| Index: webrtc/modules/audio_device/fine_audio_buffer.cc | 
| diff --git a/webrtc/modules/audio_device/fine_audio_buffer.cc b/webrtc/modules/audio_device/fine_audio_buffer.cc | 
| index c3b07eeb404ac29fd4ec8fbe3f52fcf7463f6193..7fffdd14fbcf2014b96d45869d1131a46f22c06c 100644 | 
| --- a/webrtc/modules/audio_device/fine_audio_buffer.cc | 
| +++ b/webrtc/modules/audio_device/fine_audio_buffer.cc | 
| @@ -115,7 +115,6 @@ void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer, | 
| size_t size_in_bytes, | 
| int playout_delay_ms, | 
| int record_delay_ms) { | 
| -  RTC_CHECK_EQ(size_in_bytes, desired_frame_size_bytes_); | 
| // Check if the temporary buffer can store the incoming buffer. If not, | 
| // move the remaining (old) bytes to the beginning of the temporary buffer | 
| // and start adding new samples after the old samples. | 
|  |