| Index: webrtc/modules/audio_processing/test/unpack.cc
|
| diff --git a/webrtc/modules/audio_processing/test/unpack.cc b/webrtc/modules/audio_processing/test/unpack.cc
|
| index f5c0700b3f8bb340836070daee1320c8f1311155..d3f163383246fb3b7dad913f7e9c780960496309 100644
|
| --- a/webrtc/modules/audio_processing/test/unpack.cc
|
| +++ b/webrtc/modules/audio_processing/test/unpack.cc
|
| @@ -303,13 +303,19 @@ int do_main(int argc, char* argv[]) {
|
| if (!FLAGS_raw) {
|
| // The WAV files need to be reset every time, because they cant change
|
| // their sample rate or number of channels.
|
| - reverse_wav_file.reset(new WavWriter(FLAGS_reverse_file + ".wav",
|
| + std::stringstream reverse_name;
|
| + reverse_name << FLAGS_reverse_file << frame_count << ".wav";
|
| + reverse_wav_file.reset(new WavWriter(reverse_name.str(),
|
| reverse_sample_rate,
|
| num_reverse_channels));
|
| - input_wav_file.reset(new WavWriter(FLAGS_input_file + ".wav",
|
| + std::stringstream input_name;
|
| + input_name << FLAGS_input_file << frame_count << ".wav";
|
| + input_wav_file.reset(new WavWriter(input_name.str(),
|
| input_sample_rate,
|
| num_input_channels));
|
| - output_wav_file.reset(new WavWriter(FLAGS_output_file + ".wav",
|
| + std::stringstream output_name;
|
| + output_name << FLAGS_output_file << frame_count << ".wav";
|
| + output_wav_file.reset(new WavWriter(output_name.str(),
|
| output_sample_rate,
|
| num_output_channels));
|
| }
|
|
|