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 rt->server_bw = AV_RB32(pkt->data);
954 if (rt->server_bw <= 0) {
955 av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n",
957 return AVERROR_INVALIDDATA;
959 av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);
964 static int handle_invoke(URLContext *s, RTMPPacket *pkt)
966 RTMPContext *rt = s->priv_data;
968 const uint8_t *data_end = pkt->data + pkt->data_size;
971 //TODO: check for the messages sent for wrong state?
972 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
975 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
976 "description", tmpstr, sizeof(tmpstr)))
977 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
979 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
981 case STATE_HANDSHAKED:
983 if ((ret = gen_release_stream(s, rt)) < 0)
985 if ((ret = gen_fcpublish_stream(s, rt)) < 0)
987 rt->state = STATE_RELEASING;
989 if ((ret = gen_server_bw(s, rt)) < 0)
991 rt->state = STATE_CONNECTING;
993 if ((ret = gen_create_stream(s, rt)) < 0)
996 case STATE_FCPUBLISH:
997 rt->state = STATE_CONNECTING;
999 case STATE_RELEASING:
1000 rt->state = STATE_FCPUBLISH;
1001 /* hack for Wowza Media Server, it does not send result for
1002 * releaseStream and FCPublish calls */
1003 if (!pkt->data[10]) {
1004 int pkt_id = av_int2double(AV_RB64(pkt->data + 11));
1005 if (pkt_id == rt->create_stream_invoke)
1006 rt->state = STATE_CONNECTING;
1008 if (rt->state != STATE_CONNECTING)
1010 case STATE_CONNECTING:
1011 //extract a number from the result
1012 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
1013 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
1015 rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
1018 if ((ret = gen_play(s, rt)) < 0)
1020 if ((ret = gen_buffer_time(s, rt)) < 0)
1023 if ((ret = gen_publish(s, rt)) < 0)
1026 rt->state = STATE_READY;
1029 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
1030 const uint8_t* ptr = pkt->data + 11;
1031 uint8_t tmpstr[256];
1033 for (i = 0; i < 2; i++) {
1034 t = ff_amf_tag_size(ptr, data_end);
1039 t = ff_amf_get_field_value(ptr, data_end,
1040 "level", tmpstr, sizeof(tmpstr));
1041 if (!t && !strcmp(tmpstr, "error")) {
1042 if (!ff_amf_get_field_value(ptr, data_end,
1043 "description", tmpstr, sizeof(tmpstr)))
1044 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
1047 t = ff_amf_get_field_value(ptr, data_end,
1048 "code", tmpstr, sizeof(tmpstr));
1049 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
1050 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
1051 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
1052 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
1053 } else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
1054 if ((ret = gen_check_bw(s, rt)) < 0)
1062 * Parse received packet and possibly perform some action depending on
1063 * the packet contents.
1064 * @return 0 for no errors, negative values for serious errors which prevent
1065 * further communications, positive values for uncritical errors
1067 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
1072 ff_rtmp_packet_dump(s, pkt);
1075 switch (pkt->type) {
1076 case RTMP_PT_CHUNK_SIZE:
1077 if ((ret = handle_chunk_size(s, pkt)) < 0)
1081 if ((ret = handle_ping(s, pkt)) < 0)
1084 case RTMP_PT_CLIENT_BW:
1085 if ((ret = handle_client_bw(s, pkt)) < 0)
1088 case RTMP_PT_SERVER_BW:
1089 if ((ret = handle_server_bw(s, pkt)) < 0)
1092 case RTMP_PT_INVOKE:
1093 if ((ret = handle_invoke(s, pkt)) < 0)
1098 /* Audio and Video packets are parsed in get_packet() */
1101 av_log(s, AV_LOG_VERBOSE, "Unknown packet type received 0x%02X\n", pkt->type);
1108 * Interact with the server by receiving and sending RTMP packets until
1109 * there is some significant data (media data or expected status notification).
1111 * @param s reading context
1112 * @param for_header non-zero value tells function to work until it
1113 * gets notification from the server that playing has been started,
1114 * otherwise function will work until some media data is received (or
1116 * @return 0 for successful operation, negative value in case of error
1118 static int get_packet(URLContext *s, int for_header)
1120 RTMPContext *rt = s->priv_data;
1123 const uint8_t *next;
1125 uint32_t ts, cts, pts=0;
1127 if (rt->state == STATE_STOPPED)
1131 RTMPPacket rpkt = { 0 };
1132 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
1133 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
1135 return AVERROR(EAGAIN);
1137 return AVERROR(EIO);
1140 rt->bytes_read += ret;
1141 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
1142 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
1143 if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
1145 rt->last_bytes_read = rt->bytes_read;
1148 ret = rtmp_parse_result(s, rt, &rpkt);
1149 if (ret < 0) {//serious error in current packet
1150 ff_rtmp_packet_destroy(&rpkt);
1153 if (rt->state == STATE_STOPPED) {
1154 ff_rtmp_packet_destroy(&rpkt);
1157 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
1158 ff_rtmp_packet_destroy(&rpkt);
1161 if (!rpkt.data_size || !rt->is_input) {
1162 ff_rtmp_packet_destroy(&rpkt);
1165 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
1166 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
1167 ts = rpkt.timestamp;
1169 // generate packet header and put data into buffer for FLV demuxer
1171 rt->flv_size = rpkt.data_size + 15;
1172 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
1173 bytestream_put_byte(&p, rpkt.type);
1174 bytestream_put_be24(&p, rpkt.data_size);
1175 bytestream_put_be24(&p, ts);
1176 bytestream_put_byte(&p, ts >> 24);
1177 bytestream_put_be24(&p, 0);
1178 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
1179 bytestream_put_be32(&p, 0);
1180 ff_rtmp_packet_destroy(&rpkt);
1182 } else if (rpkt.type == RTMP_PT_METADATA) {
1183 // we got raw FLV data, make it available for FLV demuxer
1185 rt->flv_size = rpkt.data_size;
1186 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
1187 /* rewrite timestamps */
1189 ts = rpkt.timestamp;
1190 while (next - rpkt.data < rpkt.data_size - 11) {
1192 data_size = bytestream_get_be24(&next);
1194 cts = bytestream_get_be24(&next);
1195 cts |= bytestream_get_byte(&next) << 24;
1200 bytestream_put_be24(&p, ts);
1201 bytestream_put_byte(&p, ts >> 24);
1202 next += data_size + 3 + 4;
1204 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
1205 ff_rtmp_packet_destroy(&rpkt);
1208 ff_rtmp_packet_destroy(&rpkt);
1212 static int rtmp_close(URLContext *h)
1214 RTMPContext *rt = h->priv_data;
1217 if (!rt->is_input) {
1218 rt->flv_data = NULL;
1219 if (rt->out_pkt.data_size)
1220 ff_rtmp_packet_destroy(&rt->out_pkt);
1221 if (rt->state > STATE_FCPUBLISH)
1222 ret = gen_fcunpublish_stream(h, rt);
1224 if (rt->state > STATE_HANDSHAKED)
1225 ret = gen_delete_stream(h, rt);
1227 av_freep(&rt->flv_data);
1228 ffurl_close(rt->stream);
1233 * Open RTMP connection and verify that the stream can be played.
1235 * URL syntax: rtmp://server[:port][/app][/playpath]
1236 * where 'app' is first one or two directories in the path
1237 * (e.g. /ondemand/, /flash/live/, etc.)
1238 * and 'playpath' is a file name (the rest of the path,
1239 * may be prefixed with "mp4:")
1241 static int rtmp_open(URLContext *s, const char *uri, int flags)
1243 RTMPContext *rt = s->priv_data;
1244 char proto[8], hostname[256], path[1024], *fname;
1248 AVDictionary *opts = NULL;
1251 rt->is_input = !(flags & AVIO_FLAG_WRITE);
1253 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
1254 path, sizeof(path), s->filename);
1256 if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
1257 if (!strcmp(proto, "rtmpts"))
1258 av_dict_set(&opts, "ffrtmphttp_tls", "1", 1);
1260 /* open the http tunneling connection */
1261 ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
1262 } else if (!strcmp(proto, "rtmps")) {
1263 /* open the tls connection */
1265 port = RTMPS_DEFAULT_PORT;
1266 ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
1267 } else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
1268 if (!strcmp(proto, "rtmpte"))
1269 av_dict_set(&opts, "ffrtmpcrypt_tunneling", "1", 1);
1271 /* open the encrypted connection */
1272 ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
1275 /* open the tcp connection */
1277 port = RTMP_DEFAULT_PORT;
1278 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
1281 if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
1282 &s->interrupt_callback, &opts)) < 0) {
1283 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
1287 rt->state = STATE_START;
1288 if ((ret = rtmp_handshake(s, rt)) < 0)
1291 rt->chunk_size = 128;
1292 rt->state = STATE_HANDSHAKED;
1294 // Keep the application name when it has been defined by the user.
1297 rt->app = av_malloc(APP_MAX_LENGTH);
1299 ret = AVERROR(ENOMEM);
1303 //extract "app" part from path
1304 if (!strncmp(path, "/ondemand/", 10)) {
1306 memcpy(rt->app, "ondemand", 9);
1308 char *next = *path ? path + 1 : path;
1309 char *p = strchr(next, '/');
1314 // make sure we do not mismatch a playpath for an application instance
1315 char *c = strchr(p + 1, ':');
1316 fname = strchr(p + 1, '/');
1317 if (!fname || (c && c < fname)) {
1319 av_strlcpy(rt->app, path + 1, p - path);
1322 av_strlcpy(rt->app, path + 1, fname - path - 1);
1328 // The name of application has been defined by the user, override it.
1333 if (!rt->playpath) {
1334 int len = strlen(fname);
1336 rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
1337 if (!rt->playpath) {
1338 ret = AVERROR(ENOMEM);
1342 if (!strchr(fname, ':') && len >= 4 &&
1343 (!strcmp(fname + len - 4, ".f4v") ||
1344 !strcmp(fname + len - 4, ".mp4"))) {
1345 memcpy(rt->playpath, "mp4:", 5);
1346 } else if (len >= 4 && !strcmp(fname + len - 4, ".flv")) {
1347 fname[len - 4] = '\0';
1349 rt->playpath[0] = 0;
1351 strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
1355 rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
1357 ret = AVERROR(ENOMEM);
1360 ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
1361 port, "/%s", rt->app);
1364 if (!rt->flashver) {
1365 rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
1366 if (!rt->flashver) {
1367 ret = AVERROR(ENOMEM);
1371 snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
1372 RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
1373 RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
1375 snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
1376 "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
1380 rt->client_report_size = 1048576;
1382 rt->last_bytes_read = 0;
1383 rt->server_bw = 2500000;
1385 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
1386 proto, path, rt->app, rt->playpath);
1387 if ((ret = gen_connect(s, rt)) < 0)
1391 ret = get_packet(s, 1);
1392 } while (ret == EAGAIN);
1397 // generate FLV header for demuxer
1399 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
1401 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
1404 rt->flv_data = NULL;
1406 rt->skip_bytes = 13;
1409 s->max_packet_size = rt->stream->max_packet_size;
1414 av_dict_free(&opts);
1419 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
1421 RTMPContext *rt = s->priv_data;
1422 int orig_size = size;
1426 int data_left = rt->flv_size - rt->flv_off;
1428 if (data_left >= size) {
1429 memcpy(buf, rt->flv_data + rt->flv_off, size);
1430 rt->flv_off += size;
1433 if (data_left > 0) {
1434 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
1437 rt->flv_off = rt->flv_size;
1440 if ((ret = get_packet(s, 0)) < 0)
1446 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
1448 RTMPContext *rt = s->priv_data;
1449 int size_temp = size;
1450 int pktsize, pkttype;
1452 const uint8_t *buf_temp = buf;
1457 if (rt->skip_bytes) {
1458 int skip = FFMIN(rt->skip_bytes, size_temp);
1461 rt->skip_bytes -= skip;
1465 if (rt->flv_header_bytes < 11) {
1466 const uint8_t *header = rt->flv_header;
1467 int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
1468 bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
1469 rt->flv_header_bytes += copy;
1471 if (rt->flv_header_bytes < 11)
1474 pkttype = bytestream_get_byte(&header);
1475 pktsize = bytestream_get_be24(&header);
1476 ts = bytestream_get_be24(&header);
1477 ts |= bytestream_get_byte(&header) << 24;
1478 bytestream_get_be24(&header);
1479 rt->flv_size = pktsize;
1481 //force 12bytes header
1482 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
1483 pkttype == RTMP_PT_NOTIFY) {
1484 if (pkttype == RTMP_PT_NOTIFY)
1486 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
1489 //this can be a big packet, it's better to send it right here
1490 if ((ret = ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL,
1491 pkttype, ts, pktsize)) < 0)
1494 rt->out_pkt.extra = rt->main_channel_id;
1495 rt->flv_data = rt->out_pkt.data;
1497 if (pkttype == RTMP_PT_NOTIFY)
1498 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
1501 if (rt->flv_size - rt->flv_off > size_temp) {
1502 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
1503 rt->flv_off += size_temp;
1506 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
1507 size_temp -= rt->flv_size - rt->flv_off;
1508 rt->flv_off += rt->flv_size - rt->flv_off;
1511 if (rt->flv_off == rt->flv_size) {
1514 if ((ret = ff_rtmp_packet_write(rt->stream, &rt->out_pkt,
1515 rt->chunk_size, rt->prev_pkt[1])) < 0)
1517 ff_rtmp_packet_destroy(&rt->out_pkt);
1520 rt->flv_header_bytes = 0;
1521 rt->flv_nb_packets++;
1523 } while (buf_temp - buf < size);
1525 if (rt->flv_nb_packets < rt->flush_interval)
1527 rt->flv_nb_packets = 0;
1529 /* set stream into nonblocking mode */
1530 rt->stream->flags |= AVIO_FLAG_NONBLOCK;
1532 /* try to read one byte from the stream */
1533 ret = ffurl_read(rt->stream, &c, 1);
1535 /* switch the stream back into blocking mode */
1536 rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
1538 if (ret == AVERROR(EAGAIN)) {
1539 /* no incoming data to handle */
1541 } else if (ret < 0) {
1543 } else if (ret == 1) {
1544 RTMPPacket rpkt = { 0 };
1546 if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
1548 rt->prev_pkt[0], c)) <= 0)
1551 if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
1554 ff_rtmp_packet_destroy(&rpkt);
1560 #define OFFSET(x) offsetof(RTMPContext, x)
1561 #define DEC AV_OPT_FLAG_DECODING_PARAM
1562 #define ENC AV_OPT_FLAG_ENCODING_PARAM
1564 static const AVOption rtmp_options[] = {
1565 {"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1566 {"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},
1567 {"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1568 {"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},
1569 {"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},
1570 {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
1571 {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
1572 {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
1573 {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
1574 {"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},
1575 {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
1576 {"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},
1577 {"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},
1581 static const AVClass rtmp_class = {
1582 .class_name = "rtmp",
1583 .item_name = av_default_item_name,
1584 .option = rtmp_options,
1585 .version = LIBAVUTIL_VERSION_INT,
1588 URLProtocol ff_rtmp_protocol = {
1590 .url_open = rtmp_open,
1591 .url_read = rtmp_read,
1592 .url_write = rtmp_write,
1593 .url_close = rtmp_close,
1594 .priv_data_size = sizeof(RTMPContext),
1595 .flags = URL_PROTOCOL_FLAG_NETWORK,
1596 .priv_data_class= &rtmp_class,
1599 static const AVClass rtmpe_class = {
1600 .class_name = "rtmpe",
1601 .item_name = av_default_item_name,
1602 .option = rtmp_options,
1603 .version = LIBAVUTIL_VERSION_INT,
1606 URLProtocol ff_rtmpe_protocol = {
1608 .url_open = rtmp_open,
1609 .url_read = rtmp_read,
1610 .url_write = rtmp_write,
1611 .url_close = rtmp_close,
1612 .priv_data_size = sizeof(RTMPContext),
1613 .flags = URL_PROTOCOL_FLAG_NETWORK,
1614 .priv_data_class = &rtmpe_class,
1617 static const AVClass rtmps_class = {
1618 .class_name = "rtmps",
1619 .item_name = av_default_item_name,
1620 .option = rtmp_options,
1621 .version = LIBAVUTIL_VERSION_INT,
1624 URLProtocol ff_rtmps_protocol = {
1626 .url_open = rtmp_open,
1627 .url_read = rtmp_read,
1628 .url_write = rtmp_write,
1629 .url_close = rtmp_close,
1630 .priv_data_size = sizeof(RTMPContext),
1631 .flags = URL_PROTOCOL_FLAG_NETWORK,
1632 .priv_data_class = &rtmps_class,
1635 static const AVClass rtmpt_class = {
1636 .class_name = "rtmpt",
1637 .item_name = av_default_item_name,
1638 .option = rtmp_options,
1639 .version = LIBAVUTIL_VERSION_INT,
1642 URLProtocol ff_rtmpt_protocol = {
1644 .url_open = rtmp_open,
1645 .url_read = rtmp_read,
1646 .url_write = rtmp_write,
1647 .url_close = rtmp_close,
1648 .priv_data_size = sizeof(RTMPContext),
1649 .flags = URL_PROTOCOL_FLAG_NETWORK,
1650 .priv_data_class = &rtmpt_class,
1653 static const AVClass rtmpte_class = {
1654 .class_name = "rtmpte",
1655 .item_name = av_default_item_name,
1656 .option = rtmp_options,
1657 .version = LIBAVUTIL_VERSION_INT,
1660 URLProtocol ff_rtmpte_protocol = {
1662 .url_open = rtmp_open,
1663 .url_read = rtmp_read,
1664 .url_write = rtmp_write,
1665 .url_close = rtmp_close,
1666 .priv_data_size = sizeof(RTMPContext),
1667 .flags = URL_PROTOCOL_FLAG_NETWORK,
1668 .priv_data_class = &rtmpte_class,
1671 static const AVClass rtmpts_class = {
1672 .class_name = "rtmpts",
1673 .item_name = av_default_item_name,
1674 .option = rtmp_options,
1675 .version = LIBAVUTIL_VERSION_INT,
1678 URLProtocol ff_rtmpts_protocol = {
1680 .url_open = rtmp_open,
1681 .url_read = rtmp_read,
1682 .url_write = rtmp_write,
1683 .url_close = rtmp_close,
1684 .priv_data_size = sizeof(RTMPContext),
1685 .flags = URL_PROTOCOL_FLAG_NETWORK,
1686 .priv_data_class = &rtmpts_class,