| Index: webrtc/modules/audio_processing/test/audioproc_float.cc
|
| diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| index 268702e5dec6ac7ebfff26d9d9f466bff876b04a..3cea5b933e2d0c1c084913af5026d6693a37ad5f 100644
|
| --- a/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| +++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| @@ -172,10 +172,10 @@ int main(int argc, char* argv[]) {
|
| StreamConfig reverse_input_config = {};
|
| StreamConfig reverse_output_config = {};
|
| if (process_reverse) {
|
| - StreamConfig reverse_input_config = {in_rev_file->sample_rate(),
|
| - in_rev_file->num_channels()};
|
| - StreamConfig reverse_output_config = {out_rev_file->sample_rate(),
|
| - out_rev_file->num_channels()};
|
| + reverse_input_config = {in_rev_file->sample_rate(),
|
| + in_rev_file->num_channels()};
|
| + reverse_output_config = {out_rev_file->sample_rate(),
|
| + out_rev_file->num_channels()};
|
| }
|
| while (in_file.ReadSamples(in_interleaved.size(),
|
| &in_interleaved[0]) == in_interleaved.size()) {
|
|
|