[asterisk-commits] dvossel: branch dvossel/hd_conferencing_ftw r307967 - /team/dvossel/hd_confer...

[SVN commits to the Asterisk project]

Author: dvossel
Date: Tue Feb 15 16:34:05 2011
New Revision: 307967

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=307967
Log:
VOLUME dialplan function is now capable of working on any samplerate

Modified:
    team/dvossel/hd_conferencing_ftw/funcs/func_volume.c

Modified: team/dvossel/hd_conferencing_ftw/funcs/func_volume.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/hd_conferencing_ftw/funcs/func_volume.c?view=diff&rev=307967&r1=307966&r2=307967
==============================================================================
--- team/dvossel/hd_conferencing_ftw/funcs/func_volume.c (original)
+++ team/dvossel/hd_conferencing_ftw/funcs/func_volume.c Tue Feb 15 16:34:05 2011
@@ -132,7 +132,7 @@
 			ast_datastore_free(datastore);
 			return 0;
 		}
-		ast_audiohook_init(&vi->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", 0);
+		ast_audiohook_init(&vi->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_NATIVE_SLIN_CAPABLE);
 		vi->audiohook.manipulate_callback = volume_callback;
 		ast_set_flag(&vi->audiohook, AST_AUDIOHOOK_WANTS_DTMF);
 		is_new = 1;