test-resample.cpp revision 47f3f5a119194d4c06815453be9950ba112c8e3e 
1/*
2 * Copyright (C) 2012 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#include "AudioResampler.h"
18#include <media/AudioBufferProvider.h>
19#include <unistd.h>
20#include <stdio.h>
21#include <stdlib.h>
22#include <fcntl.h>
23#include <string.h>
24#include <sys/mman.h>
25#include <sys/stat.h>
26#include <errno.h>
27#include <time.h>
28#include <math.h>
29#include <audio_utils/sndfile.h>
30
31using namespace android;
32
33bool gVerbose = false;
34
35static int usage(const char* name) {
36    fprintf(stderr,"Usage: %s [-p] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq}] [-i input-sample-rate] "
37                   "[-o output-sample-rate] [<input-file>] <output-file>\n", name);
38    fprintf(stderr,"    -p    enable profiling\n");
39    fprintf(stderr,"    -h    create wav file\n");
40    fprintf(stderr,"    -v    verbose : log buffer provider calls\n");
41    fprintf(stderr,"    -s    stereo (ignored if input file is specified)\n");
42    fprintf(stderr,"    -q    resampler quality\n");
43    fprintf(stderr,"              dq  : default quality\n");
44    fprintf(stderr,"              lq  : low quality\n");
45    fprintf(stderr,"              mq  : medium quality\n");
46    fprintf(stderr,"              hq  : high quality\n");
47    fprintf(stderr,"              vhq : very high quality\n");
48    fprintf(stderr,"    -i    input file sample rate (ignored if input file is specified)\n");
49    fprintf(stderr,"    -o    output file sample rate\n");
50    return -1;
51}
52
53int main(int argc, char* argv[]) {
54
55    const char* const progname = argv[0];
56    bool profiling = false;
57    bool writeHeader = false;
58    int channels = 1;
59    int input_freq = 0;
60    int output_freq = 0;
61    AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
62
63    int ch;
64    while ((ch = getopt(argc, argv, "phvsq:i:o:")) != -1) {
65        switch (ch) {
66        case 'p':
67            profiling = true;
68            break;
69        case 'h':
70            writeHeader = true;
71            break;
72        case 'v':
73            gVerbose = true;
74            break;
75        case 's':
76            channels = 2;
77            break;
78        case 'q':
79            if (!strcmp(optarg, "dq"))
80                quality = AudioResampler::DEFAULT_QUALITY;
81            else if (!strcmp(optarg, "lq"))
82                quality = AudioResampler::LOW_QUALITY;
83            else if (!strcmp(optarg, "mq"))
84                quality = AudioResampler::MED_QUALITY;
85            else if (!strcmp(optarg, "hq"))
86                quality = AudioResampler::HIGH_QUALITY;
87            else if (!strcmp(optarg, "vhq"))
88                quality = AudioResampler::VERY_HIGH_QUALITY;
89            else {
90                usage(progname);
91                return -1;
92            }
93            break;
94        case 'i':
95            input_freq = atoi(optarg);
96            break;
97        case 'o':
98            output_freq = atoi(optarg);
99            break;
100        case '?':
101        default:
102            usage(progname);
103            return -1;
104        }
105    }
106    argc -= optind;
107    argv += optind;
108
109    const char* file_in = NULL;
110    const char* file_out = NULL;
111    if (argc == 1) {
112        file_out = argv[0];
113    } else if (argc == 2) {
114        file_in = argv[0];
115        file_out = argv[1];
116    } else {
117        usage(progname);
118        return -1;
119    }
120
121    // ----------------------------------------------------------
122
123    size_t input_size;
124    void* input_vaddr;
125    if (argc == 2) {
126        SF_INFO info;
127        info.format = 0;
128        SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
129        if (sf == NULL) {
130            perror(file_in);
131            return EXIT_FAILURE;
132        }
133        input_size = info.frames * info.channels * sizeof(short);
134        input_vaddr = malloc(input_size);
135        (void) sf_readf_short(sf, (short *) input_vaddr, info.frames);
136        sf_close(sf);
137        channels = info.channels;
138        input_freq = info.samplerate;
139    } else {
140        double k = 1000; // Hz / s
141        double time = (input_freq / 2) / k;
142        size_t input_frames = size_t(input_freq * time);
143        input_size = channels * sizeof(int16_t) * input_frames;
144        input_vaddr = malloc(input_size);
145        int16_t* in = (int16_t*)input_vaddr;
146        for (size_t i=0 ; i<input_frames ; i++) {
147            double t = double(i) / input_freq;
148            double y = sin(M_PI * k * t * t);
149            int16_t yi = floor(y * 32767.0 + 0.5);
150            for (size_t j=0 ; j<(size_t)channels ; j++) {
151                in[i*channels + j] = yi / (1+j);
152            }
153        }
154    }
155
156    // ----------------------------------------------------------
157
158    class Provider: public AudioBufferProvider {
159        int16_t* const  mAddr;      // base address
160        const size_t    mNumFrames; // total frames
161        const int       mChannels;
162        size_t          mNextFrame; // index of next frame to provide
163        size_t          mUnrel;     // number of frames not yet released
164    public:
165        Provider(const void* addr, size_t size, int channels)
166          : mAddr((int16_t*) addr),
167            mNumFrames(size / (channels*sizeof(int16_t))),
168            mChannels(channels),
169            mNextFrame(0), mUnrel(0) {
170        }
171        virtual status_t getNextBuffer(Buffer* buffer,
172                int64_t pts = kInvalidPTS) {
173            size_t requestedFrames = buffer->frameCount;
174            if (requestedFrames > mNumFrames - mNextFrame) {
175                buffer->frameCount = mNumFrames - mNextFrame;
176            }
177            if (gVerbose) {
178                printf("getNextBuffer() requested %u frames out of %u frames available,"
179                        " and returned %u frames\n",
180                        requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
181            }
182            mUnrel = buffer->frameCount;
183            if (buffer->frameCount > 0) {
184                buffer->i16 = &mAddr[mChannels * mNextFrame];
185                return NO_ERROR;
186            } else {
187                buffer->i16 = NULL;
188                return NOT_ENOUGH_DATA;
189            }
190        }
191        virtual void releaseBuffer(Buffer* buffer) {
192            if (buffer->frameCount > mUnrel) {
193                fprintf(stderr, "ERROR releaseBuffer() released %u frames but only %u available "
194                        "to release\n", buffer->frameCount, mUnrel);
195                mNextFrame += mUnrel;
196                mUnrel = 0;
197            } else {
198                if (gVerbose) {
199                    printf("releaseBuffer() released %u frames out of %u frames available "
200                            "to release\n", buffer->frameCount, mUnrel);
201                }
202                mNextFrame += buffer->frameCount;
203                mUnrel -= buffer->frameCount;
204            }
205            buffer->frameCount = 0;
206            buffer->i16 = NULL;
207        }
208    } provider(input_vaddr, input_size, channels);
209
210    size_t input_frames = input_size / (channels * sizeof(int16_t));
211    if (gVerbose) {
212        printf("%u input frames\n", input_frames);
213    }
214    size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
215    output_size &= ~7; // always stereo, 32-bits
216
217    void* output_vaddr = malloc(output_size);
218
219    if (profiling) {
220        AudioResampler* resampler = AudioResampler::create(16, channels,
221                output_freq, quality);
222
223        size_t out_frames = output_size/8;
224        resampler->setSampleRate(input_freq);
225        resampler->setVolume(0x1000, 0x1000);
226
227        memset(output_vaddr, 0, output_size);
228        timespec start, end;
229        clock_gettime(CLOCK_MONOTONIC, &start);
230        resampler->resample((int*) output_vaddr, out_frames, &provider);
231        resampler->resample((int*) output_vaddr, out_frames, &provider);
232        resampler->resample((int*) output_vaddr, out_frames, &provider);
233        resampler->resample((int*) output_vaddr, out_frames, &provider);
234        clock_gettime(CLOCK_MONOTONIC, &end);
235        int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
236        int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
237        int64_t time = (end_ns - start_ns)/4;
238        printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6);
239
240        delete resampler;
241    }
242
243    AudioResampler* resampler = AudioResampler::create(16, channels,
244            output_freq, quality);
245    size_t out_frames = output_size/8;
246    resampler->setSampleRate(input_freq);
247    resampler->setVolume(0x1000, 0x1000);
248
249    memset(output_vaddr, 0, output_size);
250    if (gVerbose) {
251        printf("resample() %u output frames\n", out_frames);
252    }
253    resampler->resample((int*) output_vaddr, out_frames, &provider);
254    if (gVerbose) {
255        printf("resample() complete\n");
256    }
257    resampler->reset();
258    if (gVerbose) {
259        printf("reset() complete\n");
260    }
261
262    // down-mix (we just truncate and keep the left channel)
263    int32_t* out = (int32_t*) output_vaddr;
264    int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));
265    for (size_t i = 0; i < out_frames; i++) {
266        for (int j=0 ; j<channels ; j++) {
267            int32_t s = out[i * 2 + j] >> 12;
268            if (s > 32767)       s =  32767;
269            else if (s < -32768) s = -32768;
270            convert[i * channels + j] = int16_t(s);
271        }
272    }
273
274    // write output to disk
275    if (writeHeader) {
276        SF_INFO info;
277        info.frames = 0;
278        info.samplerate = output_freq;
279        info.channels = channels;
280        info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
281        SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info);
282        if (sf == NULL) {
283            perror(file_out);
284            return EXIT_FAILURE;
285        }
286        (void) sf_writef_short(sf, convert, out_frames);
287        sf_close(sf);
288    } else {
289        int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC,
290                S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
291        if (output_fd < 0) {
292            perror(file_out);
293            return EXIT_FAILURE;
294        }
295        write(output_fd, convert, out_frames * channels * sizeof(int16_t));
296        close(output_fd);
297    }
298
299    return EXIT_SUCCESS;
300}
301