Replace use of system_wrappers/include/logging.h by base/logging.h.

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/modules/audio_processing/aec3/echo_canceller3.h"
 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
 #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
 #include "webrtc/modules/audio_processing/gain_control_for_experimental_agc.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #endif
 #include "webrtc/modules/audio_processing/level_controller/level_controller.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 #include "webrtc/modules/audio_processing/low_cut_filter.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 #include "webrtc/modules/audio_processing/residual_echo_detector.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
 #else
 #include "webrtc/modules/audio_processing/debug.pb.h"
 #endif
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
@@ skipping 1934 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc