| commit | 6ac22e6b47f9a6ed70b0a376984b39b9a745dd94 | [log] [tgz] |
|---|---|---|
| author | henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Mon Aug 11 21:06:30 2014 +0000 |
| committer | henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Mon Aug 11 21:06:30 2014 +0000 |
| tree | 4bcd4e92220d1376dd9c0db0afac1be7371a0ee3 | |
| parent | 820f8e9ca7dddeef909cadb9411264942fba9dcb [diff] [blame] |
Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc index d06cc07..3253bbd 100644 --- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc +++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -125,7 +125,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, int channels) { - struct CodecInst recvCodec; + struct CodecInst recvCodec = CodecInst(); int noOfCodecs; EXPECT_EQ(0, acm->InitializeReceiver());