diff options
Diffstat (limited to 'services/audioflinger/tests/test-mixer.cpp')
-rw-r--r-- | services/audioflinger/tests/test-mixer.cpp | 286 |
1 files changed, 286 insertions, 0 deletions
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp new file mode 100644 index 0000000..3940702 --- /dev/null +++ b/services/audioflinger/tests/test-mixer.cpp @@ -0,0 +1,286 @@ +/* + * Copyright (C) 2014 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include <stdio.h> +#include <inttypes.h> +#include <math.h> +#include <vector> +#include <audio_utils/primitives.h> +#include <audio_utils/sndfile.h> +#include <media/AudioBufferProvider.h> +#include "AudioMixer.h" +#include "test_utils.h" + +/* Testing is typically through creation of an output WAV file from several + * source inputs, to be later analyzed by an audio program such as Audacity. + * + * Sine or chirp functions are typically more useful as input to the mixer + * as they show up as straight lines on a spectrogram if successfully mixed. + * + * A sample shell script is provided: mixer_to_wave_tests.sh + */ + +using namespace android; + +static void usage(const char* name) { + fprintf(stderr, "Usage: %s [-f] [-m]" + " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]" + " (<input-file> | <command>)+\n", name); + fprintf(stderr, " -f enable floating point input track\n"); + fprintf(stderr, " -m enable floating point mixer output\n"); + fprintf(stderr, " -s mixer sample-rate\n"); + fprintf(stderr, " -o <output-file> WAV file, pcm16 (or float if -m specified)\n"); + fprintf(stderr, " -a <aux-buffer-file>\n"); + fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n"); + fprintf(stderr, " <input-file> is a WAV file\n"); + fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n"); + fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n"); +} + +static int writeFile(const char *filename, const void *buffer, + uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) { + if (filename == NULL) { + return 0; // ok to pass in NULL filename + } + // write output to file. + SF_INFO info; + info.frames = 0; + info.samplerate = sampleRate; + info.channels = channels; + info.format = SF_FORMAT_WAV | (isBufferFloat ? SF_FORMAT_FLOAT : SF_FORMAT_PCM_16); + printf("saving file:%s channels:%d samplerate:%d frames:%d\n", + filename, info.channels, info.samplerate, frames); + SNDFILE *sf = sf_open(filename, SFM_WRITE, &info); + if (sf == NULL) { + perror(filename); + return EXIT_FAILURE; + } + if (isBufferFloat) { + (void) sf_writef_float(sf, (float*)buffer, frames); + } else { + (void) sf_writef_short(sf, (short*)buffer, frames); + } + sf_close(sf); + return EXIT_SUCCESS; +} + +int main(int argc, char* argv[]) { + const char* const progname = argv[0]; + bool useInputFloat = false; + bool useMixerFloat = false; + bool useRamp = true; + uint32_t outputSampleRate = 48000; + uint32_t outputChannels = 2; // stereo for now + std::vector<int> Pvalues; + const char* outputFilename = NULL; + const char* auxFilename = NULL; + std::vector<int32_t> Names; + std::vector<SignalProvider> Providers; + + for (int ch; (ch = getopt(argc, argv, "fms:o:a:P:")) != -1;) { + switch (ch) { + case 'f': + useInputFloat = true; + break; + case 'm': + useMixerFloat = true; + break; + case 's': + outputSampleRate = atoi(optarg); + break; + case 'o': + outputFilename = optarg; + break; + case 'a': + auxFilename = optarg; + break; + case 'P': + if (parseCSV(optarg, Pvalues) < 0) { + fprintf(stderr, "incorrect syntax for -P option\n"); + return EXIT_FAILURE; + } + break; + case '?': + default: + usage(progname); + return EXIT_FAILURE; + } + } + argc -= optind; + argv += optind; + + if (argc == 0) { + usage(progname); + return EXIT_FAILURE; + } + if ((unsigned)argc > AudioMixer::MAX_NUM_TRACKS) { + fprintf(stderr, "too many tracks: %d > %u", argc, AudioMixer::MAX_NUM_TRACKS); + return EXIT_FAILURE; + } + + size_t outputFrames = 0; + + // create providers for each track + Providers.resize(argc); + for (int i = 0; i < argc; ++i) { + static const char chirp[] = "chirp:"; + static const char sine[] = "sine:"; + static const double kSeconds = 1; + + if (!strncmp(argv[i], chirp, strlen(chirp))) { + std::vector<int> v; + + parseCSV(argv[i] + strlen(chirp), v); + if (v.size() == 2) { + printf("creating chirp(%d %d)\n", v[0], v[1]); + if (useInputFloat) { + Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds); + } else { + Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds); + } + Providers[i].setIncr(Pvalues); + } else { + fprintf(stderr, "malformed input '%s'\n", argv[i]); + } + } else if (!strncmp(argv[i], sine, strlen(sine))) { + std::vector<int> v; + + parseCSV(argv[i] + strlen(sine), v); + if (v.size() == 3) { + printf("creating sine(%d %d)\n", v[0], v[1]); + if (useInputFloat) { + Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds); + } else { + Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds); + } + Providers[i].setIncr(Pvalues); + } else { + fprintf(stderr, "malformed input '%s'\n", argv[i]); + } + } else { + printf("creating filename(%s)\n", argv[i]); + if (useInputFloat) { + Providers[i].setFile<float>(argv[i]); + } else { + Providers[i].setFile<short>(argv[i]); + } + Providers[i].setIncr(Pvalues); + } + // calculate the number of output frames + size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate + / Providers[i].getSampleRate(); + if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames + outputFrames = nframes; + } + } + + // create the output buffer. + const size_t outputFrameSize = outputChannels + * (useMixerFloat ? sizeof(float) : sizeof(int16_t)); + const size_t outputSize = outputFrames * outputFrameSize; + void *outputAddr = NULL; + (void) posix_memalign(&outputAddr, 32, outputSize); + memset(outputAddr, 0, outputSize); + + // create the aux buffer, if needed. + const size_t auxFrameSize = sizeof(int32_t); // Q4.27 always + const size_t auxSize = outputFrames * auxFrameSize; + void *auxAddr = NULL; + if (auxFilename) { + (void) posix_memalign(&auxAddr, 32, auxSize); + memset(auxAddr, 0, auxSize); + } + + // create the mixer. + const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960 + AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate); + audio_format_t inputFormat = useInputFloat + ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; + audio_format_t mixerFormat = useMixerFloat + ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; + float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks + static float f0; // zero + + // set up the tracks. + for (size_t i = 0; i < Providers.size(); ++i) { + //printf("track %d out of %d\n", i, Providers.size()); + uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels()); + int32_t name = mixer->getTrackName(channelMask, + inputFormat, AUDIO_SESSION_OUTPUT_MIX); + ALOG_ASSERT(name >= 0); + Names.push_back(name); + mixer->setBufferProvider(name, &Providers[i]); + mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, + (void *) outputAddr); + mixer->setParameter( + name, + AudioMixer::TRACK, + AudioMixer::MIXER_FORMAT, (void *)mixerFormat); + mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT, + (void *)(uintptr_t)inputFormat); + mixer->setParameter( + name, + AudioMixer::RESAMPLE, + AudioMixer::SAMPLE_RATE, + (void *)(uintptr_t)Providers[i].getSampleRate()); + if (useRamp) { + mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0); + mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0); + mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME0, &f); + mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME1, &f); + } else { + mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f); + mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f); + } + if (auxFilename) { + mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::AUX_BUFFER, + (void *) auxAddr); + mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::AUXLEVEL, &f0); + mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::AUXLEVEL, &f); + } + mixer->enable(name); + } + + // pump the mixer to process data. + size_t i; + for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) { + for (size_t j = 0; j < Names.size(); ++j) { + mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, + (char *) outputAddr + i * outputFrameSize); + if (auxFilename) { + mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER, + (char *) auxAddr + i * auxFrameSize); + } + } + mixer->process(AudioBufferProvider::kInvalidPTS); + } + outputFrames = i; // reset output frames to the data actually produced. + + // write to files + writeFile(outputFilename, outputAddr, + outputSampleRate, outputChannels, outputFrames, useMixerFloat); + if (auxFilename) { + // Aux buffer is always in q4_27 format for now. + // memcpy_to_i16_from_q4_27(), but with stereo frame count (not sample count) + ditherAndClamp((int32_t*)auxAddr, (int32_t*)auxAddr, outputFrames >> 1); + writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false); + } + + delete mixer; + free(outputAddr); + free(auxAddr); + return EXIT_SUCCESS; +} |