AudioFlinger: Split off audio processing library
Test: native AudioResampler test, general playback test
Bug: 31015569
Change-Id: Ifb248f4402a583438d756c014dcd7a4577aef713
diff --git a/media/libaudioprocessing/tests/test-resampler.cpp b/media/libaudioprocessing/tests/test-resampler.cpp
new file mode 100644
index 0000000..fbc9326
--- /dev/null
+++ b/media/libaudioprocessing/tests/test-resampler.cpp
@@ -0,0 +1,515 @@
+/*
+ * Copyright (C) 2012 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 <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 <inttypes.h>
+#include <time.h>
+#include <math.h>
+#include <audio_utils/primitives.h>
+#include <audio_utils/sndfile.h>
+#include <utils/Vector.h>
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+
+using namespace android;
+
+static bool gVerbose = false;
+
+static int usage(const char* name) {
+ fprintf(stderr,"Usage: %s [-p] [-f] [-F] [-v] [-c channels]"
+ " [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
+ " [-i input-sample-rate] [-o output-sample-rate]"
+ " [-O csv] [-P csv] [<input-file>]"
+ " <output-file>\n", name);
+ fprintf(stderr," -p enable profiling\n");
+ fprintf(stderr," -f enable filter profiling\n");
+ fprintf(stderr," -F enable floating point -q {dlq|dmq|dhq} only");
+ fprintf(stderr," -v verbose : log buffer provider calls\n");
+ fprintf(stderr," -c # channels (1-2 for lq|mq|hq; 1-8 for dlq|dmq|dhq)\n");
+ fprintf(stderr," -q resampler quality\n");
+ fprintf(stderr," dq : default quality\n");
+ fprintf(stderr," lq : low quality\n");
+ fprintf(stderr," mq : medium quality\n");
+ fprintf(stderr," hq : high quality\n");
+ fprintf(stderr," vhq : very high quality\n");
+ fprintf(stderr," dlq : dynamic low quality\n");
+ fprintf(stderr," dmq : dynamic medium quality\n");
+ fprintf(stderr," dhq : dynamic high quality\n");
+ fprintf(stderr," -i input file sample rate (ignored if input file is specified)\n");
+ fprintf(stderr," -o output file sample rate\n");
+ fprintf(stderr," -O # frames output per call to resample() in CSV format\n");
+ fprintf(stderr," -P # frames provided per call to resample() in CSV format\n");
+ return -1;
+}
+
+// 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.
+int parseCSV(const char *string, 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.editItemAt(0) = atoi(p = optarg);
+ for (size_t i = 1; i < numValues; ) {
+ if (*p++ == ',') {
+ values.editItemAt(i++) = atoi(p);
+ }
+ }
+ return numValues;
+ }
+ // fall through
+ case ',':
+ if (hadDigit) {
+ hadDigit = false;
+ numValues++;
+ break;
+ }
+ // fall through
+ default:
+ return -1;
+ }
+ }
+}
+
+int main(int argc, char* argv[]) {
+ const char* const progname = argv[0];
+ bool profileResample = false;
+ bool profileFilter = false;
+ bool useFloat = false;
+ int channels = 1;
+ int input_freq = 0;
+ int output_freq = 0;
+ AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
+ Vector<int> Ovalues;
+ Vector<int> Pvalues;
+
+ int ch;
+ while ((ch = getopt(argc, argv, "pfFvc:q:i:o:O:P:")) != -1) {
+ switch (ch) {
+ case 'p':
+ profileResample = true;
+ break;
+ case 'f':
+ profileFilter = true;
+ break;
+ case 'F':
+ useFloat = true;
+ break;
+ case 'v':
+ gVerbose = true;
+ break;
+ case 'c':
+ channels = atoi(optarg);
+ break;
+ case 'q':
+ if (!strcmp(optarg, "dq"))
+ quality = AudioResampler::DEFAULT_QUALITY;
+ else if (!strcmp(optarg, "lq"))
+ quality = AudioResampler::LOW_QUALITY;
+ else if (!strcmp(optarg, "mq"))
+ quality = AudioResampler::MED_QUALITY;
+ else if (!strcmp(optarg, "hq"))
+ quality = AudioResampler::HIGH_QUALITY;
+ else if (!strcmp(optarg, "vhq"))
+ quality = AudioResampler::VERY_HIGH_QUALITY;
+ else if (!strcmp(optarg, "dlq"))
+ quality = AudioResampler::DYN_LOW_QUALITY;
+ else if (!strcmp(optarg, "dmq"))
+ quality = AudioResampler::DYN_MED_QUALITY;
+ else if (!strcmp(optarg, "dhq"))
+ quality = AudioResampler::DYN_HIGH_QUALITY;
+ else {
+ usage(progname);
+ return -1;
+ }
+ break;
+ case 'i':
+ input_freq = atoi(optarg);
+ break;
+ case 'o':
+ output_freq = atoi(optarg);
+ break;
+ case 'O':
+ if (parseCSV(optarg, Ovalues) < 0) {
+ fprintf(stderr, "incorrect syntax for -O option\n");
+ return -1;
+ }
+ break;
+ case 'P':
+ if (parseCSV(optarg, Pvalues) < 0) {
+ fprintf(stderr, "incorrect syntax for -P option\n");
+ return -1;
+ }
+ break;
+ case '?':
+ default:
+ usage(progname);
+ return -1;
+ }
+ }
+
+ if (channels < 1
+ || channels > (quality < AudioResampler::DYN_LOW_QUALITY ? 2 : 8)) {
+ fprintf(stderr, "invalid number of audio channels %d\n", channels);
+ return -1;
+ }
+ if (useFloat && quality < AudioResampler::DYN_LOW_QUALITY) {
+ fprintf(stderr, "float processing is only possible for dynamic resamplers\n");
+ return -1;
+ }
+
+ argc -= optind;
+ argv += optind;
+
+ const char* file_in = NULL;
+ const char* file_out = NULL;
+ if (argc == 1) {
+ file_out = argv[0];
+ } else if (argc == 2) {
+ file_in = argv[0];
+ file_out = argv[1];
+ } else {
+ usage(progname);
+ return -1;
+ }
+
+ // ----------------------------------------------------------
+
+ size_t input_size;
+ void* input_vaddr;
+ if (argc == 2) {
+ SF_INFO info;
+ info.format = 0;
+ SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+ if (sf == NULL) {
+ perror(file_in);
+ return EXIT_FAILURE;
+ }
+ input_size = info.frames * info.channels * sizeof(short);
+ input_vaddr = malloc(input_size);
+ (void) sf_readf_short(sf, (short *) input_vaddr, info.frames);
+ sf_close(sf);
+ channels = info.channels;
+ input_freq = info.samplerate;
+ } else {
+ // data for testing is exactly (input sampling rate/1000)/2 seconds
+ // so 44.1khz input is 22.05 seconds
+ double k = 1000; // Hz / s
+ double time = (input_freq / 2) / k;
+ size_t input_frames = size_t(input_freq * time);
+ input_size = channels * sizeof(int16_t) * input_frames;
+ input_vaddr = malloc(input_size);
+ int16_t* in = (int16_t*)input_vaddr;
+ for (size_t i=0 ; i<input_frames ; i++) {
+ double t = double(i) / input_freq;
+ double y = sin(M_PI * k * t * t);
+ int16_t yi = floor(y * 32767.0 + 0.5);
+ for (int j = 0; j < channels; j++) {
+ in[i*channels + j] = yi / (1 + j);
+ }
+ }
+ }
+ size_t input_framesize = channels * sizeof(int16_t);
+ size_t input_frames = input_size / input_framesize;
+
+ // For float processing, convert input int16_t to float array
+ if (useFloat) {
+ void *new_vaddr;
+
+ input_framesize = channels * sizeof(float);
+ input_size = input_frames * input_framesize;
+ new_vaddr = malloc(input_size);
+ memcpy_to_float_from_i16(reinterpret_cast<float*>(new_vaddr),
+ reinterpret_cast<int16_t*>(input_vaddr), input_frames * channels);
+ free(input_vaddr);
+ input_vaddr = new_vaddr;
+ }
+
+ // ----------------------------------------------------------
+
+ class Provider: public AudioBufferProvider {
+ const void* mAddr; // base address
+ const size_t mNumFrames; // total frames
+ const size_t mFrameSize; // size of each frame in bytes
+ size_t mNextFrame; // index of next frame to provide
+ size_t mUnrel; // number of frames not yet released
+ const Vector<int> mPvalues; // number of frames provided per call
+ size_t mNextPidx; // index of next entry in mPvalues to use
+ public:
+ Provider(const void* addr, size_t frames, size_t frameSize, const Vector<int>& Pvalues)
+ : mAddr(addr),
+ mNumFrames(frames),
+ mFrameSize(frameSize),
+ mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
+ }
+ virtual status_t getNextBuffer(Buffer* buffer) {
+ size_t requestedFrames = buffer->frameCount;
+ if (requestedFrames > mNumFrames - mNextFrame) {
+ buffer->frameCount = mNumFrames - mNextFrame;
+ }
+ if (!mPvalues.isEmpty()) {
+ size_t provided = mPvalues[mNextPidx++];
+ printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount);
+ if (provided < buffer->frameCount) {
+ buffer->frameCount = provided;
+ }
+ if (mNextPidx >= mPvalues.size()) {
+ mNextPidx = 0;
+ }
+ }
+ if (gVerbose) {
+ printf("getNextBuffer() requested %zu frames out of %zu frames available,"
+ " and returned %zu frames\n",
+ requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount);
+ }
+ mUnrel = buffer->frameCount;
+ if (buffer->frameCount > 0) {
+ buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
+ return NO_ERROR;
+ } else {
+ buffer->raw = NULL;
+ return NOT_ENOUGH_DATA;
+ }
+ }
+ virtual void releaseBuffer(Buffer* buffer) {
+ if (buffer->frameCount > mUnrel) {
+ fprintf(stderr, "ERROR releaseBuffer() released %zu frames but only %zu available "
+ "to release\n", buffer->frameCount, mUnrel);
+ mNextFrame += mUnrel;
+ mUnrel = 0;
+ } else {
+ if (gVerbose) {
+ printf("releaseBuffer() released %zu frames out of %zu frames available "
+ "to release\n", buffer->frameCount, mUnrel);
+ }
+ mNextFrame += buffer->frameCount;
+ mUnrel -= buffer->frameCount;
+ }
+ buffer->frameCount = 0;
+ buffer->raw = NULL;
+ }
+ void reset() {
+ mNextFrame = 0;
+ }
+ } provider(input_vaddr, input_frames, input_framesize, Pvalues);
+
+ if (gVerbose) {
+ printf("%zu input frames\n", input_frames);
+ }
+
+ audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ int output_channels = channels > 2 ? channels : 2; // output is at least stereo samples
+ size_t output_framesize = output_channels * (useFloat ? sizeof(float) : sizeof(int32_t));
+ size_t output_frames = ((int64_t) input_frames * output_freq) / input_freq;
+ size_t output_size = output_frames * output_framesize;
+
+ if (profileFilter) {
+ // Check how fast sample rate changes are that require filter changes.
+ // The delta sample rate changes must indicate a downsampling ratio,
+ // and must be larger than 10% changes.
+ //
+ // On fast devices, filters should be generated between 0.1ms - 1ms.
+ // (single threaded).
+ AudioResampler* resampler = AudioResampler::create(format, channels,
+ 8000, quality);
+ int looplimit = 100;
+ timespec start, end;
+ clock_gettime(CLOCK_MONOTONIC, &start);
+ for (int i = 0; i < looplimit; ++i) {
+ resampler->setSampleRate(9000);
+ resampler->setSampleRate(12000);
+ resampler->setSampleRate(20000);
+ resampler->setSampleRate(30000);
+ }
+ clock_gettime(CLOCK_MONOTONIC, &end);
+ int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+ int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+ int64_t time = end_ns - start_ns;
+ printf("%.2f sample rate changes with filter calculation/sec\n",
+ looplimit * 4 / (time / 1e9));
+
+ // Check how fast sample rate changes are without filter changes.
+ // This should be very fast, probably 0.1us - 1us per sample rate
+ // change.
+ resampler->setSampleRate(1000);
+ looplimit = 1000;
+ clock_gettime(CLOCK_MONOTONIC, &start);
+ for (int i = 0; i < looplimit; ++i) {
+ resampler->setSampleRate(1000+i);
+ }
+ clock_gettime(CLOCK_MONOTONIC, &end);
+ start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+ end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+ time = end_ns - start_ns;
+ printf("%.2f sample rate changes without filter calculation/sec\n",
+ looplimit / (time / 1e9));
+ resampler->reset();
+ delete resampler;
+ }
+
+ void* output_vaddr = malloc(output_size);
+ AudioResampler* resampler = AudioResampler::create(format, channels,
+ output_freq, quality);
+
+ resampler->setSampleRate(input_freq);
+ resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);
+
+ if (profileResample) {
+ /*
+ * For profiling on mobile devices, upon experimentation
+ * it is better to run a few trials with a shorter loop limit,
+ * and take the minimum time.
+ *
+ * Long tests can cause CPU temperature to build up and thermal throttling
+ * to reduce CPU frequency.
+ *
+ * For frequency checks (index=0, or 1, etc.):
+ * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq"
+ *
+ * For temperature checks (index=0, or 1, etc.):
+ * "cat /sys/class/thermal/thermal_zone${index}/temp"
+ *
+ * Another way to avoid thermal throttling is to fix the CPU frequency
+ * at a lower level which prevents excessive temperatures.
+ */
+ const int trials = 4;
+ const int looplimit = 4;
+ timespec start, end;
+ int64_t time = 0;
+
+ for (int n = 0; n < trials; ++n) {
+ clock_gettime(CLOCK_MONOTONIC, &start);
+ for (int i = 0; i < looplimit; ++i) {
+ resampler->resample((int*) output_vaddr, output_frames, &provider);
+ provider.reset(); // during benchmarking reset only the provider
+ }
+ clock_gettime(CLOCK_MONOTONIC, &end);
+ int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+ int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+ int64_t diff_ns = end_ns - start_ns;
+ if (n == 0 || diff_ns < time) {
+ time = diff_ns; // save the best out of our trials.
+ }
+ }
+ // Mfrms/s is "Millions of output frames per second".
+ printf("quality: %d channels: %d msec: %" PRId64 " Mfrms/s: %.2lf\n",
+ quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6);
+ resampler->reset();
+
+ // TODO fix legacy bug: reset does not clear buffers.
+ // delete and recreate resampler here.
+ delete resampler;
+ resampler = AudioResampler::create(format, channels,
+ output_freq, quality);
+ resampler->setSampleRate(input_freq);
+ resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);
+ }
+
+ memset(output_vaddr, 0, output_size);
+ if (gVerbose) {
+ printf("resample() %zu output frames\n", output_frames);
+ }
+ if (Ovalues.isEmpty()) {
+ Ovalues.push(output_frames);
+ }
+ for (size_t i = 0, j = 0; i < output_frames; ) {
+ size_t thisFrames = Ovalues[j++];
+ if (j >= Ovalues.size()) {
+ j = 0;
+ }
+ if (thisFrames == 0 || thisFrames > output_frames - i) {
+ thisFrames = output_frames - i;
+ }
+ resampler->resample((int*) output_vaddr + output_channels*i, thisFrames, &provider);
+ i += thisFrames;
+ }
+ if (gVerbose) {
+ printf("resample() complete\n");
+ }
+ resampler->reset();
+ if (gVerbose) {
+ printf("reset() complete\n");
+ }
+ delete resampler;
+ resampler = NULL;
+
+ // For float processing, convert output format from float to Q4.27,
+ // which is then converted to int16_t for final storage.
+ if (useFloat) {
+ memcpy_to_q4_27_from_float(reinterpret_cast<int32_t*>(output_vaddr),
+ reinterpret_cast<float*>(output_vaddr), output_frames * output_channels);
+ }
+
+ // mono takes left channel only (out of stereo output pair)
+ // stereo and multichannel preserve all channels.
+ int32_t* out = (int32_t*) output_vaddr;
+ int16_t* convert = (int16_t*) malloc(output_frames * channels * sizeof(int16_t));
+
+ const int volumeShift = 12; // shift requirement for Q4.27 to Q.15
+ // round to half towards zero and saturate at int16 (non-dithered)
+ const int roundVal = (1<<(volumeShift-1)) - 1; // volumePrecision > 0
+
+ for (size_t i = 0; i < output_frames; i++) {
+ for (int j = 0; j < channels; j++) {
+ int32_t s = out[i * output_channels + j] + roundVal; // add offset here
+ if (s < 0) {
+ s = (s + 1) >> volumeShift; // round to 0
+ if (s < -32768) {
+ s = -32768;
+ }
+ } else {
+ s = s >> volumeShift;
+ if (s > 32767) {
+ s = 32767;
+ }
+ }
+ convert[i * channels + j] = int16_t(s);
+ }
+ }
+
+ // write output to disk
+ SF_INFO info;
+ info.frames = 0;
+ info.samplerate = output_freq;
+ info.channels = channels;
+ info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+ SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info);
+ if (sf == NULL) {
+ perror(file_out);
+ return EXIT_FAILURE;
+ }
+ (void) sf_writef_short(sf, convert, output_frames);
+ sf_close(sf);
+
+ return EXIT_SUCCESS;
+}