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