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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 "modules/audio_processing/audio_buffer.h"
#include <cmath>
#include "api/audio/audio_view.h"
#include "test/gtest.h"
#include "test/testsupport/rtc_expect_death.h"
namespace webrtc {
namespace {
const size_t kSampleRateHz = 48000u;
const size_t kStereo = 2u;
const size_t kMono = 1u;
void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
EXPECT_EQ(ab.num_channels(), num_channels);
}
} // namespace
TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
AudioBuffer ab(kSampleRateHz, kStereo, kSampleRateHz, kStereo, kSampleRateHz,
kStereo);
ExpectNumChannels(ab, kStereo);
ab.set_num_channels(1);
ExpectNumChannels(ab, kMono);
ab.RestoreNumChannels();
ExpectNumChannels(ab, kStereo);
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(AudioBufferDeathTest, SetNumChannelsDeathTest) {
AudioBuffer ab(kSampleRateHz, kMono, kSampleRateHz, kMono, kSampleRateHz,
kMono);
RTC_EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
}
#endif
TEST(AudioBufferTest, CopyWithoutResampling) {
AudioBuffer ab1(32000, 2, 32000, 2, 32000, 2);
AudioBuffer ab2(32000, 2, 32000, 2, 32000, 2);
// Fill first buffer.
for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
for (size_t i = 0; i < ab1.num_frames(); ++i) {
ab1.channels()[ch][i] = i + ch;
}
}
// Copy to second buffer.
ab1.CopyTo(&ab2);
// Verify content of second buffer.
for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
for (size_t i = 0; i < ab2.num_frames(); ++i) {
EXPECT_EQ(ab2.channels()[ch][i], i + ch);
}
}
}
TEST(AudioBufferTest, CopyWithResampling) {
AudioBuffer ab1(32000, 2, 32000, 2, 48000, 2);
AudioBuffer ab2(48000, 2, 48000, 2, 48000, 2);
float energy_ab1 = 0.f;
float energy_ab2 = 0.f;
const float pi = std::acos(-1.f);
// Put a sine and compute energy of first buffer.
for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
for (size_t i = 0; i < ab1.num_frames(); ++i) {
ab1.channels()[ch][i] = std::sin(2 * pi * 100.f / 32000.f * i);
energy_ab1 += ab1.channels()[ch][i] * ab1.channels()[ch][i];
}
}
// Copy to second buffer.
ab1.CopyTo(&ab2);
// Compute energy of second buffer.
for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
for (size_t i = 0; i < ab2.num_frames(); ++i) {
energy_ab2 += ab2.channels()[ch][i] * ab2.channels()[ch][i];
}
}
// Verify that energies match.
EXPECT_NEAR(energy_ab1, energy_ab2 * 32000.f / 48000.f, .01f * energy_ab1);
}
TEST(AudioBufferTest, DeinterleavedView) {
AudioBuffer ab(48000, 2, 48000, 2, 48000, 2);
// Fill the buffer with data.
const float pi = std::acos(-1.f);
float* const* channels = ab.channels();
for (size_t ch = 0; ch < ab.num_channels(); ++ch) {
for (size_t i = 0; i < ab.num_frames(); ++i) {
channels[ch][i] = std::sin(2 * pi * 100.f / 32000.f * i);
}
}
// Verify that the DeinterleavedView correctly maps to channels.
DeinterleavedView<float> view = ab.view();
ASSERT_EQ(view.num_channels(), ab.num_channels());
for (size_t c = 0; c < view.num_channels(); ++c) {
MonoView<float> channel = view[c];
EXPECT_EQ(SamplesPerChannel(channel), ab.num_frames());
for (size_t s = 0; s < SamplesPerChannel(channel); ++s) {
ASSERT_EQ(channel[s], channels[c][s]);
}
}
}
} // namespace webrtc