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Commit 87925675 authored by Mathias Agopian's avatar Mathias Agopian Committed by Android (Google) Code Review
Browse files

Merge changes I5d62a6c2,Iebf76958,I8068f0f7 

* changes:
  improve resample test
  change how we store the FIR coefficients
  improve SINC resampler performance
parents 48c6b262 3f71761c

services/audioflinger/AudioResamplerSinc.cpp

+309 −72

File changed.

Preview size limit exceeded, changes collapsed.

services/audioflinger/AudioResamplerSinc.h

+3 −2
Original line number Diff line number Diff line
@@ -50,12 +50,13 @@ private: 


    template<int CHANNELS>

    inline void filterCoefficient(

            int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples);

            int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples, uint32_t vRL);



    template<int CHANNELS>

    inline void interpolate(

            int32_t& l, int32_t& r,

            const int32_t* coefs, int16_t lerp, const int16_t* samples);

            const int32_t* coefs, size_t offset,

            int32_t lerp, const int16_t* samples);



    template<int CHANNELS>

    inline void read(int16_t*& impulse, uint32_t& phaseFraction,

services/audioflinger/audio-resampler/filter_coefficients.h

+260 −36

File changed.

Preview size limit exceeded, changes collapsed.

services/audioflinger/test-resample.cpp

+102 −61
Original line number Diff line number Diff line
@@ -25,6 +25,7 @@ 
#include <sys/stat.h>

#include <errno.h>

#include <time.h>

#include <math.h>



using namespace android;
@@ -61,31 +62,34 @@ struct HeaderWav { 
};



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,"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");

    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, "phq:i:o:")) != -1) {

    while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) {

        switch (ch) {

        case 'p':

            profiling = true;
@@ -93,6 +97,9 @@ int main(int argc, char* argv[]) { 
        case 'h':

            writeHeader = true;

            break;

        case 's':

            channels = 2;

            break;

        case 'q':

            if (!strcmp(optarg, "dq"))

                quality = AudioResampler::DEFAULT_QUALITY;
@@ -105,7 +112,7 @@ int main(int argc, char* argv[]) { 
            else if (!strcmp(optarg, "vhq"))

                quality = AudioResampler::VERY_HIGH_QUALITY;

            else {

                usage(argv[0]);

                usage(progname);

                return -1;

            }

            break;
@@ -117,44 +124,64 @@ int main(int argc, char* argv[]) { 
            break;

        case '?':

        default:

            usage(argv[0]);

            usage(progname);

            return -1;

        }

    }

    argc -= optind;

    argv += optind;



    if (argc != 2) {

        usage(argv[0]);

    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;

    }



    argv += optind;



    // ----------------------------------------------------------



    size_t input_size;

    void* input_vaddr;

    if (argc == 2) {

        struct stat st;

    if (stat(argv[0], &st) < 0) {

        if (stat(file_in, &st) < 0) {

            fprintf(stderr, "stat: %s\n", strerror(errno));

            return -1;

        }



    int input_fd = open(argv[0], O_RDONLY);

        int input_fd = open(file_in, 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);

        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;

        }



//    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);

    } 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<channels ; j++) {

                in[i*channels + j] = yi;

            }

        }

    }



    // ----------------------------------------------------------
@@ -162,9 +189,9 @@ int main(int argc, char* argv[]) { 
        int16_t* mAddr;

        size_t mNumFrames;

    public:

        Provider(const void* addr, size_t size) {

        Provider(const void* addr, size_t size, int channels) {

            mAddr = (int16_t*) addr;

            mNumFrames = size / sizeof(int16_t);

            mNumFrames = size / (channels*sizeof(int16_t));

        }

        virtual status_t getNextBuffer(Buffer* buffer,

                int64_t pts = kInvalidPTS) {
@@ -174,47 +201,61 @@ int main(int argc, char* argv[]) { 
        }

        virtual void releaseBuffer(Buffer* buffer) {

        }

    } provider(input_vaddr, input_size);

    } provider(input_vaddr, input_size, channels);



    size_t output_size = 2 * 2 * ((int64_t) input_size * output_freq)

            / input_freq;

    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);

    memset(output_vaddr, 0, output_size);



    AudioResampler* resampler = AudioResampler::create(16, 1, output_freq,

            quality);

    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);

    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);

        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;

        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 * sizeof(int16_t));

    int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));

    for (size_t i = 0; i < out_frames; i++) {

        int32_t s = out[i * 2] >> 12;

        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] = int16_t(s);

            convert[i * channels + j] = int16_t(s);

        }

    }



    // write output to disk

    int output_fd = open(argv[1], O_WRONLY | O_CREAT | O_TRUNC,

    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));
@@ -222,11 +263,11 @@ int main(int argc, char* argv[]) { 
    }



    if (writeHeader) {

        HeaderWav wav(out_frames*sizeof(int16_t), 1, output_freq, 16);

        HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16);

        write(output_fd, &wav, sizeof(wav));

    }



    write(output_fd, convert, out_frames * sizeof(int16_t));

    write(output_fd, convert, out_frames * channels * sizeof(int16_t));

    close(output_fd);



    return 0;

tools/resampler_tools/fir.cpp

+19 −20
Original line number Diff line number Diff line
@@ -195,7 +195,8 @@ int main(int argc, char** argv) 




    // total number of coefficients (one side)

    const int N = (1 << nz) * nzc;

    const int M = (1 << nz);

    const int N = M * nzc;



    // generate the right half of the filter

    if (!debug) {
@@ -220,13 +221,15 @@ int main(int argc, char** argv) 
    }



    if (!polyphase) {

        for (int i=0 ; i<N ; i++) {

            double x = (2.0 * M_PI * i * Fcr) / (1 << nz);

            double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;

        for (int i=0 ; i<=M ; i++) { // an extra set of coefs for interpolation

            for (int j=0 ; j<nzc ; j++) {

                int ix = j*M + i;

                double x = (2.0 * M_PI * ix * Fcr) / (1 << nz);

                double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;

                y *= atten;



                if (!debug) {

                if ((i % (1<<nz)) == 0)

                    if (j == 0)

                        printf("\n    ");

                }
@@ -238,6 +241,7 @@ int main(int argc, char** argv) 
                    printf("%.9g%s ", y, debug ? "," : "f,");

                }

            }

        }

    } else {

        for (int j=0 ; j<polyN ; j++) {

            // calculate the phase
@@ -266,11 +270,6 @@ int main(int argc, char** argv) 
    }



    if (!debug) {

        if (!format) {

            printf("\n    0x%08x ", 0);

        } else {

            printf("\n    %.9g ", 0.0f);

        }

        printf("\n};");

    }

    printf("\n");