Index: webrtc/voice_engine/utility.cc |
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc |
index b04163834cde1e4affecad3675622dc11ec2eb23..eb442ecb8f74c8ac607b1f74c85274a0566520b5 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_ = |