ACM test are modified to run with both ACM1 and ACM2.
Beside the changes in test files. acm2/acm_generic_codec.cc and acm2/audio_coding_module_impl.cc are modified to fix a bug.
Also, nack{.cc, .h, _unittest.cc} are removed form main/sourc as nack files in both ACM1 and ACM2 are essentially identical.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2192005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4908 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h
index 7b146b3..9b6d5fc 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.h
@@ -8,16 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef RTPFILE_H
-#define RTPFILE_H
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
-#include "audio_coding_module.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-#include "rw_lock_wrapper.h"
#include <stdio.h>
#include <queue>
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/typedefs.h"
+
namespace webrtc {
class RTPStream {
@@ -113,4 +114,5 @@
};
} // namespace webrtc
-#endif
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_