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/*
* 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 "webrtc/base/array_view.h"
#include "webrtc/base/optional.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/level_controller/level_controller.h"
#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
#include "webrtc/test/gtest.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,
rtc::Optional<float> initial_peak_level_dbfs,
rtc::ArrayView<const float> output_reference) {
LevelController level_controller;
level_controller.Initialize(sample_rate_hz);
if (initial_peak_level_dbfs) {
AudioProcessing::Config::LevelController config;
config.initial_peak_level_dbfs = *initial_peak_level_dbfs;
level_controller.ApplyConfig(config);
}
int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
const 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_controller.Process(&capture_buffer);
}
// Extract test results.
std::vector<float> capture_output;
test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
&capture_output);
// Compare the output with the reference. Only the first values of the output
// from last frame processed are compared in order not having to specify all
// preceding frames as testvectors. As the algorithm being tested has a
// memory, testing only the last frame implicitly also tests the preceeding
// frames.
const float kVectorElementErrorBound = 1.0f / 32768.0f;
EXPECT_TRUE(test::VerifyDeinterleavedArray(
capture_config.num_frames(), capture_config.num_channels(),
output_reference, capture_output, kVectorElementErrorBound));
}
} // namespace
TEST(LevelControllerConfig, ToString) {
AudioProcessing::Config config;
config.level_controller.enabled = true;
config.level_controller.initial_peak_level_dbfs = -6.0206f;
EXPECT_EQ("{enabled: true, initial_peak_level_dbfs: -6.0206}",
LevelController::ToString(config.level_controller));
config.level_controller.enabled = false;
config.level_controller.initial_peak_level_dbfs = -50.f;
EXPECT_EQ("{enabled: false, initial_peak_level_dbfs: -50}",
LevelController::ToString(config.level_controller));
}
TEST(LevelControlBitExactnessTest, DISABLED_Mono8kHz) {
const float kOutputReference[] = {-0.013939f, -0.012154f, -0.009054f};
RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1,
rtc::Optional<float>(), kOutputReference);
}
TEST(LevelControlBitExactnessTest, DISABLED_Mono16kHz) {
const float kOutputReference[] = {-0.013706f, -0.013215f, -0.013018f};
RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 1,
rtc::Optional<float>(), kOutputReference);
}
TEST(LevelControlBitExactnessTest, DISABLED_Mono32kHz) {
const float kOutputReference[] = {-0.014495f, -0.016425f, -0.016085f};
RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 1,
rtc::Optional<float>(), kOutputReference);
}
// TODO(peah): Investigate why this particular testcase differ between Android
// and the rest of the platforms.
TEST(LevelControlBitExactnessTest, DISABLED_Mono48kHz) {
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
const float kOutputReference[] = {-0.014277f, -0.015180f, -0.017437f};
#else
const float kOutputReference[] = {-0.015949f, -0.016957f, -0.019478f};
#endif
RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1,
rtc::Optional<float>(), kOutputReference);
}
TEST(LevelControlBitExactnessTest, DISABLED_Stereo8kHz) {
const float kOutputReference[] = {-0.014063f, -0.008450f, -0.012159f,
-0.051967f, -0.023202f, -0.047858f};
RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 2,
rtc::Optional<float>(), kOutputReference);
}
TEST(LevelControlBitExactnessTest, DISABLED_Stereo16kHz) {
const float kOutputReference[] = {-0.012714f, -0.005896f, -0.012220f,
-0.053306f, -0.024549f, -0.051527f};
RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 2,
rtc::Optional<float>(), kOutputReference);
}
TEST(LevelControlBitExactnessTest, DISABLED_Stereo32kHz) {
const float kOutputReference[] = {-0.011737f, -0.007018f, -0.013446f,
-0.053505f, -0.026292f, -0.056221f};
RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 2,
rtc::Optional<float>(), kOutputReference);
}
TEST(LevelControlBitExactnessTest, DISABLED_Stereo48kHz) {
const float kOutputReference[] = {-0.010643f, -0.006334f, -0.011377f,
-0.049088f, -0.023600f, -0.050465f};
RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2,
rtc::Optional<float>(), kOutputReference);
}
TEST(LevelControlBitExactnessTest, DISABLED_MonoInitial48kHz) {
const float kOutputReference[] = {-0.013753f, -0.014623f, -0.016797f};
RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1,
rtc::Optional<float>(-50), kOutputReference);
}
} // namespace webrtc
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