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	 82bf41f3ab
			
		
	
	82bf41f3ab
	
	
	
		
			
			It avoids leaving dangling pointers behind in memory. Also remove redundant checks for whether the URLContext to be closed is already NULL. Reviewed-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
		
			
				
	
	
		
			146 lines
		
	
	
		
			4.8 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			146 lines
		
	
	
		
			4.8 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * SRTP network protocol
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|  * Copyright (c) 2012 Martin Storsjo
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|  *
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|  * This file is part of FFmpeg.
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|  *
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|  * FFmpeg is free software; you can redistribute it and/or
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|  * modify it under the terms of the GNU Lesser General Public
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|  * License as published by the Free Software Foundation; either
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|  * version 2.1 of the License, or (at your option) any later version.
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|  *
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|  * FFmpeg 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 FFmpeg; 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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|  */
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| 
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| #include "libavutil/opt.h"
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| #include "avformat.h"
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| #include "avio_internal.h"
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| #include "url.h"
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| 
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| #include "internal.h"
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| #include "rtpdec.h"
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| #include "srtp.h"
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| 
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| typedef struct SRTPProtoContext {
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|     const AVClass *class;
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|     URLContext *rtp_hd;
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|     const char *out_suite, *out_params;
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|     const char *in_suite, *in_params;
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|     struct SRTPContext srtp_out, srtp_in;
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|     uint8_t encryptbuf[RTP_MAX_PACKET_LENGTH];
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| } SRTPProtoContext;
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| 
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| #define D AV_OPT_FLAG_DECODING_PARAM
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| #define E AV_OPT_FLAG_ENCODING_PARAM
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| static const AVOption options[] = {
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|     { "srtp_out_suite", "", offsetof(SRTPProtoContext, out_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
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|     { "srtp_out_params", "", offsetof(SRTPProtoContext, out_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
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|     { "srtp_in_suite", "", offsetof(SRTPProtoContext, in_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D },
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|     { "srtp_in_params", "", offsetof(SRTPProtoContext, in_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D },
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|     { NULL }
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| };
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| 
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| static const AVClass srtp_context_class = {
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|     .class_name     = "srtp",
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|     .item_name      = av_default_item_name,
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|     .option         = options,
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|     .version        = LIBAVUTIL_VERSION_INT,
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| };
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| 
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| static int srtp_close(URLContext *h)
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| {
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|     SRTPProtoContext *s = h->priv_data;
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|     ff_srtp_free(&s->srtp_out);
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|     ff_srtp_free(&s->srtp_in);
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|     ffurl_closep(&s->rtp_hd);
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|     return 0;
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| }
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| 
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| static int srtp_open(URLContext *h, const char *uri, int flags)
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| {
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|     SRTPProtoContext *s = h->priv_data;
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|     char hostname[256], buf[1024], path[1024];
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|     int rtp_port, ret;
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| 
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|     if (s->out_suite && s->out_params)
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|         if ((ret = ff_srtp_set_crypto(&s->srtp_out, s->out_suite, s->out_params)) < 0)
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|             goto fail;
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|     if (s->in_suite && s->in_params)
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|         if ((ret = ff_srtp_set_crypto(&s->srtp_in, s->in_suite, s->in_params)) < 0)
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|             goto fail;
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| 
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|     av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &rtp_port,
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|                  path, sizeof(path), uri);
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|     ff_url_join(buf, sizeof(buf), "rtp", NULL, hostname, rtp_port, "%s", path);
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|     if ((ret = ffurl_open_whitelist(&s->rtp_hd, buf, flags, &h->interrupt_callback,
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|                                     NULL, h->protocol_whitelist, h->protocol_blacklist, h)) < 0)
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|         goto fail;
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| 
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|     h->max_packet_size = FFMIN(s->rtp_hd->max_packet_size,
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|                                sizeof(s->encryptbuf)) - 14;
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|     h->is_streamed = 1;
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|     return 0;
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| 
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| fail:
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|     srtp_close(h);
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|     return ret;
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| }
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| 
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| static int srtp_read(URLContext *h, uint8_t *buf, int size)
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| {
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|     SRTPProtoContext *s = h->priv_data;
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|     int ret;
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| start:
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|     ret = ffurl_read(s->rtp_hd, buf, size);
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|     if (ret > 0 && s->srtp_in.aes) {
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|         if (ff_srtp_decrypt(&s->srtp_in, buf, &ret) < 0)
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|             goto start;
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|     }
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|     return ret;
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| }
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| 
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| static int srtp_write(URLContext *h, const uint8_t *buf, int size)
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| {
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|     SRTPProtoContext *s = h->priv_data;
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|     if (!s->srtp_out.aes)
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|         return ffurl_write(s->rtp_hd, buf, size);
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|     size = ff_srtp_encrypt(&s->srtp_out, buf, size, s->encryptbuf,
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|                            sizeof(s->encryptbuf));
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|     if (size < 0)
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|         return size;
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|     return ffurl_write(s->rtp_hd, s->encryptbuf, size);
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| }
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| 
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| static int srtp_get_file_handle(URLContext *h)
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| {
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|     SRTPProtoContext *s = h->priv_data;
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|     return ffurl_get_file_handle(s->rtp_hd);
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| }
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| 
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| static int srtp_get_multi_file_handle(URLContext *h, int **handles,
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|                                       int *numhandles)
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| {
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|     SRTPProtoContext *s = h->priv_data;
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|     return ffurl_get_multi_file_handle(s->rtp_hd, handles, numhandles);
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| }
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| 
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| const URLProtocol ff_srtp_protocol = {
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|     .name                      = "srtp",
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|     .url_open                  = srtp_open,
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|     .url_read                  = srtp_read,
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|     .url_write                 = srtp_write,
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|     .url_close                 = srtp_close,
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|     .url_get_file_handle       = srtp_get_file_handle,
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|     .url_get_multi_file_handle = srtp_get_multi_file_handle,
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|     .priv_data_size            = sizeof(SRTPProtoContext),
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|     .priv_data_class           = &srtp_context_class,
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|     .flags                     = URL_PROTOCOL_FLAG_NETWORK,
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| };
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