FFmpeg/libavcodec/sonic.c

/*
 * Simple free lossless/lossy audio codec
 * Copyright (c) 2004 Alex Beregszaszi
 *
 * This library 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 of the License, or (at your option) any later version.
 *
 * This library 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 this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */
#include "avcodec.h"
#include "bitstream.h"
#include "golomb.h"

/**
 * @file 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;
}

// 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);
}

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

#ifdef CONFIG_ENCODERS

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 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 -ENOMEM;
    avctx->coded_frame->key_frame = 1;
    avctx->frame_size = s->block_align*s->downsampling;

    return 0;
}

static 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_ENCODERS

static 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);

    return 0;
}

static 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,
			    uint8_t *buf, int buf_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++)
    {
	if (s->int_samples[i] > 32767)
	    samples[i] = 32767;
	else if (s->int_samples[i] < -32768)
	    samples[i] = -32768;
	else
	    samples[i] = s->int_samples[i];
    }

    align_get_bits(&gb);

    *data_size = s->frame_size * 2;

    return (get_bits_count(&gb)+7)/8;
}

#ifdef CONFIG_ENCODERS
AVCodec sonic_encoder = {
    "sonic",
    CODEC_TYPE_AUDIO,
    CODEC_ID_SONIC,
    sizeof(SonicContext),
    sonic_encode_init,
    sonic_encode_frame,
    sonic_encode_close,
    NULL,
};

AVCodec sonic_ls_encoder = {
    "sonicls",
    CODEC_TYPE_AUDIO,
    CODEC_ID_SONIC_LS,
    sizeof(SonicContext),
    sonic_encode_init,
    sonic_encode_frame,
    sonic_encode_close,
    NULL,
};
#endif

#ifdef CONFIG_DECODERS
AVCodec sonic_decoder = {
    "sonic",
    CODEC_TYPE_AUDIO,
    CODEC_ID_SONIC,
    sizeof(SonicContext),
    sonic_decode_init,
    NULL,
    sonic_decode_close,
    sonic_decode_frame,
};
#endif