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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc | 187 |
1 files changed, 187 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc new file mode 100644 index 0000000000..1183441a14 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc @@ -0,0 +1,187 @@ +/* + * 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/capture_levels_adjuster.h" + +#include <algorithm> +#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); + } +} + +float ComputeExpectedSignalGainAfterApplyPreLevelAdjustment( + bool emulated_analog_mic_gain_enabled, + int emulated_analog_mic_gain_level, + float pre_gain) { + if (!emulated_analog_mic_gain_enabled) { + return pre_gain; + } + return pre_gain * std::min(emulated_analog_mic_gain_level, 255) / 255.f; +} + +float ComputeExpectedSignalGainAfterApplyPostLevelAdjustment( + bool emulated_analog_mic_gain_enabled, + int emulated_analog_mic_gain_level, + float pre_gain, + float post_gain) { + return post_gain * ComputeExpectedSignalGainAfterApplyPreLevelAdjustment( + emulated_analog_mic_gain_enabled, + emulated_analog_mic_gain_level, pre_gain); +} + +constexpr int kNumFramesToProcess = 10; + +class CaptureLevelsAdjusterTest + : public ::testing::Test, + public ::testing::WithParamInterface< + std::tuple<int, int, bool, int, float, float>> { + protected: + int sample_rate_hz() const { return std::get<0>(GetParam()); } + int num_channels() const { return std::get<1>(GetParam()); } + bool emulated_analog_mic_gain_enabled() const { + return std::get<2>(GetParam()); + } + int emulated_analog_mic_gain_level() const { return std::get<3>(GetParam()); } + float pre_gain() const { return std::get<4>(GetParam()); } + float post_gain() const { return std::get<5>(GetParam()); } +}; + +INSTANTIATE_TEST_SUITE_P( + CaptureLevelsAdjusterTestSuite, + CaptureLevelsAdjusterTest, + ::testing::Combine(::testing::Values(16000, 32000, 48000), + ::testing::Values(1, 2, 4), + ::testing::Values(false, true), + ::testing::Values(21, 255), + ::testing::Values(0.1f, 1.f, 4.f), + ::testing::Values(0.1f, 1.f, 4.f))); + +TEST_P(CaptureLevelsAdjusterTest, InitialGainIsInstantlyAchieved) { + CaptureLevelsAdjuster adjuster(emulated_analog_mic_gain_enabled(), + emulated_analog_mic_gain_level(), pre_gain(), + post_gain()); + + AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(), + num_channels(), sample_rate_hz(), num_channels()); + + const float expected_signal_gain_after_pre_gain = + ComputeExpectedSignalGainAfterApplyPreLevelAdjustment( + emulated_analog_mic_gain_enabled(), emulated_analog_mic_gain_level(), + pre_gain()); + const float expected_signal_gain_after_post_level_adjustment = + ComputeExpectedSignalGainAfterApplyPostLevelAdjustment( + emulated_analog_mic_gain_enabled(), emulated_analog_mic_gain_level(), + pre_gain(), post_gain()); + + for (int frame = 0; frame < kNumFramesToProcess; ++frame) { + PopulateBuffer(audio_buffer); + adjuster.ApplyPreLevelAdjustment(audio_buffer); + EXPECT_FLOAT_EQ(adjuster.GetPreAdjustmentGain(), + expected_signal_gain_after_pre_gain); + + 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], + expected_signal_gain_after_pre_gain * SampleValueForChannel(ch)); + } + } + adjuster.ApplyPostLevelAdjustment(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], + expected_signal_gain_after_post_level_adjustment * + SampleValueForChannel(ch)); + } + } + } +} + +TEST_P(CaptureLevelsAdjusterTest, NewGainsAreAchieved) { + const int lower_emulated_analog_mic_gain_level = + emulated_analog_mic_gain_level(); + const float lower_pre_gain = pre_gain(); + const float lower_post_gain = post_gain(); + const int higher_emulated_analog_mic_gain_level = + std::min(lower_emulated_analog_mic_gain_level * 2, 255); + const float higher_pre_gain = lower_pre_gain * 2.f; + const float higher_post_gain = lower_post_gain * 2.f; + + CaptureLevelsAdjuster adjuster(emulated_analog_mic_gain_enabled(), + lower_emulated_analog_mic_gain_level, + lower_pre_gain, lower_post_gain); + + AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(), + num_channels(), sample_rate_hz(), num_channels()); + + const float expected_signal_gain_after_pre_gain = + ComputeExpectedSignalGainAfterApplyPreLevelAdjustment( + emulated_analog_mic_gain_enabled(), + higher_emulated_analog_mic_gain_level, higher_pre_gain); + const float expected_signal_gain_after_post_level_adjustment = + ComputeExpectedSignalGainAfterApplyPostLevelAdjustment( + emulated_analog_mic_gain_enabled(), + higher_emulated_analog_mic_gain_level, higher_pre_gain, + higher_post_gain); + + adjuster.SetPreGain(higher_pre_gain); + adjuster.SetPostGain(higher_post_gain); + adjuster.SetAnalogMicGainLevel(higher_emulated_analog_mic_gain_level); + + PopulateBuffer(audio_buffer); + adjuster.ApplyPreLevelAdjustment(audio_buffer); + adjuster.ApplyPostLevelAdjustment(audio_buffer); + EXPECT_EQ(adjuster.GetAnalogMicGainLevel(), + higher_emulated_analog_mic_gain_level); + + for (int frame = 1; frame < kNumFramesToProcess; ++frame) { + PopulateBuffer(audio_buffer); + adjuster.ApplyPreLevelAdjustment(audio_buffer); + EXPECT_FLOAT_EQ(adjuster.GetPreAdjustmentGain(), + expected_signal_gain_after_pre_gain); + 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], + expected_signal_gain_after_pre_gain * SampleValueForChannel(ch)); + } + } + + adjuster.ApplyPostLevelAdjustment(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], + expected_signal_gain_after_post_level_adjustment * + SampleValueForChannel(ch)); + } + } + + EXPECT_EQ(adjuster.GetAnalogMicGainLevel(), + higher_emulated_analog_mic_gain_level); + } +} + +} // namespace +} // namespace webrtc |