Changed the names of some of the bitexactness unittests to be similar to the other test file names

webrtc/modules/audio_processing/level_estimator_bitexactness_unittest.cc

Index: webrtc/modules/audio_processing/level_estimator_bitexactness_unittest.cc
diff --git a/webrtc/modules/audio_processing/level_estimator_bitexactness_unittest.cc b/webrtc/modules/audio_processing/level_estimator_bitexactness_unittest.cc
deleted file mode 100644
index cb3bdf50969c88856d0b55041b1f5d27ec434deb..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/level_estimator_bitexactness_unittest.cc
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- *  Copyright (c) 2016 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.
- */
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/array_view.h"
-#include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/modules/audio_processing/level_estimator_impl.h"
-#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
-#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
-
-namespace webrtc {
-namespace {
-
-const int kNumFramesToProcess = 1000;
-
-// Processes a specified amount of frames, verifies the results and reports
-// any errors.
-void RunBitexactnessTest(int sample_rate_hz,
-                         size_t num_channels,
-                         int rms_reference) {
-  rtc::CriticalSection crit_capture;
-  LevelEstimatorImpl level_estimator(&crit_capture);
-  level_estimator.Initialize();
-  level_estimator.Enable(true);
-
-  int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
-  StreamConfig capture_config(sample_rate_hz, num_channels, false);
-  AudioBuffer capture_buffer(
-      capture_config.num_frames(), capture_config.num_channels(),
-      capture_config.num_frames(), capture_config.num_channels(),
-      capture_config.num_frames());
-
-  test::InputAudioFile capture_file(
-      test::GetApmCaptureTestVectorFileName(sample_rate_hz));
-  std::vector<float> capture_input(samples_per_channel * num_channels);
-  for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
-    ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
-                                   &capture_file, capture_input);
-
-    test::CopyVectorToAudioBuffer(capture_config, capture_input,
-                                  &capture_buffer);
-
-    level_estimator.ProcessStream(&capture_buffer);
-  }
-
-  // Extract test results.
-  int rms = level_estimator.RMS();
-
-  // Compare the output to the reference.
-  EXPECT_EQ(rms_reference, rms);
-}
-
-}  // namespace
-
-TEST(LevelEstimatorBitExactnessTest, Mono8kHz) {
-  const int kRmsReference = 31;
-
-  RunBitexactnessTest(8000, 1, kRmsReference);
-}
-
-TEST(LevelEstimatorBitExactnessTest, Mono16kHz) {
-  const int kRmsReference = 31;
-
-  RunBitexactnessTest(16000, 1, kRmsReference);
-}
-
-TEST(LevelEstimatorBitExactnessTest, Mono32kHz) {
-  const int kRmsReference = 31;
-
-  RunBitexactnessTest(32000, 1, kRmsReference);
-}
-
-TEST(LevelEstimatorBitExactnessTest, Mono48kHz) {
-  const int kRmsReference = 31;
-
-  RunBitexactnessTest(48000, 1, kRmsReference);
-}
-
-TEST(LevelEstimatorBitExactnessTest, Stereo16kHz) {
-  const int kRmsReference = 30;
-
-  RunBitexactnessTest(16000, 2, kRmsReference);
-}
-
-}  // namespace webrtc