diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 0a3e69654e..90ac9d02ea 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -14,6 +14,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
@@ -27,7 +28,6 @@
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
@@ -1020,11 +1020,11 @@ Channel::Init()
     }
 
     if (rx_audioproc_->noise_suppression()->set_level(kDefaultNsMode) != 0) {
-      LOG_FERR1(LS_ERROR, noise_suppression()->set_level, kDefaultNsMode);
+      LOG(LS_ERROR) << "noise_suppression()->set_level(kDefaultNsMode) failed.";
       return -1;
     }
     if (rx_audioproc_->gain_control()->set_mode(kDefaultRxAgcMode) != 0) {
-      LOG_FERR1(LS_ERROR, gain_control()->set_mode, kDefaultRxAgcMode);
+      LOG(LS_ERROR) << "gain_control()->set_mode(kDefaultRxAgcMode) failed.";
       return -1;
     }
 
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 3af4c07ed3..2241f46bba 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -11,10 +11,10 @@
 #include "webrtc/voice_engine/transmit_mixer.h"
 
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/channel_manager.h"
@@ -1241,15 +1241,13 @@ int32_t TransmitMixer::MixOrReplaceAudioWithFile(
 void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
                                  int current_mic_level, bool key_pressed) {
   if (audioproc_->set_stream_delay_ms(delay_ms) != 0) {
-    // A redundant warning is reported in AudioDevice, which we've throttled
-    // to avoid flooding the logs. Relegate this one to LS_VERBOSE to avoid
-    // repeating the problem here.
-    LOG_FERR1(LS_VERBOSE, set_stream_delay_ms, delay_ms);
+    // Silently ignore this failure to avoid flooding the logs.
   }
 
   GainControl* agc = audioproc_->gain_control();
   if (agc->set_stream_analog_level(current_mic_level) != 0) {
-    LOG_FERR1(LS_ERROR, set_stream_analog_level, current_mic_level);
+    LOG(LS_ERROR) << "set_stream_analog_level failed: current_mic_level = "
+                  << current_mic_level;
     assert(false);
   }
 
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index b04163834c..eb442ecb8f 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -10,12 +10,12 @@
 
 #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 {
@@ -52,8 +52,10 @@ void RemixAndResample(const int16_t* src_data,
 
   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);
   }
 
@@ -61,7 +63,9 @@ void RemixAndResample(const int16_t* src_data,
   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_ =
diff --git a/webrtc/voice_engine/voe_audio_processing_impl.cc b/webrtc/voice_engine/voe_audio_processing_impl.cc
index 83f70fe68e..c95726339c 100644
--- a/webrtc/voice_engine/voe_audio_processing_impl.cc
+++ b/webrtc/voice_engine/voe_audio_processing_impl.cc
@@ -10,9 +10,9 @@
 
 #include "webrtc/voice_engine/voe_audio_processing_impl.h"
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
@@ -317,7 +317,6 @@ int VoEAudioProcessingImpl::GetAgcConfig(AgcConfig& config) {
 int VoEAudioProcessingImpl::SetRxNsStatus(int channel,
                                           bool enable,
                                           NsModes mode) {
-  LOG_API3(channel, enable, mode);
 #ifdef WEBRTC_VOICE_ENGINE_NR
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -469,7 +468,6 @@ bool VoEAudioProcessing::DriftCompensationSupported() {
 }
 
 int VoEAudioProcessingImpl::EnableDriftCompensation(bool enable) {
-  LOG_API1(enable);
   WEBRTC_VOICE_INIT_CHECK();
 
   if (!DriftCompensationSupported()) {
@@ -489,7 +487,6 @@ int VoEAudioProcessingImpl::EnableDriftCompensation(bool enable) {
 }
 
 bool VoEAudioProcessingImpl::DriftCompensationEnabled() {
-  LOG_API0();
   WEBRTC_VOICE_INIT_CHECK_BOOL();
 
   EchoCancellation* aec = _shared->audio_processing()->echo_cancellation();
@@ -1038,12 +1035,10 @@ int VoEAudioProcessingImpl::SetTypingDetectionParameters(int timeWindow,
 }
 
 void VoEAudioProcessingImpl::EnableStereoChannelSwapping(bool enable) {
-  LOG_API1(enable);
   _shared->transmit_mixer()->EnableStereoChannelSwapping(enable);
 }
 
 bool VoEAudioProcessingImpl::IsStereoChannelSwappingEnabled() {
-  LOG_API0();
   return _shared->transmit_mixer()->IsStereoChannelSwappingEnabled();
 }
 
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 2b5587ddef..60a649eefd 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -11,6 +11,7 @@
 #include "webrtc/voice_engine/voe_base_impl.h"
 
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/common.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
@@ -18,7 +19,6 @@
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/output_mixer.h"
diff --git a/webrtc/voice_engine/voe_network_impl.cc b/webrtc/voice_engine/voe_network_impl.cc
index 4a0c3f7861..0574aa9f05 100644
--- a/webrtc/voice_engine/voe_network_impl.cc
+++ b/webrtc/voice_engine/voe_network_impl.cc
@@ -12,8 +12,8 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"