| Index: webrtc/modules/audio_coding/acm2/acm_resampler.cc
|
| diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
|
| index dfc3ef7e2746fd4d28ea62d51f058e9afb499a2f..3146f675707b286a313b4a4f66780c83f9e5751e 100644
|
| --- a/webrtc/modules/audio_coding/acm2/acm_resampler.cc
|
| +++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc
|
| @@ -14,7 +14,6 @@
|
| #include <string.h>
|
|
|
| #include "webrtc/common_audio/resampler/include/resampler.h"
|
| -#include "webrtc/system_wrappers/include/logging.h"
|
|
|
| namespace webrtc {
|
| namespace acm2 {
|
| @@ -43,16 +42,12 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
|
|
|
| if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
|
| num_audio_channels) != 0) {
|
| - LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", " << out_freq_hz
|
| - << ", " << num_audio_channels << ") failed.";
|
| return -1;
|
| }
|
|
|
| int out_length =
|
| resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
|
| if (out_length == -1) {
|
| - LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
|
| - << out_audio << ", " << out_capacity_samples << ") failed.";
|
| return -1;
|
| }
|
|
|
|
|