| 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()));
|
|
|