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/*
* Copyright (c) 2012 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.
*/
// Use CreateHistUnittestFile.m to generate the input file.
#include "webrtc/modules/audio_processing/agc/histogram.h"
#include <stdio.h>
#include <cmath>
#include "gtest/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/modules/audio_processing/agc/utility.h"
namespace webrtc {
struct InputOutput {
double rms;
double activity_probability;
double audio_content;
double loudness;
};
const double kRelativeErrTol = 1e-10;
class HistogramTest : public ::testing::Test {
protected:
void RunTest(bool enable_circular_buff,
const char* filename);
private:
void TestClean();
scoped_ptr<Histogram> hist_;
};
void HistogramTest::TestClean() {
EXPECT_EQ(hist_->CurrentRms(), 7.59621091765857e-02);
EXPECT_EQ(hist_->AudioContent(), 0);
EXPECT_EQ(hist_->num_updates(), 0);
}
void HistogramTest::RunTest(bool enable_circular_buff, const char* filename) {
FILE* in_file = fopen(filename, "rb");
ASSERT_TRUE(in_file != NULL);
if (enable_circular_buff) {
int buffer_size;
EXPECT_EQ(fread(&buffer_size, sizeof(buffer_size), 1, in_file), 1u);
hist_.reset(Histogram::Create(buffer_size));
} else {
hist_.reset(Histogram::Create());
}
TestClean();
InputOutput io;
int num_updates = 0;
int num_reset = 0;
while (fread(&io, sizeof(InputOutput), 1, in_file) == 1) {
if (io.rms < 0) {
// We have to reset.
hist_->Reset();
TestClean();
num_updates = 0;
num_reset++;
// Read the next chunk of input.
if (fread(&io, sizeof(InputOutput), 1, in_file) != 1)
break;
}
hist_->Update(io.rms, io.activity_probability);
num_updates++;
EXPECT_EQ(hist_->num_updates(), num_updates);
double audio_content = hist_->AudioContent();
double abs_err = std::min(audio_content, io.audio_content) *
kRelativeErrTol;
ASSERT_NEAR(audio_content, io.audio_content, abs_err);
double current_loudness = Linear2Loudness(hist_->CurrentRms());
abs_err = std::min(fabs(current_loudness), fabs(io.loudness)) *
kRelativeErrTol;
ASSERT_NEAR(current_loudness, io.loudness, abs_err);
}
fclose(in_file);
}
TEST_F(HistogramTest, ActiveCircularBuffer) {
RunTest(true,
test::ResourcePath("audio_processing/agc/agc_with_circular_buffer",
"dat").c_str());
}
TEST_F(HistogramTest, InactiveCircularBuffer) {
RunTest(false,
test::ResourcePath("audio_processing/agc/agc_no_circular_buffer",
"dat").c_str());
}
} // namespace webrtc
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