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 #include <stdio.h> 18 #include <inttypes.h> 19 #include <math.h> 20 #include <vector> 21 #include <audio_utils/primitives.h> 22 #include <audio_utils/sndfile.h> 23 #include <media/AudioBufferProvider.h> 24 #include "AudioMixer.h" 25 #include "test_utils.h" 26 27 /* Testing is typically through creation of an output WAV file from several 28 * source inputs, to be later analyzed by an audio program such as Audacity. 29 * 30 * Sine or chirp functions are typically more useful as input to the mixer 31 * as they show up as straight lines on a spectrogram if successfully mixed. 32 * 33 * A sample shell script is provided: mixer_to_wave_tests.sh 34 */ 35 36 using namespace android; 37 38 static void usage(const char* name) { 39 fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]" 40 " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]" 41 " (<input-file> | <command>)+\n", name); 42 fprintf(stderr, " -f enable floating point input track\n"); 43 fprintf(stderr, " -m enable floating point mixer output\n"); 44 fprintf(stderr, " -c number of mixer output channels\n"); 45 fprintf(stderr, " -s mixer sample-rate\n"); 46 fprintf(stderr, " -o <output-file> WAV file, pcm16 (or float if -m specified)\n"); 47 fprintf(stderr, " -a <aux-buffer-file>\n"); 48 fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n"); 49 fprintf(stderr, " <input-file> is a WAV file\n"); 50 fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n"); 51 fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n"); 52 } 53 54 static int writeFile(const char *filename, const void *buffer, 55 uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) { 56 if (filename == NULL) { 57 return 0; // ok to pass in NULL filename 58 } 59 // write output to file. 60 SF_INFO info; 61 info.frames = 0; 62 info.samplerate = sampleRate; 63 info.channels = channels; 64 info.format = SF_FORMAT_WAV | (isBufferFloat ? SF_FORMAT_FLOAT : SF_FORMAT_PCM_16); 65 printf("saving file:%s channels:%u samplerate:%u frames:%zu\n", 66 filename, info.channels, info.samplerate, frames); 67 SNDFILE *sf = sf_open(filename, SFM_WRITE, &info); 68 if (sf == NULL) { 69 perror(filename); 70 return EXIT_FAILURE; 71 } 72 if (isBufferFloat) { 73 (void) sf_writef_float(sf, (float*)buffer, frames); 74 } else { 75 (void) sf_writef_short(sf, (short*)buffer, frames); 76 } 77 sf_close(sf); 78 return EXIT_SUCCESS; 79 } 80 81 int main(int argc, char* argv[]) { 82 const char* const progname = argv[0]; 83 bool useInputFloat = false; 84 bool useMixerFloat = false; 85 bool useRamp = true; 86 uint32_t outputSampleRate = 48000; 87 uint32_t outputChannels = 2; // stereo for now 88 std::vector<int> Pvalues; 89 const char* outputFilename = NULL; 90 const char* auxFilename = NULL; 91 std::vector<int32_t> Names; 92 std::vector<SignalProvider> Providers; 93 94 for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) { 95 switch (ch) { 96 case 'f': 97 useInputFloat = true; 98 break; 99 case 'm': 100 useMixerFloat = true; 101 break; 102 case 'c': 103 outputChannels = atoi(optarg); 104 break; 105 case 's': 106 outputSampleRate = atoi(optarg); 107 break; 108 case 'o': 109 outputFilename = optarg; 110 break; 111 case 'a': 112 auxFilename = optarg; 113 break; 114 case 'P': 115 if (parseCSV(optarg, Pvalues) < 0) { 116 fprintf(stderr, "incorrect syntax for -P option\n"); 117 return EXIT_FAILURE; 118 } 119 break; 120 case '?': 121 default: 122 usage(progname); 123 return EXIT_FAILURE; 124 } 125 } 126 argc -= optind; 127 argv += optind; 128 129 if (argc == 0) { 130 usage(progname); 131 return EXIT_FAILURE; 132 } 133 if ((unsigned)argc > AudioMixer::MAX_NUM_TRACKS) { 134 fprintf(stderr, "too many tracks: %d > %u", argc, AudioMixer::MAX_NUM_TRACKS); 135 return EXIT_FAILURE; 136 } 137 138 size_t outputFrames = 0; 139 140 // create providers for each track 141 Providers.resize(argc); 142 for (int i = 0; i < argc; ++i) { 143 static const char chirp[] = "chirp:"; 144 static const char sine[] = "sine:"; 145 static const double kSeconds = 1; 146 147 if (!strncmp(argv[i], chirp, strlen(chirp))) { 148 std::vector<int> v; 149 150 parseCSV(argv[i] + strlen(chirp), v); 151 if (v.size() == 2) { 152 printf("creating chirp(%d %d)\n", v[0], v[1]); 153 if (useInputFloat) { 154 Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds); 155 } else { 156 Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds); 157 } 158 Providers[i].setIncr(Pvalues); 159 } else { 160 fprintf(stderr, "malformed input '%s'\n", argv[i]); 161 } 162 } else if (!strncmp(argv[i], sine, strlen(sine))) { 163 std::vector<int> v; 164 165 parseCSV(argv[i] + strlen(sine), v); 166 if (v.size() == 3) { 167 printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]); 168 if (useInputFloat) { 169 Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds); 170 } else { 171 Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds); 172 } 173 Providers[i].setIncr(Pvalues); 174 } else { 175 fprintf(stderr, "malformed input '%s'\n", argv[i]); 176 } 177 } else { 178 printf("creating filename(%s)\n", argv[i]); 179 if (useInputFloat) { 180 Providers[i].setFile<float>(argv[i]); 181 } else { 182 Providers[i].setFile<short>(argv[i]); 183 } 184 Providers[i].setIncr(Pvalues); 185 } 186 // calculate the number of output frames 187 size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate 188 / Providers[i].getSampleRate(); 189 if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames 190 outputFrames = nframes; 191 } 192 } 193 194 // create the output buffer. 195 const size_t outputFrameSize = outputChannels 196 * (useMixerFloat ? sizeof(float) : sizeof(int16_t)); 197 const size_t outputSize = outputFrames * outputFrameSize; 198 const audio_channel_mask_t outputChannelMask = 199 audio_channel_out_mask_from_count(outputChannels); 200 void *outputAddr = NULL; 201 (void) posix_memalign(&outputAddr, 32, outputSize); 202 memset(outputAddr, 0, outputSize); 203 204 // create the aux buffer, if needed. 205 const size_t auxFrameSize = sizeof(int32_t); // Q4.27 always 206 const size_t auxSize = outputFrames * auxFrameSize; 207 void *auxAddr = NULL; 208 if (auxFilename) { 209 (void) posix_memalign(&auxAddr, 32, auxSize); 210 memset(auxAddr, 0, auxSize); 211 } 212 213 // create the mixer. 214 const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960 215 AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate); 216 audio_format_t inputFormat = useInputFloat 217 ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; 218 audio_format_t mixerFormat = useMixerFloat 219 ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; 220 float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks 221 static float f0; // zero 222 223 // set up the tracks. 224 for (size_t i = 0; i < Providers.size(); ++i) { 225 //printf("track %d out of %d\n", i, Providers.size()); 226 uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels()); 227 int32_t name = mixer->getTrackName(channelMask, 228 inputFormat, AUDIO_SESSION_OUTPUT_MIX); 229 ALOG_ASSERT(name >= 0); 230 Names.push_back(name); 231 mixer->setBufferProvider(name, &Providers[i]); 232 mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, 233 (void *)outputAddr); 234 mixer->setParameter( 235 name, 236 AudioMixer::TRACK, 237 AudioMixer::MIXER_FORMAT, 238 (void *)(uintptr_t)mixerFormat); 239 mixer->setParameter( 240 name, 241 AudioMixer::TRACK, 242 AudioMixer::FORMAT, 243 (void *)(uintptr_t)inputFormat); 244 mixer->setParameter( 245 name, 246 AudioMixer::TRACK, 247 AudioMixer::MIXER_CHANNEL_MASK, 248 (void *)(uintptr_t)outputChannelMask); 249 mixer->setParameter( 250 name, 251 AudioMixer::TRACK, 252 AudioMixer::CHANNEL_MASK, 253 (void *)(uintptr_t)channelMask); 254 mixer->setParameter( 255 name, 256 AudioMixer::RESAMPLE, 257 AudioMixer::SAMPLE_RATE, 258 (void *)(uintptr_t)Providers[i].getSampleRate()); 259 if (useRamp) { 260 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0); 261 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0); 262 mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME0, &f); 263 mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME1, &f); 264 } else { 265 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f); 266 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f); 267 } 268 if (auxFilename) { 269 mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::AUX_BUFFER, 270 (void *) auxAddr); 271 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::AUXLEVEL, &f0); 272 mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::AUXLEVEL, &f); 273 } 274 mixer->enable(name); 275 } 276 277 // pump the mixer to process data. 278 size_t i; 279 for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) { 280 for (size_t j = 0; j < Names.size(); ++j) { 281 mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, 282 (char *) outputAddr + i * outputFrameSize); 283 if (auxFilename) { 284 mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER, 285 (char *) auxAddr + i * auxFrameSize); 286 } 287 } 288 mixer->process(AudioBufferProvider::kInvalidPTS); 289 } 290 outputFrames = i; // reset output frames to the data actually produced. 291 292 // write to files 293 writeFile(outputFilename, outputAddr, 294 outputSampleRate, outputChannels, outputFrames, useMixerFloat); 295 if (auxFilename) { 296 // Aux buffer is always in q4_27 format for now. 297 // memcpy_to_i16_from_q4_27(), but with stereo frame count (not sample count) 298 ditherAndClamp((int32_t*)auxAddr, (int32_t*)auxAddr, outputFrames >> 1); 299 writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false); 300 } 301 302 delete mixer; 303 free(outputAddr); 304 free(auxAddr); 305 return EXIT_SUCCESS; 306 } 307
