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

webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 183 matching lines @@
     time_now = next_send_time = last_lost_time_ =
         rtp_generator_->GetRtpHeader(kPayloadType, frame_size_samples_,
                                      &rtp_header_);
     for (int k = 0; k < num_loops; ++k) {
       // Delay by one frame such that the FEC can come in.
       while (time_now + kFrameSizeMs >= next_send_time) {
         next_send_time = rtp_generator_->GetRtpHeader(kPayloadType,
                                                       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_);
       time_now += kOutputLengthMs;
     }
     CheckNetworkStatistics(expects);
     neteq()->FlushBuffers();
   }
 
   void DecodeFecTest() {
@@ skipping 55 matching lines @@
   }
 
  private:
   MockAudioDecoder* external_decoder_;
   const int samples_per_ms_;
   const size_t frame_size_samples_;
   std::unique_ptr<test::RtpGenerator> rtp_generator_;
   WebRtcRTPHeader rtp_header_;
   uint32_t last_lost_time_;
   uint32_t packet_loss_interval_;
-  uint8_t payload_[kPayloadSizeByte];
   AudioFrame output_frame_;
 };
 
 TEST(NetEqNetworkStatsTest, DecodeFec) {
   MockAudioDecoder decoder(48000, 1);
   NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, 48000, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, StereoDecodeFec) {
   MockAudioDecoder decoder(48000, 2);
   NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, 48000, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
   MockAudioDecoder decoder(48000, 1);
   NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, 48000, &decoder);
   test.NoiseExpansionTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 }  // namespace test
 }  // namespace webrtc