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; |