diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
new file mode 100644
index 0000000..b082e8c
--- /dev/null
+++ b/services/audioflinger/test-resample.cpp
@@ -0,0 +1,274 @@
+/*
+ * 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>
+#include <math.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] [-s] [-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,"    -s    stereo\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[]) {
+
+    const char* const progname = argv[0];
+    bool profiling = false;
+    bool writeHeader = false;
+    int channels = 1;
+    int input_freq = 0;
+    int output_freq = 0;
+    AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
+
+    int ch;
+    while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) {
+        switch (ch) {
+        case 'p':
+            profiling = true;
+            break;
+        case 'h':
+            writeHeader = true;
+            break;
+        case 's':
+            channels = 2;
+            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(progname);
+                return -1;
+            }
+            break;
+        case 'i':
+            input_freq = atoi(optarg);
+            break;
+        case 'o':
+            output_freq = atoi(optarg);
+            break;
+        case '?':
+        default:
+            usage(progname);
+            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) {
+        struct stat st;
+        if (stat(file_in, &st) < 0) {
+            fprintf(stderr, "stat: %s\n", strerror(errno));
+            return -1;
+        }
+
+        int input_fd = open(file_in, O_RDONLY);
+        if (input_fd < 0) {
+            fprintf(stderr, "open: %s\n", strerror(errno));
+            return -1;
+        }
+
+        input_size = st.st_size;
+        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;
+        }
+    } else {
+        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 (size_t j=0 ; j<(size_t)channels ; j++) {
+                in[i*channels + j] = yi / (1+j);
+            }
+        }
+    }
+
+    // ----------------------------------------------------------
+
+    class Provider: public AudioBufferProvider {
+        int16_t* mAddr;
+        size_t mNumFrames;
+    public:
+        Provider(const void* addr, size_t size, int channels) {
+            mAddr = (int16_t*) addr;
+            mNumFrames = size / (channels*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, channels);
+
+    size_t input_frames = input_size / (channels * sizeof(int16_t));
+    size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
+    output_size &= ~7; // always stereo, 32-bits
+
+    void* output_vaddr = malloc(output_size);
+
+    if (profiling) {
+        AudioResampler* resampler = AudioResampler::create(16, channels,
+                output_freq, quality);
+
+        size_t out_frames = output_size/8;
+        resampler->setSampleRate(input_freq);
+        resampler->setVolume(0x1000, 0x1000);
+
+        memset(output_vaddr, 0, output_size);
+        timespec start, end;
+        clock_gettime(CLOCK_MONOTONIC_HR, &start);
+        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        resampler->resample((int*) output_vaddr, out_frames, &provider);
+        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)/4;
+        printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6);
+
+        delete resampler;
+    }
+
+    AudioResampler* resampler = AudioResampler::create(16, channels,
+            output_freq, quality);
+    size_t out_frames = output_size/8;
+    resampler->setSampleRate(input_freq);
+    resampler->setVolume(0x1000, 0x1000);
+
+    memset(output_vaddr, 0, output_size);
+    resampler->resample((int*) output_vaddr, out_frames, &provider);
+
+    // 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 * channels * sizeof(int16_t));
+    for (size_t i = 0; i < out_frames; i++) {
+        for (int j=0 ; j<channels ; j++) {
+            int32_t s = out[i * 2 + j] >> 12;
+            if (s > 32767)       s =  32767;
+            else if (s < -32768) s = -32768;
+            convert[i * channels + j] = int16_t(s);
+        }
+    }
+
+    // write output to disk
+    int output_fd = open(file_out, 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 * channels * sizeof(int16_t), channels, output_freq, 16);
+        write(output_fd, &wav, sizeof(wav));
+    }
+
+    write(output_fd, convert, out_frames * channels * sizeof(int16_t));
+    close(output_fd);
+
+    return 0;
+}