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)); |
} |