| 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"
|
|
|