Using ring buffer for AudioVector in NetEq.

webrtc/modules/audio_coding/neteq/expand_unittest.cc

 /*
  *  Copyright (c) 2012 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 75 matching lines @@
   }
 
   void SetUp() override {
     // Fast-forward the input file until there is speech (about 1.1 second into
     // the file).
     const size_t speech_start_samples =
         static_cast<size_t>(test_sample_rate_hz_ * 1.1f);
     ASSERT_TRUE(input_file_.Seek(speech_start_samples));
 
     // Pre-load the sync buffer with speech data.
-    ASSERT_TRUE(
-        input_file_.Read(sync_buffer_.Size(), &sync_buffer_.Channel(0)[0]));
+    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
+    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
+    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
     ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
   }
 
   test::ResampleInputAudioFile input_file_;
   int test_sample_rate_hz_;
   size_t num_channels_;
   BackgroundNoise background_noise_;
   SyncBuffer sync_buffer_;
   RandomVector random_vector_;
   FakeStatisticsCalculator statistics_;
@@ skipping 57 matching lines @@
   expand_.SetParametersForNormalAfterExpand();
   // Convert |sum_output_len_samples| to milliseconds.
   EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
                                    (test_sample_rate_hz_ / 1000)),
             statistics_.last_outage_duration_ms());
 }
 
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc