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/*
* Copyright (C) 2021 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#include <vector>
#include "EffectDownmix.h"
#include <audio_utils/channels.h>
#include <audio_utils/primitives.h>
#include <audio_utils/Statistics.h>
#include <gtest/gtest.h>
#include <log/log.h>
extern audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM;
static constexpr audio_channel_mask_t kChannelPositionMasks[] = {
AUDIO_CHANNEL_OUT_FRONT_LEFT, // Legacy: the downmix effect treats MONO as FRONT_LEFT only.
// The AudioMixer interprets MONO as a special case requiring
// channel replication, bypassing the downmix effect.
AUDIO_CHANNEL_OUT_FRONT_CENTER,
AUDIO_CHANNEL_OUT_STEREO,
AUDIO_CHANNEL_OUT_2POINT1,
AUDIO_CHANNEL_OUT_2POINT0POINT2,
AUDIO_CHANNEL_OUT_QUAD, // AUDIO_CHANNEL_OUT_QUAD_BACK
AUDIO_CHANNEL_OUT_QUAD_SIDE,
AUDIO_CHANNEL_OUT_SURROUND,
AUDIO_CHANNEL_OUT_2POINT1POINT2,
AUDIO_CHANNEL_OUT_3POINT0POINT2,
AUDIO_CHANNEL_OUT_PENTA,
AUDIO_CHANNEL_OUT_3POINT1POINT2,
AUDIO_CHANNEL_OUT_5POINT1, // AUDIO_CHANNEL_OUT_5POINT1_BACK
AUDIO_CHANNEL_OUT_5POINT1_SIDE,
AUDIO_CHANNEL_OUT_6POINT1,
AUDIO_CHANNEL_OUT_5POINT1POINT2,
AUDIO_CHANNEL_OUT_7POINT1,
AUDIO_CHANNEL_OUT_5POINT1POINT4,
AUDIO_CHANNEL_OUT_7POINT1POINT2,
AUDIO_CHANNEL_OUT_7POINT1POINT4,
AUDIO_CHANNEL_OUT_13POINT_360RA,
AUDIO_CHANNEL_OUT_22POINT2,
audio_channel_mask_t(AUDIO_CHANNEL_OUT_22POINT2
| AUDIO_CHANNEL_OUT_FRONT_WIDE_LEFT | AUDIO_CHANNEL_OUT_FRONT_WIDE_RIGHT),
};
constexpr float COEF_25 = 0.2508909536f;
constexpr float COEF_35 = 0.3543928915f;
constexpr float COEF_36 = 0.3552343859f;
constexpr float COEF_61 = 0.6057043428f;
constexpr inline float kScaleFromChannelIdxLeft[] = {
1.f, // AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1u,
0.f, // AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_CENTER = 0x4u,
0.5f, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 0x8u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_BACK_LEFT = 0x10u,
0.f, // AUDIO_CHANNEL_OUT_BACK_RIGHT = 0x20u,
COEF_61, // AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40u,
COEF_25, // AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80u,
0.5f, // AUDIO_CHANNEL_OUT_BACK_CENTER = 0x100u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_SIDE_LEFT = 0x200u,
0.f, // AUDIO_CHANNEL_OUT_SIDE_RIGHT = 0x400u,
COEF_36, // AUDIO_CHANNEL_OUT_TOP_CENTER = 0x800u,
1.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 0x1000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 0x2000u,
0.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 0x4000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 0x8000u,
COEF_35, // AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000u,
0.f, // AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000u,
COEF_61, // AUDIO_CHANNEL_OUT_TOP_SIDE_LEFT = 0x40000u,
0.f, // AUDIO_CHANNEL_OUT_TOP_SIDE_RIGHT = 0x80000u,
1.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_LEFT = 0x100000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_CENTER = 0x200000u,
0.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_RIGHT = 0x400000u,
0.f, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2 = 0x800000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_WIDE_LEFT = 0x1000000u,
0.f, // AUDIO_CHANNEL_OUT_FRONT_WIDE_RIGHT = 0x2000000u,
};
constexpr inline float kScaleFromChannelIdxRight[] = {
0.f, // AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1u,
1.f, // AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_CENTER = 0x4u,
0.5f, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 0x8u,
0.f, // AUDIO_CHANNEL_OUT_BACK_LEFT = 0x10u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_BACK_RIGHT = 0x20u,
COEF_25, // AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40u,
COEF_61, // AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80u,
0.5f, // AUDIO_CHANNEL_OUT_BACK_CENTER = 0x100u,
0.f, // AUDIO_CHANNEL_OUT_SIDE_LEFT = 0x200u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_SIDE_RIGHT = 0x400u,
COEF_36, // AUDIO_CHANNEL_OUT_TOP_CENTER = 0x800u,
0.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 0x1000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 0x2000u,
1.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 0x4000u,
0.f, // AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 0x8000u,
COEF_35, // AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000u,
0.f, // AUDIO_CHANNEL_OUT_TOP_SIDE_LEFT = 0x40000u,
COEF_61, // AUDIO_CHANNEL_OUT_TOP_SIDE_RIGHT = 0x80000u,
0.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_LEFT = 0x100000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_CENTER = 0x200000u,
1.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_RIGHT = 0x400000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2 = 0x800000u,
0.f, // AUDIO_CHANNEL_OUT_FRONT_WIDE_LEFT = 0x1000000u,
M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_WIDE_RIGHT = 0x2000000u,
};
// Downmix doesn't change with sample rate
static constexpr size_t kSampleRates[] = {
48000,
};
// Our near expectation is 16x the bit that doesn't fit the mantissa.
// this works so long as we add values close in exponent with each other
// realizing that errors accumulate as the sqrt of N (random walk, lln, etc).
#define EXPECT_NEAR_EPSILON(e, v) EXPECT_NEAR((e), (v), \
abs((e) * std::numeric_limits<std::decay_t<decltype(e)>>::epsilon() * 8))
template<typename T>
static auto channelStatistics(const std::vector<T>& input, size_t channels) {
std::vector<android::audio_utils::Statistics<T>> result(channels);
const size_t frames = input.size() / channels;
if (frames > 0) {
const float *fptr = input.data();
for (size_t i = 0; i < frames; ++i) {
for (size_t j = 0; j < channels; ++j) {
result[j].add(*fptr++);
}
}
}
return result;
}
using DownmixParam = std::tuple<int /* sample rate */, int /* channel mask */>;
class DownmixTest : public ::testing::TestWithParam<DownmixParam> {
public:
static constexpr effect_uuid_t downmix_uuid_ = {
0x93f04452, 0xe4fe, 0x41cc, 0x91f9, {0xe4, 0x75, 0xb6, 0xd1, 0xd6, 0x9f}};
static constexpr size_t FRAME_LENGTH = 256;
void testBalance(int sampleRate, audio_channel_mask_t channelMask) {
using namespace ::android::audio_utils::channels;
size_t frames = 100; // set to an even number (2, 4, 6 ... ) stream alternates +1, -1.
constexpr unsigned outChannels = 2;
unsigned inChannels = audio_channel_count_from_out_mask(channelMask);
std::vector<float> input(frames * inChannels);
std::vector<float> output(frames * outChannels);
double savedPower[32][2]{};
for (unsigned i = 0, channel = channelMask; channel != 0; ++i) {
const int index = __builtin_ctz(channel);
ASSERT_LT(index, FCC_26);
const int pairIndex = pairIdxFromChannelIdx(index);
const AUDIO_GEOMETRY_SIDE side = sideFromChannelIdx(index);
const int channelBit = 1 << index;
channel &= ~channelBit;
// Generate a +1, -1 alternating stream in one channel, which has variance 1.
auto indata = input.data();
for (unsigned j = 0; j < frames; ++j) {
for (unsigned k = 0; k < inChannels; ++k) {
*indata++ = (k == i) ? (j & 1 ? -1 : 1) : 0;
}
}
run(sampleRate, channelMask, input, output, frames);
auto stats = channelStatistics(output, 2 /* channels */);
// printf("power: %s %s\n", stats[0].toString().c_str(), stats[1].toString().c_str());
double power[2] = { stats[0].getPopVariance(), stats[1].getPopVariance() };
// Check symmetric power for pair channels on exchange of left/right position.
// to do this, we save previous power measurements.
if (pairIndex >= 0 && pairIndex < index) {
EXPECT_NEAR_EPSILON(power[0], savedPower[pairIndex][1]);
EXPECT_NEAR_EPSILON(power[1], savedPower[pairIndex][0]);
}
savedPower[index][0] = power[0];
savedPower[index][1] = power[1];
constexpr float POWER_TOLERANCE = 0.001;
const float expectedPower =
kScaleFromChannelIdxLeft[index] * kScaleFromChannelIdxLeft[index]
+ kScaleFromChannelIdxRight[index] * kScaleFromChannelIdxRight[index];
EXPECT_NEAR(expectedPower, power[0] + power[1], POWER_TOLERANCE);
switch (side) {
case AUDIO_GEOMETRY_SIDE_LEFT:
if (channelBit == AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) {
break;
}
EXPECT_EQ(0.f, power[1]);
break;
case AUDIO_GEOMETRY_SIDE_RIGHT:
if (channelBit == AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) {
break;
}
EXPECT_EQ(0.f, power[0]);
break;
case AUDIO_GEOMETRY_SIDE_CENTER:
if (channelBit == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) {
if (channelMask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) {
EXPECT_EQ(0.f, power[1]);
break;
} else {
EXPECT_NEAR_EPSILON(power[0], power[1]); // always true
EXPECT_NEAR(expectedPower, power[0] + power[1], POWER_TOLERANCE);
break;
}
} else if (channelBit == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) {
EXPECT_EQ(0.f, power[0]);
EXPECT_NEAR(expectedPower, power[1], POWER_TOLERANCE);
break;
}
EXPECT_NEAR_EPSILON(power[0], power[1]);
break;
}
}
}
void run(int sampleRate, audio_channel_mask_t channelMask,
std::vector<float>& input, std::vector<float>& output, size_t frames) {
reconfig(sampleRate, channelMask);
ASSERT_EQ(frames * inputChannelCount_, input.size());
ASSERT_EQ(frames * outputChannelCount_, output.size());
const int32_t sessionId = 0;
const int32_t ioId = 0;
int32_t err = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(
&downmix_uuid_, sessionId, ioId, &handle_);
ASSERT_EQ(0, err);
const struct effect_interface_s * const downmixApi = *handle_;
int32_t reply = 0;
uint32_t replySize = (uint32_t)sizeof(reply);
err = (downmixApi->command)(
handle_, EFFECT_CMD_SET_CONFIG,
sizeof(effect_config_t), &config_, &replySize, &reply);
ASSERT_EQ(0, err);
ASSERT_EQ(0, reply);
err = (downmixApi->command)(
handle_, EFFECT_CMD_ENABLE,
0, nullptr, &replySize, &reply);
ASSERT_EQ(0, err);
process(input, output, frames);
err = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(handle_);
ASSERT_EQ(0, err);
}
// This test assumes the channel mask is invalid.
void testInvalidChannelMask(audio_channel_mask_t invalidChannelMask) {
reconfig(48000 /* sampleRate */, invalidChannelMask);
const int32_t sessionId = 0;
const int32_t ioId = 0;
int32_t err = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(
&downmix_uuid_, sessionId, ioId, &handle_);
ASSERT_EQ(0, err);
const struct effect_interface_s * const downmixApi = *handle_;
int32_t reply = 0;
uint32_t replySize = (uint32_t)sizeof(reply);
err = (downmixApi->command)(
handle_, EFFECT_CMD_SET_CONFIG,
sizeof(effect_config_t), &config_, &replySize, &reply);
ASSERT_EQ(0, err);
ASSERT_NE(0, reply); // error has occurred.
err = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(handle_);
ASSERT_EQ(0, err);
}
private:
void reconfig(int sampleRate, audio_channel_mask_t channelMask) {
config_.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
config_.inputCfg.format = AUDIO_FORMAT_PCM_FLOAT;
config_.inputCfg.bufferProvider.getBuffer = nullptr;
config_.inputCfg.bufferProvider.releaseBuffer = nullptr;
config_.inputCfg.bufferProvider.cookie = nullptr;
config_.inputCfg.mask = EFFECT_CONFIG_ALL;
config_.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
config_.outputCfg.format = AUDIO_FORMAT_PCM_FLOAT;
config_.outputCfg.bufferProvider.getBuffer = nullptr;
config_.outputCfg.bufferProvider.releaseBuffer = nullptr;
config_.outputCfg.bufferProvider.cookie = nullptr;
config_.outputCfg.mask = EFFECT_CONFIG_ALL;
config_.inputCfg.samplingRate = sampleRate;
config_.inputCfg.channels = channelMask;
inputChannelCount_ = audio_channel_count_from_out_mask(config_.inputCfg.channels);
config_.outputCfg.samplingRate = sampleRate;
config_.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO; // output always stereo
outputChannelCount_ = audio_channel_count_from_out_mask(config_.outputCfg.channels);
}
void process(std::vector<float> &input, std::vector<float> &output, size_t frames) const {
const struct effect_interface_s * const downmixApi = *handle_;
for (size_t pos = 0; pos < frames;) {
const size_t transfer = std::min(frames - pos, FRAME_LENGTH);
audio_buffer_t inbuffer{.frameCount = transfer,
.f32 = input.data() + pos * inputChannelCount_};
audio_buffer_t outbuffer{.frameCount = transfer,
.f32 = output.data() + pos * outputChannelCount_};
const int32_t err = (downmixApi->process)(handle_, &inbuffer, &outbuffer);
ASSERT_EQ(0, err);
pos += transfer;
}
}
effect_handle_t handle_{};
effect_config_t config_{};
int outputChannelCount_{};
int inputChannelCount_{};
};
TEST(DownmixTestSimple, invalidChannelMask) {
// Fill in a dummy test method to use DownmixTest outside of a parameterized test.
class DownmixTestComplete : public DownmixTest {
void TestBody() override {}
} downmixtest;
constexpr auto INVALID_CHANNEL_MASK = audio_channel_mask_t(1 << 31);
downmixtest.testInvalidChannelMask(INVALID_CHANNEL_MASK);
}
TEST_P(DownmixTest, basic) {
testBalance(kSampleRates[std::get<0>(GetParam())],
kChannelPositionMasks[std::get<1>(GetParam())]);
}
INSTANTIATE_TEST_SUITE_P(
DownmixTestAll, DownmixTest,
::testing::Combine(
::testing::Range(0, (int)std::size(kSampleRates)),
::testing::Range(0, (int)std::size(kChannelPositionMasks))
),
[](const testing::TestParamInfo<DownmixTest::ParamType>& info) {
const int index = std::get<1>(info.param);
const audio_channel_mask_t channelMask = kChannelPositionMasks[index];
const std::string name = std::string(audio_channel_out_mask_to_string(channelMask))
+ "_" + std::to_string(std::get<0>(info.param)) + "_" + std::to_string(index);
return name;
});