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1
/*
2
 * RTMP network protocol
3
 * Copyright (c) 2009 Kostya Shishkov
4
 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21

    
22
/**
23
 * @file
24
 * RTMP protocol
25
 */
26

    
27
#include "libavcodec/bytestream.h"
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#include "libavutil/avstring.h"
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#include "libavutil/lfg.h"
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#include "libavutil/sha.h"
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#include "avformat.h"
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#include "internal.h"
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... */
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#if FF_API_URL_CLASS
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#define LOG_CONTEXT s
43
#else
44
#define LOG_CONTEXT NULL
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#endif
46

    
47
//#define DEBUG
48

    
49
/** RTMP protocol handler state */
50
typedef enum {
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    STATE_START,      ///< client has not done anything yet
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    STATE_HANDSHAKED, ///< client has performed handshake
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    STATE_RELEASING,  ///< client releasing stream before publish it (for output)
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    STATE_FCPUBLISH,  ///< client FCPublishing stream (for output)
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    STATE_CONNECTING, ///< client connected to server successfully
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    STATE_READY,      ///< client has sent all needed commands and waits for server reply
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    STATE_PLAYING,    ///< client has started receiving multimedia data from server
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    STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
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    STATE_STOPPED,    ///< the broadcast has been stopped
60
} ClientState;
61

    
62
/** protocol handler context */
63
typedef struct RTMPContext {
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    URLContext*   stream;                     ///< TCP stream used in interactions with RTMP server
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    RTMPPacket    prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
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    int           chunk_size;                 ///< size of the chunks RTMP packets are divided into
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    int           is_input;                   ///< input/output flag
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    char          playpath[256];              ///< path to filename to play (with possible "mp4:" prefix)
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    char          app[128];                   ///< application
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    ClientState   state;                      ///< current state
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    int           main_channel_id;            ///< an additional channel ID which is used for some invocations
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    uint8_t*      flv_data;                   ///< buffer with data for demuxer
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    int           flv_size;                   ///< current buffer size
74
    int           flv_off;                    ///< number of bytes read from current buffer
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    RTMPPacket    out_pkt;                    ///< rtmp packet, created from flv a/v or metadata (for output)
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    uint32_t      client_report_size;         ///< number of bytes after which client should report to server
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    uint32_t      bytes_read;                 ///< number of bytes read from server
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    uint32_t      last_bytes_read;            ///< number of bytes read last reported to server
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', ' ',
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    '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,
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    0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
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    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', ' ',
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    'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
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    'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
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    0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
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    0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
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    0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
102
};
103

    
104
/**
105
 * Generate 'connect' call and send it to the server.
106
 */
107
static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
108
                        const char *host, int port)
109
{
110
    RTMPPacket pkt;
111
    uint8_t ver[64], *p;
112
    char tcurl[512];
113

    
114
    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
115
    p = pkt.data;
116

    
117
    ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
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");
122
    ff_amf_write_string(&p, rt->app);
123

    
124
    if (rt->is_input) {
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);
127
    } else {
128
        snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
129
        ff_amf_write_field_name(&p, "type");
130
        ff_amf_write_string(&p, "nonprivate");
131
    }
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);
136
    if (rt->is_input) {
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);
147
    }
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]);
153
    ff_rtmp_packet_destroy(&pkt);
154
}
155

    
156
/**
157
 * Generate 'releaseStream' call and send it to the server. It should make
158
 * the server release some channel for media streams.
159
 */
160
static void gen_release_stream(URLContext *s, RTMPContext *rt)
161
{
162
    RTMPPacket pkt;
163
    uint8_t *p;
164

    
165
    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
166
                          29 + strlen(rt->playpath));
167

    
168
    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n");
169
    p = pkt.data;
170
    ff_amf_write_string(&p, "releaseStream");
171
    ff_amf_write_number(&p, 2.0);
172
    ff_amf_write_null(&p);
173
    ff_amf_write_string(&p, rt->playpath);
174

    
175
    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
176
    ff_rtmp_packet_destroy(&pkt);
177
}
178

    
179
/**
180
 * Generate 'FCPublish' call and send it to the server. It should make
181
 * the server preapare for receiving media streams.
182
 */
183
static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
184
{
185
    RTMPPacket pkt;
186
    uint8_t *p;
187

    
188
    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
189
                          25 + strlen(rt->playpath));
190

    
191
    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n");
192
    p = pkt.data;
193
    ff_amf_write_string(&p, "FCPublish");
194
    ff_amf_write_number(&p, 3.0);
195
    ff_amf_write_null(&p);
196
    ff_amf_write_string(&p, rt->playpath);
197

    
198
    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
199
    ff_rtmp_packet_destroy(&pkt);
200
}
201

    
202
/**
203
 * Generate 'FCUnpublish' call and send it to the server. It should make
204
 * the server destroy stream.
205
 */
206
static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
207
{
208
    RTMPPacket pkt;
209
    uint8_t *p;
210

    
211
    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
212
                          27 + strlen(rt->playpath));
213

    
214
    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n");
215
    p = pkt.data;
216
    ff_amf_write_string(&p, "FCUnpublish");
217
    ff_amf_write_number(&p, 5.0);
218
    ff_amf_write_null(&p);
219
    ff_amf_write_string(&p, rt->playpath);
220

    
221
    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
222
    ff_rtmp_packet_destroy(&pkt);
223
}
224

    
225
/**
226
 * Generate 'createStream' call and send 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");
235
    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
236

    
237
    p = pkt.data;
238
    ff_amf_write_string(&p, "createStream");
239
    ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
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

    
246

    
247
/**
248
 * Generate 'deleteStream' call and send it to the server. It should make
249
 * the server remove some channel for media streams.
250
 */
251
static void gen_delete_stream(URLContext *s, RTMPContext *rt)
252
{
253
    RTMPPacket pkt;
254
    uint8_t *p;
255

    
256
    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n");
257
    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
258

    
259
    p = pkt.data;
260
    ff_amf_write_string(&p, "deleteStream");
261
    ff_amf_write_number(&p, 0.0);
262
    ff_amf_write_null(&p);
263
    ff_amf_write_number(&p, rt->main_channel_id);
264

    
265
    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
266
    ff_rtmp_packet_destroy(&pkt);
267
}
268

    
269
/**
270
 * Generate 'play' call and send it to the server, then ping 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,
280
                          20 + strlen(rt->playpath));
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
/**
305
 * Generate 'publish' call and send it to the server.
306
 */
307
static void gen_publish(URLContext *s, RTMPContext *rt)
308
{
309
    RTMPPacket pkt;
310
    uint8_t *p;
311

    
312
    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
313
    ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
314
                          30 + strlen(rt->playpath));
315
    pkt.extra = rt->main_channel_id;
316

    
317
    p = pkt.data;
318
    ff_amf_write_string(&p, "publish");
319
    ff_amf_write_number(&p, 0.0);
320
    ff_amf_write_null(&p);
321
    ff_amf_write_string(&p, rt->playpath);
322
    ff_amf_write_string(&p, "live");
323

    
324
    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
325
    ff_rtmp_packet_destroy(&pkt);
326
}
327

    
328
/**
329
 * Generate ping reply and send 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);
339
    bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
340
    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
341
    ff_rtmp_packet_destroy(&pkt);
342
}
343

    
344
/**
345
 * Generate report on bytes read so far and send 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

    
359
//TODO: Move HMAC code somewhere. Eventually.
360
#define HMAC_IPAD_VAL 0x36
361
#define HMAC_OPAD_VAL 0x5C
362

    
363
/**
364
 * Calculate 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
 * Put 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
 * Verify 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
 * Perform 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

    
503
    if (rt->is_input && serverdata[5] >= 3) {
504
        server_pos = rtmp_validate_digest(serverdata + 1, 772);
505
        if (!server_pos) {
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");
521
            return -1;
522
        }
523

    
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);
532

    
533
        // write reply back to the server
534
        url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
535
    } else {
536
        url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
537
    }
538

    
539
    return 0;
540
}
541

    
542
/**
543
 * Parse received packet and possibly 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

    
553
#ifdef DEBUG
554
    ff_rtmp_packet_dump(LOG_CONTEXT, pkt);
555
#endif
556

    
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
        }
564
        if (!rt->is_input)
565
            ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
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;
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;
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:
600
                if (!rt->is_input) {
601
                    gen_release_stream(s, rt);
602
                    gen_fcpublish_stream(s, rt);
603
                    rt->state = STATE_RELEASING;
604
                } else {
605
                    rt->state = STATE_CONNECTING;
606
                }
607
                gen_create_stream(s, rt);
608
                break;
609
            case STATE_FCPUBLISH:
610
                rt->state = STATE_CONNECTING;
611
                break;
612
            case STATE_RELEASING:
613
                rt->state = STATE_FCPUBLISH;
614
                /* hack for Wowza Media Server, it does not send result for
615
                 * releaseStream and FCPublish calls */
616
                if (!pkt->data[10]) {
617
                    int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
618
                    if (pkt_id == 4)
619
                        rt->state = STATE_CONNECTING;
620
                }
621
                if (rt->state != STATE_CONNECTING)
622
                    break;
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
                }
630
                if (rt->is_input) {
631
                    gen_play(s, rt);
632
                } else {
633
                    gen_publish(s, rt);
634
                }
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));
658
            if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
659
            if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
660
            if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
661
            if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
662
        }
663
        break;
664
    }
665
    return 0;
666
}
667

    
668
/**
669
 * Interact 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
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)
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;
683
    uint8_t *p;
684
    const uint8_t *next;
685
    uint32_t data_size;
686
    uint32_t ts, cts, pts=0;
687

    
688
    if (rt->state == STATE_STOPPED)
689
        return AVERROR_EOF;
690

    
691
    for (;;) {
692
        RTMPPacket rpkt;
693
        if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
694
                                       rt->chunk_size, rt->prev_pkt[0])) <= 0) {
695
            if (ret == 0) {
696
                return AVERROR(EAGAIN);
697
            } else {
698
                return AVERROR(EIO);
699
            }
700
        }
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
        }
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
        }
713
        if (rt->state == STATE_STOPPED) {
714
            ff_rtmp_packet_destroy(&rpkt);
715
            return AVERROR_EOF;
716
        }
717
        if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
718
            ff_rtmp_packet_destroy(&rpkt);
719
            return 0;
720
        }
721
        if (!rpkt.data_size || !rt->is_input) {
722
            ff_rtmp_packet_destroy(&rpkt);
723
            continue;
724
        }
725
        if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
726
           (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
727
            ts = rpkt.timestamp;
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);
747
            /* rewrite timestamps */
748
            next = rpkt.data;
749
            ts = rpkt.timestamp;
750
            while (next - rpkt.data < rpkt.data_size - 11) {
751
                next++;
752
                data_size = bytestream_get_be24(&next);
753
                p=next;
754
                cts = bytestream_get_be24(&next);
755
                cts |= bytestream_get_byte(&next) << 24;
756
                if (pts==0)
757
                    pts=cts;
758
                ts += cts - pts;
759
                pts = cts;
760
                bytestream_put_be24(&p, ts);
761
                bytestream_put_byte(&p, ts >> 24);
762
                next += data_size + 3 + 4;
763
            }
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

    
777
    if (!rt->is_input) {
778
        rt->flv_data = NULL;
779
        if (rt->out_pkt.data_size)
780
            ff_rtmp_packet_destroy(&rt->out_pkt);
781
        if (rt->state > STATE_FCPUBLISH)
782
            gen_fcunpublish_stream(h, rt);
783
    }
784
    if (rt->state > STATE_HANDSHAKED)
785
        gen_delete_stream(h, rt);
786

    
787
    av_freep(&rt->flv_data);
788
    url_close(rt->stream);
789
    av_free(rt);
790
    return 0;
791
}
792

    
793
/**
794
 * Open RTMP connection and verify 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;
805
    char proto[8], hostname[256], path[1024], *fname;
806
    uint8_t buf[2048];
807
    int port;
808
    int ret;
809

    
810
    rt = av_mallocz(sizeof(RTMPContext));
811
    if (!rt)
812
        return AVERROR(ENOMEM);
813
    s->priv_data = rt;
814
    rt->is_input = !(flags & URL_WRONLY);
815

    
816
    av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
817
                 path, sizeof(path), s->filename);
818

    
819
    if (port < 0)
820
        port = RTMP_DEFAULT_PORT;
821
    ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
822

    
823
    if (url_open(&rt->stream, buf, URL_RDWR) < 0) {
824
        av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);
825
        goto fail;
826
    }
827

    
828
    rt->state = STATE_START;
829
    if (rtmp_handshake(s, rt))
830
        return -1;
831

    
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';
843
        } else {
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);
849
            } else {
850
                fname++;
851
                av_strlcpy(rt->app, path + 1, fname - path - 1);
852
            }
853
        }
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);
863

    
864
    rt->client_report_size = 1048576;
865
    rt->bytes_read = 0;
866
    rt->last_bytes_read = 0;
867

    
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);
871

    
872
    do {
873
        ret = get_packet(s, 1);
874
    } while (ret == EAGAIN);
875
    if (ret < 0)
876
        goto fail;
877

    
878
    if (rt->is_input) {
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);
884
    } else {
885
        rt->flv_size = 0;
886
        rt->flv_data = NULL;
887
        rt->flv_off  = 0;
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
            return data_left;
919
        }
920
        if ((ret = get_packet(s, 0)) < 0)
921
           return ret;
922
    }
923
    return orig_size;
924
}
925

    
926
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
927
{
928
    RTMPContext *rt = s->priv_data;
929
    int size_temp = size;
930
    int pktsize, pkttype;
931
    uint32_t ts;
932
    const uint8_t *buf_temp = buf;
933

    
934
    if (size < 11) {
935
        av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
936
        return 0;
937
    }
938

    
939
    do {
940
        if (!rt->flv_off) {
941
            //skip flv header
942
            if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
943
                buf_temp += 9 + 4;
944
                size_temp -= 9 + 4;
945
            }
946

    
947
            pkttype = bytestream_get_byte(&buf_temp);
948
            pktsize = bytestream_get_be24(&buf_temp);
949
            ts = bytestream_get_be24(&buf_temp);
950
            ts |= bytestream_get_byte(&buf_temp) << 24;
951
            bytestream_get_be24(&buf_temp);
952
            size_temp -= 11;
953
            rt->flv_size = pktsize;
954

    
955
            //force 12bytes header
956
            if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
957
                pkttype == RTMP_PT_NOTIFY) {
958
                if (pkttype == RTMP_PT_NOTIFY)
959
                    pktsize += 16;
960
                rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
961
            }
962

    
963
            //this can be a big packet, it's better to send it right here
964
            ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
965
            rt->out_pkt.extra = rt->main_channel_id;
966
            rt->flv_data = rt->out_pkt.data;
967

    
968
            if (pkttype == RTMP_PT_NOTIFY)
969
                ff_amf_write_string(&rt->flv_data, "@setDataFrame");
970
        }
971

    
972
        if (rt->flv_size - rt->flv_off > size_temp) {
973
            bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
974
            rt->flv_off += size_temp;
975
        } else {
976
            bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
977
            rt->flv_off += rt->flv_size - rt->flv_off;
978
        }
979

    
980
        if (rt->flv_off == rt->flv_size) {
981
            bytestream_get_be32(&buf_temp);
982

    
983
            ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
984
            ff_rtmp_packet_destroy(&rt->out_pkt);
985
            rt->flv_size = 0;
986
            rt->flv_off = 0;
987
        }
988
    } while (buf_temp - buf < size_temp);
989
    return size;
990
}
991

    
992
URLProtocol ff_rtmp_protocol = {
993
    "rtmp",
994
    rtmp_open,
995
    rtmp_read,
996
    rtmp_write,
997
    NULL, /* seek */
998
    rtmp_close,
999
};