annotate sonic.c @ 9675:2e84b386a8b6 libavcodec

Fix for a problem with inverted sign of output data from ffvorbis decoder. Now the sign of audio samples in ffvorbis output is the same as in original uncompressed audio file and this also allows the use of tiny_psnr to compare ffvorbis with libvorbis/tremor.

/*
 * Simple free lossless/lossy audio codec
 * Copyright (c) 2004 Alex Beregszaszi
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
#include "avcodec.h"
#include "get_bits.h"
#include "golomb.h"

/**
 * @file libavcodec/sonic.c
 * Simple free lossless/lossy audio codec
 * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
 * Written and designed by Alex Beregszaszi
 *
 * TODO:
 *  - CABAC put/get_symbol
 *  - independent quantizer for channels
 *  - >2 channels support
 *  - more decorrelation types
 *  - more tap_quant tests
 *  - selectable intlist writers/readers (bonk-style, golomb, cabac)
 */

#define MAX_CHANNELS 2

#define MID_SIDE 0
#define LEFT_SIDE 1
#define RIGHT_SIDE 2

typedef struct SonicContext {
    int lossless, decorrelation;

    int num_taps, downsampling;
    double quantization;

    int channels, samplerate, block_align, frame_size;

    int *tap_quant;
    int *int_samples;
    int *coded_samples[MAX_CHANNELS];

    // for encoding
    int *tail;
    int tail_size;
    int *window;
    int window_size;

    // for decoding
    int *predictor_k;
    int *predictor_state[MAX_CHANNELS];
} SonicContext;

#define LATTICE_SHIFT   10
#define SAMPLE_SHIFT    4
#define LATTICE_FACTOR  (1 << LATTICE_SHIFT)
#define SAMPLE_FACTOR   (1 << SAMPLE_SHIFT)

#define BASE_QUANT      0.6
#define RATE_VARIATION  3.0

static inline int divide(int a, int b)
{
    if (a < 0)
        return -( (-a + b/2)/b );
    else
        return (a + b/2)/b;
}

static inline int shift(int a,int b)
{
    return (a+(1<<(b-1))) >> b;
}

static inline int shift_down(int a,int b)
{
    return (a>>b)+((a<0)?1:0);
}

#if 1
static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
{
    int i;

    for (i = 0; i < entries; i++)
        set_se_golomb(pb, buf[i]);

    return 1;
}

static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
{
    int i;

    for (i = 0; i < entries; i++)
        buf[i] = get_se_golomb(gb);

    return 1;
}

#else

#define ADAPT_LEVEL 8

static int bits_to_store(uint64_t x)
{
    int res = 0;

    while(x)
    {
        res++;
        x >>= 1;
    }
    return res;
}

static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max)
{
    int i, bits;

    if (!max)
        return;

    bits = bits_to_store(max);

    for (i = 0; i < bits-1; i++)
        put_bits(pb, 1, value & (1 << i));

    if ( (value | (1 << (bits-1))) <= max)
        put_bits(pb, 1, value & (1 << (bits-1)));
}

static unsigned int read_uint_max(GetBitContext *gb, int max)
{
    int i, bits, value = 0;

    if (!max)
        return 0;

    bits = bits_to_store(max);

    for (i = 0; i < bits-1; i++)
        if (get_bits1(gb))
            value += 1 << i;

    if ( (value | (1<<(bits-1))) <= max)
        if (get_bits1(gb))
            value += 1 << (bits-1);

    return value;
}

static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
{
    int i, j, x = 0, low_bits = 0, max = 0;
    int step = 256, pos = 0, dominant = 0, any = 0;
    int *copy, *bits;

    copy = av_mallocz(4* entries);
    if (!copy)
        return -1;

    if (base_2_part)
    {
        int energy = 0;

        for (i = 0; i < entries; i++)
            energy += abs(buf[i]);

        low_bits = bits_to_store(energy / (entries * 2));
        if (low_bits > 15)
            low_bits = 15;

        put_bits(pb, 4, low_bits);
    }

    for (i = 0; i < entries; i++)
    {
        put_bits(pb, low_bits, abs(buf[i]));
        copy[i] = abs(buf[i]) >> low_bits;
        if (copy[i] > max)
            max = abs(copy[i]);
    }

    bits = av_mallocz(4* entries*max);
    if (!bits)
    {
//        av_free(copy);
        return -1;
    }

    for (i = 0; i <= max; i++)
    {
        for (j = 0; j < entries; j++)
            if (copy[j] >= i)
                bits[x++] = copy[j] > i;
    }

    // store bitstream
    while (pos < x)
    {
        int steplet = step >> 8;

        if (pos + steplet > x)
            steplet = x - pos;

        for (i = 0; i < steplet; i++)
            if (bits[i+pos] != dominant)
                any = 1;

        put_bits(pb, 1, any);

        if (!any)
        {
            pos += steplet;
            step += step / ADAPT_LEVEL;
        }
        else
        {
            int interloper = 0;

            while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
                interloper++;

            // note change
            write_uint_max(pb, interloper, (step >> 8) - 1);

            pos += interloper + 1;
            step -= step / ADAPT_LEVEL;
        }

        if (step < 256)
        {
            step = 65536 / step;
            dominant = !dominant;
        }
    }

    // store signs
    for (i = 0; i < entries; i++)
        if (buf[i])
            put_bits(pb, 1, buf[i] < 0);

//    av_free(bits);
//    av_free(copy);

    return 0;
}

static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
{
    int i, low_bits = 0, x = 0;
    int n_zeros = 0, step = 256, dominant = 0;
    int pos = 0, level = 0;
    int *bits = av_mallocz(4* entries);

    if (!bits)
        return -1;

    if (base_2_part)
    {
        low_bits = get_bits(gb, 4);

        if (low_bits)
            for (i = 0; i < entries; i++)
                buf[i] = get_bits(gb, low_bits);
    }

//    av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits);

    while (n_zeros < entries)
    {
        int steplet = step >> 8;

        if (!get_bits1(gb))
        {
            for (i = 0; i < steplet; i++)
                bits[x++] = dominant;

            if (!dominant)
                n_zeros += steplet;

            step += step / ADAPT_LEVEL;
        }
        else
        {
            int actual_run = read_uint_max(gb, steplet-1);

//            av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run);

            for (i = 0; i < actual_run; i++)
                bits[x++] = dominant;

            bits[x++] = !dominant;

            if (!dominant)
                n_zeros += actual_run;
            else
                n_zeros++;

            step -= step / ADAPT_LEVEL;
        }

        if (step < 256)
        {
            step = 65536 / step;
            dominant = !dominant;
        }
    }

    // reconstruct unsigned values
    n_zeros = 0;
    for (i = 0; n_zeros < entries; i++)
    {
        while(1)
        {
            if (pos >= entries)
            {
                pos = 0;
                level += 1 << low_bits;
            }

            if (buf[pos] >= level)
                break;

            pos++;
        }

        if (bits[i])
            buf[pos] += 1 << low_bits;
        else
            n_zeros++;

        pos++;
    }
//    av_free(bits);

    // read signs
    for (i = 0; i < entries; i++)
        if (buf[i] && get_bits1(gb))
            buf[i] = -buf[i];

//    av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos);

    return 0;
}
#endif

static void predictor_init_state(int *k, int *state, int order)
{
    int i;

    for (i = order-2; i >= 0; i--)
    {
        int j, p, x = state[i];

        for (j = 0, p = i+1; p < order; j++,p++)
            {
            int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);
            state[p] += shift_down(k[j]*x, LATTICE_SHIFT);
            x = tmp;
        }
    }
}

static int predictor_calc_error(int *k, int *state, int order, int error)
{
    int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT);

#if 1
    int *k_ptr = &(k[order-2]),
        *state_ptr = &(state[order-2]);
    for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--)
    {
        int k_value = *k_ptr, state_value = *state_ptr;
        x -= shift_down(k_value * state_value, LATTICE_SHIFT);
        state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
    }
#else
    for (i = order-2; i >= 0; i--)
    {
        x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
        state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT);
    }
#endif

    // don't drift too far, to avoid overflows
    if (x >  (SAMPLE_FACTOR<<16)) x =  (SAMPLE_FACTOR<<16);
    if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16);

    state[0] = x;

    return x;
}

#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
// Heavily modified Levinson-Durbin algorithm which
// copes better with quantization, and calculates the
// actual whitened result as it goes.

static void modified_levinson_durbin(int *window, int window_entries,
        int *out, int out_entries, int channels, int *tap_quant)
{
    int i;
    int *state = av_mallocz(4* window_entries);

    memcpy(state, window, 4* window_entries);

    for (i = 0; i < out_entries; i++)
    {
        int step = (i+1)*channels, k, j;
        double xx = 0.0, xy = 0.0;
#if 1
        int *x_ptr = &(window[step]), *state_ptr = &(state[0]);
        j = window_entries - step;
        for (;j>=0;j--,x_ptr++,state_ptr++)
        {
            double x_value = *x_ptr, state_value = *state_ptr;
            xx += state_value*state_value;
            xy += x_value*state_value;
        }
#else
        for (j = 0; j <= (window_entries - step); j++);
        {
            double stepval = window[step+j], stateval = window[j];
//            xx += (double)window[j]*(double)window[j];
//            xy += (double)window[step+j]*(double)window[j];
            xx += stateval*stateval;
            xy += stepval*stateval;
        }
#endif
        if (xx == 0.0)
            k = 0;
        else
            k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));

        if (k > (LATTICE_FACTOR/tap_quant[i]))
            k = LATTICE_FACTOR/tap_quant[i];
        if (-k > (LATTICE_FACTOR/tap_quant[i]))
            k = -(LATTICE_FACTOR/tap_quant[i]);

        out[i] = k;
        k *= tap_quant[i];

#if 1
        x_ptr = &(window[step]);
        state_ptr = &(state[0]);
        j = window_entries - step;
        for (;j>=0;j--,x_ptr++,state_ptr++)
        {
            int x_value = *x_ptr, state_value = *state_ptr;
            *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
            *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
        }
#else
        for (j=0; j <= (window_entries - step); j++)
        {
            int stepval = window[step+j], stateval=state[j];
            window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
            state[j] += shift_down(k * stepval, LATTICE_SHIFT);
        }
#endif
    }

    av_free(state);
}
#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */

static const int samplerate_table[] =
    { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };

#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
static inline int code_samplerate(int samplerate)
{
    switch (samplerate)
    {
        case 44100: return 0;
        case 22050: return 1;
        case 11025: return 2;
        case 96000: return 3;
        case 48000: return 4;
        case 32000: return 5;
        case 24000: return 6;
        case 16000: return 7;
        case 8000: return 8;
    }
    return -1;
}

static av_cold int sonic_encode_init(AVCodecContext *avctx)
{
    SonicContext *s = avctx->priv_data;
    PutBitContext pb;
    int i, version = 0;

    if (avctx->channels > MAX_CHANNELS)
    {
        av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
        return -1; /* only stereo or mono for now */
    }

    if (avctx->channels == 2)
        s->decorrelation = MID_SIDE;

    if (avctx->codec->id == CODEC_ID_SONIC_LS)
    {
        s->lossless = 1;
        s->num_taps = 32;
        s->downsampling = 1;
        s->quantization = 0.0;
    }
    else
    {
        s->num_taps = 128;
        s->downsampling = 2;
        s->quantization = 1.0;
    }

    // max tap 2048
    if ((s->num_taps < 32) || (s->num_taps > 1024) ||
        ((s->num_taps>>5)<<5 != s->num_taps))
    {
        av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
        return -1;
    }

    // generate taps
    s->tap_quant = av_mallocz(4* s->num_taps);
    for (i = 0; i < s->num_taps; i++)
        s->tap_quant[i] = (int)(sqrt(i+1));

    s->channels = avctx->channels;
    s->samplerate = avctx->sample_rate;

    s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling;
    s->frame_size = s->channels*s->block_align*s->downsampling;

    s->tail = av_mallocz(4* s->num_taps*s->channels);
    if (!s->tail)
        return -1;
    s->tail_size = s->num_taps*s->channels;

    s->predictor_k = av_mallocz(4 * s->num_taps);
    if (!s->predictor_k)
        return -1;

    for (i = 0; i < s->channels; i++)
    {
        s->coded_samples[i] = av_mallocz(4* s->block_align);
        if (!s->coded_samples[i])
            return -1;
    }

    s->int_samples = av_mallocz(4* s->frame_size);

    s->window_size = ((2*s->tail_size)+s->frame_size);
    s->window = av_mallocz(4* s->window_size);
    if (!s->window)
        return -1;

    avctx->extradata = av_mallocz(16);
    if (!avctx->extradata)
        return -1;
    init_put_bits(&pb, avctx->extradata, 16*8);

    put_bits(&pb, 2, version); // version
    if (version == 1)
    {
        put_bits(&pb, 2, s->channels);
        put_bits(&pb, 4, code_samplerate(s->samplerate));
    }
    put_bits(&pb, 1, s->lossless);
    if (!s->lossless)
        put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
    put_bits(&pb, 2, s->decorrelation);
    put_bits(&pb, 2, s->downsampling);
    put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024
    put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table

    flush_put_bits(&pb);
    avctx->extradata_size = put_bits_count(&pb)/8;

    av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
        version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);

    avctx->coded_frame = avcodec_alloc_frame();
    if (!avctx->coded_frame)
        return AVERROR(ENOMEM);
    avctx->coded_frame->key_frame = 1;
    avctx->frame_size = s->block_align*s->downsampling;

    return 0;
}

static av_cold int sonic_encode_close(AVCodecContext *avctx)
{
    SonicContext *s = avctx->priv_data;
    int i;

    av_freep(&avctx->coded_frame);

    for (i = 0; i < s->channels; i++)
        av_free(s->coded_samples[i]);

    av_free(s->predictor_k);
    av_free(s->tail);
    av_free(s->tap_quant);
    av_free(s->window);
    av_free(s->int_samples);

    return 0;
}

static int sonic_encode_frame(AVCodecContext *avctx,
                            uint8_t *buf, int buf_size, void *data)
{
    SonicContext *s = avctx->priv_data;
    PutBitContext pb;
    int i, j, ch, quant = 0, x = 0;
    short *samples = data;

    init_put_bits(&pb, buf, buf_size*8);

    // short -> internal
    for (i = 0; i < s->frame_size; i++)
        s->int_samples[i] = samples[i];

    if (!s->lossless)
        for (i = 0; i < s->frame_size; i++)
            s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;

    switch(s->decorrelation)
    {
        case MID_SIDE:
            for (i = 0; i < s->frame_size; i += s->channels)
            {
                s->int_samples[i] += s->int_samples[i+1];
                s->int_samples[i+1] -= shift(s->int_samples[i], 1);
            }
            break;
        case LEFT_SIDE:
            for (i = 0; i < s->frame_size; i += s->channels)
                s->int_samples[i+1] -= s->int_samples[i];
            break;
        case RIGHT_SIDE:
            for (i = 0; i < s->frame_size; i += s->channels)
                s->int_samples[i] -= s->int_samples[i+1];
            break;
    }

    memset(s->window, 0, 4* s->window_size);

    for (i = 0; i < s->tail_size; i++)
        s->window[x++] = s->tail[i];

    for (i = 0; i < s->frame_size; i++)
        s->window[x++] = s->int_samples[i];

    for (i = 0; i < s->tail_size; i++)
        s->window[x++] = 0;

    for (i = 0; i < s->tail_size; i++)
        s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];

    // generate taps
    modified_levinson_durbin(s->window, s->window_size,
                s->predictor_k, s->num_taps, s->channels, s->tap_quant);
    if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0)
        return -1;

    for (ch = 0; ch < s->channels; ch++)
    {
        x = s->tail_size+ch;
        for (i = 0; i < s->block_align; i++)
        {
            int sum = 0;
            for (j = 0; j < s->downsampling; j++, x += s->channels)
                sum += s->window[x];
            s->coded_samples[ch][i] = sum;
        }
    }

    // simple rate control code
    if (!s->lossless)
    {
        double energy1 = 0.0, energy2 = 0.0;
        for (ch = 0; ch < s->channels; ch++)
        {
            for (i = 0; i < s->block_align; i++)
            {
                double sample = s->coded_samples[ch][i];
                energy2 += sample*sample;
                energy1 += fabs(sample);
            }
        }

        energy2 = sqrt(energy2/(s->channels*s->block_align));
        energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align);

        // increase bitrate when samples are like a gaussian distribution
        // reduce bitrate when samples are like a two-tailed exponential distribution

        if (energy2 > energy1)
            energy2 += (energy2-energy1)*RATE_VARIATION;

        quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
//        av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);

        if (quant < 1)
            quant = 1;
        if (quant > 65535)
            quant = 65535;

        set_ue_golomb(&pb, quant);

        quant *= SAMPLE_FACTOR;
    }

    // write out coded samples
    for (ch = 0; ch < s->channels; ch++)
    {
        if (!s->lossless)
            for (i = 0; i < s->block_align; i++)
                s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant);

        if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0)
            return -1;
    }

//    av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);

    flush_put_bits(&pb);
    return (put_bits_count(&pb)+7)/8;
}
#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */

#if CONFIG_SONIC_DECODER
static av_cold int sonic_decode_init(AVCodecContext *avctx)
{
    SonicContext *s = avctx->priv_data;
    GetBitContext gb;
    int i, version;

    s->channels = avctx->channels;
    s->samplerate = avctx->sample_rate;

    if (!avctx->extradata)
    {
        av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
        return -1;
    }

    init_get_bits(&gb, avctx->extradata, avctx->extradata_size);

    version = get_bits(&gb, 2);
    if (version > 1)
    {
        av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
        return -1;
    }

    if (version == 1)
    {
        s->channels = get_bits(&gb, 2);
        s->samplerate = samplerate_table[get_bits(&gb, 4)];
        av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
            s->channels, s->samplerate);
    }

    if (s->channels > MAX_CHANNELS)
    {
        av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
        return -1;
    }

    s->lossless = get_bits1(&gb);
    if (!s->lossless)
        skip_bits(&gb, 3); // XXX FIXME
    s->decorrelation = get_bits(&gb, 2);

    s->downsampling = get_bits(&gb, 2);
    s->num_taps = (get_bits(&gb, 5)+1)<<5;
    if (get_bits1(&gb)) // XXX FIXME
        av_log(avctx, AV_LOG_INFO, "Custom quant table\n");

    s->block_align = (int)(2048.0*(s->samplerate/44100))/s->downsampling;
    s->frame_size = s->channels*s->block_align*s->downsampling;
//    avctx->frame_size = s->block_align;

    av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
        version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);

    // generate taps
    s->tap_quant = av_mallocz(4* s->num_taps);
    for (i = 0; i < s->num_taps; i++)
        s->tap_quant[i] = (int)(sqrt(i+1));

    s->predictor_k = av_mallocz(4* s->num_taps);

    for (i = 0; i < s->channels; i++)
    {
        s->predictor_state[i] = av_mallocz(4* s->num_taps);
        if (!s->predictor_state[i])
            return -1;
    }

    for (i = 0; i < s->channels; i++)
    {
        s->coded_samples[i] = av_mallocz(4* s->block_align);
        if (!s->coded_samples[i])
            return -1;
    }
    s->int_samples = av_mallocz(4* s->frame_size);

    avctx->sample_fmt = SAMPLE_FMT_S16;
    return 0;
}

static av_cold int sonic_decode_close(AVCodecContext *avctx)
{
    SonicContext *s = avctx->priv_data;
    int i;

    av_free(s->int_samples);
    av_free(s->tap_quant);
    av_free(s->predictor_k);

    for (i = 0; i < s->channels; i++)
    {
        av_free(s->predictor_state[i]);
        av_free(s->coded_samples[i]);
    }

    return 0;
}

static int sonic_decode_frame(AVCodecContext *avctx,
                            void *data, int *data_size,
                            AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    SonicContext *s = avctx->priv_data;
    GetBitContext gb;
    int i, quant, ch, j;
    short *samples = data;

    if (buf_size == 0) return 0;

//    av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);

    init_get_bits(&gb, buf, buf_size*8);

    intlist_read(&gb, s->predictor_k, s->num_taps, 0);

    // dequantize
    for (i = 0; i < s->num_taps; i++)
        s->predictor_k[i] *= s->tap_quant[i];

    if (s->lossless)
        quant = 1;
    else
        quant = get_ue_golomb(&gb) * SAMPLE_FACTOR;

//    av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);

    for (ch = 0; ch < s->channels; ch++)
    {
        int x = ch;

        predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);

        intlist_read(&gb, s->coded_samples[ch], s->block_align, 1);

        for (i = 0; i < s->block_align; i++)
        {
            for (j = 0; j < s->downsampling - 1; j++)
            {
                s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
                x += s->channels;
            }

            s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
            x += s->channels;
        }

        for (i = 0; i < s->num_taps; i++)
            s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
    }

    switch(s->decorrelation)
    {
        case MID_SIDE:
            for (i = 0; i < s->frame_size; i += s->channels)
            {
                s->int_samples[i+1] += shift(s->int_samples[i], 1);
                s->int_samples[i] -= s->int_samples[i+1];
            }
            break;
        case LEFT_SIDE:
            for (i = 0; i < s->frame_size; i += s->channels)
                s->int_samples[i+1] += s->int_samples[i];
            break;
        case RIGHT_SIDE:
            for (i = 0; i < s->frame_size; i += s->channels)
                s->int_samples[i] += s->int_samples[i+1];
            break;
    }

    if (!s->lossless)
        for (i = 0; i < s->frame_size; i++)
            s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);

    // internal -> short
    for (i = 0; i < s->frame_size; i++)
        samples[i] = av_clip_int16(s->int_samples[i]);

    align_get_bits(&gb);

    *data_size = s->frame_size * 2;

    return (get_bits_count(&gb)+7)/8;
}
#endif /* CONFIG_SONIC_DECODER */

#if CONFIG_SONIC_ENCODER
AVCodec sonic_encoder = {
    "sonic",
    CODEC_TYPE_AUDIO,
    CODEC_ID_SONIC,
    sizeof(SonicContext),
    sonic_encode_init,
    sonic_encode_frame,
    sonic_encode_close,
    NULL,
    .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
};
#endif

#if CONFIG_SONIC_LS_ENCODER
AVCodec sonic_ls_encoder = {
    "sonicls",
    CODEC_TYPE_AUDIO,
    CODEC_ID_SONIC_LS,
    sizeof(SonicContext),
    sonic_encode_init,
    sonic_encode_frame,
    sonic_encode_close,
    NULL,
    .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
};
#endif

#if CONFIG_SONIC_DECODER
AVCodec sonic_decoder = {
    "sonic",
    CODEC_TYPE_AUDIO,
    CODEC_ID_SONIC,
    sizeof(SonicContext),
    sonic_decode_init,
    NULL,
    sonic_decode_close,
    sonic_decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
};
#endif