1 files changed, 204 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc
new file mode 100644
index 0000000000..6e5fc2cbe3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2021 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 "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"
+
+#include <tuple>
+
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+float SampleValueForChannel(int channel) {
+  constexpr float kSampleBaseValue = 100.f;
+  constexpr float kSampleChannelOffset = 1.f;
+  return kSampleBaseValue + channel * kSampleChannelOffset;
+}
+
+void PopulateBuffer(AudioBuffer& audio_buffer) {
+  for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+    test::FillBufferChannel(SampleValueForChannel(ch), ch, audio_buffer);
+  }
+}
+
+constexpr int kNumFramesToProcess = 10;
+
+class AudioSamplesScalerTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<int, int, float>> {
+ protected:
+  int sample_rate_hz() const { return std::get<0>(GetParam()); }
+  int num_channels() const { return std::get<1>(GetParam()); }
+  float initial_gain() const { return std::get<2>(GetParam()); }
+};
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioSamplesScalerTestSuite,
+    AudioSamplesScalerTest,
+    ::testing::Combine(::testing::Values(16000, 32000, 48000),
+                       ::testing::Values(1, 2, 4),
+                       ::testing::Values(0.1f, 1.f, 2.f, 4.f)));
+
+TEST_P(AudioSamplesScalerTest, InitialGainIsRespected) {
+  AudioSamplesScaler scaler(initial_gain());
+
+  AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
+                           num_channels(), sample_rate_hz(), num_channels());
+
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    scaler.Process(audio_buffer);
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        initial_gain() * SampleValueForChannel(ch));
+      }
+    }
+  }
+}
+
+TEST_P(AudioSamplesScalerTest, VerifyGainAdjustment) {
+  const float higher_gain = initial_gain();
+  const float lower_gain = higher_gain / 2.f;
+
+  AudioSamplesScaler scaler(lower_gain);
+
+  AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
+                           num_channels(), sample_rate_hz(), num_channels());
+
+  // Allow the intial, lower, gain to take effect.
+  PopulateBuffer(audio_buffer);
+
+  scaler.Process(audio_buffer);
+
+  // Set the new, higher, gain.
+  scaler.SetGain(higher_gain);
+
+  // Ensure that the new, higher, gain is achieved gradually over one frame.
+  PopulateBuffer(audio_buffer);
+
+  scaler.Process(audio_buffer);
+  for (int ch = 0; ch < num_channels(); ++ch) {
+    for (size_t i = 0; i < audio_buffer.num_frames() - 1; ++i) {
+      EXPECT_LT(audio_buffer.channels_const()[ch][i],
+                higher_gain * SampleValueForChannel(ch));
+      EXPECT_LE(audio_buffer.channels_const()[ch][i],
+                audio_buffer.channels_const()[ch][i + 1]);
+    }
+    EXPECT_LE(audio_buffer.channels_const()[ch][audio_buffer.num_frames() - 1],
+              higher_gain * SampleValueForChannel(ch));
+  }
+
+  // Ensure that the new, higher, gain is achieved and stay unchanged.
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    scaler.Process(audio_buffer);
+
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        higher_gain * SampleValueForChannel(ch));
+      }
+    }
+  }
+
+  // Set the new, lower, gain.
+  scaler.SetGain(lower_gain);
+
+  // Ensure that the new, lower, gain is achieved gradually over one frame.
+  PopulateBuffer(audio_buffer);
+  scaler.Process(audio_buffer);
+  for (int ch = 0; ch < num_channels(); ++ch) {
+    for (size_t i = 0; i < audio_buffer.num_frames() - 1; ++i) {
+      EXPECT_GT(audio_buffer.channels_const()[ch][i],
+                lower_gain * SampleValueForChannel(ch));
+      EXPECT_GE(audio_buffer.channels_const()[ch][i],
+                audio_buffer.channels_const()[ch][i + 1]);
+    }
+    EXPECT_GE(audio_buffer.channels_const()[ch][audio_buffer.num_frames() - 1],
+              lower_gain * SampleValueForChannel(ch));
+  }
+
+  // Ensure that the new, lower, gain is achieved and stay unchanged.
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    scaler.Process(audio_buffer);
+
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        lower_gain * SampleValueForChannel(ch));
+      }
+    }
+  }
+}
+
+TEST(AudioSamplesScaler, UpwardsClamping) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  constexpr float kGain = 10.f;
+  constexpr float kMaxClampedSampleValue = 32767.f;
+  static_assert(kGain > 1.f, "");
+
+  AudioSamplesScaler scaler(kGain);
+
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+      test::FillBufferChannel(
+          kMaxClampedSampleValue - audio_buffer.num_channels() + 1.f + ch, ch,
+          audio_buffer);
+    }
+
+    scaler.Process(audio_buffer);
+    for (int ch = 0; ch < kNumChannels; ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        kMaxClampedSampleValue);
+      }
+    }
+  }
+}
+
+TEST(AudioSamplesScaler, DownwardsClamping) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  constexpr float kGain = 10.f;
+  constexpr float kMinClampedSampleValue = -32768.f;
+  static_assert(kGain > 1.f, "");
+
+  AudioSamplesScaler scaler(kGain);
+
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+      test::FillBufferChannel(
+          kMinClampedSampleValue + audio_buffer.num_channels() - 1.f + ch, ch,
+          audio_buffer);
+    }
+
+    scaler.Process(audio_buffer);
+    for (int ch = 0; ch < kNumChannels; ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        kMinClampedSampleValue);
+      }
+    }
+  }
+}
+
+}  // namespace
+}  // namespace webrtc