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 <media/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
usage(const char * name)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 by default\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:[(i|f),]<channels>,<frequency>,<samplerate>'\n");
51 fprintf(stderr, " 'chirp:[(i|f),]<channels>,<samplerate>'\n");
52 }
53
writeFile(const char * filename,const void * buffer,uint32_t sampleRate,uint32_t channels,size_t frames,bool isBufferFloat)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
parseFormat(const char * s,bool * useFloat)81 const char *parseFormat(const char *s, bool *useFloat) {
82 if (!strncmp(s, "f,", 2)) {
83 *useFloat = true;
84 return s + 2;
85 }
86 if (!strncmp(s, "i,", 2)) {
87 *useFloat = false;
88 return s + 2;
89 }
90 return s;
91 }
92
main(int argc,char * argv[])93 int main(int argc, char* argv[]) {
94 const char* const progname = argv[0];
95 bool useInputFloat = false;
96 bool useMixerFloat = false;
97 bool useRamp = true;
98 uint32_t outputSampleRate = 48000;
99 uint32_t outputChannels = 2; // stereo for now
100 std::vector<int> Pvalues;
101 const char* outputFilename = NULL;
102 const char* auxFilename = NULL;
103 std::vector<int32_t> names;
104 std::vector<SignalProvider> providers;
105 std::vector<audio_format_t> formats;
106
107 for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
108 switch (ch) {
109 case 'f':
110 useInputFloat = true;
111 break;
112 case 'm':
113 useMixerFloat = true;
114 break;
115 case 'c':
116 outputChannels = atoi(optarg);
117 break;
118 case 's':
119 outputSampleRate = atoi(optarg);
120 break;
121 case 'o':
122 outputFilename = optarg;
123 break;
124 case 'a':
125 auxFilename = optarg;
126 break;
127 case 'P':
128 if (parseCSV(optarg, Pvalues) < 0) {
129 fprintf(stderr, "incorrect syntax for -P option\n");
130 return EXIT_FAILURE;
131 }
132 break;
133 case '?':
134 default:
135 usage(progname);
136 return EXIT_FAILURE;
137 }
138 }
139 argc -= optind;
140 argv += optind;
141
142 if (argc == 0) {
143 usage(progname);
144 return EXIT_FAILURE;
145 }
146
147 size_t outputFrames = 0;
148
149 // create providers for each track
150 names.resize(argc);
151 providers.resize(argc);
152 formats.resize(argc);
153 for (int i = 0; i < argc; ++i) {
154 static const char chirp[] = "chirp:";
155 static const char sine[] = "sine:";
156 static const double kSeconds = 1;
157 bool useFloat = useInputFloat;
158
159 if (!strncmp(argv[i], chirp, strlen(chirp))) {
160 std::vector<int> v;
161 const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat);
162
163 parseCSV(s, v);
164 if (v.size() == 2) {
165 printf("creating chirp(%d %d)\n", v[0], v[1]);
166 if (useFloat) {
167 providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
168 formats[i] = AUDIO_FORMAT_PCM_FLOAT;
169 } else {
170 providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
171 formats[i] = AUDIO_FORMAT_PCM_16_BIT;
172 }
173 providers[i].setIncr(Pvalues);
174 } else {
175 fprintf(stderr, "malformed input '%s'\n", argv[i]);
176 }
177 } else if (!strncmp(argv[i], sine, strlen(sine))) {
178 std::vector<int> v;
179 const char *s = parseFormat(argv[i] + strlen(sine), &useFloat);
180
181 parseCSV(s, v);
182 if (v.size() == 3) {
183 printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]);
184 if (useFloat) {
185 providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
186 formats[i] = AUDIO_FORMAT_PCM_FLOAT;
187 } else {
188 providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
189 formats[i] = AUDIO_FORMAT_PCM_16_BIT;
190 }
191 providers[i].setIncr(Pvalues);
192 } else {
193 fprintf(stderr, "malformed input '%s'\n", argv[i]);
194 }
195 } else {
196 printf("creating filename(%s)\n", argv[i]);
197 if (useInputFloat) {
198 providers[i].setFile<float>(argv[i]);
199 formats[i] = AUDIO_FORMAT_PCM_FLOAT;
200 } else {
201 providers[i].setFile<short>(argv[i]);
202 formats[i] = AUDIO_FORMAT_PCM_16_BIT;
203 }
204 providers[i].setIncr(Pvalues);
205 }
206 // calculate the number of output frames
207 size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate
208 / providers[i].getSampleRate();
209 if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
210 outputFrames = nframes;
211 }
212 }
213
214 // create the output buffer.
215 const size_t outputFrameSize = outputChannels
216 * (useMixerFloat ? sizeof(float) : sizeof(int16_t));
217 const size_t outputSize = outputFrames * outputFrameSize;
218 const audio_channel_mask_t outputChannelMask =
219 audio_channel_out_mask_from_count(outputChannels);
220 void *outputAddr = NULL;
221 (void) posix_memalign(&outputAddr, 32, outputSize);
222 memset(outputAddr, 0, outputSize);
223
224 // create the aux buffer, if needed.
225 const size_t auxFrameSize = sizeof(int32_t); // Q4.27 always
226 const size_t auxSize = outputFrames * auxFrameSize;
227 void *auxAddr = NULL;
228 if (auxFilename) {
229 (void) posix_memalign(&auxAddr, 32, auxSize);
230 memset(auxAddr, 0, auxSize);
231 }
232
233 // create the mixer.
234 const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
235 AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
236 audio_format_t mixerFormat = useMixerFloat
237 ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
238 float f = AudioMixer::UNITY_GAIN_FLOAT / providers.size(); // normalize volume by # tracks
239 static float f0; // zero
240
241 // set up the tracks.
242 for (size_t i = 0; i < providers.size(); ++i) {
243 //printf("track %d out of %d\n", i, providers.size());
244 uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels());
245 const int name = i;
246 const status_t status = mixer->create(
247 name, channelMask, formats[i], AUDIO_SESSION_OUTPUT_MIX);
248 LOG_ALWAYS_FATAL_IF(status != OK);
249 names[i] = name;
250 mixer->setBufferProvider(name, &providers[i]);
251 mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
252 (void *)outputAddr);
253 mixer->setParameter(
254 name,
255 AudioMixer::TRACK,
256 AudioMixer::MIXER_FORMAT,
257 (void *)(uintptr_t)mixerFormat);
258 mixer->setParameter(
259 name,
260 AudioMixer::TRACK,
261 AudioMixer::FORMAT,
262 (void *)(uintptr_t)formats[i]);
263 mixer->setParameter(
264 name,
265 AudioMixer::TRACK,
266 AudioMixer::MIXER_CHANNEL_MASK,
267 (void *)(uintptr_t)outputChannelMask);
268 mixer->setParameter(
269 name,
270 AudioMixer::TRACK,
271 AudioMixer::CHANNEL_MASK,
272 (void *)(uintptr_t)channelMask);
273 mixer->setParameter(
274 name,
275 AudioMixer::RESAMPLE,
276 AudioMixer::SAMPLE_RATE,
277 (void *)(uintptr_t)providers[i].getSampleRate());
278 if (useRamp) {
279 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
280 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
281 mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME0, &f);
282 mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME1, &f);
283 } else {
284 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
285 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
286 }
287 if (auxFilename) {
288 mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
289 (void *) auxAddr);
290 mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::AUXLEVEL, &f0);
291 mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::AUXLEVEL, &f);
292 }
293 mixer->enable(name);
294 }
295
296 // pump the mixer to process data.
297 size_t i;
298 for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
299 for (size_t j = 0; j < names.size(); ++j) {
300 mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
301 (char *) outputAddr + i * outputFrameSize);
302 if (auxFilename) {
303 mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
304 (char *) auxAddr + i * auxFrameSize);
305 }
306 }
307 mixer->process();
308 }
309 outputFrames = i; // reset output frames to the data actually produced.
310
311 // write to files
312 writeFile(outputFilename, outputAddr,
313 outputSampleRate, outputChannels, outputFrames, useMixerFloat);
314 if (auxFilename) {
315 // Aux buffer is always in q4_27 format for O and earlier.
316 // memcpy_to_i16_from_q4_27((int16_t*)auxAddr, (const int32_t*)auxAddr, outputFrames);
317 // Aux buffer is always in float format for P.
318 memcpy_to_i16_from_float((int16_t*)auxAddr, (const float*)auxAddr, outputFrames);
319 writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false);
320 }
321
322 delete mixer;
323 free(outputAddr);
324 free(auxAddr);
325 return EXIT_SUCCESS;
326 }
327