diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -12,24 +12,16 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/linux/audio_device_alsa_linux.h"
 
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/trace.h"
- 
-#include "Latency.h"
-
-#define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
-                                     reinterpret_cast<uint64_t>(x), 0)
-#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
-                                                 reinterpret_cast<uint64_t>(x), frames)
-
 
 webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
 
 // Accesses ALSA functions through our late-binding symbol table instead of
 // directly. This way we don't have to link to libasound, which means our binary
 // will work on systems that don't have it.
 #define LATE(sym) \
   LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
@@ -2138,20 +2130,18 @@ bool AudioDeviceLinuxALSA::RecThreadProc
                buffer, size);
         _recordingFramesLeft -= frames;
 
         if (!_recordingFramesLeft)
         { // buf is full
             _recordingFramesLeft = _recordingFramesIn10MS;
 
             if (_firstRecord) {
-              LOG_FIRST_CAPTURE(this);
               _firstRecord = false;
             }
-            LOG_CAPTURE_FRAMES(this, _recordingFramesIn10MS);
             // store the recorded buffer (no action will be taken if the
             // #recorded samples is not a full buffer)
             _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
                                                _recordingFramesIn10MS);
 
             uint32_t currentMicLevel = 0;
             uint32_t newMicLevel = 0;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
--- a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
@@ -13,22 +13,16 @@
 
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/sndio/audio_device_sndio.h"
 
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
-#include "Latency.h"
-
-#define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
-                                     reinterpret_cast<uint64_t>(x), 0)
-#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
-                                                 reinterpret_cast<uint64_t>(x), frames)
 extern "C"
 {
     static void playOnmove(void *arg, int delta)
     {
         static_cast<webrtc::AudioDeviceSndio *>(arg)->_playDelay -= delta;
     }
 
     static void recOnmove(void *arg, int delta)