| Index: webrtc/modules/audio_processing/test/conversational_speech/timing.cc
|
| diff --git a/webrtc/modules/audio_processing/test/conversational_speech/timing.cc b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
|
| index 0aa44fa42c864aad43eaa5927297778756b1fe3c..53076f1cacdaf50780b137fb733fb47cc0724625 100644
|
| --- a/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
|
| +++ b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
|
| @@ -53,7 +53,7 @@ std::vector<Turn> LoadTiming(const std::string& timing_filepath) {
|
| void SaveTiming(const std::string& timing_filepath,
|
| rtc::ArrayView<const Turn> timing) {
|
| std::ofstream outfile(timing_filepath);
|
| - // TODO(alessio): check if file open for writing.
|
| + RTC_CHECK(outfile.is_open());
|
| for (const Turn& turn : timing) {
|
| outfile << turn.speaker_name << " " << turn.audiotrack_file_name
|
| << " " << turn.offset << std::endl;
|
|
|