| Index: webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| index cd4b31c2c8215168a4de7baf05bc5597a7bb5bb1..7e9677446626a1c76160760c2f875cb35b7479c3 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
|
| @@ -478,7 +478,7 @@ void PopulateAudioFrame(AudioFrame* frame,
|
| ASSERT_GT(amplitude, 0);
|
| ASSERT_LE(amplitude, 32767);
|
| for (int ch = 0; ch < frame->num_channels_; ch++) {
|
| - for (int k = 0; k < static_cast<int>(frame->samples_per_channel_); k++) {
|
| + for (size_t k = 0; k < frame->samples_per_channel_; k++) {
|
| // Store random 16 bit number between -(amplitude+1) and
|
| // amplitude.
|
| frame->data_[k * ch] =
|
|
|