Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc |
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc |
index f32c7fb9563108030ca6169430164090530a27fd..6375aed99f4945983644bf8a078d53fcb7f062bc 100644 |
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc |
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc |
@@ -62,8 +62,11 @@ TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) { |
// Send a packet via |track_| and it data should reach the sink of the |
// |adapter_|. |
- scoped_ptr<int16[]> data( |
- new int16[params_.frames_per_buffer() * params_.channels()]); |
+ const int length = params_.frames_per_buffer() * params_.channels(); |
+ scoped_ptr<int16[]> data(new int16[length]); |
+ // Initialize the data to 0 to avoid Memcheck:Uninitialized warning. |
+ memset(data.get(), 0, length * sizeof(data[0])); |
+ |
EXPECT_CALL(*sink, |
OnData(_, 16, params_.sample_rate(), params_.channels(), |
params_.frames_per_buffer())); |