Fixed the Memcheck:Uninitialized warning in WebRtcLocalAudioTrackAdapterTest_AddAndRemoveSink

content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
@@ skipping 44 matching lines @@
 // Adds and Removes a WebRtcAudioSink to a local audio track.
 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
   // Add a sink to the webrtc track.
   scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink());
   webrtc::AudioTrackInterface* webrtc_track =
       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
   webrtc_track->AddSink(sink.get());
 
   // 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()));
   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
 
   // Remove the sink from the webrtc track.
   webrtc_track->RemoveSink(sink.get());
   sink.reset();
 
   // Verify that no more callback gets into the sink.
   track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
 }
 
 }  // namespace content