| Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| index be8b1d91d861b73f747aa433f8f84fdeb6ede4b6..14958181859501459181f56380e79bf885e40376 100644
|
| --- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| @@ -10,6 +10,8 @@
|
|
|
| #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
|
|
|
| +#include <algorithm>
|
| +
|
| #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
|
| #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
|
| #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
|
| @@ -335,7 +337,7 @@ int DownMix(const AudioFrame& frame,
|
| static_cast<int32_t>(frame_data[2 * n + 1])) >> 1);
|
| }
|
| } else {
|
| - memset(out_buff, 0, frame.samples_per_channel_);
|
| + std::fill(out_buff, out_buff + frame.samples_per_channel_, 0);
|
| }
|
| return 0;
|
| }
|
| @@ -354,7 +356,7 @@ int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {
|
| out_buff[2 * i] = sample;
|
| }
|
| } else {
|
| - memset(out_buff, 0, 2 * frame.samples_per_channel_);
|
| + std::fill(out_buff, out_buff + frame.samples_per_channel_ * 2, 0);
|
| }
|
| return 0;
|
| }
|
|
|