Add functions to interact with ASan and MSan, and some sample uses

webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc

Index: webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index ca7c45967696627c1484922acae13b986f704d93..15c60a83b5dedf5f3a44210a702580479e23a3c7 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -201,7 +201,8 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
                                                       frame_size_samples_,
                                                       &rtp_header_);
         if (!Lost(next_send_time)) {
-          InsertPacket(rtp_header_, payload_, next_send_time);
+          static const uint8_t payload[kPayloadSizeByte] = {0};
+          InsertPacket(rtp_header_, payload, next_send_time);
         }
       }
       GetOutputAudio(&output_frame_);
@@ -277,7 +278,6 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
   WebRtcRTPHeader rtp_header_;
   uint32_t last_lost_time_;
   uint32_t packet_loss_interval_;
-  uint8_t payload_[kPayloadSizeByte];
   AudioFrame output_frame_;
 };