Index: webrtc/voice_engine/channel.h |
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h |
index 0f7b5435a98f76a1e7673b2024af9a11178e7282..ba18aaa8dd7692cb5817f71bad54a1826f990da7 100644 |
--- a/webrtc/voice_engine/channel.h |
+++ b/webrtc/voice_engine/channel.h |
@@ -15,14 +15,14 @@ |
#include "webrtc/common_audio/resampler/include/push_resampler.h" |
#include "webrtc/common_types.h" |
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" |
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h" |
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" |
#include "webrtc/modules/audio_processing/rms_level.h" |
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" |
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
-#include "webrtc/modules/utility/interface/file_player.h" |
-#include "webrtc/modules/utility/interface/file_recorder.h" |
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
+#include "webrtc/modules/utility/include/file_player.h" |
+#include "webrtc/modules/utility/include/file_recorder.h" |
#include "webrtc/voice_engine/dtmf_inband.h" |
#include "webrtc/voice_engine/dtmf_inband_queue.h" |
#include "webrtc/voice_engine/include/voe_audio_processing.h" |