--- /dev/null
+/*
+ * 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 "AudioResampler.h"
+#include <media/AudioBufferProvider.h>
+#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>
+
+using namespace android;
+
+struct HeaderWav {
+ HeaderWav(size_t size, int nc, int sr, int bits) {
+ strncpy(RIFF, "RIFF", 4);
+ chunkSize = size + sizeof(HeaderWav);
+ strncpy(WAVE, "WAVE", 4);
+ strncpy(fmt, "fmt ", 4);
+ fmtSize = 16;
+ audioFormat = 1;
+ numChannels = nc;
+ samplesRate = sr;
+ byteRate = sr * numChannels * (bits/8);
+ align = nc*(bits/8);
+ bitsPerSample = bits;
+ strncpy(data, "data", 4);
+ dataSize = size;
+ }
+
+ char RIFF[4]; // RIFF
+ uint32_t chunkSize; // File size
+ char WAVE[4]; // WAVE
+ char fmt[4]; // fmt\0
+ uint32_t fmtSize; // fmt size
+ uint16_t audioFormat; // 1=PCM
+ uint16_t numChannels; // num channels
+ uint32_t samplesRate; // sample rate in hz
+ uint32_t byteRate; // Bps
+ uint16_t align; // 2=16-bit mono, 4=16-bit stereo
+ uint16_t bitsPerSample; // bits per sample
+ char data[4]; // "data"
+ uint32_t dataSize; // size
+};
+
+static int usage(const char* name) {
+ fprintf(stderr,"Usage: %s [-p] [-h] [-q <dq|lq|mq|hq|vhq>] [-i <input-sample-rate>] [-o <output-sample-rate>] <input-file> <output-file>\n", name);
+ fprintf(stderr,"-p - enable profiling\n");
+ fprintf(stderr,"-h - create wav file\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,"-i - input file sample rate\n");
+ fprintf(stderr,"-o - output file sample rate\n");
+ return -1;
+}
+
+int main(int argc, char* argv[]) {
+
+ bool profiling = false;
+ bool writeHeader = false;
+ int input_freq = 0;
+ int output_freq = 0;
+ AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
+
+ int ch;
+ while ((ch = getopt(argc, argv, "phq:i:o:")) != -1) {
+ switch (ch) {
+ case 'p':
+ profiling = true;
+ break;
+ case 'h':
+ writeHeader = true;
+ 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 {
+ usage(argv[0]);
+ return -1;
+ }
+ break;
+ case 'i':
+ input_freq = atoi(optarg);
+ break;
+ case 'o':
+ output_freq = atoi(optarg);
+ break;
+ case '?':
+ default:
+ usage(argv[0]);
+ return -1;
+ }
+ }
+ argc -= optind;
+
+ if (argc != 2) {
+ usage(argv[0]);
+ return -1;
+ }
+
+ argv += optind;
+
+ // ----------------------------------------------------------
+
+ struct stat st;
+ if (stat(argv[0], &st) < 0) {
+ fprintf(stderr, "stat: %s\n", strerror(errno));
+ return -1;
+ }
+
+ int input_fd = open(argv[0], O_RDONLY);
+ if (input_fd < 0) {
+ fprintf(stderr, "open: %s\n", strerror(errno));
+ return -1;
+ }
+
+ size_t input_size = st.st_size;
+ void* input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd,
+ 0);
+ if (input_vaddr == MAP_FAILED ) {
+ fprintf(stderr, "mmap: %s\n", strerror(errno));
+ return -1;
+ }
+
+// printf("input sample rate: %d Hz\n", input_freq);
+// printf("output sample rate: %d Hz\n", output_freq);
+// printf("input mmap: %p, size=%u\n", input_vaddr, input_size);
+
+ // ----------------------------------------------------------
+
+ class Provider: public AudioBufferProvider {
+ int16_t* mAddr;
+ size_t mNumFrames;
+ public:
+ Provider(const void* addr, size_t size) {
+ mAddr = (int16_t*) addr;
+ mNumFrames = size / sizeof(int16_t);
+ }
+ virtual status_t getNextBuffer(Buffer* buffer,
+ int64_t pts = kInvalidPTS) {
+ buffer->frameCount = mNumFrames;
+ buffer->i16 = mAddr;
+ return NO_ERROR;
+ }
+ virtual void releaseBuffer(Buffer* buffer) {
+ }
+ } provider(input_vaddr, input_size);
+
+ size_t output_size = 2 * 2 * ((int64_t) input_size * output_freq)
+ / input_freq;
+ output_size &= ~7; // always stereo, 32-bits
+
+ void* output_vaddr = malloc(output_size);
+ memset(output_vaddr, 0, output_size);
+
+ AudioResampler* resampler = AudioResampler::create(16, 1, output_freq,
+ quality);
+
+ size_t out_frames = output_size/8;
+ resampler->setSampleRate(input_freq);
+ resampler->setVolume(0x1000, 0x1000);
+ resampler->resample((int*) output_vaddr, out_frames, &provider);
+
+ if (profiling) {
+ memset(output_vaddr, 0, output_size);
+ timespec start, end;
+ clock_gettime(CLOCK_MONOTONIC_HR, &start);
+ resampler->resample((int*) output_vaddr, out_frames, &provider);
+ clock_gettime(CLOCK_MONOTONIC_HR, &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("%f Mspl/s\n", out_frames/(time/1e9)/1e6);
+ }
+
+ // down-mix (we just truncate and keep the left channel)
+ int32_t* out = (int32_t*) output_vaddr;
+ int16_t* convert = (int16_t*) malloc(out_frames * sizeof(int16_t));
+ for (size_t i = 0; i < out_frames; i++) {
+ convert[i] = out[i * 2] >> 12;
+ }
+
+ // write output to disk
+ int output_fd = open(argv[1], O_WRONLY | O_CREAT | O_TRUNC,
+ S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
+ if (output_fd < 0) {
+ fprintf(stderr, "open: %s\n", strerror(errno));
+ return -1;
+ }
+
+ if (writeHeader) {
+ HeaderWav wav(out_frames*sizeof(int16_t), 1, output_freq, 16);
+ write(output_fd, &wav, sizeof(wav));
+ }
+
+ write(output_fd, convert, out_frames * sizeof(int16_t));
+ close(output_fd);
+
+ return 0;
+}
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