invoke; ///< invoke id for the create stream command
90 char* tcurl; ///< url of the target stream
91 char* flashver; ///< version of the flash plugin
92 char* swfurl; ///< url of the swf player
93 char* pageurl; ///< url of the web page
94 int server_bw; ///< server bandwidth
95 int client_buffer_time; ///< client buffer time in ms
96 int flush_interval; ///< number of packets flushed in the same request (RTMPT only)
97 int encrypted; ///< use an encrypted connection (RTMPE only)
100 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
101 /** Client key used for digest signing */
102 static const uint8_t rtmp_player_key[] = {
103 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
104 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
106 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
107 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
108 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
111 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
112 /** Key used for RTMP server digest signing */
113 static const uint8_t rtmp_server_key[] = {
114 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
115 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
116 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
118 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
119 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
120 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
123 static int rtmp_write_amf_data(URLContext *s, char *param, uint8_t **p)
128 /* The type must be B for Boolean, N for number, S for string, O for
129 * object, or Z for null. For Booleans the data must be either 0 or 1 for
130 * FALSE or TRUE, respectively. Likewise for Objects the data must be
131 * 0 or 1 to end or begin an object, respectively. Data items in subobjects
132 * may be named, by prefixing the type with 'N' and specifying the name
133 * before the value (ie. NB:myFlag:1). This option may be used multiple times
134 * to construct arbitrary AMF sequences. */
135 if (param[0] && param[1] == ':') {
138 } else if (param[0] == 'N' && param[1] && param[2] == ':') {
141 value = strchr(field, ':');
147 if (!field || !value)
150 ff_amf_write_field_name(p, field);
157 ff_amf_write_bool(p, value[0] != '0');
160 ff_amf_write_string(p, value);
163 ff_amf_write_number(p, strtod(value, NULL));
166 ff_amf_write_null(p);
170 ff_amf_write_object_start(p);
172 ff_amf_write_object_end(p);
182 av_log(s, AV_LOG_ERROR, "Invalid AMF parameter: %s\n", param);
183 return AVERROR(EINVAL);
187 * Generate 'connect' call and send it to the server.
189 static int gen_connect(URLContext *s, RTMPContext *rt)
195 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
201 ff_amf_write_string(&p, "connect");
202 ff_amf_write_number(&p, ++rt->nb_invokes);
203 ff_amf_write_object_start(&p);
204 ff_amf_write_field_name(&p, "app");
205 ff_amf_write_string(&p, rt->app);
208 ff_amf_write_field_name(&p, "type");
209 ff_amf_write_string(&p, "nonprivate");
211 ff_amf_write_field_name(&p, "flashVer");
212 ff_amf_write_string(&p, rt->flashver);
215 ff_amf_write_field_name(&p, "swfUrl");
216 ff_amf_write_string(&p, rt->swfurl);
219 ff_amf_write_field_name(&p, "tcUrl");
220 ff_amf_write_string(&p, rt->tcurl);
222 ff_amf_write_field_name(&p, "fpad");
223 ff_amf_write_bool(&p, 0);
224 ff_amf_write_field_name(&p, "capabilities");
225 ff_amf_write_number(&p, 15.0);
227 /* Tell the server we support all the audio codecs except
228 * SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010)
229 * which are unused in the RTMP protocol implementation. */
230 ff_amf_write_field_name(&p, "audioCodecs");
231 ff_amf_write_number(&p, 4071.0);
232 ff_amf_write_field_name(&p, "videoCodecs");
233 ff_amf_write_number(&p, 252.0);
234 ff_amf_write_field_name(&p, "videoFunction");
235 ff_amf_write_number(&p, 1.0);
238 ff_amf_write_field_name(&p, "pageUrl");
239 ff_amf_write_string(&p, rt->pageurl);
242 ff_amf_write_object_end(&p);
245 char *param = rt->conn;
247 // Write arbitrary AMF data to the Connect message.
248 while (param != NULL) {
250 param += strspn(param, " ");
253 sep = strchr(param, ' ');
256 if ((ret = rtmp_write_amf_data(s, param, &p)) < 0) {
257 // Invalid AMF parameter.
258 ff_rtmp_packet_destroy(&pkt);
269 pkt.data_size = p - pkt.data;
271 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
273 ff_rtmp_packet_destroy(&pkt);
279 * Generate 'releaseStream' call and send it to the server. It should make
280 * the server release some channel for media streams.
282 static int gen_release_stream(URLContext *s, RTMPContext *rt)
288 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
289 0, 29 + strlen(rt->playpath))) < 0)
292 av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
294 ff_amf_write_string(&p, "releaseStream");
295 ff_amf_write_number(&p, ++rt->nb_invokes);
296 ff_amf_write_null(&p);
297 ff_amf_write_string(&p, rt->playpath);
299 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
301 ff_rtmp_packet_destroy(&pkt);
307 * Generate 'FCPublish' call and send it to the server. It should make
308 * the server preapare for receiving media streams.
310 static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
316 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
317 0, 25 + strlen(rt->playpath))) < 0)
320 av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
322 ff_amf_write_string(&p, "FCPublish");
323 ff_amf_write_number(&p, ++rt->nb_invokes);
324 ff_amf_write_null(&p);
325 ff_amf_write_string(&p, rt->playpath);
327 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
329 ff_rtmp_packet_destroy(&pkt);
335 * Generate 'FCUnpublish' call and send it to the server. It should make
336 * the server destroy stream.
338 static int gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
344 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
345 0, 27 + strlen(rt->playpath))) < 0)
348 av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
350 ff_amf_write_string(&p, "FCUnpublish");
351 ff_amf_write_number(&p, ++rt->nb_invokes);
352 ff_amf_write_null(&p);
353 ff_amf_write_string(&p, rt->playpath);
355 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
357 ff_rtmp_packet_destroy(&pkt);
363 * Generate 'createStream' call and send it to the server. It should make
364 * the server allocate some channel for media streams.
366 static int gen_create_stream(URLContext *s, RTMPContext *rt)
372 av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
374 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
379 ff_amf_write_string(&p, "createStream");
380 ff_amf_write_number(&p, ++rt->nb_invokes);
381 ff_amf_write_null(&p);
382 rt->create_stream_invoke = rt->nb_invokes;
384 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
386 ff_rtmp_packet_destroy(&pkt);
393 * Generate 'deleteStream' call and send it to the server. It should make
394 * the server remove some channel for media streams.
396 static int gen_delete_stream(URLContext *s, RTMPContext *rt)
402 av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
404 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
409 ff_amf_write_string(&p, "deleteStream");
410 ff_amf_write_number(&p, ++rt->nb_invokes);
411 ff_amf_write_null(&p);
412 ff_amf_write_number(&p, rt->main_channel_id);
414 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
416 ff_rtmp_packet_destroy(&pkt);
422 * Generate client buffer time and send it to the server.
424 static int gen_buffer_time(URLContext *s, RTMPContext *rt)
430 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
435 bytestream_put_be16(&p, 3);
436 bytestream_put_be32(&p, rt->main_channel_id);
437 bytestream_put_be32(&p, rt->client_buffer_time);
439 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
441 ff_rtmp_packet_destroy(&pkt);
447 * Generate 'play' call and send it to the server, then ping the server
448 * to start actual playing.
450 static int gen_play(URLContext *s, RTMPContext *rt)
456 av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
458 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE,
459 0, 29 + strlen(rt->playpath))) < 0)
462 pkt.extra = rt->main_channel_id;
465 ff_amf_write_string(&p, "play");
466 ff_amf_write_number(&p, ++rt->nb_invokes);
467 ff_amf_write_null(&p);
468 ff_amf_write_string(&p, rt->playpath);
469 ff_amf_write_number(&p, rt->live);
471 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
473 ff_rtmp_packet_destroy(&pkt);
479 * Generate 'publish' call and send it to the server.
481 static int gen_publish(URLContext *s, RTMPContext *rt)
487 av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
489 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
490 0, 30 + strlen(rt->playpath))) < 0)
493 pkt.extra = rt->main_channel_id;
496 ff_amf_write_string(&p, "publish");
497 ff_amf_write_number(&p, ++rt->nb_invokes);
498 ff_amf_write_null(&p);
499 ff_amf_write_string(&p, rt->playpath);
500 ff_amf_write_string(&p, "live");
502 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
504 ff_rtmp_packet_destroy(&pkt);
510 * Generate ping reply and send it to the server.
512 static int gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
518 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
519 ppkt->timestamp + 1, 6)) < 0)
523 bytestream_put_be16(&p, 7);
524 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
525 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
527 ff_rtmp_packet_destroy(&pkt);
533 * Generate server bandwidth message and send it to the server.
535 static int gen_server_bw(URLContext *s, RTMPContext *rt)
541 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW,
546 bytestream_put_be32(&p, rt->server_bw);
547 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
549 ff_rtmp_packet_destroy(&pkt);
555 * Generate check bandwidth message and send it to the server.
557 static int gen_check_bw(URLContext *s, RTMPContext *rt)
563 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
568 ff_amf_write_string(&p, "_checkbw");
569 ff_amf_write_number(&p, ++rt->nb_invokes);
570 ff_amf_write_null(&p);
572 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
574 ff_rtmp_packet_destroy(&pkt);
580 * Generate report on bytes read so far and send it to the server.
582 static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
588 if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ,
593 bytestream_put_be32(&p, rt->bytes_read);
594 ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
596 ff_rtmp_packet_destroy(&pkt);
601 int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap,
602 const uint8_t *key, int keylen, uint8_t *dst)
605 uint8_t hmac_buf[64+32] = {0};
608 sha = av_mallocz(av_sha_size);
610 return AVERROR(ENOMEM);
613 memcpy(hmac_buf, key, keylen);
615 av_sha_init(sha, 256);
616 av_sha_update(sha,key, keylen);
617 av_sha_final(sha, hmac_buf);
619 for (i = 0; i < 64; i++)
620 hmac_buf[i] ^= HMAC_IPAD_VAL;
622 av_sha_init(sha, 256);
623 av_sha_update(sha, hmac_buf, 64);
625 av_sha_update(sha, src, len);
626 } else { //skip 32 bytes used for storing digest
627 av_sha_update(sha, src, gap);
628 av_sha_update(sha, src + gap + 32, len - gap - 32);
630 av_sha_final(sha, hmac_buf + 64);
632 for (i = 0; i < 64; i++)
633 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
634 av_sha_init(sha, 256);
635 av_sha_update(sha, hmac_buf, 64+32);
636 av_sha_final(sha, dst);
643 int ff_rtmp_calc_digest_pos(const uint8_t *buf, int off, int mod_val,
646 int i, digest_pos = 0;
648 for (i = 0; i < 4; i++)
649 digest_pos += buf[i + off];
650 digest_pos = digest_pos % mod_val + add_val;
656 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
657 * will be stored) into that packet.
659 * @param buf handshake data (1536 bytes)
660 * @param encrypted use an encrypted connection (RTMPE)
661 * @return offset to the digest inside input data
663 static int rtmp_handshake_imprint_with_digest(uint8_t *buf, int encrypted)
668 digest_pos = ff_rtmp_calc_digest_pos(buf, 772, 728, 776);
670 digest_pos = ff_rtmp_calc_digest_pos(buf, 8, 728, 12);
672 ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
673 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
682 * Verify that the received server response has the expected digest value.
684 * @param buf handshake data received from the server (1536 bytes)
685 * @param off position to search digest offset from
686 * @return 0 if digest is valid, digest position otherwise
688 static int rtmp_validate_digest(uint8_t *buf, int off)
693 digest_pos = ff_rtmp_calc_digest_pos(buf, off, 728, off + 4);
695 ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
696 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
701 if (!memcmp(digest, buf + digest_pos, 32))
707 * Perform handshake with the server by means of exchanging pseudorandom data
708 * signed with HMAC-SHA2 digest.
710 * @return 0 if handshake succeeds, negative value otherwise
712 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
715 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
716 3, // unencrypted data
717 0, 0, 0, 0, // client uptime
723 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
724 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
726 int server_pos, client_pos;
727 uint8_t digest[32], signature[32];
730 av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
732 av_lfg_init(&rnd, 0xDEADC0DE);
733 // generate handshake packet - 1536 bytes of pseudorandom data
734 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
735 tosend[i] = av_lfg_get(&rnd) >> 24;
737 if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
738 /* When the client wants to use RTMPE, we have to change the command
739 * byte to 0x06 which means to use encrypted data and we have to set
740 * the flash version to at least 9.0.115.0. */
747 /* Initialize the Diffie-Hellmann context and generate the public key
748 * to send to the server. */
749 if ((ret = ff_rtmpe_gen_pub_key(rt->stream, tosend + 1)) < 0)
753 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
757 if ((ret = ffurl_write(rt->stream, tosend,
758 RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
759 av_log(s, AV_LOG_ERROR, "Cannot write RTMP handshake request\n");
763 if ((ret = ffurl_read_complete(rt->stream, serverdata,
764 RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
765 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
769 if ((ret = ffurl_read_complete(rt->stream, clientdata,
770 RTMP_HANDSHAKE_PACKET_SIZE)) < 0) {
771 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
775 av_log(s, AV_LOG_DEBUG, "Type answer %d\n", serverdata[0]);
776 av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
777 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
779 if (rt->is_input && serverdata[5] >= 3) {
780 server_pos = rtmp_validate_digest(serverdata + 1, 772);
786 server_pos = rtmp_validate_digest(serverdata + 1, 8);
791 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
796 ret = ff_rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
797 rtmp_server_key, sizeof(rtmp_server_key),
802 ret = ff_rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32,
803 0, digest, 32, signature);
807 if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
808 /* Compute the shared secret key sent by the server and initialize
809 * the RC4 encryption. */
810 if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
811 tosend + 1, type)) < 0)
814 /* Encrypt the signature received by the server. */
815 ff_rtmpe_encrypt_sig(rt->stream, signature, digest, serverdata[0]);
818 if (memcmp(signature, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
819 av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
823 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
824 tosend[i] = av_lfg_get(&rnd) >> 24;
825 ret = ff_rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
826 rtmp_player_key, sizeof(rtmp_player_key),
831 ret = ff_rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
833 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
837 if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
838 /* Encrypt the signature to be send to the server. */
839 ff_rtmpe_encrypt_sig(rt->stream, tosend +
840 RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
844 // write reply back to the server
845 if ((ret = ffurl_write(rt->stream, tosend,
846 RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
849 if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
850 /* Set RC4 keys for encryption and update the keystreams. */
851 if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
855 if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
856 /* Compute the shared secret key sent by the server and initialize
857 * the RC4 encryption. */
858 if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
862 if (serverdata[0] == 9) {
863 /* Encrypt the signature received by the server. */
864 ff_rtmpe_encrypt_sig(rt->stream, signature, digest,
869 if ((ret = ffurl_write(rt->stream, serverdata + 1,
870 RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
873 if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
874 /* Set RC4 keys for encryption and update the keystreams. */
875 if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
883 static int handle_chunk_size(URLContext *s, RTMPPacket *pkt)
885 RTMPContext *rt = s->priv_data;
888 if (pkt->data_size != 4) {
889 av_log(s, AV_LOG_ERROR,
890 "Chunk size change packet is not 4 bytes long (%d)\n",
892 return AVERROR_INVALIDDATA;
896 if ((ret = ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size,
897 rt->prev_pkt[1])) < 0)
901 rt->chunk_size = AV_RB32(pkt->data);
902 if (rt->chunk_size <= 0) {
903 av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
904 return AVERROR_INVALIDDATA;
906 av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
911 static int handle_ping(URLContext *s, RTMPPacket *pkt)
913 RTMPContext *rt = s->priv_data;
916 t = AV_RB16(pkt->data);
918 if ((ret = gen_pong(s, rt, pkt)) < 0)
925 static int handle_client_bw(URLContext *s, RTMPPacket *pkt)
927 RTMPContext *rt = s->priv_data;
929 if (pkt->data_size < 4) {
930 av_log(s, AV_LOG_ERROR,
931 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
935 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
936 rt->client_report_size = AV_RB32(pkt->data) >> 1;
941 static int handle_server_bw(URLContext *s, RTMPPacket *pkt)
943 RTMPContext *rt = s->priv_data;
945 rt->server_bw = AV_RB32(pkt->data);
946 if (rt->server_bw <= 0) {
947 av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n",
949 return AVERROR(EINVAL);
951 av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);
956 static int handle_invoke(URLContext *s, RTMPPacket *pkt)
958 RTMPContext *rt = s->priv_data;
960 const uint8_t *data_end = pkt->data + pkt->data_size;
963 //TODO: check for the messages sent for wrong state?
964 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
967 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
968 "description", tmpstr, sizeof(tmpstr)))
969 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
971 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
973 case STATE_HANDSHAKED:
975 if ((ret = gen_release_stream(s, rt)) < 0)
977 if ((ret = gen_fcpublish_stream(s, rt)) < 0)
979 rt->state = STATE_RELEASING;
981 if ((ret = gen_server_bw(s, rt)) < 0)
983 rt->state = STATE_CONNECTING;
985 if ((ret = gen_create_stream(s, rt)) < 0)
988 case STATE_FCPUBLISH:
989 rt->state = STATE_CONNECTING;
991 case STATE_RELEASING:
992 rt->state = STATE_FCPUBLISH;
993 /* hack for Wowza Media Server, it does not send result for
994 * releaseStream and FCPublish calls */
995 if (!pkt->data[10]) {
996 int pkt_id = av_int2double(AV_RB64(pkt->data + 11));
997 if (pkt_id == rt->create_stream_invoke)
998 rt->state = STATE_CONNECTING;
1000 if (rt->state != STATE_CONNECTING)
1002 case STATE_CONNECTING:
1003 //extract a number from the result
1004 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
1005 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
1007 rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
1010 if ((ret = gen_play(s, rt)) < 0)
1012 if ((ret = gen_buffer_time(s, rt)) < 0)
1015 if ((ret = gen_publish(s, rt)) < 0)
1018 rt->state = STATE_READY;
1021 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
1022 const uint8_t* ptr = pkt->data + 11;
1023 uint8_t tmpstr[256];
1025 for (i = 0; i < 2; i++) {
1026 t = ff_amf_tag_size(ptr, data_end);
1031 t = ff_amf_get_field_value(ptr, data_end,
1032 "level", tmpstr, sizeof(tmpstr));
1033 if (!t && !strcmp(tmpstr, "error")) {
1034 if (!ff_amf_get_field_value(ptr, data_end,
1035 "description", tmpstr, sizeof(tmpstr)))
1036 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
1039 t = ff_amf_get_field_value(ptr, data_end,
1040 "code", tmpstr, sizeof(tmpstr));
1041 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
1042 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
1043 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
1044 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
1045 } else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
1046 if ((ret = gen_check_bw(s, rt)) < 0)
1054 * Parse received packet and possibly perform some action depending on
1055 * the packet contents.
1056 * @return 0 for no errors, negative values for serious errors which prevent
1057 * further communications, positive values for uncritical errors
1059 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
1064 ff_rtmp_packet_dump(s, pkt);
1067 switch (pkt->type) {
1068 case RTMP_PT_CHUNK_SIZE:
1069 if ((ret = handle_chunk_size(s, pkt)) < 0)
1073 if ((ret = handle_ping(s, pkt)) < 0)
1076 case RTMP_PT_CLIENT_BW:
1077 if ((ret = handle_client_bw(s, pkt)) < 0)
1080 case RTMP_PT_SERVER_BW:
1081 if ((ret = handle_server_bw(s, pkt)) < 0)
1084 case RTMP_PT_INVOKE:
1085 if ((ret = handle_invoke(s, pkt)) < 0)
1090 /* Audio and Video packets are parsed in get_packet() */
1093 av_log(s, AV_LOG_VERBOSE, "Unknown packet type received 0x%02X\n", pkt->type);
1100 * Interact with the server by receiving and sending RTMP packets until
1101 * there is some significant data (media data or expected status notification).
1103 * @param s reading context
1104 * @param for_header non-zero value tells function to work until it
1105 * gets notification from the server that playing has been started,
1106 * otherwise function will work until some media data is received (or
1108 * @return 0 for successful operation, negative value in case of error
1110 static int get_packet(URLContext *s, int for_header)
1112 RTMPContext *rt = s->priv_data;
1115 const uint8_t *next;
1117 uint32_t ts, cts, pts=0;
1119 if (rt->state == STATE_STOPPED)
1123 RTMPPacket rpkt = { 0 };
1124 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
1125 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
1127 return AVERROR(EAGAIN);
1129 return AVERROR(EIO);
1132 rt->bytes_read += ret;
1133 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
1134 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
1135 if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
1137 rt->last_bytes_read = rt->bytes_read;
1140 ret = rtmp_parse_result(s, rt, &rpkt);
1141 if (ret < 0) {//serious error in current packet
1142 ff_rtmp_packet_destroy(&rpkt);
1145 if (rt->state == STATE_STOPPED) {
1146 ff_rtmp_packet_destroy(&rpkt);
1149 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
1150 ff_rtmp_packet_destroy(&rpkt);
1153 if (!rpkt.data_size || !rt->is_input) {
1154 ff_rtmp_packet_destroy(&rpkt);
1157 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
1158 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
1159 ts = rpkt.timestamp;
1161 // generate packet header and put data into buffer for FLV demuxer
1163 rt->flv_size = rpkt.data_size + 15;
1164 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
1165 bytestream_put_byte(&p, rpkt.type);
1166 bytestream_put_be24(&p, rpkt.data_size);
1167 bytestream_put_be24(&p, ts);
1168 bytestream_put_byte(&p, ts >> 24);
1169 bytestream_put_be24(&p, 0);
1170 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
1171 bytestream_put_be32(&p, 0);
1172 ff_rtmp_packet_destroy(&rpkt);
1174 } else if (rpkt.type == RTMP_PT_METADATA) {
1175 // we got raw FLV data, make it available for FLV demuxer
1177 rt->flv_size = rpkt.data_size;
1178 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
1179 /* rewrite timestamps */
1181 ts = rpkt.timestamp;
1182 while (next - rpkt.data < rpkt.data_size - 11) {
1184 data_size = bytestream_get_be24(&next);
1186 cts = bytestream_get_be24(&next);
1187 cts |= bytestream_get_byte(&next) << 24;
1192 bytestream_put_be24(&p, ts);
1193 bytestream_put_byte(&p, ts >> 24);
1194 next += data_size + 3 + 4;
1196 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
1197 ff_rtmp_packet_destroy(&rpkt);
1200 ff_rtmp_packet_destroy(&rpkt);
1204 static int rtmp_close(URLContext *h)
1206 RTMPContext *rt = h->priv_data;
1209 if (!rt->is_input) {
1210 rt->flv_data = NULL;
1211 if (rt->out_pkt.data_size)
1212 ff_rtmp_packet_destroy(&rt->out_pkt);
1213 if (rt->state > STATE_FCPUBLISH)
1214 ret = gen_fcunpublish_stream(h, rt);
1216 if (rt->state > STATE_HANDSHAKED)
1217 ret = gen_delete_stream(h, rt);
1219 av_freep(&rt->flv_data);
1220 ffurl_close(rt->stream);
1225 * Open RTMP connection and verify that the stream can be played.
1227 * URL syntax: rtmp://server[:port][/app][/playpath]
1228 * where 'app' is first one or two directories in the path
1229 * (e.g. /ondemand/, /flash/live/, etc.)
1230 * and 'playpath' is a file name (the rest of the path,
1231 * may be prefixed with "mp4:")
1233 static int rtmp_open(URLContext *s, const char *uri, int flags)
1235 RTMPContext *rt = s->priv_data;
1236 char proto[8], hostname[256], path[1024], *fname;
1240 AVDictionary *opts = NULL;
1243 rt->is_input = !(flags & AVIO_FLAG_WRITE);
1245 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
1246 path, sizeof(path), s->filename);
1248 if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
1249 if (!strcmp(proto, "rtmpts"))
1250 av_dict_set(&opts, "ffrtmphttp_tls", "1", 1);
1252 /* open the http tunneling connection */
1253 ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
1254 } else if (!strcmp(proto, "rtmps")) {
1255 /* open the tls connection */
1257 port = RTMPS_DEFAULT_PORT;
1258 ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
1259 } else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
1260 if (!strcmp(proto, "rtmpte"))
1261 av_dict_set(&opts, "ffrtmpcrypt_tunneling", "1", 1);
1263 /* open the encrypted connection */
1264 ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
1267 /* open the tcp connection */
1269 port = RTMP_DEFAULT_PORT;
1270 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
1273 if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
1274 &s->interrupt_callback, &opts)) < 0) {
1275 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
1279 rt->state = STATE_START;
1280 if ((ret = rtmp_handshake(s, rt)) < 0)
1283 rt->chunk_size = 128;
1284 rt->state = STATE_HANDSHAKED;
1286 // Keep the application name when it has been defined by the user.
1289 rt->app = av_malloc(APP_MAX_LENGTH);
1291 ret = AVERROR(ENOMEM);
1295 //extract "app" part from path
1296 if (!strncmp(path, "/ondemand/", 10)) {
1298 memcpy(rt->app, "ondemand", 9);
1300 char *next = *path ? path + 1 : path;
1301 char *p = strchr(next, '/');
1306 // make sure we do not mismatch a playpath for an application instance
1307 char *c = strchr(p + 1, ':');
1308 fname = strchr(p + 1, '/');
1309 if (!fname || (c && c < fname)) {
1311 av_strlcpy(rt->app, path + 1, p - path);
1314 av_strlcpy(rt->app, path + 1, fname - path - 1);
1320 // The name of application has been defined by the user, override it.
1325 if (!rt->playpath) {
1326 int len = strlen(fname);
1328 rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
1329 if (!rt->playpath) {
1330 ret = AVERROR(ENOMEM);
1334 if (!strchr(fname, ':') && len >= 4 &&
1335 (!strcmp(fname + len - 4, ".f4v") ||
1336 !strcmp(fname + len - 4, ".mp4"))) {
1337 memcpy(rt->playpath, "mp4:", 5);
1338 } else if (len >= 4 && !strcmp(fname + len - 4, ".flv")) {
1339 fname[len - 4] = '\0';
1341 rt->playpath[0] = 0;
1343 strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
1347 rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
1349 ret = AVERROR(ENOMEM);
1352 ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
1353 port, "/%s", rt->app);
1356 if (!rt->flashver) {
1357 rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
1358 if (!rt->flashver) {
1359 ret = AVERROR(ENOMEM);
1363 snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
1364 RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
1365 RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
1367 snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
1368 "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
1372 rt->client_report_size = 1048576;
1374 rt->last_bytes_read = 0;
1375 rt->server_bw = 2500000;
1377 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
1378 proto, path, rt->app, rt->playpath);
1379 if ((ret = gen_connect(s, rt)) < 0)
1383 ret = get_packet(s, 1);
1384 } while (ret == EAGAIN);
1389 // generate FLV header for demuxer
1391 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
1393 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
1396 rt->flv_data = NULL;
1398 rt->skip_bytes = 13;
1401 s->max_packet_size = rt->stream->max_packet_size;
1406 av_dict_free(&opts);
1411 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
1413 RTMPContext *rt = s->priv_data;
1414 int orig_size = size;
1418 int data_left = rt->flv_size - rt->flv_off;
1420 if (data_left >= size) {
1421 memcpy(buf, rt->flv_data + rt->flv_off, size);
1422 rt->flv_off += size;
1425 if (data_left > 0) {
1426 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
1429 rt->flv_off = rt->flv_size;
1432 if ((ret = get_packet(s, 0)) < 0)
1438 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
1440 RTMPContext *rt = s->priv_data;
1441 int size_temp = size;
1442 int pktsize, pkttype;
1444 const uint8_t *buf_temp = buf;
1449 if (rt->skip_bytes) {
1450 int skip = FFMIN(rt->skip_bytes, size_temp);
1453 rt->skip_bytes -= skip;
1457 if (rt->flv_header_bytes < 11) {
1458 const uint8_t *header = rt->flv_header;
1459 int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
1460 bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
1461 rt->flv_header_bytes += copy;
1463 if (rt->flv_header_bytes < 11)
1466 pkttype = bytestream_get_byte(&header);
1467 pktsize = bytestream_get_be24(&header);
1468 ts = bytestream_get_be24(&header);
1469 ts |= bytestream_get_byte(&header) << 24;
1470 bytestream_get_be24(&header);
1471 rt->flv_size = pktsize;
1473 //force 12bytes header
1474 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
1475 pkttype == RTMP_PT_NOTIFY) {
1476 if (pkttype == RTMP_PT_NOTIFY)
1478 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
1481 //this can be a big packet, it's better to send it right here
1482 if ((ret = ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL,
1483 pkttype, ts, pktsize)) < 0)
1486 rt->out_pkt.extra = rt->main_channel_id;
1487 rt->flv_data = rt->out_pkt.data;
1489 if (pkttype == RTMP_PT_NOTIFY)
1490 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
1493 if (rt->flv_size - rt->flv_off > size_temp) {
1494 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
1495 rt->flv_off += size_temp;
1498 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
1499 size_temp -= rt->flv_size - rt->flv_off;
1500 rt->flv_off += rt->flv_size - rt->flv_off;
1503 if (rt->flv_off == rt->flv_size) {
1506 if ((ret = ff_rtmp_packet_write(rt->stream, &rt->out_pkt,
1507 rt->chunk_size, rt->prev_pkt[1])) < 0)
1509 ff_rtmp_packet_destroy(&rt->out_pkt);
1512 rt->flv_header_bytes = 0;
1513 rt->flv_nb_packets++;
1515 } while (buf_temp - buf < size);
1517 if (rt->flv_nb_packets < rt->flush_interval)
1519 rt->flv_nb_packets = 0;
1521 /* set stream into nonblocking mode */
1522 rt->stream->flags |= AVIO_FLAG_NONBLOCK;
1524 /* try to read one byte from the stream */
1525 ret = ffurl_read(rt->stream, &c, 1);
1527 /* switch the stream back into blocking mode */
1528 rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
1530 if (ret == AVERROR(EAGAIN)) {
1531 /* no incoming data to handle */
1533 } else if (ret < 0) {
1535 } else if (ret == 1) {
1536 RTMPPacket rpkt = { 0 };
1538 if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
1540 rt->prev_pkt[0], c)) <= 0)
1543 if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
1546 ff_rtmp_packet_destroy(&rpkt);
1552 #define OFFSET(x) offsetof(RTMPContext, x)
1553 #define DEC AV_OPT_FLAG_DECODING_PARAM
1554 #define ENC AV_OPT_FLAG_ENCODING_PARAM
1556 static const AVOption rtmp_options[] = {
1557 {"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1558 {"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {3000}, 0, INT_MAX, DEC|ENC},
1559 {"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1560 {"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1561 {"rtmp_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {10}, 0, INT_MAX, ENC},
1562 {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
1563 {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
1564 {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
1565 {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
1566 {"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
1567 {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1568 {"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1569 {"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1573 static const AVClass rtmp_class = {
1574 .class_name = "rtmp",
1575 .item_name = av_default_item_name,
1576 .option = rtmp_options,
1577 .version = LIBAVUTIL_VERSION_INT,
1580 URLProtocol ff_rtmp_protocol = {
1582 .url_open = rtmp_open,
1583 .url_read = rtmp_read,
1584 .url_write = rtmp_write,
1585 .url_close = rtmp_close,
1586 .priv_data_size = sizeof(RTMPContext),
1587 .flags = URL_PROTOCOL_FLAG_NETWORK,
1588 .priv_data_class= &rtmp_class,
1591 static const AVClass rtmpe_class = {
1592 .class_name = "rtmpe",
1593 .item_name = av_default_item_name,
1594 .option = rtmp_options,
1595 .version = LIBAVUTIL_VERSION_INT,
1598 URLProtocol ff_rtmpe_protocol = {
1600 .url_open = rtmp_open,
1601 .url_read = rtmp_read,
1602 .url_write = rtmp_write,
1603 .url_close = rtmp_close,
1604 .priv_data_size = sizeof(RTMPContext),
1605 .flags = URL_PROTOCOL_FLAG_NETWORK,
1606 .priv_data_class = &rtmpe_class,
1609 static const AVClass rtmps_class = {
1610 .class_name = "rtmps",
1611 .item_name = av_default_item_name,
1612 .option = rtmp_options,
1613 .version = LIBAVUTIL_VERSION_INT,
1616 URLProtocol ff_rtmps_protocol = {
1618 .url_open = rtmp_open,
1619 .url_read = rtmp_read,
1620 .url_write = rtmp_write,
1621 .url_close = rtmp_close,
1622 .priv_data_size = sizeof(RTMPContext),
1623 .flags = URL_PROTOCOL_FLAG_NETWORK,
1624 .priv_data_class = &rtmps_class,
1627 static const AVClass rtmpt_class = {
1628 .class_name = "rtmpt",
1629 .item_name = av_default_item_name,
1630 .option = rtmp_options,
1631 .version = LIBAVUTIL_VERSION_INT,
1634 URLProtocol ff_rtmpt_protocol = {
1636 .url_open = rtmp_open,
1637 .url_read = rtmp_read,
1638 .url_write = rtmp_write,
1639 .url_close = rtmp_close,
1640 .priv_data_size = sizeof(RTMPContext),
1641 .flags = URL_PROTOCOL_FLAG_NETWORK,
1642 .priv_data_class = &rtmpt_class,
1645 static const AVClass rtmpte_class = {
1646 .class_name = "rtmpte",
1647 .item_name = av_default_item_name,
1648 .option = rtmp_options,
1649 .version = LIBAVUTIL_VERSION_INT,
1652 URLProtocol ff_rtmpte_protocol = {
1654 .url_open = rtmp_open,
1655 .url_read = rtmp_read,
1656 .url_write = rtmp_write,
1657 .url_close = rtmp_close,
1658 .priv_data_size = sizeof(RTMPContext),
1659 .flags = URL_PROTOCOL_FLAG_NETWORK,
1660 .priv_data_class = &rtmpte_class,
1663 static const AVClass rtmpts_class = {
1664 .class_name = "rtmpts",
1665 .item_name = av_default_item_name,
1666 .option = rtmp_options,
1667 .version = LIBAVUTIL_VERSION_INT,
1670 URLProtocol ff_rtmpts_protocol = {
1672 .url_open = rtmp_open,
1673 .url_read = rtmp_read,
1674 .url_write = rtmp_write,
1675 .url_close = rtmp_close,
1676 .priv_data_size = sizeof(RTMPContext),
1677 .flags = URL_PROTOCOL_FLAG_NETWORK,
1678 .priv_data_class = &rtmpts_class,