commit | a1ef7bfa156c513d215cc22ab721328e387af547 | [log] [tgz] |
---|---|---|
author | kwiberg@webrtc.org <kwiberg@webrtc.org> | Mon Dec 08 17:53:10 2014 +0000 |
committer | kwiberg@webrtc.org <kwiberg@webrtc.org> | Mon Dec 08 17:53:10 2014 +0000 |
tree | f828ed09d00eddfbb93e64d1f628cdde84cddee6 | |
parent | 3b3c4069082e00d0430e75ac242c6f0578e7a528 [diff] |
ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable. Ideally, this is a stopgap fix until ATTRIBUTE_UNUSED can be given a proper definition. TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7830 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc index cf1d632..d6bc4b3 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -55,9 +55,9 @@ uint8_t* encoded, size_t* encoded_bytes, EncodedInfo* info) { - const size_t expected_output_len ATTRIBUTE_UNUSED = + const size_t expected_output_len = num_10ms_frames_per_packet_ == 2 ? 38 : 50; - DCHECK_GE(max_encoded_bytes, expected_output_len); + CHECK_GE(max_encoded_bytes, expected_output_len); // Save timestamp if starting a new packet. if (num_10ms_frames_buffered_ == 0)