summaryrefslogtreecommitdiffstats
path: root/services/audioflinger/tests/resampler_tests.cpp
blob: d6217ba53c50c699097d1efddfae58816e0fc21c (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
/*
 * Copyright (C) 2014 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.
 */

//#define LOG_NDEBUG 0
#define LOG_TAG "audioflinger_resampler_tests"

#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <fcntl.h>
#include <string.h>
#include <sys/mman.h>
#include <sys/stat.h>
#include <errno.h>
#include <time.h>
#include <math.h>
#include <vector>
#include <utility>
#include <iostream>
#include <cutils/log.h>
#include <gtest/gtest.h>
#include <media/AudioBufferProvider.h>
#include "AudioResampler.h"
#include "test_utils.h"

void resample(int channels, void *output,
        size_t outputFrames, const std::vector<size_t> &outputIncr,
        android::AudioBufferProvider *provider, android::AudioResampler *resampler)
{
    for (size_t i = 0, j = 0; i < outputFrames; ) {
        size_t thisFrames = outputIncr[j++];
        if (j >= outputIncr.size()) {
            j = 0;
        }
        if (thisFrames == 0 || thisFrames > outputFrames - i) {
            thisFrames = outputFrames - i;
        }
        resampler->resample((int32_t*) output + channels*i, thisFrames, provider);
        i += thisFrames;
    }
}

void buffercmp(const void *reference, const void *test,
        size_t outputFrameSize, size_t outputFrames)
{
    for (size_t i = 0; i < outputFrames; ++i) {
        int check = memcmp((const char*)reference + i * outputFrameSize,
                (const char*)test + i * outputFrameSize, outputFrameSize);
        if (check) {
            ALOGE("Failure at frame %zu", i);
            ASSERT_EQ(check, 0); /* fails */
        }
    }
}

void testBufferIncrement(size_t channels, bool useFloat,
        unsigned inputFreq, unsigned outputFreq,
        enum android::AudioResampler::src_quality quality)
{
    const audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
    // create the provider
    std::vector<int> inputIncr;
    SignalProvider provider;
    if (useFloat) {
        provider.setChirp<float>(channels,
                0., outputFreq/2., outputFreq, outputFreq/2000.);
    } else {
        provider.setChirp<int16_t>(channels,
                0., outputFreq/2., outputFreq, outputFreq/2000.);
    }
    provider.setIncr(inputIncr);

    // calculate the output size
    size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
    size_t outputFrameSize = channels * (useFloat ? sizeof(float) : sizeof(int32_t));
    size_t outputSize = outputFrameSize * outputFrames;
    outputSize &= ~7;

    // create the resampler
    android::AudioResampler* resampler;

    resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
    resampler->setSampleRate(inputFreq);
    resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
            android::AudioResampler::UNITY_GAIN_FLOAT);

    // set up the reference run
    std::vector<size_t> refIncr;
    refIncr.push_back(outputFrames);
    void* reference = malloc(outputSize);
    resample(channels, reference, outputFrames, refIncr, &provider, resampler);

    provider.reset();

#if 0
    /* this test will fail - API interface issue: reset() does not clear internal buffers */
    resampler->reset();
#else
    delete resampler;
    resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
    resampler->setSampleRate(inputFreq);
    resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
            android::AudioResampler::UNITY_GAIN_FLOAT);
#endif

    // set up the test run
    std::vector<size_t> outIncr;
    outIncr.push_back(1);
    outIncr.push_back(2);
    outIncr.push_back(3);
    void* test = malloc(outputSize);
    inputIncr.push_back(1);
    inputIncr.push_back(3);
    provider.setIncr(inputIncr);
    resample(channels, test, outputFrames, outIncr, &provider, resampler);

    // check
    buffercmp(reference, test, outputFrameSize, outputFrames);

    free(reference);
    free(test);
    delete resampler;
}

template <typename T>
inline double sqr(T v)
{
    double dv = static_cast<double>(v);
    return dv * dv;
}

template <typename T>
double signalEnergy(T *start, T *end, unsigned stride)
{
    double accum = 0;

    for (T *p = start; p < end; p += stride) {
        accum += sqr(*p);
    }
    unsigned count = (end - start + stride - 1) / stride;
    return accum / count;
}

// TI = resampler input type, int16_t or float
// TO = resampler output type, int32_t or float
template <typename TI, typename TO>
void testStopbandDownconversion(size_t channels,
        unsigned inputFreq, unsigned outputFreq,
        unsigned passband, unsigned stopband,
        enum android::AudioResampler::src_quality quality)
{
    // create the provider
    std::vector<int> inputIncr;
    SignalProvider provider;
    provider.setChirp<TI>(channels,
            0., inputFreq/2., inputFreq, inputFreq/2000.);
    provider.setIncr(inputIncr);

    // calculate the output size
    size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
    size_t outputFrameSize = channels * sizeof(TO);
    size_t outputSize = outputFrameSize * outputFrames;
    outputSize &= ~7;

    // create the resampler
    android::AudioResampler* resampler;

    resampler = android::AudioResampler::create(
            is_same<TI, int16_t>::value ? AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_FLOAT,
            channels, outputFreq, quality);
    resampler->setSampleRate(inputFreq);
    resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
            android::AudioResampler::UNITY_GAIN_FLOAT);

    // set up the reference run
    std::vector<size_t> refIncr;
    refIncr.push_back(outputFrames);
    void* reference = malloc(outputSize);
    resample(channels, reference, outputFrames, refIncr, &provider, resampler);

    TO *out = reinterpret_cast<TO *>(reference);

    // check signal energy in passband
    const unsigned passbandFrame = passband * outputFreq / 1000.;
    const unsigned stopbandFrame = stopband * outputFreq / 1000.;

    // check each channel separately
    for (size_t i = 0; i < channels; ++i) {
        double passbandEnergy = signalEnergy(out, out + passbandFrame * channels, channels);
        double stopbandEnergy = signalEnergy(out + stopbandFrame * channels,
                out + outputFrames * channels, channels);
        double dbAtten = -10. * log10(stopbandEnergy / passbandEnergy);
        ASSERT_GT(dbAtten, 60.);

#if 0
        // internal verification
        printf("if:%d  of:%d  pbf:%d  sbf:%d  sbe: %f  pbe: %f  db: %.2f\n",
                provider.getNumFrames(), outputFrames,
                passbandFrame, stopbandFrame, stopbandEnergy, passbandEnergy, dbAtten);
        for (size_t i = 0; i < 10; ++i) {
            std::cout << out[i+passbandFrame*channels] << std::endl;
        }
        for (size_t i = 0; i < 10; ++i) {
            std::cout << out[i+stopbandFrame*channels] << std::endl;
        }
#endif
    }

    free(reference);
    delete resampler;
}

/* Buffer increment test
 *
 * We compare a reference output, where we consume and process the entire
 * buffer at a time, and a test output, where we provide small chunks of input
 * data and process small chunks of output (which may not be equivalent in size).
 *
 * Two subtests - fixed phase (3:2 down) and interpolated phase (147:320 up)
 */
TEST(audioflinger_resampler, bufferincrement_fixedphase) {
    // all of these work
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
            android::AudioResampler::LOW_QUALITY,
            android::AudioResampler::MED_QUALITY,
            android::AudioResampler::HIGH_QUALITY,
            android::AudioResampler::VERY_HIGH_QUALITY,
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testBufferIncrement(2, false, 48000, 32000, kQualityArray[i]);
    }
}

TEST(audioflinger_resampler, bufferincrement_interpolatedphase) {
    // all of these work except low quality
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
//           android::AudioResampler::LOW_QUALITY,
            android::AudioResampler::MED_QUALITY,
            android::AudioResampler::HIGH_QUALITY,
            android::AudioResampler::VERY_HIGH_QUALITY,
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testBufferIncrement(2, false, 22050, 48000, kQualityArray[i]);
    }
}

TEST(audioflinger_resampler, bufferincrement_fixedphase_multi) {
    // only dynamic quality
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testBufferIncrement(4, false, 48000, 32000, kQualityArray[i]);
    }
}

TEST(audioflinger_resampler, bufferincrement_interpolatedphase_multi_float) {
    // only dynamic quality
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testBufferIncrement(8, true, 22050, 48000, kQualityArray[i]);
    }
}

/* Simple aliasing test
 *
 * This checks stopband response of the chirp signal to make sure frequencies
 * are properly suppressed.  It uses downsampling because the stopband can be
 * clearly isolated by input frequencies exceeding the output sample rate (nyquist).
 */
TEST(audioflinger_resampler, stopbandresponse_integer) {
    // not all of these may work (old resamplers fail on downsampling)
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
            //android::AudioResampler::LOW_QUALITY,
            //android::AudioResampler::MED_QUALITY,
            //android::AudioResampler::HIGH_QUALITY,
            //android::AudioResampler::VERY_HIGH_QUALITY,
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    // in this test we assume a maximum transition band between 12kHz and 20kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<int16_t, int32_t>(
                2, 48000, 32000, 12000, 20000, kQualityArray[i]);
    }

    // in this test we assume a maximum transition band between 7kHz and 15kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    // (the weird ratio triggers interpolative resampling)
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<int16_t, int32_t>(
                2, 48000, 22101, 7000, 15000, kQualityArray[i]);
    }
}

TEST(audioflinger_resampler, stopbandresponse_integer_multichannel) {
    // not all of these may work (old resamplers fail on downsampling)
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
            //android::AudioResampler::LOW_QUALITY,
            //android::AudioResampler::MED_QUALITY,
            //android::AudioResampler::HIGH_QUALITY,
            //android::AudioResampler::VERY_HIGH_QUALITY,
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    // in this test we assume a maximum transition band between 12kHz and 20kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<int16_t, int32_t>(
                8, 48000, 32000, 12000, 20000, kQualityArray[i]);
    }

    // in this test we assume a maximum transition band between 7kHz and 15kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    // (the weird ratio triggers interpolative resampling)
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<int16_t, int32_t>(
                8, 48000, 22101, 7000, 15000, kQualityArray[i]);
    }
}

TEST(audioflinger_resampler, stopbandresponse_float) {
    // not all of these may work (old resamplers fail on downsampling)
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
            //android::AudioResampler::LOW_QUALITY,
            //android::AudioResampler::MED_QUALITY,
            //android::AudioResampler::HIGH_QUALITY,
            //android::AudioResampler::VERY_HIGH_QUALITY,
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    // in this test we assume a maximum transition band between 12kHz and 20kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<float, float>(
                2, 48000, 32000, 12000, 20000, kQualityArray[i]);
    }

    // in this test we assume a maximum transition band between 7kHz and 15kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    // (the weird ratio triggers interpolative resampling)
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<float, float>(
                2, 48000, 22101, 7000, 15000, kQualityArray[i]);
    }
}

TEST(audioflinger_resampler, stopbandresponse_float_multichannel) {
    // not all of these may work (old resamplers fail on downsampling)
    static const enum android::AudioResampler::src_quality kQualityArray[] = {
            //android::AudioResampler::LOW_QUALITY,
            //android::AudioResampler::MED_QUALITY,
            //android::AudioResampler::HIGH_QUALITY,
            //android::AudioResampler::VERY_HIGH_QUALITY,
            android::AudioResampler::DYN_LOW_QUALITY,
            android::AudioResampler::DYN_MED_QUALITY,
            android::AudioResampler::DYN_HIGH_QUALITY,
    };

    // in this test we assume a maximum transition band between 12kHz and 20kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<float, float>(
                8, 48000, 32000, 12000, 20000, kQualityArray[i]);
    }

    // in this test we assume a maximum transition band between 7kHz and 15kHz.
    // there must be at least 60dB relative attenuation between stopband and passband.
    // (the weird ratio triggers interpolative resampling)
    for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
        testStopbandDownconversion<float, float>(
                8, 48000, 22101, 7000, 15000, kQualityArray[i]);
    }
}