Only reinitialize AudioProcessing when needed.
This takes away the burden from the user, resulting in cleaner code.
Review URL: https://webrtc-codereview.appspot.com/941005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3010 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 75b3e20..a70dd2c 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -13,8 +13,8 @@
#include <stddef.h> // size_t
-#include "typedefs.h"
#include "module.h"
+#include "typedefs.h"
namespace webrtc {
@@ -124,6 +124,10 @@
// should be called before beginning to process a new audio stream. However,
// it is not necessary to call before processing the first stream after
// creation.
+ //
+ // set_sample_rate_hz(), set_num_channels() and set_num_reverse_channels()
+ // will trigger a full initialization if the settings are changed from their
+ // existing values. Otherwise they are no-ops.
virtual int Initialize() = 0;
// Sets the sample |rate| in Hz for both the primary and reverse audio