Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (C) 2014 The Android Open Source Project |
| 3 | * |
| 4 | * Licensed under the Apache License, Version 2.0 (the "License"); |
| 5 | * you may not use this file except in compliance with the License. |
| 6 | * You may obtain a copy of the License at |
| 7 | * |
| 8 | * http://www.apache.org/licenses/LICENSE-2.0 |
| 9 | * |
| 10 | * Unless required by applicable law or agreed to in writing, software |
| 11 | * distributed under the License is distributed on an "AS IS" BASIS, |
| 12 | * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| 13 | * See the License for the specific language governing permissions and |
| 14 | * limitations under the License. |
| 15 | */ |
| 16 | |
| 17 | //#define LOG_NDEBUG 0 |
| 18 | #define LOG_TAG "audioflinger_resampler_tests" |
| 19 | |
| 20 | #include <unistd.h> |
| 21 | #include <stdio.h> |
| 22 | #include <stdlib.h> |
| 23 | #include <fcntl.h> |
| 24 | #include <string.h> |
| 25 | #include <sys/mman.h> |
| 26 | #include <sys/stat.h> |
| 27 | #include <errno.h> |
| 28 | #include <time.h> |
| 29 | #include <math.h> |
| 30 | #include <vector> |
| 31 | #include <utility> |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 32 | #include <iostream> |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 33 | #include <cutils/log.h> |
| 34 | #include <gtest/gtest.h> |
| 35 | #include <media/AudioBufferProvider.h> |
| 36 | #include "AudioResampler.h" |
Andy Hung | c0e5ec8 | 2014-06-17 14:33:39 -0700 | [diff] [blame] | 37 | #include "test_utils.h" |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 38 | |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 39 | void resample(int channels, void *output, |
| 40 | size_t outputFrames, const std::vector<size_t> &outputIncr, |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 41 | android::AudioBufferProvider *provider, android::AudioResampler *resampler) |
| 42 | { |
| 43 | for (size_t i = 0, j = 0; i < outputFrames; ) { |
| 44 | size_t thisFrames = outputIncr[j++]; |
| 45 | if (j >= outputIncr.size()) { |
| 46 | j = 0; |
| 47 | } |
| 48 | if (thisFrames == 0 || thisFrames > outputFrames - i) { |
| 49 | thisFrames = outputFrames - i; |
| 50 | } |
Andy Hung | 6b3b7e3 | 2015-03-29 00:49:22 -0700 | [diff] [blame] | 51 | size_t framesResampled = resampler->resample( |
| 52 | (int32_t*) output + channels*i, thisFrames, provider); |
| 53 | // we should have enough buffer space, so there is no short count. |
| 54 | ASSERT_EQ(thisFrames, framesResampled); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 55 | i += thisFrames; |
| 56 | } |
| 57 | } |
| 58 | |
| 59 | void buffercmp(const void *reference, const void *test, |
| 60 | size_t outputFrameSize, size_t outputFrames) |
| 61 | { |
| 62 | for (size_t i = 0; i < outputFrames; ++i) { |
| 63 | int check = memcmp((const char*)reference + i * outputFrameSize, |
| 64 | (const char*)test + i * outputFrameSize, outputFrameSize); |
| 65 | if (check) { |
Glenn Kasten | a4daf0b | 2014-07-28 16:34:45 -0700 | [diff] [blame] | 66 | ALOGE("Failure at frame %zu", i); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 67 | ASSERT_EQ(check, 0); /* fails */ |
| 68 | } |
| 69 | } |
| 70 | } |
| 71 | |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 72 | void testBufferIncrement(size_t channels, bool useFloat, |
| 73 | unsigned inputFreq, unsigned outputFreq, |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 74 | enum android::AudioResampler::src_quality quality) |
| 75 | { |
Andy Hung | 3348e36 | 2014-07-07 10:21:44 -0700 | [diff] [blame] | 76 | const audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 77 | // create the provider |
Andy Hung | c0e5ec8 | 2014-06-17 14:33:39 -0700 | [diff] [blame] | 78 | std::vector<int> inputIncr; |
| 79 | SignalProvider provider; |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 80 | if (useFloat) { |
| 81 | provider.setChirp<float>(channels, |
| 82 | 0., outputFreq/2., outputFreq, outputFreq/2000.); |
| 83 | } else { |
| 84 | provider.setChirp<int16_t>(channels, |
| 85 | 0., outputFreq/2., outputFreq, outputFreq/2000.); |
| 86 | } |
Andy Hung | c0e5ec8 | 2014-06-17 14:33:39 -0700 | [diff] [blame] | 87 | provider.setIncr(inputIncr); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 88 | |
| 89 | // calculate the output size |
| 90 | size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq; |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 91 | size_t outputFrameSize = channels * (useFloat ? sizeof(float) : sizeof(int32_t)); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 92 | size_t outputSize = outputFrameSize * outputFrames; |
| 93 | outputSize &= ~7; |
| 94 | |
| 95 | // create the resampler |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 96 | android::AudioResampler* resampler; |
| 97 | |
Andy Hung | 3348e36 | 2014-07-07 10:21:44 -0700 | [diff] [blame] | 98 | resampler = android::AudioResampler::create(format, channels, outputFreq, quality); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 99 | resampler->setSampleRate(inputFreq); |
Andy Hung | 5e58b0a | 2014-06-23 19:07:29 -0700 | [diff] [blame] | 100 | resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT, |
| 101 | android::AudioResampler::UNITY_GAIN_FLOAT); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 102 | |
| 103 | // set up the reference run |
| 104 | std::vector<size_t> refIncr; |
| 105 | refIncr.push_back(outputFrames); |
| 106 | void* reference = malloc(outputSize); |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 107 | resample(channels, reference, outputFrames, refIncr, &provider, resampler); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 108 | |
| 109 | provider.reset(); |
| 110 | |
| 111 | #if 0 |
| 112 | /* this test will fail - API interface issue: reset() does not clear internal buffers */ |
| 113 | resampler->reset(); |
| 114 | #else |
| 115 | delete resampler; |
Andy Hung | 3348e36 | 2014-07-07 10:21:44 -0700 | [diff] [blame] | 116 | resampler = android::AudioResampler::create(format, channels, outputFreq, quality); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 117 | resampler->setSampleRate(inputFreq); |
Andy Hung | 5e58b0a | 2014-06-23 19:07:29 -0700 | [diff] [blame] | 118 | resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT, |
| 119 | android::AudioResampler::UNITY_GAIN_FLOAT); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 120 | #endif |
| 121 | |
| 122 | // set up the test run |
| 123 | std::vector<size_t> outIncr; |
| 124 | outIncr.push_back(1); |
| 125 | outIncr.push_back(2); |
| 126 | outIncr.push_back(3); |
| 127 | void* test = malloc(outputSize); |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 128 | inputIncr.push_back(1); |
| 129 | inputIncr.push_back(3); |
| 130 | provider.setIncr(inputIncr); |
| 131 | resample(channels, test, outputFrames, outIncr, &provider, resampler); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 132 | |
| 133 | // check |
| 134 | buffercmp(reference, test, outputFrameSize, outputFrames); |
| 135 | |
| 136 | free(reference); |
| 137 | free(test); |
| 138 | delete resampler; |
| 139 | } |
| 140 | |
| 141 | template <typename T> |
| 142 | inline double sqr(T v) |
| 143 | { |
| 144 | double dv = static_cast<double>(v); |
| 145 | return dv * dv; |
| 146 | } |
| 147 | |
| 148 | template <typename T> |
| 149 | double signalEnergy(T *start, T *end, unsigned stride) |
| 150 | { |
| 151 | double accum = 0; |
| 152 | |
| 153 | for (T *p = start; p < end; p += stride) { |
| 154 | accum += sqr(*p); |
| 155 | } |
| 156 | unsigned count = (end - start + stride - 1) / stride; |
| 157 | return accum / count; |
| 158 | } |
| 159 | |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 160 | // TI = resampler input type, int16_t or float |
| 161 | // TO = resampler output type, int32_t or float |
| 162 | template <typename TI, typename TO> |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 163 | void testStopbandDownconversion(size_t channels, |
| 164 | unsigned inputFreq, unsigned outputFreq, |
| 165 | unsigned passband, unsigned stopband, |
| 166 | enum android::AudioResampler::src_quality quality) |
| 167 | { |
| 168 | // create the provider |
Andy Hung | c0e5ec8 | 2014-06-17 14:33:39 -0700 | [diff] [blame] | 169 | std::vector<int> inputIncr; |
| 170 | SignalProvider provider; |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 171 | provider.setChirp<TI>(channels, |
Andy Hung | c0e5ec8 | 2014-06-17 14:33:39 -0700 | [diff] [blame] | 172 | 0., inputFreq/2., inputFreq, inputFreq/2000.); |
| 173 | provider.setIncr(inputIncr); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 174 | |
| 175 | // calculate the output size |
| 176 | size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq; |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 177 | size_t outputFrameSize = channels * sizeof(TO); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 178 | size_t outputSize = outputFrameSize * outputFrames; |
| 179 | outputSize &= ~7; |
| 180 | |
| 181 | // create the resampler |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 182 | android::AudioResampler* resampler; |
| 183 | |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 184 | resampler = android::AudioResampler::create( |
| 185 | is_same<TI, int16_t>::value ? AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_FLOAT, |
Andy Hung | 3348e36 | 2014-07-07 10:21:44 -0700 | [diff] [blame] | 186 | channels, outputFreq, quality); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 187 | resampler->setSampleRate(inputFreq); |
Andy Hung | 5e58b0a | 2014-06-23 19:07:29 -0700 | [diff] [blame] | 188 | resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT, |
| 189 | android::AudioResampler::UNITY_GAIN_FLOAT); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 190 | |
| 191 | // set up the reference run |
| 192 | std::vector<size_t> refIncr; |
| 193 | refIncr.push_back(outputFrames); |
| 194 | void* reference = malloc(outputSize); |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 195 | resample(channels, reference, outputFrames, refIncr, &provider, resampler); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 196 | |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 197 | TO *out = reinterpret_cast<TO *>(reference); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 198 | |
| 199 | // check signal energy in passband |
| 200 | const unsigned passbandFrame = passband * outputFreq / 1000.; |
| 201 | const unsigned stopbandFrame = stopband * outputFreq / 1000.; |
| 202 | |
| 203 | // check each channel separately |
| 204 | for (size_t i = 0; i < channels; ++i) { |
| 205 | double passbandEnergy = signalEnergy(out, out + passbandFrame * channels, channels); |
| 206 | double stopbandEnergy = signalEnergy(out + stopbandFrame * channels, |
| 207 | out + outputFrames * channels, channels); |
| 208 | double dbAtten = -10. * log10(stopbandEnergy / passbandEnergy); |
| 209 | ASSERT_GT(dbAtten, 60.); |
| 210 | |
| 211 | #if 0 |
| 212 | // internal verification |
| 213 | printf("if:%d of:%d pbf:%d sbf:%d sbe: %f pbe: %f db: %.2f\n", |
| 214 | provider.getNumFrames(), outputFrames, |
| 215 | passbandFrame, stopbandFrame, stopbandEnergy, passbandEnergy, dbAtten); |
| 216 | for (size_t i = 0; i < 10; ++i) { |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 217 | std::cout << out[i+passbandFrame*channels] << std::endl; |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 218 | } |
| 219 | for (size_t i = 0; i < 10; ++i) { |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 220 | std::cout << out[i+stopbandFrame*channels] << std::endl; |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 221 | } |
| 222 | #endif |
| 223 | } |
| 224 | |
| 225 | free(reference); |
| 226 | delete resampler; |
| 227 | } |
| 228 | |
| 229 | /* Buffer increment test |
| 230 | * |
| 231 | * We compare a reference output, where we consume and process the entire |
| 232 | * buffer at a time, and a test output, where we provide small chunks of input |
| 233 | * data and process small chunks of output (which may not be equivalent in size). |
| 234 | * |
| 235 | * Two subtests - fixed phase (3:2 down) and interpolated phase (147:320 up) |
| 236 | */ |
| 237 | TEST(audioflinger_resampler, bufferincrement_fixedphase) { |
| 238 | // all of these work |
| 239 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 240 | android::AudioResampler::LOW_QUALITY, |
| 241 | android::AudioResampler::MED_QUALITY, |
| 242 | android::AudioResampler::HIGH_QUALITY, |
| 243 | android::AudioResampler::VERY_HIGH_QUALITY, |
| 244 | android::AudioResampler::DYN_LOW_QUALITY, |
| 245 | android::AudioResampler::DYN_MED_QUALITY, |
| 246 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 247 | }; |
| 248 | |
| 249 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 250 | testBufferIncrement(2, false, 48000, 32000, kQualityArray[i]); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 251 | } |
| 252 | } |
| 253 | |
| 254 | TEST(audioflinger_resampler, bufferincrement_interpolatedphase) { |
| 255 | // all of these work except low quality |
| 256 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 257 | // android::AudioResampler::LOW_QUALITY, |
| 258 | android::AudioResampler::MED_QUALITY, |
| 259 | android::AudioResampler::HIGH_QUALITY, |
| 260 | android::AudioResampler::VERY_HIGH_QUALITY, |
| 261 | android::AudioResampler::DYN_LOW_QUALITY, |
| 262 | android::AudioResampler::DYN_MED_QUALITY, |
| 263 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 264 | }; |
| 265 | |
| 266 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
Andy Hung | 075abae | 2014-04-09 19:36:43 -0700 | [diff] [blame] | 267 | testBufferIncrement(2, false, 22050, 48000, kQualityArray[i]); |
| 268 | } |
| 269 | } |
| 270 | |
| 271 | TEST(audioflinger_resampler, bufferincrement_fixedphase_multi) { |
| 272 | // only dynamic quality |
| 273 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 274 | android::AudioResampler::DYN_LOW_QUALITY, |
| 275 | android::AudioResampler::DYN_MED_QUALITY, |
| 276 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 277 | }; |
| 278 | |
| 279 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 280 | testBufferIncrement(4, false, 48000, 32000, kQualityArray[i]); |
| 281 | } |
| 282 | } |
| 283 | |
| 284 | TEST(audioflinger_resampler, bufferincrement_interpolatedphase_multi_float) { |
| 285 | // only dynamic quality |
| 286 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 287 | android::AudioResampler::DYN_LOW_QUALITY, |
| 288 | android::AudioResampler::DYN_MED_QUALITY, |
| 289 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 290 | }; |
| 291 | |
| 292 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 293 | testBufferIncrement(8, true, 22050, 48000, kQualityArray[i]); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 294 | } |
| 295 | } |
| 296 | |
| 297 | /* Simple aliasing test |
| 298 | * |
| 299 | * This checks stopband response of the chirp signal to make sure frequencies |
| 300 | * are properly suppressed. It uses downsampling because the stopband can be |
| 301 | * clearly isolated by input frequencies exceeding the output sample rate (nyquist). |
| 302 | */ |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 303 | TEST(audioflinger_resampler, stopbandresponse_integer) { |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 304 | // not all of these may work (old resamplers fail on downsampling) |
| 305 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 306 | //android::AudioResampler::LOW_QUALITY, |
| 307 | //android::AudioResampler::MED_QUALITY, |
| 308 | //android::AudioResampler::HIGH_QUALITY, |
| 309 | //android::AudioResampler::VERY_HIGH_QUALITY, |
| 310 | android::AudioResampler::DYN_LOW_QUALITY, |
| 311 | android::AudioResampler::DYN_MED_QUALITY, |
| 312 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 313 | }; |
| 314 | |
| 315 | // in this test we assume a maximum transition band between 12kHz and 20kHz. |
| 316 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 317 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 318 | testStopbandDownconversion<int16_t, int32_t>( |
| 319 | 2, 48000, 32000, 12000, 20000, kQualityArray[i]); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 320 | } |
| 321 | |
| 322 | // in this test we assume a maximum transition band between 7kHz and 15kHz. |
| 323 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 324 | // (the weird ratio triggers interpolative resampling) |
| 325 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 326 | testStopbandDownconversion<int16_t, int32_t>( |
| 327 | 2, 48000, 22101, 7000, 15000, kQualityArray[i]); |
Andy Hung | 546734b | 2014-04-01 18:31:42 -0700 | [diff] [blame] | 328 | } |
| 329 | } |
Andy Hung | 42b0111 | 2014-07-20 14:04:19 -0700 | [diff] [blame] | 330 | |
| 331 | TEST(audioflinger_resampler, stopbandresponse_integer_multichannel) { |
| 332 | // not all of these may work (old resamplers fail on downsampling) |
| 333 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 334 | //android::AudioResampler::LOW_QUALITY, |
| 335 | //android::AudioResampler::MED_QUALITY, |
| 336 | //android::AudioResampler::HIGH_QUALITY, |
| 337 | //android::AudioResampler::VERY_HIGH_QUALITY, |
| 338 | android::AudioResampler::DYN_LOW_QUALITY, |
| 339 | android::AudioResampler::DYN_MED_QUALITY, |
| 340 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 341 | }; |
| 342 | |
| 343 | // in this test we assume a maximum transition band between 12kHz and 20kHz. |
| 344 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 345 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 346 | testStopbandDownconversion<int16_t, int32_t>( |
| 347 | 8, 48000, 32000, 12000, 20000, kQualityArray[i]); |
| 348 | } |
| 349 | |
| 350 | // in this test we assume a maximum transition band between 7kHz and 15kHz. |
| 351 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 352 | // (the weird ratio triggers interpolative resampling) |
| 353 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 354 | testStopbandDownconversion<int16_t, int32_t>( |
| 355 | 8, 48000, 22101, 7000, 15000, kQualityArray[i]); |
| 356 | } |
| 357 | } |
| 358 | |
| 359 | TEST(audioflinger_resampler, stopbandresponse_float) { |
| 360 | // not all of these may work (old resamplers fail on downsampling) |
| 361 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 362 | //android::AudioResampler::LOW_QUALITY, |
| 363 | //android::AudioResampler::MED_QUALITY, |
| 364 | //android::AudioResampler::HIGH_QUALITY, |
| 365 | //android::AudioResampler::VERY_HIGH_QUALITY, |
| 366 | android::AudioResampler::DYN_LOW_QUALITY, |
| 367 | android::AudioResampler::DYN_MED_QUALITY, |
| 368 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 369 | }; |
| 370 | |
| 371 | // in this test we assume a maximum transition band between 12kHz and 20kHz. |
| 372 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 373 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 374 | testStopbandDownconversion<float, float>( |
| 375 | 2, 48000, 32000, 12000, 20000, kQualityArray[i]); |
| 376 | } |
| 377 | |
| 378 | // in this test we assume a maximum transition band between 7kHz and 15kHz. |
| 379 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 380 | // (the weird ratio triggers interpolative resampling) |
| 381 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 382 | testStopbandDownconversion<float, float>( |
| 383 | 2, 48000, 22101, 7000, 15000, kQualityArray[i]); |
| 384 | } |
| 385 | } |
| 386 | |
| 387 | TEST(audioflinger_resampler, stopbandresponse_float_multichannel) { |
| 388 | // not all of these may work (old resamplers fail on downsampling) |
| 389 | static const enum android::AudioResampler::src_quality kQualityArray[] = { |
| 390 | //android::AudioResampler::LOW_QUALITY, |
| 391 | //android::AudioResampler::MED_QUALITY, |
| 392 | //android::AudioResampler::HIGH_QUALITY, |
| 393 | //android::AudioResampler::VERY_HIGH_QUALITY, |
| 394 | android::AudioResampler::DYN_LOW_QUALITY, |
| 395 | android::AudioResampler::DYN_MED_QUALITY, |
| 396 | android::AudioResampler::DYN_HIGH_QUALITY, |
| 397 | }; |
| 398 | |
| 399 | // in this test we assume a maximum transition band between 12kHz and 20kHz. |
| 400 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 401 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 402 | testStopbandDownconversion<float, float>( |
| 403 | 8, 48000, 32000, 12000, 20000, kQualityArray[i]); |
| 404 | } |
| 405 | |
| 406 | // in this test we assume a maximum transition band between 7kHz and 15kHz. |
| 407 | // there must be at least 60dB relative attenuation between stopband and passband. |
| 408 | // (the weird ratio triggers interpolative resampling) |
| 409 | for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { |
| 410 | testStopbandDownconversion<float, float>( |
| 411 | 8, 48000, 22101, 7000, 15000, kQualityArray[i]); |
| 412 | } |
| 413 | } |
| 414 | |