[asterisk-dev] [asterisk-commits] coreyfarrell: branch 11 r427024 - in /branches/11: ./ apps/ main/

[Matthew Jordan]

Whoops - looks like this broke the build:

https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD3-460

On Sun, Nov 2, 2014 at 2:03 AM, SVN commits to the Asterisk project
<asterisk-commits at lists.digium.com> wrote:
> Author: coreyfarrell
> Date: Sun Nov  2 02:03:18 2014
> New Revision: 427024
>
> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=427024
> Log:
> Fix ast_writestream leaks
>
> Fix cleanup in __ast_play_and_record where others[x] may be leaked.
> This was caught where prepend != NULL && outmsg != NULL, once
> realfile[x] == NULL any further others[x] would be leaked. A cleanup
> block was also added for prepend != NULL && outmsg == NULL.
>
> 11+: Fix leak of ast_writestream recording_fs in
> app_voicemail:leave_voicemail.
>
> ASTERISK-24476 #close
> Reported by: Corey Farrell
> Review: https://reviewboard.asterisk.org/r/4138/
> ........
>
> Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>
> Modified:
>     branches/11/   (props changed)
>     branches/11/apps/app_voicemail.c
>     branches/11/main/app.c
>
> Propchange: branches/11/
> ------------------------------------------------------------------------------
> Binary property 'branch-1.8-merged' - no diff available.
>
> Modified: branches/11/apps/app_voicemail.c
> URL: http://svnview.digium.com/svn/asterisk/branches/11/apps/app_voicemail.c?view=diff&rev=427024&r1=427023&r2=427024
> ==============================================================================
> --- branches/11/apps/app_voicemail.c (original)
> +++ branches/11/apps/app_voicemail.c Sun Nov  2 02:03:18 2014
> @@ -6041,6 +6041,7 @@
>                         ast_getformatbyname(recdata->recording_ext, &result);
>                         duration = (int) (framelength / ast_format_rate(&result));
>                 }
> +               ast_closeframe(recording_fs);
>         }
>
>         /* If the duration was below the minimum duration for the user, let's just drop the whole thing now */
>
> Modified: branches/11/main/app.c
> URL: http://svnview.digium.com/svn/asterisk/branches/11/main/app.c?view=diff&rev=427024&r1=427023&r2=427024
> ==============================================================================
> --- branches/11/main/app.c (original)
> +++ branches/11/main/app.c Sun Nov  2 02:03:18 2014
> @@ -1385,17 +1385,19 @@
>                         ast_truncstream(others[x]);
>                         ast_closestream(others[x]);
>                 }
> -       }
> -
> -       if (prepend && outmsg) {
> +       } else if (prepend && outmsg) {
>                 struct ast_filestream *realfiles[AST_MAX_FORMATS];
>                 struct ast_frame *fr;
>
>                 for (x = 0; x < fmtcnt; x++) {
>                         snprintf(comment, sizeof(comment), "Opening the real file %s.%s\n", recordfile, sfmt[x]);
>                         realfiles[x] = ast_readfile(recordfile, sfmt[x], comment, O_RDONLY, 0, 0);
> -                       if (!others[x] || !realfiles[x]) {
> +                       if (!others[x]) {
>                                 break;
> +                       }
> +                       if (!realfiles[x]) {
> +                               ast_closestream(others[x]);
> +                               continue;
>                         }
>                         /*!\note Same logic as above. */
>                         if (dspsilence) {
> @@ -1413,7 +1415,15 @@
>                         ast_verb(4, "Recording Format: sfmts=%s, prependfile %s, recordfile %s\n", sfmt[x], prependfile, recordfile);
>                         ast_filedelete(prependfile, sfmt[x]);
>                 }
> -       }
> +       } else {
> +               for (x = 0; x < fmtcnt; x++) {
> +                       if (!others[x]) {
> +                               break;
> +                       }
> +                       ast_closestream(others[x]);
> +               }
> +       }
> +
>         if (rfmt.id && ast_set_read_format(chan, &rfmt)) {
>                 ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(&rfmt), ast_channel_name(chan));
>         }
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-commits mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-commits



-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org