Mercurial > libavformat.hg
annotate rtmpproto.c @ 6114:10c9d304794f libavformat
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author | reimar |
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date | Tue, 08 Jun 2010 19:31:08 +0000 |
parents | 7fdda2416684 |
children | bd851edc58ac |
rev | line source |
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5123 | 1 /* |
2 * RTMP network protocol | |
3 * Copyright (c) 2009 Kostya Shishkov | |
4 * | |
5 * This file is part of FFmpeg. | |
6 * | |
7 * FFmpeg is free software; you can redistribute it and/or | |
8 * modify it under the terms of the GNU Lesser General Public | |
9 * License as published by the Free Software Foundation; either | |
10 * version 2.1 of the License, or (at your option) any later version. | |
11 * | |
12 * FFmpeg is distributed in the hope that it will be useful, | |
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
18 * License along with FFmpeg; if not, write to the Free Software | |
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
20 */ | |
21 | |
22 /** | |
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23 * @file |
5123 | 24 * RTMP protocol |
25 */ | |
26 | |
27 #include "libavcodec/bytestream.h" | |
28 #include "libavutil/avstring.h" | |
29 #include "libavutil/lfg.h" | |
30 #include "libavutil/sha.h" | |
31 #include "avformat.h" | |
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32 #include "internal.h" |
5123 | 33 |
34 #include "network.h" | |
35 | |
36 #include "flv.h" | |
37 #include "rtmp.h" | |
38 #include "rtmppkt.h" | |
39 | |
40 /* we can't use av_log() with URLContext yet... */ | |
41 #if LIBAVFORMAT_VERSION_MAJOR < 53 | |
42 #define LOG_CONTEXT NULL | |
43 #else | |
44 #define LOG_CONTEXT s | |
45 #endif | |
46 | |
5432 | 47 //#define DEBUG |
48 | |
5123 | 49 /** RTMP protocol handler state */ |
50 typedef enum { | |
51 STATE_START, ///< client has not done anything yet | |
52 STATE_HANDSHAKED, ///< client has performed handshake | |
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53 STATE_RELEASING, ///< client releasing stream before publish it (for output) |
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54 STATE_FCPUBLISH, ///< client FCPublishing stream (for output) |
5123 | 55 STATE_CONNECTING, ///< client connected to server successfully |
56 STATE_READY, ///< client has sent all needed commands and waits for server reply | |
57 STATE_PLAYING, ///< client has started receiving multimedia data from server | |
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58 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output) |
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59 STATE_STOPPED, ///< the broadcast has been stopped |
5123 | 60 } ClientState; |
61 | |
62 /** protocol handler context */ | |
63 typedef struct RTMPContext { | |
64 URLContext* stream; ///< TCP stream used in interactions with RTMP server | |
65 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets | |
66 int chunk_size; ///< size of the chunks RTMP packets are divided into | |
5407 | 67 int is_input; ///< input/output flag |
5123 | 68 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix) |
5411 | 69 char app[128]; ///< application |
5123 | 70 ClientState state; ///< current state |
71 int main_channel_id; ///< an additional channel ID which is used for some invocations | |
72 uint8_t* flv_data; ///< buffer with data for demuxer | |
73 int flv_size; ///< current buffer size | |
74 int flv_off; ///< number of bytes read from current buffer | |
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75 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output) |
5668 | 76 uint32_t client_report_size; ///< number of bytes after which client should report to server |
77 uint32_t bytes_read; ///< number of bytes read from server | |
78 uint32_t last_bytes_read; ///< number of bytes read last reported to server | |
5123 | 79 } RTMPContext; |
80 | |
81 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing | |
82 /** Client key used for digest signing */ | |
83 static const uint8_t rtmp_player_key[] = { | |
84 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ', | |
85 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1', | |
86 | |
87 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02, | |
88 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8, | |
89 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE | |
90 }; | |
91 | |
92 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing | |
93 /** Key used for RTMP server digest signing */ | |
94 static const uint8_t rtmp_server_key[] = { | |
95 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ', | |
96 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ', | |
97 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1', | |
98 | |
99 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02, | |
100 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8, | |
101 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE | |
102 }; | |
103 | |
104 /** | |
105 * Generates 'connect' call and sends it to the server. | |
106 */ | |
107 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto, | |
5411 | 108 const char *host, int port) |
5123 | 109 { |
110 RTMPPacket pkt; | |
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111 uint8_t ver[64], *p; |
5123 | 112 char tcurl[512]; |
113 | |
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114 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096); |
5123 | 115 p = pkt.data; |
116 | |
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117 ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app); |
5123 | 118 ff_amf_write_string(&p, "connect"); |
119 ff_amf_write_number(&p, 1.0); | |
120 ff_amf_write_object_start(&p); | |
121 ff_amf_write_field_name(&p, "app"); | |
5411 | 122 ff_amf_write_string(&p, rt->app); |
5123 | 123 |
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124 if (rt->is_input) { |
5417 | 125 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, |
126 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4); | |
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127 } else { |
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128 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT); |
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129 ff_amf_write_field_name(&p, "type"); |
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130 ff_amf_write_string(&p, "nonprivate"); |
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131 } |
5123 | 132 ff_amf_write_field_name(&p, "flashVer"); |
133 ff_amf_write_string(&p, ver); | |
134 ff_amf_write_field_name(&p, "tcUrl"); | |
135 ff_amf_write_string(&p, tcurl); | |
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136 if (rt->is_input) { |
5417 | 137 ff_amf_write_field_name(&p, "fpad"); |
138 ff_amf_write_bool(&p, 0); | |
139 ff_amf_write_field_name(&p, "capabilities"); | |
140 ff_amf_write_number(&p, 15.0); | |
141 ff_amf_write_field_name(&p, "audioCodecs"); | |
142 ff_amf_write_number(&p, 1639.0); | |
143 ff_amf_write_field_name(&p, "videoCodecs"); | |
144 ff_amf_write_number(&p, 252.0); | |
145 ff_amf_write_field_name(&p, "videoFunction"); | |
146 ff_amf_write_number(&p, 1.0); | |
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147 } |
5123 | 148 ff_amf_write_object_end(&p); |
149 | |
150 pkt.data_size = p - pkt.data; | |
151 | |
152 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); | |
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153 ff_rtmp_packet_destroy(&pkt); |
5123 | 154 } |
155 | |
156 /** | |
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157 * Generates 'releaseStream' call and sends it to the server. It should make |
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158 * the server release some channel for media streams. |
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159 */ |
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160 static void gen_release_stream(URLContext *s, RTMPContext *rt) |
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161 { |
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162 RTMPPacket pkt; |
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163 uint8_t *p; |
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164 |
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165 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, |
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166 29 + strlen(rt->playpath)); |
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167 |
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168 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n"); |
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169 p = pkt.data; |
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170 ff_amf_write_string(&p, "releaseStream"); |
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171 ff_amf_write_number(&p, 2.0); |
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172 ff_amf_write_null(&p); |
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173 ff_amf_write_string(&p, rt->playpath); |
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174 |
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175 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); |
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176 ff_rtmp_packet_destroy(&pkt); |
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177 } |
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178 |
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179 /** |
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180 * Generates 'FCPublish' call and sends it to the server. It should make |
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181 * the server preapare for receiving media streams. |
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182 */ |
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183 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt) |
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184 { |
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185 RTMPPacket pkt; |
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186 uint8_t *p; |
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187 |
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188 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, |
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189 25 + strlen(rt->playpath)); |
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190 |
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191 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n"); |
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192 p = pkt.data; |
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193 ff_amf_write_string(&p, "FCPublish"); |
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194 ff_amf_write_number(&p, 3.0); |
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195 ff_amf_write_null(&p); |
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196 ff_amf_write_string(&p, rt->playpath); |
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197 |
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198 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); |
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199 ff_rtmp_packet_destroy(&pkt); |
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200 } |
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201 |
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202 /** |
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203 * Generates 'FCUnpublish' call and sends it to the server. It should make |
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204 * the server destroy stream. |
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205 */ |
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206 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt) |
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207 { |
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208 RTMPPacket pkt; |
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209 uint8_t *p; |
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210 |
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211 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, |
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212 27 + strlen(rt->playpath)); |
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213 |
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214 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n"); |
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215 p = pkt.data; |
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216 ff_amf_write_string(&p, "FCUnpublish"); |
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217 ff_amf_write_number(&p, 5.0); |
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218 ff_amf_write_null(&p); |
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219 ff_amf_write_string(&p, rt->playpath); |
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220 |
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221 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); |
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222 ff_rtmp_packet_destroy(&pkt); |
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223 } |
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224 |
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225 /** |
5123 | 226 * Generates 'createStream' call and sends it to the server. It should make |
227 * the server allocate some channel for media streams. | |
228 */ | |
229 static void gen_create_stream(URLContext *s, RTMPContext *rt) | |
230 { | |
231 RTMPPacket pkt; | |
232 uint8_t *p; | |
233 | |
234 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n"); | |
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235 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25); |
5123 | 236 |
237 p = pkt.data; | |
238 ff_amf_write_string(&p, "createStream"); | |
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239 ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0); |
5123 | 240 ff_amf_write_null(&p); |
241 | |
242 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); | |
243 ff_rtmp_packet_destroy(&pkt); | |
244 } | |
245 | |
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246 |
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247 /** |
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248 * Generates 'deleteStream' call and sends it to the server. It should make |
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249 * the server remove some channel for media streams. |
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250 */ |
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251 static void gen_delete_stream(URLContext *s, RTMPContext *rt) |
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252 { |
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253 RTMPPacket pkt; |
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254 uint8_t *p; |
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255 |
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256 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n"); |
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257 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34); |
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258 |
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259 p = pkt.data; |
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260 ff_amf_write_string(&p, "deleteStream"); |
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261 ff_amf_write_number(&p, 0.0); |
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262 ff_amf_write_null(&p); |
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263 ff_amf_write_number(&p, rt->main_channel_id); |
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264 |
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265 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); |
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266 ff_rtmp_packet_destroy(&pkt); |
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267 } |
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268 |
5123 | 269 /** |
270 * Generates 'play' call and sends it to the server, then pings the server | |
271 * to start actual playing. | |
272 */ | |
273 static void gen_play(URLContext *s, RTMPContext *rt) | |
274 { | |
275 RTMPPacket pkt; | |
276 uint8_t *p; | |
277 | |
278 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath); | |
279 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, | |
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280 20 + strlen(rt->playpath)); |
5123 | 281 pkt.extra = rt->main_channel_id; |
282 | |
283 p = pkt.data; | |
284 ff_amf_write_string(&p, "play"); | |
285 ff_amf_write_number(&p, 0.0); | |
286 ff_amf_write_null(&p); | |
287 ff_amf_write_string(&p, rt->playpath); | |
288 | |
289 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); | |
290 ff_rtmp_packet_destroy(&pkt); | |
291 | |
292 // set client buffer time disguised in ping packet | |
293 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10); | |
294 | |
295 p = pkt.data; | |
296 bytestream_put_be16(&p, 3); | |
297 bytestream_put_be32(&p, 1); | |
298 bytestream_put_be32(&p, 256); //TODO: what is a good value here? | |
299 | |
300 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); | |
301 ff_rtmp_packet_destroy(&pkt); | |
302 } | |
303 | |
304 /** | |
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305 * Generates 'publish' call and sends it to the server. |
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306 */ |
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307 static void gen_publish(URLContext *s, RTMPContext *rt) |
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308 { |
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309 RTMPPacket pkt; |
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310 uint8_t *p; |
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311 |
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312 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath); |
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313 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0, |
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314 30 + strlen(rt->playpath)); |
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315 pkt.extra = rt->main_channel_id; |
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316 |
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317 p = pkt.data; |
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318 ff_amf_write_string(&p, "publish"); |
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319 ff_amf_write_number(&p, 0.0); |
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320 ff_amf_write_null(&p); |
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321 ff_amf_write_string(&p, rt->playpath); |
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322 ff_amf_write_string(&p, "live"); |
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323 |
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324 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); |
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325 ff_rtmp_packet_destroy(&pkt); |
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326 } |
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327 |
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328 /** |
5123 | 329 * Generates ping reply and sends it to the server. |
330 */ | |
331 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt) | |
332 { | |
333 RTMPPacket pkt; | |
334 uint8_t *p; | |
335 | |
336 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6); | |
337 p = pkt.data; | |
338 bytestream_put_be16(&p, 7); | |
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339 bytestream_put_be32(&p, AV_RB32(ppkt->data+2)); |
5123 | 340 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); |
341 ff_rtmp_packet_destroy(&pkt); | |
342 } | |
343 | |
5668 | 344 /** |
345 * Generates report on bytes read so far and sends it to the server. | |
346 */ | |
347 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts) | |
348 { | |
349 RTMPPacket pkt; | |
350 uint8_t *p; | |
351 | |
352 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4); | |
353 p = pkt.data; | |
354 bytestream_put_be32(&p, rt->bytes_read); | |
355 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); | |
356 ff_rtmp_packet_destroy(&pkt); | |
357 } | |
358 | |
5123 | 359 //TODO: Move HMAC code somewhere. Eventually. |
360 #define HMAC_IPAD_VAL 0x36 | |
361 #define HMAC_OPAD_VAL 0x5C | |
362 | |
363 /** | |
364 * Calculates HMAC-SHA2 digest for RTMP handshake packets. | |
365 * | |
366 * @param src input buffer | |
367 * @param len input buffer length (should be 1536) | |
368 * @param gap offset in buffer where 32 bytes should not be taken into account | |
369 * when calculating digest (since it will be used to store that digest) | |
370 * @param key digest key | |
371 * @param keylen digest key length | |
372 * @param dst buffer where calculated digest will be stored (32 bytes) | |
373 */ | |
374 static void rtmp_calc_digest(const uint8_t *src, int len, int gap, | |
375 const uint8_t *key, int keylen, uint8_t *dst) | |
376 { | |
377 struct AVSHA *sha; | |
378 uint8_t hmac_buf[64+32] = {0}; | |
379 int i; | |
380 | |
381 sha = av_mallocz(av_sha_size); | |
382 | |
383 if (keylen < 64) { | |
384 memcpy(hmac_buf, key, keylen); | |
385 } else { | |
386 av_sha_init(sha, 256); | |
387 av_sha_update(sha,key, keylen); | |
388 av_sha_final(sha, hmac_buf); | |
389 } | |
390 for (i = 0; i < 64; i++) | |
391 hmac_buf[i] ^= HMAC_IPAD_VAL; | |
392 | |
393 av_sha_init(sha, 256); | |
394 av_sha_update(sha, hmac_buf, 64); | |
395 if (gap <= 0) { | |
396 av_sha_update(sha, src, len); | |
397 } else { //skip 32 bytes used for storing digest | |
398 av_sha_update(sha, src, gap); | |
399 av_sha_update(sha, src + gap + 32, len - gap - 32); | |
400 } | |
401 av_sha_final(sha, hmac_buf + 64); | |
402 | |
403 for (i = 0; i < 64; i++) | |
404 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad | |
405 av_sha_init(sha, 256); | |
406 av_sha_update(sha, hmac_buf, 64+32); | |
407 av_sha_final(sha, dst); | |
408 | |
409 av_free(sha); | |
410 } | |
411 | |
412 /** | |
413 * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest | |
414 * will be stored) into that packet. | |
415 * | |
416 * @param buf handshake data (1536 bytes) | |
417 * @return offset to the digest inside input data | |
418 */ | |
419 static int rtmp_handshake_imprint_with_digest(uint8_t *buf) | |
420 { | |
421 int i, digest_pos = 0; | |
422 | |
423 for (i = 8; i < 12; i++) | |
424 digest_pos += buf[i]; | |
425 digest_pos = (digest_pos % 728) + 12; | |
426 | |
427 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos, | |
428 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN, | |
429 buf + digest_pos); | |
430 return digest_pos; | |
431 } | |
432 | |
433 /** | |
434 * Verifies that the received server response has the expected digest value. | |
435 * | |
436 * @param buf handshake data received from the server (1536 bytes) | |
437 * @param off position to search digest offset from | |
438 * @return 0 if digest is valid, digest position otherwise | |
439 */ | |
440 static int rtmp_validate_digest(uint8_t *buf, int off) | |
441 { | |
442 int i, digest_pos = 0; | |
443 uint8_t digest[32]; | |
444 | |
445 for (i = 0; i < 4; i++) | |
446 digest_pos += buf[i + off]; | |
447 digest_pos = (digest_pos % 728) + off + 4; | |
448 | |
449 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos, | |
450 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN, | |
451 digest); | |
452 if (!memcmp(digest, buf + digest_pos, 32)) | |
453 return digest_pos; | |
454 return 0; | |
455 } | |
456 | |
457 /** | |
458 * Performs handshake with the server by means of exchanging pseudorandom data | |
459 * signed with HMAC-SHA2 digest. | |
460 * | |
461 * @return 0 if handshake succeeds, negative value otherwise | |
462 */ | |
463 static int rtmp_handshake(URLContext *s, RTMPContext *rt) | |
464 { | |
465 AVLFG rnd; | |
466 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = { | |
467 3, // unencrypted data | |
468 0, 0, 0, 0, // client uptime | |
469 RTMP_CLIENT_VER1, | |
470 RTMP_CLIENT_VER2, | |
471 RTMP_CLIENT_VER3, | |
472 RTMP_CLIENT_VER4, | |
473 }; | |
474 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE]; | |
475 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1]; | |
476 int i; | |
477 int server_pos, client_pos; | |
478 uint8_t digest[32]; | |
479 | |
480 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n"); | |
481 | |
482 av_lfg_init(&rnd, 0xDEADC0DE); | |
483 // generate handshake packet - 1536 bytes of pseudorandom data | |
484 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++) | |
485 tosend[i] = av_lfg_get(&rnd) >> 24; | |
486 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1); | |
487 | |
488 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1); | |
489 i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1); | |
490 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) { | |
491 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n"); | |
492 return -1; | |
493 } | |
494 i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE); | |
495 if (i != RTMP_HANDSHAKE_PACKET_SIZE) { | |
496 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n"); | |
497 return -1; | |
498 } | |
499 | |
500 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n", | |
501 serverdata[5], serverdata[6], serverdata[7], serverdata[8]); | |
502 | |
5523 | 503 if (rt->is_input && serverdata[5] >= 3) { |
5417 | 504 server_pos = rtmp_validate_digest(serverdata + 1, 772); |
5123 | 505 if (!server_pos) { |
5417 | 506 server_pos = rtmp_validate_digest(serverdata + 1, 8); |
507 if (!server_pos) { | |
508 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n"); | |
509 return -1; | |
510 } | |
511 } | |
512 | |
513 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0, | |
514 rtmp_server_key, sizeof(rtmp_server_key), | |
515 digest); | |
516 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0, | |
517 digest, 32, | |
518 digest); | |
519 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) { | |
520 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n"); | |
5123 | 521 return -1; |
522 } | |
523 | |
5417 | 524 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++) |
525 tosend[i] = av_lfg_get(&rnd) >> 24; | |
526 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0, | |
527 rtmp_player_key, sizeof(rtmp_player_key), | |
528 digest); | |
529 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0, | |
530 digest, 32, | |
531 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32); | |
5123 | 532 |
5417 | 533 // write reply back to the server |
534 url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE); | |
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535 } else { |
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536 url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE); |
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537 } |
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538 |
5123 | 539 return 0; |
540 } | |
541 | |
542 /** | |
543 * Parses received packet and may perform some action depending on | |
544 * the packet contents. | |
545 * @return 0 for no errors, negative values for serious errors which prevent | |
546 * further communications, positive values for uncritical errors | |
547 */ | |
548 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt) | |
549 { | |
550 int i, t; | |
551 const uint8_t *data_end = pkt->data + pkt->data_size; | |
552 | |
5432 | 553 #ifdef DEBUG |
554 ff_rtmp_packet_dump(LOG_CONTEXT, pkt); | |
555 #endif | |
556 | |
5123 | 557 switch (pkt->type) { |
558 case RTMP_PT_CHUNK_SIZE: | |
559 if (pkt->data_size != 4) { | |
560 av_log(LOG_CONTEXT, AV_LOG_ERROR, | |
561 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size); | |
562 return -1; | |
563 } | |
5416
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564 if (!rt->is_input) |
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565 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]); |
5123 | 566 rt->chunk_size = AV_RB32(pkt->data); |
567 if (rt->chunk_size <= 0) { | |
568 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size); | |
569 return -1; | |
570 } | |
571 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size); | |
572 break; | |
573 case RTMP_PT_PING: | |
574 t = AV_RB16(pkt->data); | |
575 if (t == 6) | |
576 gen_pong(s, rt, pkt); | |
577 break; | |
5668 | 578 case RTMP_PT_CLIENT_BW: |
579 if (pkt->data_size < 4) { | |
580 av_log(LOG_CONTEXT, AV_LOG_ERROR, | |
581 "Client bandwidth report packet is less than 4 bytes long (%d)\n", | |
582 pkt->data_size); | |
583 return -1; | |
584 } | |
585 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data)); | |
586 rt->client_report_size = AV_RB32(pkt->data) >> 1; | |
587 break; | |
5123 | 588 case RTMP_PT_INVOKE: |
589 //TODO: check for the messages sent for wrong state? | |
590 if (!memcmp(pkt->data, "\002\000\006_error", 9)) { | |
591 uint8_t tmpstr[256]; | |
592 | |
593 if (!ff_amf_get_field_value(pkt->data + 9, data_end, | |
594 "description", tmpstr, sizeof(tmpstr))) | |
595 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr); | |
596 return -1; | |
597 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) { | |
598 switch (rt->state) { | |
599 case STATE_HANDSHAKED: | |
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600 if (!rt->is_input) { |
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601 gen_release_stream(s, rt); |
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602 gen_fcpublish_stream(s, rt); |
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603 rt->state = STATE_RELEASING; |
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604 } else { |
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605 rt->state = STATE_CONNECTING; |
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606 } |
5123 | 607 gen_create_stream(s, rt); |
5416
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608 break; |
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609 case STATE_FCPUBLISH: |
5123 | 610 rt->state = STATE_CONNECTING; |
611 break; | |
5416
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612 case STATE_RELEASING: |
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613 rt->state = STATE_FCPUBLISH; |
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614 /* hack for Wowza Media Server, it does not send result for |
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615 * releaseStream and FCPublish calls */ |
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616 if (!pkt->data[10]) { |
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617 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11)); |
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618 if (pkt_id == 4) |
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619 rt->state = STATE_CONNECTING; |
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620 } |
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621 if (rt->state != STATE_CONNECTING) |
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622 break; |
5123 | 623 case STATE_CONNECTING: |
624 //extract a number from the result | |
625 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) { | |
626 av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n"); | |
627 } else { | |
628 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21)); | |
629 } | |
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630 if (rt->is_input) { |
5417 | 631 gen_play(s, rt); |
5416
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632 } else { |
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633 gen_publish(s, rt); |
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634 } |
5123 | 635 rt->state = STATE_READY; |
636 break; | |
637 } | |
638 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) { | |
639 const uint8_t* ptr = pkt->data + 11; | |
640 uint8_t tmpstr[256]; | |
641 | |
642 for (i = 0; i < 2; i++) { | |
643 t = ff_amf_tag_size(ptr, data_end); | |
644 if (t < 0) | |
645 return 1; | |
646 ptr += t; | |
647 } | |
648 t = ff_amf_get_field_value(ptr, data_end, | |
649 "level", tmpstr, sizeof(tmpstr)); | |
650 if (!t && !strcmp(tmpstr, "error")) { | |
651 if (!ff_amf_get_field_value(ptr, data_end, | |
652 "description", tmpstr, sizeof(tmpstr))) | |
653 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr); | |
654 return -1; | |
655 } | |
656 t = ff_amf_get_field_value(ptr, data_end, | |
657 "code", tmpstr, sizeof(tmpstr)); | |
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658 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING; |
5430
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659 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED; |
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660 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED; |
5416
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661 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING; |
5123 | 662 } |
663 break; | |
664 } | |
665 return 0; | |
666 } | |
667 | |
668 /** | |
669 * Interacts with the server by receiving and sending RTMP packets until | |
670 * there is some significant data (media data or expected status notification). | |
671 * | |
672 * @param s reading context | |
5367 | 673 * @param for_header non-zero value tells function to work until it |
674 * gets notification from the server that playing has been started, | |
675 * otherwise function will work until some media data is received (or | |
676 * an error happens) | |
5123 | 677 * @return 0 for successful operation, negative value in case of error |
678 */ | |
679 static int get_packet(URLContext *s, int for_header) | |
680 { | |
681 RTMPContext *rt = s->priv_data; | |
682 int ret; | |
5524
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683 uint8_t *p; |
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684 const uint8_t *next; |
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685 uint32_t data_size; |
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686 uint32_t ts, cts, pts=0; |
5123 | 687 |
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688 if (rt->state == STATE_STOPPED) |
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689 return AVERROR_EOF; |
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690 |
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691 for (;;) { |
5123 | 692 RTMPPacket rpkt; |
693 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt, | |
5606
2023dcca33ff
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694 rt->chunk_size, rt->prev_pkt[0])) <= 0) { |
5605
32fe8ab7e6b6
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|
695 if (ret == 0) { |
5123 | 696 return AVERROR(EAGAIN); |
697 } else { | |
698 return AVERROR(EIO); | |
699 } | |
700 } | |
5668 | 701 rt->bytes_read += ret; |
702 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) { | |
703 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending bytes read report\n"); | |
704 gen_bytes_read(s, rt, rpkt.timestamp + 1); | |
705 rt->last_bytes_read = rt->bytes_read; | |
706 } | |
5123 | 707 |
708 ret = rtmp_parse_result(s, rt, &rpkt); | |
709 if (ret < 0) {//serious error in current packet | |
710 ff_rtmp_packet_destroy(&rpkt); | |
711 return -1; | |
712 } | |
5430
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713 if (rt->state == STATE_STOPPED) { |
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714 ff_rtmp_packet_destroy(&rpkt); |
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715 return AVERROR_EOF; |
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|
716 } |
5416
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717 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) { |
5123 | 718 ff_rtmp_packet_destroy(&rpkt); |
719 return 0; | |
720 } | |
5416
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721 if (!rpkt.data_size || !rt->is_input) { |
5123 | 722 ff_rtmp_packet_destroy(&rpkt); |
723 continue; | |
724 } | |
725 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO || | |
5376 | 726 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) { |
5524
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|
727 ts = rpkt.timestamp; |
5123 | 728 |
729 // generate packet header and put data into buffer for FLV demuxer | |
730 rt->flv_off = 0; | |
731 rt->flv_size = rpkt.data_size + 15; | |
732 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size); | |
733 bytestream_put_byte(&p, rpkt.type); | |
734 bytestream_put_be24(&p, rpkt.data_size); | |
735 bytestream_put_be24(&p, ts); | |
736 bytestream_put_byte(&p, ts >> 24); | |
737 bytestream_put_be24(&p, 0); | |
738 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size); | |
739 bytestream_put_be32(&p, 0); | |
740 ff_rtmp_packet_destroy(&rpkt); | |
741 return 0; | |
742 } else if (rpkt.type == RTMP_PT_METADATA) { | |
743 // we got raw FLV data, make it available for FLV demuxer | |
744 rt->flv_off = 0; | |
745 rt->flv_size = rpkt.data_size; | |
746 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size); | |
5524
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|
747 /* rewrite timestamps */ |
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748 next = rpkt.data; |
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|
749 ts = rpkt.timestamp; |
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diff
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750 while (next - rpkt.data < rpkt.data_size - 11) { |
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diff
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|
751 next++; |
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|
752 data_size = bytestream_get_be24(&next); |
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|
753 p=next; |
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diff
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|
754 cts = bytestream_get_be24(&next); |
6055
1e49d06baf93
24l trocadero: RTMP reader forgot to shift high byte of timestamp to its
kostya
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diff
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|
755 cts |= bytestream_get_byte(&next) << 24; |
5524
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diff
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756 if (pts==0) |
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|
757 pts=cts; |
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|
758 ts += cts - pts; |
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|
759 pts = cts; |
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|
760 bytestream_put_be24(&p, ts); |
043b36cefe89
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|
761 bytestream_put_byte(&p, ts >> 24); |
043b36cefe89
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|
762 next += data_size + 3 + 4; |
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|
763 } |
5123 | 764 memcpy(rt->flv_data, rpkt.data, rpkt.data_size); |
765 ff_rtmp_packet_destroy(&rpkt); | |
766 return 0; | |
767 } | |
768 ff_rtmp_packet_destroy(&rpkt); | |
769 } | |
770 return 0; | |
771 } | |
772 | |
773 static int rtmp_close(URLContext *h) | |
774 { | |
775 RTMPContext *rt = h->priv_data; | |
776 | |
5420
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777 if (!rt->is_input) { |
5416
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778 rt->flv_data = NULL; |
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779 if (rt->out_pkt.data_size) |
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780 ff_rtmp_packet_destroy(&rt->out_pkt); |
5419
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781 if (rt->state > STATE_FCPUBLISH) |
006c95a7a3d2
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diff
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|
782 gen_fcunpublish_stream(h, rt); |
5416
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783 } |
5419
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784 if (rt->state > STATE_HANDSHAKED) |
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785 gen_delete_stream(h, rt); |
5416
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|
786 |
5123 | 787 av_freep(&rt->flv_data); |
788 url_close(rt->stream); | |
789 av_free(rt); | |
790 return 0; | |
791 } | |
792 | |
793 /** | |
794 * Opens RTMP connection and verifies that the stream can be played. | |
795 * | |
796 * URL syntax: rtmp://server[:port][/app][/playpath] | |
797 * where 'app' is first one or two directories in the path | |
798 * (e.g. /ondemand/, /flash/live/, etc.) | |
799 * and 'playpath' is a file name (the rest of the path, | |
800 * may be prefixed with "mp4:") | |
801 */ | |
802 static int rtmp_open(URLContext *s, const char *uri, int flags) | |
803 { | |
804 RTMPContext *rt; | |
5411 | 805 char proto[8], hostname[256], path[1024], *fname; |
5123 | 806 uint8_t buf[2048]; |
5407 | 807 int port; |
5123 | 808 int ret; |
809 | |
810 rt = av_mallocz(sizeof(RTMPContext)); | |
811 if (!rt) | |
812 return AVERROR(ENOMEM); | |
813 s->priv_data = rt; | |
5407 | 814 rt->is_input = !(flags & URL_WRONLY); |
5123 | 815 |
5775 | 816 ff_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port, |
5776 | 817 path, sizeof(path), s->filename); |
5123 | 818 |
819 if (port < 0) | |
820 port = RTMP_DEFAULT_PORT; | |
5756
7c7fe75728dd
Use ff_url_join for assembling URLs, instead of snprintf
mstorsjo
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diff
changeset
|
821 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL); |
5123 | 822 |
5377
9313acce85fc
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changeset
|
823 if (url_open(&rt->stream, buf, URL_RDWR) < 0) { |
9313acce85fc
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diff
changeset
|
824 av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf); |
5123 | 825 goto fail; |
5377
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diff
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|
826 } |
5123 | 827 |
5417 | 828 rt->state = STATE_START; |
829 if (rtmp_handshake(s, rt)) | |
830 return -1; | |
5123 | 831 |
5417 | 832 rt->chunk_size = 128; |
833 rt->state = STATE_HANDSHAKED; | |
834 //extract "app" part from path | |
835 if (!strncmp(path, "/ondemand/", 10)) { | |
836 fname = path + 10; | |
837 memcpy(rt->app, "ondemand", 9); | |
838 } else { | |
839 char *p = strchr(path + 1, '/'); | |
840 if (!p) { | |
841 fname = path + 1; | |
842 rt->app[0] = '\0'; | |
5123 | 843 } else { |
5417 | 844 char *c = strchr(p + 1, ':'); |
845 fname = strchr(p + 1, '/'); | |
846 if (!fname || c < fname) { | |
847 fname = p + 1; | |
848 av_strlcpy(rt->app, path + 1, p - path); | |
5123 | 849 } else { |
5417 | 850 fname++; |
851 av_strlcpy(rt->app, path + 1, fname - path - 1); | |
5123 | 852 } |
853 } | |
5417 | 854 } |
855 if (!strchr(fname, ':') && | |
856 (!strcmp(fname + strlen(fname) - 4, ".f4v") || | |
857 !strcmp(fname + strlen(fname) - 4, ".mp4"))) { | |
858 memcpy(rt->playpath, "mp4:", 5); | |
859 } else { | |
860 rt->playpath[0] = 0; | |
861 } | |
862 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5); | |
5123 | 863 |
5668 | 864 rt->client_report_size = 1048576; |
865 rt->bytes_read = 0; | |
866 rt->last_bytes_read = 0; | |
867 | |
5417 | 868 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n", |
869 proto, path, rt->app, rt->playpath); | |
870 gen_connect(s, rt, proto, hostname, port); | |
5123 | 871 |
5417 | 872 do { |
873 ret = get_packet(s, 1); | |
874 } while (ret == EAGAIN); | |
875 if (ret < 0) | |
876 goto fail; | |
5416
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877 |
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878 if (rt->is_input) { |
5123 | 879 // generate FLV header for demuxer |
880 rt->flv_size = 13; | |
881 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size); | |
882 rt->flv_off = 0; | |
883 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size); | |
5416
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884 } else { |
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885 rt->flv_size = 0; |
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886 rt->flv_data = NULL; |
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887 rt->flv_off = 0; |
5123 | 888 } |
889 | |
890 s->max_packet_size = url_get_max_packet_size(rt->stream); | |
891 s->is_streamed = 1; | |
892 return 0; | |
893 | |
894 fail: | |
895 rtmp_close(s); | |
896 return AVERROR(EIO); | |
897 } | |
898 | |
899 static int rtmp_read(URLContext *s, uint8_t *buf, int size) | |
900 { | |
901 RTMPContext *rt = s->priv_data; | |
902 int orig_size = size; | |
903 int ret; | |
904 | |
905 while (size > 0) { | |
906 int data_left = rt->flv_size - rt->flv_off; | |
907 | |
908 if (data_left >= size) { | |
909 memcpy(buf, rt->flv_data + rt->flv_off, size); | |
910 rt->flv_off += size; | |
911 return orig_size; | |
912 } | |
913 if (data_left > 0) { | |
914 memcpy(buf, rt->flv_data + rt->flv_off, data_left); | |
915 buf += data_left; | |
916 size -= data_left; | |
917 rt->flv_off = rt->flv_size; | |
918 } | |
919 if ((ret = get_packet(s, 0)) < 0) | |
920 return ret; | |
921 } | |
922 return orig_size; | |
923 } | |
924 | |
6068 | 925 static int rtmp_write(URLContext *h, const uint8_t *buf, int size) |
5123 | 926 { |
5416
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927 RTMPContext *rt = h->priv_data; |
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928 int size_temp = size; |
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929 int pktsize, pkttype; |
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930 uint32_t ts; |
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931 const uint8_t *buf_temp = buf; |
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932 |
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933 if (size < 11) { |
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934 av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size); |
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935 return 0; |
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936 } |
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937 |
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938 do { |
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939 if (!rt->flv_off) { |
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940 //skip flv header |
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941 if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') { |
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942 buf_temp += 9 + 4; |
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943 size_temp -= 9 + 4; |
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944 } |
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945 |
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946 pkttype = bytestream_get_byte(&buf_temp); |
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947 pktsize = bytestream_get_be24(&buf_temp); |
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948 ts = bytestream_get_be24(&buf_temp); |
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949 ts |= bytestream_get_byte(&buf_temp) << 24; |
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950 bytestream_get_be24(&buf_temp); |
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951 size_temp -= 11; |
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952 rt->flv_size = pktsize; |
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953 |
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954 //force 12bytes header |
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955 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) || |
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956 pkttype == RTMP_PT_NOTIFY) { |
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957 if (pkttype == RTMP_PT_NOTIFY) |
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958 pktsize += 16; |
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959 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0; |
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960 } |
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961 |
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962 //this can be a big packet, it's better to send it right here |
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963 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize); |
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964 rt->out_pkt.extra = rt->main_channel_id; |
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965 rt->flv_data = rt->out_pkt.data; |
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966 |
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967 if (pkttype == RTMP_PT_NOTIFY) |
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968 ff_amf_write_string(&rt->flv_data, "@setDataFrame"); |
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969 } |
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970 |
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971 if (rt->flv_size - rt->flv_off > size_temp) { |
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972 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp); |
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973 rt->flv_off += size_temp; |
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974 } else { |
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975 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off); |
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976 rt->flv_off += rt->flv_size - rt->flv_off; |
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977 } |
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978 |
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979 if (rt->flv_off == rt->flv_size) { |
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980 bytestream_get_be32(&buf_temp); |
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981 |
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982 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]); |
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983 ff_rtmp_packet_destroy(&rt->out_pkt); |
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984 rt->flv_size = 0; |
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985 rt->flv_off = 0; |
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986 } |
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987 } while (buf_temp - buf < size_temp); |
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988 return size; |
5123 | 989 } |
990 | |
991 URLProtocol rtmp_protocol = { | |
992 "rtmp", | |
993 rtmp_open, | |
994 rtmp_read, | |
995 rtmp_write, | |
996 NULL, /* seek */ | |
997 rtmp_close, | |
998 }; |