Added a bitexactness test for the level estimator in the audio processing module

webrtc/modules/audio_processing/level_estimator_bitexactness_unittest.cc

+/*
+ *  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();

[hlundin-webrtc, 2016/03/17 15:20:57]

Skip this line and simply do a one-liner:
EXPECT_EQ(rms_reference, 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