mirror of
https://github.com/FFmpeg/FFmpeg.git
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e85d38c20a
The exit condition was missing. Signed-off-by: Timothy Gu <timothygu99@gmail.com>
415 lines
13 KiB
C
415 lines
13 KiB
C
/*
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* RTMP network protocol
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* Copyright (c) 2010 Howard Chu
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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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* @file
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* RTMP protocol based on http://rtmpdump.mplayerhq.hu/ librtmp
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*/
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#include "libavutil/avstring.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "avformat.h"
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#if CONFIG_NETWORK
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#include "network.h"
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#endif
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#include "url.h"
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#include <librtmp/rtmp.h>
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#include <librtmp/log.h>
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typedef struct LibRTMPContext {
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const AVClass *class;
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RTMP rtmp;
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char *app;
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char *conn;
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char *subscribe;
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char *playpath;
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char *tcurl;
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char *flashver;
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char *swfurl;
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char *swfverify;
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char *pageurl;
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char *client_buffer_time;
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int live;
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char *temp_filename;
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int buffer_size;
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} LibRTMPContext;
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static void rtmp_log(int level, const char *fmt, va_list args)
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{
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switch (level) {
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default:
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case RTMP_LOGCRIT: level = AV_LOG_FATAL; break;
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case RTMP_LOGERROR: level = AV_LOG_ERROR; break;
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case RTMP_LOGWARNING: level = AV_LOG_WARNING; break;
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case RTMP_LOGINFO: level = AV_LOG_INFO; break;
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case RTMP_LOGDEBUG: level = AV_LOG_VERBOSE; break;
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case RTMP_LOGDEBUG2: level = AV_LOG_DEBUG; break;
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}
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av_vlog(NULL, level, fmt, args);
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av_log(NULL, level, "\n");
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}
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static int rtmp_close(URLContext *s)
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{
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LibRTMPContext *ctx = s->priv_data;
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RTMP *r = &ctx->rtmp;
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RTMP_Close(r);
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av_freep(&ctx->temp_filename);
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return 0;
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}
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/**
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* Open RTMP connection and verify that the stream can be played.
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*
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* URL syntax: rtmp://server[:port][/app][/playpath][ keyword=value]...
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* where 'app' is first one or two directories in the path
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* (e.g. /ondemand/, /flash/live/, etc.)
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* and 'playpath' is a file name (the rest of the path,
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* may be prefixed with "mp4:")
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*
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* Additional RTMP library options may be appended as
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* space-separated key-value pairs.
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*/
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static int rtmp_open(URLContext *s, const char *uri, int flags)
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{
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LibRTMPContext *ctx = s->priv_data;
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RTMP *r = &ctx->rtmp;
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int rc = 0, level;
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char *filename = s->filename;
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int len = strlen(s->filename) + 1;
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switch (av_log_get_level()) {
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default:
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case AV_LOG_FATAL: level = RTMP_LOGCRIT; break;
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case AV_LOG_ERROR: level = RTMP_LOGERROR; break;
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case AV_LOG_WARNING: level = RTMP_LOGWARNING; break;
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case AV_LOG_INFO: level = RTMP_LOGINFO; break;
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case AV_LOG_VERBOSE: level = RTMP_LOGDEBUG; break;
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case AV_LOG_DEBUG: level = RTMP_LOGDEBUG2; break;
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}
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RTMP_LogSetLevel(level);
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RTMP_LogSetCallback(rtmp_log);
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if (ctx->app) len += strlen(ctx->app) + sizeof(" app=");
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if (ctx->tcurl) len += strlen(ctx->tcurl) + sizeof(" tcUrl=");
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if (ctx->pageurl) len += strlen(ctx->pageurl) + sizeof(" pageUrl=");
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if (ctx->flashver) len += strlen(ctx->flashver) + sizeof(" flashver=");
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if (ctx->conn) {
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char *sep, *p = ctx->conn;
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int options = 0;
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while (p) {
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options++;
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p += strspn(p, " ");
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if (!*p)
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break;
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sep = strchr(p, ' ');
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if (sep)
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p = sep + 1;
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else
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break;
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}
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len += options * sizeof(" conn=");
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len += strlen(ctx->conn);
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}
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if (ctx->playpath)
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len += strlen(ctx->playpath) + sizeof(" playpath=");
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if (ctx->live)
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len += sizeof(" live=1");
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if (ctx->subscribe)
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len += strlen(ctx->subscribe) + sizeof(" subscribe=");
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if (ctx->client_buffer_time)
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len += strlen(ctx->client_buffer_time) + sizeof(" buffer=");
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if (ctx->swfurl || ctx->swfverify) {
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len += sizeof(" swfUrl=");
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if (ctx->swfverify)
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len += strlen(ctx->swfverify) + sizeof(" swfVfy=1");
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else
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len += strlen(ctx->swfurl);
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}
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if (!(ctx->temp_filename = filename = av_malloc(len)))
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return AVERROR(ENOMEM);
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av_strlcpy(filename, s->filename, len);
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if (ctx->app) {
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av_strlcat(filename, " app=", len);
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av_strlcat(filename, ctx->app, len);
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}
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if (ctx->tcurl) {
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av_strlcat(filename, " tcUrl=", len);
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av_strlcat(filename, ctx->tcurl, len);
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}
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if (ctx->pageurl) {
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av_strlcat(filename, " pageUrl=", len);
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av_strlcat(filename, ctx->pageurl, len);
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}
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if (ctx->swfurl) {
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av_strlcat(filename, " swfUrl=", len);
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av_strlcat(filename, ctx->swfurl, len);
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}
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if (ctx->flashver) {
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av_strlcat(filename, " flashVer=", len);
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av_strlcat(filename, ctx->flashver, len);
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}
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if (ctx->conn) {
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char *sep, *p = ctx->conn;
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while (p) {
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av_strlcat(filename, " conn=", len);
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p += strspn(p, " ");
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if (!*p)
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break;
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sep = strchr(p, ' ');
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if (sep)
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*sep = '\0';
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av_strlcat(filename, p, len);
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if (sep)
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p = sep + 1;
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else
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break;
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}
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}
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if (ctx->playpath) {
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av_strlcat(filename, " playpath=", len);
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av_strlcat(filename, ctx->playpath, len);
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}
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if (ctx->live)
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av_strlcat(filename, " live=1", len);
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if (ctx->subscribe) {
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av_strlcat(filename, " subscribe=", len);
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av_strlcat(filename, ctx->subscribe, len);
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}
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if (ctx->client_buffer_time) {
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av_strlcat(filename, " buffer=", len);
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av_strlcat(filename, ctx->client_buffer_time, len);
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}
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if (ctx->swfurl || ctx->swfverify) {
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av_strlcat(filename, " swfUrl=", len);
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if (ctx->swfverify) {
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av_strlcat(filename, ctx->swfverify, len);
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av_strlcat(filename, " swfVfy=1", len);
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} else {
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av_strlcat(filename, ctx->swfurl, len);
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}
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}
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RTMP_Init(r);
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if (!RTMP_SetupURL(r, filename)) {
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rc = AVERROR_UNKNOWN;
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goto fail;
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}
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if (flags & AVIO_FLAG_WRITE)
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RTMP_EnableWrite(r);
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if (!RTMP_Connect(r, NULL) || !RTMP_ConnectStream(r, 0)) {
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rc = AVERROR_UNKNOWN;
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goto fail;
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}
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#if CONFIG_NETWORK
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if (ctx->buffer_size >= 0 && (flags & AVIO_FLAG_WRITE)) {
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int tmp = ctx->buffer_size;
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setsockopt(r->m_sb.sb_socket, SOL_SOCKET, SO_SNDBUF, &tmp, sizeof(tmp));
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}
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#endif
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s->is_streamed = 1;
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return 0;
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fail:
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av_freep(&ctx->temp_filename);
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if (rc)
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RTMP_Close(r);
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return rc;
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}
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static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
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{
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LibRTMPContext *ctx = s->priv_data;
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RTMP *r = &ctx->rtmp;
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return RTMP_Write(r, buf, size);
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}
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static int rtmp_read(URLContext *s, uint8_t *buf, int size)
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{
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LibRTMPContext *ctx = s->priv_data;
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RTMP *r = &ctx->rtmp;
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return RTMP_Read(r, buf, size);
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}
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static int rtmp_read_pause(URLContext *s, int pause)
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{
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LibRTMPContext *ctx = s->priv_data;
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RTMP *r = &ctx->rtmp;
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if (!RTMP_Pause(r, pause))
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return AVERROR_UNKNOWN;
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return 0;
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}
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static int64_t rtmp_read_seek(URLContext *s, int stream_index,
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int64_t timestamp, int flags)
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{
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LibRTMPContext *ctx = s->priv_data;
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RTMP *r = &ctx->rtmp;
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if (flags & AVSEEK_FLAG_BYTE)
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return AVERROR(ENOSYS);
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/* seeks are in milliseconds */
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if (stream_index < 0)
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timestamp = av_rescale_rnd(timestamp, 1000, AV_TIME_BASE,
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flags & AVSEEK_FLAG_BACKWARD ? AV_ROUND_DOWN : AV_ROUND_UP);
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if (!RTMP_SendSeek(r, timestamp))
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return AVERROR_UNKNOWN;
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return timestamp;
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}
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static int rtmp_get_file_handle(URLContext *s)
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{
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LibRTMPContext *ctx = s->priv_data;
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RTMP *r = &ctx->rtmp;
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return RTMP_Socket(r);
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}
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#define OFFSET(x) offsetof(LibRTMPContext, x)
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#define DEC AV_OPT_FLAG_DECODING_PARAM
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#define ENC AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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{"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
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{"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_STRING, {.str = "3000"}, 0, 0, DEC|ENC},
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{"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
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{"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
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{"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
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{"any", "both", 0, AV_OPT_TYPE_CONST, {.i64 = -2}, 0, 0, DEC, "rtmp_live"},
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{"live", "live stream", 0, AV_OPT_TYPE_CONST, {.i64 = -1}, 0, 0, DEC, "rtmp_live"},
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{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, 0, 0, DEC, "rtmp_live"},
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{"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
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{"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
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{"rtmp_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
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{"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
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{"rtmp_swfverify", "URL to player swf file, compute hash/size automatically. (unimplemented)", OFFSET(swfverify), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
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{"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
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#if CONFIG_NETWORK
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{"rtmp_buffer_size", "set buffer size in bytes", OFFSET(buffer_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, DEC|ENC },
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#endif
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{ NULL },
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};
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#define RTMP_CLASS(flavor)\
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static const AVClass lib ## flavor ## _class = {\
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.class_name = "lib" #flavor " protocol",\
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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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RTMP_CLASS(rtmp)
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const URLProtocol ff_librtmp_protocol = {
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.name = "rtmp",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(LibRTMPContext),
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.priv_data_class = &librtmp_class,
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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RTMP_CLASS(rtmpt)
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const URLProtocol ff_librtmpt_protocol = {
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.name = "rtmpt",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(LibRTMPContext),
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.priv_data_class = &librtmpt_class,
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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RTMP_CLASS(rtmpe)
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const URLProtocol ff_librtmpe_protocol = {
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.name = "rtmpe",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(LibRTMPContext),
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.priv_data_class = &librtmpe_class,
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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RTMP_CLASS(rtmpte)
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const URLProtocol ff_librtmpte_protocol = {
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.name = "rtmpte",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(LibRTMPContext),
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.priv_data_class = &librtmpte_class,
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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RTMP_CLASS(rtmps)
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const URLProtocol ff_librtmps_protocol = {
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.name = "rtmps",
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.url_open = rtmp_open,
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.url_read = rtmp_read,
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.url_write = rtmp_write,
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.url_close = rtmp_close,
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.url_read_pause = rtmp_read_pause,
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.url_read_seek = rtmp_read_seek,
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.url_get_file_handle = rtmp_get_file_handle,
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.priv_data_size = sizeof(LibRTMPContext),
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.priv_data_class = &librtmps_class,
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.flags = URL_PROTOCOL_FLAG_NETWORK,
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};
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