Use webrtc/base/logging.h for voice_engine.

webrtc/voice_engine/transmit_mixer.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/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"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/utility.h"
 #include "webrtc/voice_engine/voe_base_impl.h"
 
 namespace webrtc {
 namespace voe {
@@ skipping 1206 matching lines @@
                                 AudioFrame::kNormalSpeech,
                                 AudioFrame::kVadUnknown,
                                 1);
     }
     return 0;
 }
 
 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);
   }
 
   EchoCancellation* aec = audioproc_->echo_cancellation();
   if (aec->is_drift_compensation_enabled()) {
     aec->set_stream_drift_samples(clock_drift);
   }
 
   audioproc_->set_stream_key_pressed(key_pressed);
 
@@ skipping 72 matching lines @@
 void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
   swap_stereo_channels_ = enable;
 }
 
 bool TransmitMixer::IsStereoChannelSwappingEnabled() {
   return swap_stereo_channels_;
 }
 
 }  // namespace voe
 }  // namespace webrtc