Fix ast_writestream leaks
authorCorey Farrell <git@cfware.com>
Sun, 2 Nov 2014 08:13:52 +0000 (08:13 +0000)
committerCorey Farrell <git@cfware.com>
Sun, 2 Nov 2014 08:13:52 +0000 (08:13 +0000)
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/
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Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 427026 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427027 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_voicemail.c
main/app.c

index af4e7f0..d99bced 100644 (file)
@@ -6158,6 +6158,7 @@ static int msg_create_from_file(struct ast_vm_recording_data *recdata)
                                ast_log(LOG_ERROR,"Unable to determine sample rate of recording %s\n", recdata->recording_file);
                        }
                }
+               ast_closeframe(recording_fs);
        }
 
        /* If the duration was below the minimum duration for the user, let's just drop the whole thing now */
index fd19bee..dea5ee6 100644 (file)
@@ -1782,18 +1782,20 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
                        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[0]);
+                               continue;
+                       }
                        /*!\note Same logic as above. */
                        if (dspsilence) {
                                ast_stream_rewind(others[x], dspsilence - 200);
@@ -1810,7 +1812,15 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
                        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 && ast_set_read_format(chan, rfmt)) {
                ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_format_get_name(rfmt), ast_channel_name(chan));
        }