| 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_ =
|
|
|