Use webrtc/base/logging.h for voice_engine.

webrtc/voice_engine/utility.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/voice_engine/utility.h"
 
+#include "webrtc/base/logging.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 namespace voe {
 
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
   RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
                    src_frame.num_channels_, src_frame.sample_rate_hz_,
@@ skipping 16 matching lines @@
   // Downmix before resampling.
   if (num_channels == 2 && dst_frame->num_channels_ == 1) {
     AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
                                        mono_audio);
     audio_ptr = mono_audio;
     audio_ptr_num_channels = 1;
   }
 
   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
-    LOG_FERR3(LS_ERROR, InitializeIfNeeded, sample_rate_hz,
-              dst_frame->sample_rate_hz_, audio_ptr_num_channels);
+    LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
+                  << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
+                  << dst_frame->sample_rate_hz_
+                  << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
     assert(false);
   }
 
   const size_t src_length = samples_per_channel * audio_ptr_num_channels;
   int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
                                        AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
-    LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_);
+    LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
+                  << ", src_length = " << src_length
+                  << ", dst_frame->data_ = " << dst_frame->data_;
     assert(false);
   }
   dst_frame->samples_per_channel_ =
       static_cast<size_t>(out_length / audio_ptr_num_channels);
 
   // Upmix after resampling.
   if (num_channels == 1 && dst_frame->num_channels_ == 2) {
     // The audio in dst_frame really is mono at this point; MonoToStereo will
     // set this back to stereo.
     dst_frame->num_channels_ = 1;
@@ skipping 30 matching lines @@
     int32_t temp = 0;
     for (size_t i = 0; i < source_len; ++i) {
       temp = source[i] + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   }
 }
 
 }  // namespace voe
 }  // namespace webrtc