diff options
Diffstat (limited to 'services/audioflinger/tests')
| -rw-r--r-- | services/audioflinger/tests/Android.mk | 73 | ||||
| -rwxr-xr-x | services/audioflinger/tests/build_and_run_all_unit_tests.sh | 22 | ||||
| -rwxr-xr-x | services/audioflinger/tests/mixer_to_wav_tests.sh | 134 | ||||
| -rw-r--r-- | services/audioflinger/tests/resampler_tests.cpp | 280 | ||||
| -rwxr-xr-x | services/audioflinger/tests/run_all_unit_tests.sh | 11 | ||||
| -rw-r--r-- | services/audioflinger/tests/test-mixer.cpp | 286 | ||||
| -rw-r--r-- | services/audioflinger/tests/test_utils.h | 307 |
7 files changed, 1113 insertions, 0 deletions
diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk new file mode 100644 index 0000000..7bba05b --- /dev/null +++ b/services/audioflinger/tests/Android.mk @@ -0,0 +1,73 @@ +# Build the unit tests for audioflinger + +# +# resampler unit test +# +LOCAL_PATH:= $(call my-dir) +include $(CLEAR_VARS) + +LOCAL_SHARED_LIBRARIES := \ + liblog \ + libutils \ + libcutils \ + libstlport \ + libaudioutils \ + libaudioresampler + +LOCAL_STATIC_LIBRARIES := \ + libgtest \ + libgtest_main + +LOCAL_C_INCLUDES := \ + bionic \ + bionic/libstdc++/include \ + external/gtest/include \ + external/stlport/stlport \ + $(call include-path-for, audio-utils) \ + frameworks/av/services/audioflinger + +LOCAL_SRC_FILES := \ + resampler_tests.cpp + +LOCAL_MODULE := resampler_tests +LOCAL_MODULE_TAGS := tests + +include $(BUILD_EXECUTABLE) + +# +# audio mixer test tool +# +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + test-mixer.cpp \ + ../AudioMixer.cpp.arm \ + +LOCAL_C_INCLUDES := \ + bionic \ + bionic/libstdc++/include \ + external/stlport/stlport \ + $(call include-path-for, audio-effects) \ + $(call include-path-for, audio-utils) \ + frameworks/av/services/audioflinger + +LOCAL_STATIC_LIBRARIES := \ + libsndfile + +LOCAL_SHARED_LIBRARIES := \ + libstlport \ + libeffects \ + libnbaio \ + libcommon_time_client \ + libaudioresampler \ + libaudioutils \ + libdl \ + libcutils \ + libutils \ + liblog + +LOCAL_MODULE:= test-mixer + +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) diff --git a/services/audioflinger/tests/build_and_run_all_unit_tests.sh b/services/audioflinger/tests/build_and_run_all_unit_tests.sh new file mode 100755 index 0000000..2c453b0 --- /dev/null +++ b/services/audioflinger/tests/build_and_run_all_unit_tests.sh @@ -0,0 +1,22 @@ +#!/bin/bash + +if [ -z "$ANDROID_BUILD_TOP" ]; then + echo "Android build environment not set" + exit -1 +fi + +# ensure we have mm +. $ANDROID_BUILD_TOP/build/envsetup.sh + +pushd $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/ +pwd +mm + +echo "waiting for device" +adb root && adb wait-for-device remount +adb push $OUT/system/lib/libaudioresampler.so /system/lib +adb push $OUT/system/bin/resampler_tests /system/bin + +sh $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/tests/run_all_unit_tests.sh + +popd diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh new file mode 100755 index 0000000..93bff47 --- /dev/null +++ b/services/audioflinger/tests/mixer_to_wav_tests.sh @@ -0,0 +1,134 @@ +#!/bin/bash +# +# This script uses test-mixer to generate WAV files +# for evaluation of the AudioMixer component. +# +# Sine and chirp signals are used for input because they +# show up as clear lines, either horizontal or diagonal, +# on a spectrogram. This means easy verification of multiple +# track mixing. +# +# After execution, look for created subdirectories like +# mixer_i_i +# mixer_i_f +# mixer_f_f +# +# Recommend using a program such as audacity to evaluate +# the output WAV files, e.g. +# +# cd testdir +# audacity *.wav +# +# Using Audacity: +# +# Under "Waveform" view mode you can zoom into the +# start of the WAV file to verify proper ramping. +# +# Select "Spectrogram" to see verify the lines +# (sine = horizontal, chirp = diagonal) which should +# be clear (except for around the start as the volume +# ramping causes spectral distortion). + +if [ -z "$ANDROID_BUILD_TOP" ]; then + echo "Android build environment not set" + exit -1 +fi + +# ensure we have mm +. $ANDROID_BUILD_TOP/build/envsetup.sh + +pushd $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/ + +# build +pwd +mm + +# send to device +echo "waiting for device" +adb root && adb wait-for-device remount +adb push $OUT/system/lib/libaudioresampler.so /system/lib +adb push $OUT/system/bin/test-mixer /system/bin + +# createwav creates a series of WAV files testing various +# mixer settings +# $1 = flags +# $2 = directory +function createwav() { +# create directory if it doesn't exist + if [ ! -d $2 ]; then + mkdir $2 + fi + +# Test: +# process__genericResampling +# track__Resample / track__genericResample + adb shell test-mixer $1 -s 48000 \ + -o /sdcard/tm48000gr.wav \ + sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 + adb pull /sdcard/tm48000gr.wav $2 + +# Test: +# process__genericResample +# track__Resample / track__genericResample +# track__NoResample / track__16BitsStereo / track__16BitsMono +# Aux buffer + adb shell test-mixer $1 -s 9307 \ + -a /sdcard/aux9307gra.wav -o /sdcard/tm9307gra.wav \ + sine:2,1000,3000 sine:1,2000,9307 chirp:2,9307 + adb pull /sdcard/tm9307gra.wav $2 + adb pull /sdcard/aux9307gra.wav $2 + +# Test: +# process__genericNoResampling +# track__NoResample / track__16BitsStereo / track__16BitsMono + adb shell test-mixer $1 -s 32000 \ + -o /sdcard/tm32000gnr.wav \ + sine:2,1000,32000 chirp:2,32000 sine:1,3000,32000 + adb pull /sdcard/tm32000gnr.wav $2 + +# Test: +# process__genericNoResampling +# track__NoResample / track__16BitsStereo / track__16BitsMono +# Aux buffer + adb shell test-mixer $1 -s 32000 \ + -a /sdcard/aux32000gnra.wav -o /sdcard/tm32000gnra.wav \ + sine:2,1000,32000 chirp:2,32000 sine:1,3000,32000 + adb pull /sdcard/tm32000gnra.wav $2 + adb pull /sdcard/aux32000gnra.wav $2 + +# Test: +# process__NoResampleOneTrack / process__OneTrack16BitsStereoNoResampling +# Downmixer + adb shell test-mixer $1 -s 32000 \ + -o /sdcard/tm32000nrot.wav \ + sine:6,1000,32000 + adb pull /sdcard/tm32000nrot.wav $2 + +# Test: +# process__NoResampleOneTrack / OneTrack16BitsStereoNoResampling +# Aux buffer + adb shell test-mixer $1 -s 44100 \ + -a /sdcard/aux44100nrota.wav -o /sdcard/tm44100nrota.wav \ + sine:2,2000,44100 + adb pull /sdcard/tm44100nrota.wav $2 + adb pull /sdcard/aux44100nrota.wav $2 +} + +# +# Call createwav to generate WAV files in various combinations +# +# i_i = integer input track, integer mixer output +# f_f = float input track, float mixer output +# i_f = integer input track, float_mixer output +# +# If the mixer output is float, then the output WAV file is pcm float. +# +# TODO: create a "snr" like "diff" to automatically +# compare files in these directories together. +# + +createwav "" "tests/mixer_i_i" +createwav "-f -m" "tests/mixer_f_f" +createwav "-m" "tests/mixer_i_f" + +popd diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp new file mode 100644 index 0000000..4a67d0b --- /dev/null +++ b/services/audioflinger/tests/resampler_tests.cpp @@ -0,0 +1,280 @@ +/* + * 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. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "audioflinger_resampler_tests" + +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <fcntl.h> +#include <string.h> +#include <sys/mman.h> +#include <sys/stat.h> +#include <errno.h> +#include <time.h> +#include <math.h> +#include <vector> +#include <utility> +#include <cutils/log.h> +#include <gtest/gtest.h> +#include <media/AudioBufferProvider.h> +#include "AudioResampler.h" +#include "test_utils.h" + +void resample(void *output, size_t outputFrames, const std::vector<size_t> &outputIncr, + android::AudioBufferProvider *provider, android::AudioResampler *resampler) +{ + for (size_t i = 0, j = 0; i < outputFrames; ) { + size_t thisFrames = outputIncr[j++]; + if (j >= outputIncr.size()) { + j = 0; + } + if (thisFrames == 0 || thisFrames > outputFrames - i) { + thisFrames = outputFrames - i; + } + resampler->resample((int32_t*) output + 2*i, thisFrames, provider); + i += thisFrames; + } +} + +void buffercmp(const void *reference, const void *test, + size_t outputFrameSize, size_t outputFrames) +{ + for (size_t i = 0; i < outputFrames; ++i) { + int check = memcmp((const char*)reference + i * outputFrameSize, + (const char*)test + i * outputFrameSize, outputFrameSize); + if (check) { + ALOGE("Failure at frame %d", i); + ASSERT_EQ(check, 0); /* fails */ + } + } +} + +void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFreq, + enum android::AudioResampler::src_quality quality) +{ + // create the provider + std::vector<int> inputIncr; + SignalProvider provider; + provider.setChirp<int16_t>(channels, + 0., outputFreq/2., outputFreq, outputFreq/2000.); + provider.setIncr(inputIncr); + + // calculate the output size + size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq; + size_t outputFrameSize = 2 * sizeof(int32_t); + size_t outputSize = outputFrameSize * outputFrames; + outputSize &= ~7; + + // create the resampler + const int volumePrecision = 12; /* typical unity gain */ + android::AudioResampler* resampler; + + resampler = android::AudioResampler::create(16, channels, outputFreq, quality); + resampler->setSampleRate(inputFreq); + resampler->setVolume(1 << volumePrecision, 1 << volumePrecision); + + // set up the reference run + std::vector<size_t> refIncr; + refIncr.push_back(outputFrames); + void* reference = malloc(outputSize); + resample(reference, outputFrames, refIncr, &provider, resampler); + + provider.reset(); + +#if 0 + /* this test will fail - API interface issue: reset() does not clear internal buffers */ + resampler->reset(); +#else + delete resampler; + resampler = android::AudioResampler::create(16, channels, outputFreq, quality); + resampler->setSampleRate(inputFreq); + resampler->setVolume(1 << volumePrecision, 1 << volumePrecision); +#endif + + // set up the test run + std::vector<size_t> outIncr; + outIncr.push_back(1); + outIncr.push_back(2); + outIncr.push_back(3); + void* test = malloc(outputSize); + resample(test, outputFrames, outIncr, &provider, resampler); + + // check + buffercmp(reference, test, outputFrameSize, outputFrames); + + free(reference); + free(test); + delete resampler; +} + +template <typename T> +inline double sqr(T v) +{ + double dv = static_cast<double>(v); + return dv * dv; +} + +template <typename T> +double signalEnergy(T *start, T *end, unsigned stride) +{ + double accum = 0; + + for (T *p = start; p < end; p += stride) { + accum += sqr(*p); + } + unsigned count = (end - start + stride - 1) / stride; + return accum / count; +} + +void testStopbandDownconversion(size_t channels, + unsigned inputFreq, unsigned outputFreq, + unsigned passband, unsigned stopband, + enum android::AudioResampler::src_quality quality) +{ + // create the provider + std::vector<int> inputIncr; + SignalProvider provider; + provider.setChirp<int16_t>(channels, + 0., inputFreq/2., inputFreq, inputFreq/2000.); + provider.setIncr(inputIncr); + + // calculate the output size + size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq; + size_t outputFrameSize = 2 * sizeof(int32_t); + size_t outputSize = outputFrameSize * outputFrames; + outputSize &= ~7; + + // create the resampler + const int volumePrecision = 12; /* typical unity gain */ + android::AudioResampler* resampler; + + resampler = android::AudioResampler::create(16, channels, outputFreq, quality); + resampler->setSampleRate(inputFreq); + resampler->setVolume(1 << volumePrecision, 1 << volumePrecision); + + // set up the reference run + std::vector<size_t> refIncr; + refIncr.push_back(outputFrames); + void* reference = malloc(outputSize); + resample(reference, outputFrames, refIncr, &provider, resampler); + + int32_t *out = reinterpret_cast<int32_t *>(reference); + + // check signal energy in passband + const unsigned passbandFrame = passband * outputFreq / 1000.; + const unsigned stopbandFrame = stopband * outputFreq / 1000.; + + // check each channel separately + for (size_t i = 0; i < channels; ++i) { + double passbandEnergy = signalEnergy(out, out + passbandFrame * channels, channels); + double stopbandEnergy = signalEnergy(out + stopbandFrame * channels, + out + outputFrames * channels, channels); + double dbAtten = -10. * log10(stopbandEnergy / passbandEnergy); + ASSERT_GT(dbAtten, 60.); + +#if 0 + // internal verification + printf("if:%d of:%d pbf:%d sbf:%d sbe: %f pbe: %f db: %.2f\n", + provider.getNumFrames(), outputFrames, + passbandFrame, stopbandFrame, stopbandEnergy, passbandEnergy, dbAtten); + for (size_t i = 0; i < 10; ++i) { + printf("%d\n", out[i+passbandFrame*channels]); + } + for (size_t i = 0; i < 10; ++i) { + printf("%d\n", out[i+stopbandFrame*channels]); + } +#endif + } + + free(reference); + delete resampler; +} + +/* Buffer increment test + * + * We compare a reference output, where we consume and process the entire + * buffer at a time, and a test output, where we provide small chunks of input + * data and process small chunks of output (which may not be equivalent in size). + * + * Two subtests - fixed phase (3:2 down) and interpolated phase (147:320 up) + */ +TEST(audioflinger_resampler, bufferincrement_fixedphase) { + // all of these work + static const enum android::AudioResampler::src_quality kQualityArray[] = { + android::AudioResampler::LOW_QUALITY, + android::AudioResampler::MED_QUALITY, + android::AudioResampler::HIGH_QUALITY, + android::AudioResampler::VERY_HIGH_QUALITY, + android::AudioResampler::DYN_LOW_QUALITY, + android::AudioResampler::DYN_MED_QUALITY, + android::AudioResampler::DYN_HIGH_QUALITY, + }; + + for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { + testBufferIncrement(2, 48000, 32000, kQualityArray[i]); + } +} + +TEST(audioflinger_resampler, bufferincrement_interpolatedphase) { + // all of these work except low quality + static const enum android::AudioResampler::src_quality kQualityArray[] = { +// android::AudioResampler::LOW_QUALITY, + android::AudioResampler::MED_QUALITY, + android::AudioResampler::HIGH_QUALITY, + android::AudioResampler::VERY_HIGH_QUALITY, + android::AudioResampler::DYN_LOW_QUALITY, + android::AudioResampler::DYN_MED_QUALITY, + android::AudioResampler::DYN_HIGH_QUALITY, + }; + + for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { + testBufferIncrement(2, 22050, 48000, kQualityArray[i]); + } +} + +/* Simple aliasing test + * + * This checks stopband response of the chirp signal to make sure frequencies + * are properly suppressed. It uses downsampling because the stopband can be + * clearly isolated by input frequencies exceeding the output sample rate (nyquist). + */ +TEST(audioflinger_resampler, stopbandresponse) { + // not all of these may work (old resamplers fail on downsampling) + static const enum android::AudioResampler::src_quality kQualityArray[] = { + //android::AudioResampler::LOW_QUALITY, + //android::AudioResampler::MED_QUALITY, + //android::AudioResampler::HIGH_QUALITY, + //android::AudioResampler::VERY_HIGH_QUALITY, + android::AudioResampler::DYN_LOW_QUALITY, + android::AudioResampler::DYN_MED_QUALITY, + android::AudioResampler::DYN_HIGH_QUALITY, + }; + + // in this test we assume a maximum transition band between 12kHz and 20kHz. + // there must be at least 60dB relative attenuation between stopband and passband. + for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { + testStopbandDownconversion(2, 48000, 32000, 12000, 20000, kQualityArray[i]); + } + + // in this test we assume a maximum transition band between 7kHz and 15kHz. + // there must be at least 60dB relative attenuation between stopband and passband. + // (the weird ratio triggers interpolative resampling) + for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) { + testStopbandDownconversion(2, 48000, 22101, 7000, 15000, kQualityArray[i]); + } +} diff --git a/services/audioflinger/tests/run_all_unit_tests.sh b/services/audioflinger/tests/run_all_unit_tests.sh new file mode 100755 index 0000000..ffae6ae --- /dev/null +++ b/services/audioflinger/tests/run_all_unit_tests.sh @@ -0,0 +1,11 @@ +#!/bin/bash + +if [ -z "$ANDROID_BUILD_TOP" ]; then + echo "Android build environment not set" + exit -1 +fi + +echo "waiting for device" +adb root && adb wait-for-device remount + +adb shell /system/bin/resampler_tests 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; +} diff --git a/services/audioflinger/tests/test_utils.h b/services/audioflinger/tests/test_utils.h new file mode 100644 index 0000000..f954292 --- /dev/null +++ b/services/audioflinger/tests/test_utils.h @@ -0,0 +1,307 @@ +/* + * 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. + */ + +#ifndef ANDROID_AUDIO_TEST_UTILS_H +#define ANDROID_AUDIO_TEST_UTILS_H + +#include <audio_utils/sndfile.h> + +#ifndef ARRAY_SIZE +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) +#endif + +template<typename T, typename U> +struct is_same +{ + static const bool value = false; +}; + +template<typename T> +struct is_same<T, T> // partial specialization +{ + static const bool value = true; +}; + +template<typename T> +static inline T convertValue(double val) +{ + if (is_same<T, int16_t>::value) { + return floor(val * 32767.0 + 0.5); + } else if (is_same<T, int32_t>::value) { + return floor(val * (1UL<<31) + 0.5); + } + return val; // assume float or double +} + +// Convert a list of integers in CSV format to a Vector of those values. +// Returns the number of elements in the list, or -1 on error. +static inline int parseCSV(const char *string, std::vector<int>& values) +{ + // pass 1: count the number of values and do syntax check + size_t numValues = 0; + bool hadDigit = false; + for (const char *p = string; ; ) { + switch (*p++) { + case '0': case '1': case '2': case '3': case '4': + case '5': case '6': case '7': case '8': case '9': + hadDigit = true; + break; + case '\0': + if (hadDigit) { + // pass 2: allocate and initialize vector of values + values.resize(++numValues); + values[0] = atoi(p = string); + for (size_t i = 1; i < numValues; ) { + if (*p++ == ',') { + values[i++] = atoi(p); + } + } + return numValues; + } + // fall through + case ',': + if (hadDigit) { + hadDigit = false; + numValues++; + break; + } + // fall through + default: + return -1; + } + } +} + +/* Creates a type-independent audio buffer provider from + * a buffer base address, size, framesize, and input increment array. + * + * No allocation or deallocation of the provided buffer is done. + */ +class TestProvider : public android::AudioBufferProvider { +public: + TestProvider(void* addr, size_t frames, size_t frameSize, + const std::vector<int>& inputIncr) + : mAddr(addr), + mNumFrames(frames), + mFrameSize(frameSize), + mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0) + { + } + + TestProvider() + : mAddr(NULL), mNumFrames(0), mFrameSize(0), + mNextFrame(0), mUnrel(0), mNextIdx(0) + { + } + + void setIncr(const std::vector<int>& inputIncr) { + mInputIncr = inputIncr; + mNextIdx = 0; + } + + virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS) + { + size_t requestedFrames = buffer->frameCount; + if (requestedFrames > mNumFrames - mNextFrame) { + buffer->frameCount = mNumFrames - mNextFrame; + } + if (!mInputIncr.empty()) { + size_t provided = mInputIncr[mNextIdx++]; + ALOGV("getNextBuffer() mValue[%d]=%u not %u", + mNextIdx-1, provided, buffer->frameCount); + if (provided < buffer->frameCount) { + buffer->frameCount = provided; + } + if (mNextIdx >= mInputIncr.size()) { + mNextIdx = 0; + } + } + ALOGV("getNextBuffer() requested %u frames out of %u frames available" + " and returned %u frames\n", + requestedFrames, mNumFrames - mNextFrame, buffer->frameCount); + mUnrel = buffer->frameCount; + if (buffer->frameCount > 0) { + buffer->raw = (char *)mAddr + mFrameSize * mNextFrame; + return android::NO_ERROR; + } else { + buffer->raw = NULL; + return android::NOT_ENOUGH_DATA; + } + } + + virtual void releaseBuffer(Buffer* buffer) + { + if (buffer->frameCount > mUnrel) { + ALOGE("releaseBuffer() released %u frames but only %u available " + "to release\n", buffer->frameCount, mUnrel); + mNextFrame += mUnrel; + mUnrel = 0; + } else { + + ALOGV("releaseBuffer() released %u frames out of %u frames available " + "to release\n", buffer->frameCount, mUnrel); + mNextFrame += buffer->frameCount; + mUnrel -= buffer->frameCount; + } + buffer->frameCount = 0; + buffer->raw = NULL; + } + + void reset() + { + mNextFrame = 0; + } + + size_t getNumFrames() + { + return mNumFrames; + } + + +protected: + void* mAddr; // base address + size_t mNumFrames; // total frames + int mFrameSize; // frame size (# channels * bytes per sample) + size_t mNextFrame; // index of next frame to provide + size_t mUnrel; // number of frames not yet released + std::vector<int> mInputIncr; // number of frames provided per call + size_t mNextIdx; // index of next entry in mInputIncr to use +}; + +/* Creates a buffer filled with a sine wave. + */ +template<typename T> +static void createSine(void *vbuffer, size_t frames, + size_t channels, double sampleRate, double freq) +{ + double tscale = 1. / sampleRate; + T* buffer = reinterpret_cast<T*>(vbuffer); + for (size_t i = 0; i < frames; ++i) { + double t = i * tscale; + double y = sin(2. * M_PI * freq * t); + T yt = convertValue<T>(y); + + for (size_t j = 0; j < channels; ++j) { + buffer[i*channels + j] = yt / (j + 1); + } + } +} + +/* Creates a buffer filled with a chirp signal (a sine wave sweep). + * + * When creating the Chirp, note that the frequency is the true sinusoidal + * frequency not the sampling rate. + * + * http://en.wikipedia.org/wiki/Chirp + */ +template<typename T> +static void createChirp(void *vbuffer, size_t frames, + size_t channels, double sampleRate, double minfreq, double maxfreq) +{ + double tscale = 1. / sampleRate; + T *buffer = reinterpret_cast<T*>(vbuffer); + // note the chirp constant k has a divide-by-two. + double k = (maxfreq - minfreq) / (2. * tscale * frames); + for (size_t i = 0; i < frames; ++i) { + double t = i * tscale; + double y = sin(2. * M_PI * (k * t + minfreq) * t); + T yt = convertValue<T>(y); + + for (size_t j = 0; j < channels; ++j) { + buffer[i*channels + j] = yt / (j + 1); + } + } +} + +/* This derived class creates a buffer provider of datatype T, + * consisting of an input signal, e.g. from createChirp(). + * The number of frames can be obtained from the base class + * TestProvider::getNumFrames(). + */ + +class SignalProvider : public TestProvider { +public: + SignalProvider() + : mSampleRate(0), + mChannels(0) + { + } + + virtual ~SignalProvider() + { + free(mAddr); + mAddr = NULL; + } + + template <typename T> + void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time) + { + createBufferByFrames<T>(channels, sampleRate, sampleRate*time); + createChirp<T>(mAddr, mNumFrames, mChannels, mSampleRate, minfreq, maxfreq); + } + + template <typename T> + void setSine(size_t channels, + double freq, double sampleRate, double time) + { + createBufferByFrames<T>(channels, sampleRate, sampleRate*time); + createSine<T>(mAddr, mNumFrames, mChannels, mSampleRate, freq); + } + + template <typename T> + void setFile(const char *file_in) + { + SF_INFO info; + info.format = 0; + SNDFILE *sf = sf_open(file_in, SFM_READ, &info); + if (sf == NULL) { + perror(file_in); + return; + } + createBufferByFrames<T>(info.channels, info.samplerate, info.frames); + if (is_same<T, float>::value) { + (void) sf_readf_float(sf, (float *) mAddr, mNumFrames); + } else if (is_same<T, short>::value) { + (void) sf_readf_short(sf, (short *) mAddr, mNumFrames); + } + sf_close(sf); + } + + template <typename T> + void createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames) + { + mNumFrames = frames; + mChannels = channels; + mFrameSize = mChannels * sizeof(T); + free(mAddr); + mAddr = malloc(mFrameSize * mNumFrames); + mSampleRate = sampleRate; + } + + uint32_t getSampleRate() const { + return mSampleRate; + } + + uint32_t getNumChannels() const { + return mChannels; + } + +protected: + uint32_t mSampleRate; + uint32_t mChannels; +}; + +#endif // ANDROID_AUDIO_TEST_UTILS_H |