diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
new file mode 100644
index 0000000..a55a32b
--- /dev/null
+++ b/services/audioflinger/test-resample.cpp
@@ -0,0 +1,229 @@
+/*
+ * 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;
+}