srtp.h
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1 /*
2  * SRTP encryption/decryption
3  * Copyright (c) 2012 Martin Storsjo
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 #ifndef AVFORMAT_SRTP_H
23 #define AVFORMAT_SRTP_H
24 
25 #include <stdint.h>
26 
27 struct AVAES;
28 struct AVHMAC;
29 
30 struct SRTPContext {
31  struct AVAES *aes;
32  struct AVHMAC *hmac;
40  uint32_t roc;
41 
42  uint32_t rtcp_index;
43 };
44 
45 int ff_srtp_set_crypto(struct SRTPContext *s, const char *suite,
46  const char *params);
47 void ff_srtp_free(struct SRTPContext *s);
48 int ff_srtp_decrypt(struct SRTPContext *s, uint8_t *buf, int *lenptr);
49 int ff_srtp_encrypt(struct SRTPContext *s, const uint8_t *in, int len,
50  uint8_t *out, int outlen);
51 
52 #endif /* AVFORMAT_SRTP_H */
int rtp_hmac_size
Definition: srtp.h:33
const char * s
Definition: avisynth_c.h:668
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
uint8_t rtcp_auth[20]
Definition: srtp.h:38
uint8_t rtp_auth[20]
Definition: srtp.h:38
uint8_t rtcp_salt[14]
Definition: srtp.h:37
int seq_initialized
Definition: srtp.h:39
uint8_t
struct AVHMAC * hmac
Definition: srtp.h:32
uint8_t master_salt[14]
Definition: srtp.h:35
struct AVAES * aes
Definition: srtp.h:31
uint8_t rtcp_key[16]
Definition: srtp.h:36
int seq_largest
Definition: srtp.h:39
FFmpeg currently uses a custom build this text attempts to document some of its obscure features and options Makefile the full command issued by make and its output will be shown on the screen DESTDIR Destination directory for the install useful to prepare packages or install FFmpeg in cross environments Makefile builds all the libraries and the executables fate Run the fate test suite
Definition: build_system.txt:1
int ff_srtp_decrypt(struct SRTPContext *s, uint8_t *buf, int *lenptr)
Definition: srtp.c:126
Definition: hmac.c:31
void ff_srtp_free(struct SRTPContext *s)
Definition: srtp.c:31
uint8_t rtp_key[16]
Definition: srtp.h:36
void * buf
Definition: avisynth_c.h:594
uint8_t master_key[16]
Definition: srtp.h:34
int ff_srtp_set_crypto(struct SRTPContext *s, const char *suite, const char *params)
Definition: srtp.c:65
const char const char * params
Definition: avisynth_c.h:675
uint32_t rtcp_index
Definition: srtp.h:42
int len
int rtcp_hmac_size
Definition: srtp.h:33
uint8_t rtp_salt[14]
Definition: srtp.h:37
int ff_srtp_encrypt(struct SRTPContext *s, const uint8_t *in, int len, uint8_t *out, int outlen)
Definition: srtp.c:238
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
uint32_t roc
Definition: srtp.h:40
Definition: aes.c:34