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. |