test-mixer.cpp revision a4daf0b4f934b800a49f199fb8c09409391c8fc0 
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
36using namespace android;
37
38static 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
54static 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
81int 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