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Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -125,7 +125,6 @@ static int timed_read(int fd, void *data, int datalen, int timeout)
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static int mp3_exec(struct ast_channel *chan, void *data)
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{
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int res=0;
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struct ast_module_user *u;
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int fds[2];
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int ms = -1;
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int pid = -1;
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@@ -144,11 +143,8 @@ static int mp3_exec(struct ast_channel *chan, void *data)
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return -1;
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}
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u = ast_module_user_add(chan);
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if (pipe(fds)) {
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ast_log(LOG_WARNING, "Unable to create pipe\n");
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ast_module_user_remove(u);
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return -1;
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}
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@@ -158,7 +154,6 @@ static int mp3_exec(struct ast_channel *chan, void *data)
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res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
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if (res < 0) {
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ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
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ast_module_user_remove(u);
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return -1;
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}
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@@ -230,8 +225,6 @@ static int mp3_exec(struct ast_channel *chan, void *data)
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kill(pid, SIGKILL);
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if (!res && owriteformat)
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ast_set_write_format(chan, owriteformat);
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ast_module_user_remove(u);
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return res;
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}
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