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 30 using namespace android; 31 32 struct HeaderWav { 33 HeaderWav(size_t size, int nc, int sr, int bits) { 34 strncpy(RIFF, "RIFF", 4); 35 chunkSize = size + sizeof(HeaderWav); 36 strncpy(WAVE, "WAVE", 4); 37 strncpy(fmt, "fmt ", 4); 38 fmtSize = 16; 39 audioFormat = 1; 40 numChannels = nc; 41 samplesRate = sr; 42 byteRate = sr * numChannels * (bits/8); 43 align = nc*(bits/8); 44 bitsPerSample = bits; 45 strncpy(data, "data", 4); 46 dataSize = size; 47 } 48 49 char RIFF[4]; // RIFF 50 uint32_t chunkSize; // File size 51 char WAVE[4]; // WAVE 52 char fmt[4]; // fmt\0 53 uint32_t fmtSize; // fmt size 54 uint16_t audioFormat; // 1=PCM 55 uint16_t numChannels; // num channels 56 uint32_t samplesRate; // sample rate in hz 57 uint32_t byteRate; // Bps 58 uint16_t align; // 2=16-bit mono, 4=16-bit stereo 59 uint16_t bitsPerSample; // bits per sample 60 char data[4]; // "data" 61 uint32_t dataSize; // size 62 }; 63 64 static int usage(const char* name) { 65 fprintf(stderr,"Usage: %s [-p] [-h] [-s] [-q {dq|lq|mq|hq|vhq}] [-i input-sample-rate] " 66 "[-o output-sample-rate] [<input-file>] <output-file>\n", name); 67 fprintf(stderr," -p enable profiling\n"); 68 fprintf(stderr," -h create wav file\n"); 69 fprintf(stderr," -s stereo\n"); 70 fprintf(stderr," -q resampler quality\n"); 71 fprintf(stderr," dq : default quality\n"); 72 fprintf(stderr," lq : low quality\n"); 73 fprintf(stderr," mq : medium quality\n"); 74 fprintf(stderr," hq : high quality\n"); 75 fprintf(stderr," vhq : very high quality\n"); 76 fprintf(stderr," -i input file sample rate\n"); 77 fprintf(stderr," -o output file sample rate\n"); 78 return -1; 79 } 80 81 int main(int argc, char* argv[]) { 82 83 const char* const progname = argv[0]; 84 bool profiling = false; 85 bool writeHeader = false; 86 int channels = 1; 87 int input_freq = 0; 88 int output_freq = 0; 89 AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY; 90 91 int ch; 92 while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) { 93 switch (ch) { 94 case 'p': 95 profiling = true; 96 break; 97 case 'h': 98 writeHeader = true; 99 break; 100 case 's': 101 channels = 2; 102 break; 103 case 'q': 104 if (!strcmp(optarg, "dq")) 105 quality = AudioResampler::DEFAULT_QUALITY; 106 else if (!strcmp(optarg, "lq")) 107 quality = AudioResampler::LOW_QUALITY; 108 else if (!strcmp(optarg, "mq")) 109 quality = AudioResampler::MED_QUALITY; 110 else if (!strcmp(optarg, "hq")) 111 quality = AudioResampler::HIGH_QUALITY; 112 else if (!strcmp(optarg, "vhq")) 113 quality = AudioResampler::VERY_HIGH_QUALITY; 114 else { 115 usage(progname); 116 return -1; 117 } 118 break; 119 case 'i': 120 input_freq = atoi(optarg); 121 break; 122 case 'o': 123 output_freq = atoi(optarg); 124 break; 125 case '?': 126 default: 127 usage(progname); 128 return -1; 129 } 130 } 131 argc -= optind; 132 argv += optind; 133 134 const char* file_in = NULL; 135 const char* file_out = NULL; 136 if (argc == 1) { 137 file_out = argv[0]; 138 } else if (argc == 2) { 139 file_in = argv[0]; 140 file_out = argv[1]; 141 } else { 142 usage(progname); 143 return -1; 144 } 145 146 // ---------------------------------------------------------- 147 148 size_t input_size; 149 void* input_vaddr; 150 if (argc == 2) { 151 struct stat st; 152 if (stat(file_in, &st) < 0) { 153 fprintf(stderr, "stat: %s\n", strerror(errno)); 154 return -1; 155 } 156 157 int input_fd = open(file_in, O_RDONLY); 158 if (input_fd < 0) { 159 fprintf(stderr, "open: %s\n", strerror(errno)); 160 return -1; 161 } 162 163 input_size = st.st_size; 164 input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd, 0); 165 if (input_vaddr == MAP_FAILED ) { 166 fprintf(stderr, "mmap: %s\n", strerror(errno)); 167 return -1; 168 } 169 } else { 170 double k = 1000; // Hz / s 171 double time = (input_freq / 2) / k; 172 size_t input_frames = size_t(input_freq * time); 173 input_size = channels * sizeof(int16_t) * input_frames; 174 input_vaddr = malloc(input_size); 175 int16_t* in = (int16_t*)input_vaddr; 176 for (size_t i=0 ; i<input_frames ; i++) { 177 double t = double(i) / input_freq; 178 double y = sin(M_PI * k * t * t); 179 int16_t yi = floor(y * 32767.0 + 0.5); 180 for (size_t j=0 ; j<(size_t)channels ; j++) { 181 in[i*channels + j] = yi / (1+j); 182 } 183 } 184 } 185 186 // ---------------------------------------------------------- 187 188 class Provider: public AudioBufferProvider { 189 int16_t* mAddr; 190 size_t mNumFrames; 191 public: 192 Provider(const void* addr, size_t size, int channels) { 193 mAddr = (int16_t*) addr; 194 mNumFrames = size / (channels*sizeof(int16_t)); 195 } 196 virtual status_t getNextBuffer(Buffer* buffer, 197 int64_t pts = kInvalidPTS) { 198 buffer->frameCount = mNumFrames; 199 buffer->i16 = mAddr; 200 return NO_ERROR; 201 } 202 virtual void releaseBuffer(Buffer* buffer) { 203 } 204 } provider(input_vaddr, input_size, channels); 205 206 size_t input_frames = input_size / (channels * sizeof(int16_t)); 207 size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq; 208 output_size &= ~7; // always stereo, 32-bits 209 210 void* output_vaddr = malloc(output_size); 211 212 if (profiling) { 213 AudioResampler* resampler = AudioResampler::create(16, channels, 214 output_freq, quality); 215 216 size_t out_frames = output_size/8; 217 resampler->setSampleRate(input_freq); 218 resampler->setVolume(0x1000, 0x1000); 219 220 memset(output_vaddr, 0, output_size); 221 timespec start, end; 222 clock_gettime(CLOCK_MONOTONIC, &start); 223 resampler->resample((int*) output_vaddr, out_frames, &provider); 224 resampler->resample((int*) output_vaddr, out_frames, &provider); 225 resampler->resample((int*) output_vaddr, out_frames, &provider); 226 resampler->resample((int*) output_vaddr, out_frames, &provider); 227 clock_gettime(CLOCK_MONOTONIC, &end); 228 int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; 229 int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; 230 int64_t time = (end_ns - start_ns)/4; 231 printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6); 232 233 delete resampler; 234 } 235 236 AudioResampler* resampler = AudioResampler::create(16, channels, 237 output_freq, quality); 238 size_t out_frames = output_size/8; 239 resampler->setSampleRate(input_freq); 240 resampler->setVolume(0x1000, 0x1000); 241 242 memset(output_vaddr, 0, output_size); 243 resampler->resample((int*) output_vaddr, out_frames, &provider); 244 245 // down-mix (we just truncate and keep the left channel) 246 int32_t* out = (int32_t*) output_vaddr; 247 int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t)); 248 for (size_t i = 0; i < out_frames; i++) { 249 for (int j=0 ; j<channels ; j++) { 250 int32_t s = out[i * 2 + j] >> 12; 251 if (s > 32767) s = 32767; 252 else if (s < -32768) s = -32768; 253 convert[i * channels + j] = int16_t(s); 254 } 255 } 256 257 // write output to disk 258 int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC, 259 S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); 260 if (output_fd < 0) { 261 fprintf(stderr, "open: %s\n", strerror(errno)); 262 return -1; 263 } 264 265 if (writeHeader) { 266 HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16); 267 write(output_fd, &wav, sizeof(wav)); 268 } 269 270 write(output_fd, convert, out_frames * channels * sizeof(int16_t)); 271 close(output_fd); 272 273 return 0; 274 } 275
