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",
933 return AVERROR_INVALIDDATA;
936 rt->client_report_size = AV_RB32(pkt->data);
937 if (rt->client_report_size <= 0) {
938 av_log(s, AV_LOG_ERROR, "Incorrect client bandwidth %d\n",
939 rt->client_report_size);
940 return AVERROR_INVALIDDATA;
943 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", rt->client_report_size);
944 rt->client_report_size >>= 1;
949 static int handle_server_bw(URLContext *s, RTMPPacket *pkt)
951 RTMPContext *rt = s->priv_data;
953 if (pkt->data_size < 4) {
954 av_log(s, AV_LOG_ERROR,
955 "Too short server bandwidth report packet (%d)\n",
957 return AVERROR_INVALIDDATA;
960 rt->server_bw = AV_RB32(pkt->data);
961 if (rt->server_bw <= 0) {
962 av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n",
964 return AVERROR_INVALIDDATA;
966 av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);
971 static int handle_invoke(URLContext *s, RTMPPacket *pkt)
973 RTMPContext *rt = s->priv_data;
975 const uint8_t *data_end = pkt->data + pkt->data_size;
978 //TODO: check for the messages sent for wrong state?
979 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
982 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
983 "description", tmpstr, sizeof(tmpstr)))
984 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
986 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
988 case STATE_HANDSHAKED:
990 if ((ret = gen_release_stream(s, rt)) < 0)
992 if ((ret = gen_fcpublish_stream(s, rt)) < 0)
994 rt->state = STATE_RELEASING;
996 if ((ret = gen_server_bw(s, rt)) < 0)
998 rt->state = STATE_CONNECTING;
1000 if ((ret = gen_create_stream(s, rt)) < 0)
1003 case STATE_FCPUBLISH:
1004 rt->state = STATE_CONNECTING;
1006 case STATE_RELEASING:
1007 rt->state = STATE_FCPUBLISH;
1008 /* hack for Wowza Media Server, it does not send result for
1009 * releaseStream and FCPublish calls */
1010 if (!pkt->data[10]) {
1011 int pkt_id = av_int2double(AV_RB64(pkt->data + 11));
1012 if (pkt_id == rt->create_stream_invoke)
1013 rt->state = STATE_CONNECTING;
1015 if (rt->state != STATE_CONNECTING)
1017 case STATE_CONNECTING:
1018 //extract a number from the result
1019 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
1020 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
1022 rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
1025 if ((ret = gen_play(s, rt)) < 0)
1027 if ((ret = gen_buffer_time(s, rt)) < 0)
1030 if ((ret = gen_publish(s, rt)) < 0)
1033 rt->state = STATE_READY;
1036 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
1037 const uint8_t* ptr = pkt->data + 11;
1038 uint8_t tmpstr[256];
1040 for (i = 0; i < 2; i++) {
1041 t = ff_amf_tag_size(ptr, data_end);
1046 t = ff_amf_get_field_value(ptr, data_end,
1047 "level", tmpstr, sizeof(tmpstr));
1048 if (!t && !strcmp(tmpstr, "error")) {
1049 if (!ff_amf_get_field_value(ptr, data_end,
1050 "description", tmpstr, sizeof(tmpstr)))
1051 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
1054 t = ff_amf_get_field_value(ptr, data_end,
1055 "code", tmpstr, sizeof(tmpstr));
1056 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
1057 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
1058 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
1059 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
1060 } else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
1061 if ((ret = gen_check_bw(s, rt)) < 0)
1069 * Parse received packet and possibly perform some action depending on
1070 * the packet contents.
1071 * @return 0 for no errors, negative values for serious errors which prevent
1072 * further communications, positive values for uncritical errors
1074 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
1079 ff_rtmp_packet_dump(s, pkt);
1082 switch (pkt->type) {
1083 case RTMP_PT_CHUNK_SIZE:
1084 if ((ret = handle_chunk_size(s, pkt)) < 0)
1088 if ((ret = handle_ping(s, pkt)) < 0)
1091 case RTMP_PT_CLIENT_BW:
1092 if ((ret = handle_client_bw(s, pkt)) < 0)
1095 case RTMP_PT_SERVER_BW:
1096 if ((ret = handle_server_bw(s, pkt)) < 0)
1099 case RTMP_PT_INVOKE:
1100 if ((ret = handle_invoke(s, pkt)) < 0)
1105 /* Audio and Video packets are parsed in get_packet() */
1108 av_log(s, AV_LOG_VERBOSE, "Unknown packet type received 0x%02X\n", pkt->type);
1115 * Interact with the server by receiving and sending RTMP packets until
1116 * there is some significant data (media data or expected status notification).
1118 * @param s reading context
1119 * @param for_header non-zero value tells function to work until it
1120 * gets notification from the server that playing has been started,
1121 * otherwise function will work until some media data is received (or
1123 * @return 0 for successful operation, negative value in case of error
1125 static int get_packet(URLContext *s, int for_header)
1127 RTMPContext *rt = s->priv_data;
1130 const uint8_t *next;
1132 uint32_t ts, cts, pts=0;
1134 if (rt->state == STATE_STOPPED)
1138 RTMPPacket rpkt = { 0 };
1139 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
1140 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
1142 return AVERROR(EAGAIN);
1144 return AVERROR(EIO);
1147 rt->bytes_read += ret;
1148 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
1149 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
1150 if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
1152 rt->last_bytes_read = rt->bytes_read;
1155 ret = rtmp_parse_result(s, rt, &rpkt);
1156 if (ret < 0) {//serious error in current packet
1157 ff_rtmp_packet_destroy(&rpkt);
1160 if (rt->state == STATE_STOPPED) {
1161 ff_rtmp_packet_destroy(&rpkt);
1164 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
1165 ff_rtmp_packet_destroy(&rpkt);
1168 if (!rpkt.data_size || !rt->is_input) {
1169 ff_rtmp_packet_destroy(&rpkt);
1172 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
1173 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
1174 ts = rpkt.timestamp;
1176 // generate packet header and put data into buffer for FLV demuxer
1178 rt->flv_size = rpkt.data_size + 15;
1179 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
1180 bytestream_put_byte(&p, rpkt.type);
1181 bytestream_put_be24(&p, rpkt.data_size);
1182 bytestream_put_be24(&p, ts);
1183 bytestream_put_byte(&p, ts >> 24);
1184 bytestream_put_be24(&p, 0);
1185 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
1186 bytestream_put_be32(&p, 0);
1187 ff_rtmp_packet_destroy(&rpkt);
1189 } else if (rpkt.type == RTMP_PT_METADATA) {
1190 // we got raw FLV data, make it available for FLV demuxer
1192 rt->flv_size = rpkt.data_size;
1193 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
1194 /* rewrite timestamps */
1196 ts = rpkt.timestamp;
1197 while (next - rpkt.data < rpkt.data_size - 11) {
1199 data_size = bytestream_get_be24(&next);
1201 cts = bytestream_get_be24(&next);
1202 cts |= bytestream_get_byte(&next) << 24;
1207 bytestream_put_be24(&p, ts);
1208 bytestream_put_byte(&p, ts >> 24);
1209 next += data_size + 3 + 4;
1211 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
1212 ff_rtmp_packet_destroy(&rpkt);
1215 ff_rtmp_packet_destroy(&rpkt);
1219 static int rtmp_close(URLContext *h)
1221 RTMPContext *rt = h->priv_data;
1224 if (!rt->is_input) {
1225 rt->flv_data = NULL;
1226 if (rt->out_pkt.data_size)
1227 ff_rtmp_packet_destroy(&rt->out_pkt);
1228 if (rt->state > STATE_FCPUBLISH)
1229 ret = gen_fcunpublish_stream(h, rt);
1231 if (rt->state > STATE_HANDSHAKED)
1232 ret = gen_delete_stream(h, rt);
1234 av_freep(&rt->flv_data);
1235 ffurl_close(rt->stream);
1240 * Open RTMP connection and verify that the stream can be played.
1242 * URL syntax: rtmp://server[:port][/app][/playpath]
1243 * where 'app' is first one or two directories in the path
1244 * (e.g. /ondemand/, /flash/live/, etc.)
1245 * and 'playpath' is a file name (the rest of the path,
1246 * may be prefixed with "mp4:")
1248 static int rtmp_open(URLContext *s, const char *uri, int flags)
1250 RTMPContext *rt = s->priv_data;
1251 char proto[8], hostname[256], path[1024], *fname;
1255 AVDictionary *opts = NULL;
1258 rt->is_input = !(flags & AVIO_FLAG_WRITE);
1260 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
1261 path, sizeof(path), s->filename);
1263 if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
1264 if (!strcmp(proto, "rtmpts"))
1265 av_dict_set(&opts, "ffrtmphttp_tls", "1", 1);
1267 /* open the http tunneling connection */
1268 ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
1269 } else if (!strcmp(proto, "rtmps")) {
1270 /* open the tls connection */
1272 port = RTMPS_DEFAULT_PORT;
1273 ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
1274 } else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
1275 if (!strcmp(proto, "rtmpte"))
1276 av_dict_set(&opts, "ffrtmpcrypt_tunneling", "1", 1);
1278 /* open the encrypted connection */
1279 ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
1282 /* open the tcp connection */
1284 port = RTMP_DEFAULT_PORT;
1285 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
1288 if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
1289 &s->interrupt_callback, &opts)) < 0) {
1290 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
1294 rt->state = STATE_START;
1295 if ((ret = rtmp_handshake(s, rt)) < 0)
1298 rt->chunk_size = 128;
1299 rt->state = STATE_HANDSHAKED;
1301 // Keep the application name when it has been defined by the user.
1304 rt->app = av_malloc(APP_MAX_LENGTH);
1306 ret = AVERROR(ENOMEM);
1310 //extract "app" part from path
1311 if (!strncmp(path, "/ondemand/", 10)) {
1313 memcpy(rt->app, "ondemand", 9);
1315 char *next = *path ? path + 1 : path;
1316 char *p = strchr(next, '/');
1321 // make sure we do not mismatch a playpath for an application instance
1322 char *c = strchr(p + 1, ':');
1323 fname = strchr(p + 1, '/');
1324 if (!fname || (c && c < fname)) {
1326 av_strlcpy(rt->app, path + 1, p - path);
1329 av_strlcpy(rt->app, path + 1, fname - path - 1);
1335 // The name of application has been defined by the user, override it.
1340 if (!rt->playpath) {
1341 int len = strlen(fname);
1343 rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
1344 if (!rt->playpath) {
1345 ret = AVERROR(ENOMEM);
1349 if (!strchr(fname, ':') && len >= 4 &&
1350 (!strcmp(fname + len - 4, ".f4v") ||
1351 !strcmp(fname + len - 4, ".mp4"))) {
1352 memcpy(rt->playpath, "mp4:", 5);
1353 } else if (len >= 4 && !strcmp(fname + len - 4, ".flv")) {
1354 fname[len - 4] = '\0';
1356 rt->playpath[0] = 0;
1358 strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
1362 rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
1364 ret = AVERROR(ENOMEM);
1367 ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
1368 port, "/%s", rt->app);
1371 if (!rt->flashver) {
1372 rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
1373 if (!rt->flashver) {
1374 ret = AVERROR(ENOMEM);
1378 snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
1379 RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
1380 RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
1382 snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
1383 "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
1387 rt->client_report_size = 1048576;
1389 rt->last_bytes_read = 0;
1390 rt->server_bw = 2500000;
1392 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
1393 proto, path, rt->app, rt->playpath);
1394 if ((ret = gen_connect(s, rt)) < 0)
1398 ret = get_packet(s, 1);
1399 } while (ret == EAGAIN);
1404 // generate FLV header for demuxer
1406 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
1408 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
1411 rt->flv_data = NULL;
1413 rt->skip_bytes = 13;
1416 s->max_packet_size = rt->stream->max_packet_size;
1421 av_dict_free(&opts);
1426 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
1428 RTMPContext *rt = s->priv_data;
1429 int orig_size = size;
1433 int data_left = rt->flv_size - rt->flv_off;
1435 if (data_left >= size) {
1436 memcpy(buf, rt->flv_data + rt->flv_off, size);
1437 rt->flv_off += size;
1440 if (data_left > 0) {
1441 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
1444 rt->flv_off = rt->flv_size;
1447 if ((ret = get_packet(s, 0)) < 0)
1453 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
1455 RTMPContext *rt = s->priv_data;
1456 int size_temp = size;
1457 int pktsize, pkttype;
1459 const uint8_t *buf_temp = buf;
1464 if (rt->skip_bytes) {
1465 int skip = FFMIN(rt->skip_bytes, size_temp);
1468 rt->skip_bytes -= skip;
1472 if (rt->flv_header_bytes < 11) {
1473 const uint8_t *header = rt->flv_header;
1474 int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
1475 bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
1476 rt->flv_header_bytes += copy;
1478 if (rt->flv_header_bytes < 11)
1481 pkttype = bytestream_get_byte(&header);
1482 pktsize = bytestream_get_be24(&header);
1483 ts = bytestream_get_be24(&header);
1484 ts |= bytestream_get_byte(&header) << 24;
1485 bytestream_get_be24(&header);
1486 rt->flv_size = pktsize;
1488 //force 12bytes header
1489 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
1490 pkttype == RTMP_PT_NOTIFY) {
1491 if (pkttype == RTMP_PT_NOTIFY)
1493 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
1496 //this can be a big packet, it's better to send it right here
1497 if ((ret = ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL,
1498 pkttype, ts, pktsize)) < 0)
1501 rt->out_pkt.extra = rt->main_channel_id;
1502 rt->flv_data = rt->out_pkt.data;
1504 if (pkttype == RTMP_PT_NOTIFY)
1505 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
1508 if (rt->flv_size - rt->flv_off > size_temp) {
1509 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
1510 rt->flv_off += size_temp;
1513 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
1514 size_temp -= rt->flv_size - rt->flv_off;
1515 rt->flv_off += rt->flv_size - rt->flv_off;
1518 if (rt->flv_off == rt->flv_size) {
1521 if ((ret = ff_rtmp_packet_write(rt->stream, &rt->out_pkt,
1522 rt->chunk_size, rt->prev_pkt[1])) < 0)
1524 ff_rtmp_packet_destroy(&rt->out_pkt);
1527 rt->flv_header_bytes = 0;
1528 rt->flv_nb_packets++;
1530 } while (buf_temp - buf < size);
1532 if (rt->flv_nb_packets < rt->flush_interval)
1534 rt->flv_nb_packets = 0;
1536 /* set stream into nonblocking mode */
1537 rt->stream->flags |= AVIO_FLAG_NONBLOCK;
1539 /* try to read one byte from the stream */
1540 ret = ffurl_read(rt->stream, &c, 1);
1542 /* switch the stream back into blocking mode */
1543 rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
1545 if (ret == AVERROR(EAGAIN)) {
1546 /* no incoming data to handle */
1548 } else if (ret < 0) {
1550 } else if (ret == 1) {
1551 RTMPPacket rpkt = { 0 };
1553 if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
1555 rt->prev_pkt[0], c)) <= 0)
1558 if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
1561 ff_rtmp_packet_destroy(&rpkt);
1567 #define OFFSET(x) offsetof(RTMPContext, x)
1568 #define DEC AV_OPT_FLAG_DECODING_PARAM
1569 #define ENC AV_OPT_FLAG_ENCODING_PARAM
1571 static const AVOption rtmp_options[] = {
1572 {"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1573 {"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},
1574 {"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1575 {"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},
1576 {"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},
1577 {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
1578 {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
1579 {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
1580 {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
1581 {"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},
1582 {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1583 {"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},
1584 {"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},
1588 static const AVClass rtmp_class = {
1589 .class_name = "rtmp",
1590 .item_name = av_default_item_name,
1591 .option = rtmp_options,
1592 .version = LIBAVUTIL_VERSION_INT,
1595 URLProtocol ff_rtmp_protocol = {
1597 .url_open = rtmp_open,
1598 .url_read = rtmp_read,
1599 .url_write = rtmp_write,
1600 .url_close = rtmp_close,
1601 .priv_data_size = sizeof(RTMPContext),
1602 .flags = URL_PROTOCOL_FLAG_NETWORK,
1603 .priv_data_class= &rtmp_class,
1606 static const AVClass rtmpe_class = {
1607 .class_name = "rtmpe",
1608 .item_name = av_default_item_name,
1609 .option = rtmp_options,
1610 .version = LIBAVUTIL_VERSION_INT,
1613 URLProtocol ff_rtmpe_protocol = {
1615 .url_open = rtmp_open,
1616 .url_read = rtmp_read,
1617 .url_write = rtmp_write,
1618 .url_close = rtmp_close,
1619 .priv_data_size = sizeof(RTMPContext),
1620 .flags = URL_PROTOCOL_FLAG_NETWORK,
1621 .priv_data_class = &rtmpe_class,
1624 static const AVClass rtmps_class = {
1625 .class_name = "rtmps",
1626 .item_name = av_default_item_name,
1627 .option = rtmp_options,
1628 .version = LIBAVUTIL_VERSION_INT,
1631 URLProtocol ff_rtmps_protocol = {
1633 .url_open = rtmp_open,
1634 .url_read = rtmp_read,
1635 .url_write = rtmp_write,
1636 .url_close = rtmp_close,
1637 .priv_data_size = sizeof(RTMPContext),
1638 .flags = URL_PROTOCOL_FLAG_NETWORK,
1639 .priv_data_class = &rtmps_class,
1642 static const AVClass rtmpt_class = {
1643 .class_name = "rtmpt",
1644 .item_name = av_default_item_name,
1645 .option = rtmp_options,
1646 .version = LIBAVUTIL_VERSION_INT,
1649 URLProtocol ff_rtmpt_protocol = {
1651 .url_open = rtmp_open,
1652 .url_read = rtmp_read,
1653 .url_write = rtmp_write,
1654 .url_close = rtmp_close,
1655 .priv_data_size = sizeof(RTMPContext),
1656 .flags = URL_PROTOCOL_FLAG_NETWORK,
1657 .priv_data_class = &rtmpt_class,
1660 static const AVClass rtmpte_class = {
1661 .class_name = "rtmpte",
1662 .item_name = av_default_item_name,
1663 .option = rtmp_options,
1664 .version = LIBAVUTIL_VERSION_INT,
1667 URLProtocol ff_rtmpte_protocol = {
1669 .url_open = rtmp_open,
1670 .url_read = rtmp_read,
1671 .url_write = rtmp_write,
1672 .url_close = rtmp_close,
1673 .priv_data_size = sizeof(RTMPContext),
1674 .flags = URL_PROTOCOL_FLAG_NETWORK,
1675 .priv_data_class = &rtmpte_class,
1678 static const AVClass rtmpts_class = {
1679 .class_name = "rtmpts",
1680 .item_name = av_default_item_name,
1681 .option = rtmp_options,
1682 .version = LIBAVUTIL_VERSION_INT,
1685 URLProtocol ff_rtmpts_protocol = {
1687 .url_open = rtmp_open,
1688 .url_read = rtmp_read,
1689 .url_write = rtmp_write,
1690 .url_close = rtmp_close,
1691 .priv_data_size = sizeof(RTMPContext),
1692 .flags = URL_PROTOCOL_FLAG_NETWORK,
1693 .priv_data_class = &rtmpts_class,