Modify headers and macros, according to Russell's suggestions on the -dev list

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher
2009-04-10 03:55:27 +00:00
parent 4d74179f20
commit 1030a25ac9
10 changed files with 413 additions and 406 deletions

View File

@@ -592,7 +592,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
ast_audiohook_write_frame(audiohook, direction, middle_frame);
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END
AST_LIST_TRAVERSE_SAFE_END;
/* If this frame is being written out to the channel then we need to use whisper sources */
if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
@@ -615,7 +615,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END
AST_LIST_TRAVERSE_SAFE_END;
/* We take all of the combined whisper sources and combine them into the audio being written out */
for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++)
ast_slinear_saturated_add(data1, data2);
@@ -638,7 +638,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END
AST_LIST_TRAVERSE_SAFE_END;
end_frame = middle_frame;
}