Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
index 8529e0d48c4f41fde20c508952c9a56a94a19a85..cf79120bb9ae932fa7627a6fab4ee9305b916111 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
@@ -12,13 +12,13 @@ |
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ |
#include "webrtc/common_types.h" |
+#include "webrtc/base/constructormagic.h" |
+#include "webrtc/base/criticalsection.h" |
+#include "webrtc/base/onetimeevent.h" |
#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
-#include "webrtc/rtc_base/constructormagic.h" |
-#include "webrtc/rtc_base/criticalsection.h" |
-#include "webrtc/rtc_base/onetimeevent.h" |
#include "webrtc/typedefs.h" |
namespace webrtc { |