view sonic.c @ 2601:c31a28f27d9a libavcodec

increasing precission of the quantization parameter this is needed as the quantization stepsize for each subband is also in this precission and insignificant changes to the wavelet like scaling its coefficients slightly differently would lead to wildly variing PSNR and bitrate note, a encoder could also simply choose to leave the least significant bits of the quantization parameters zero which would give the exact previous behaviour except a y very tiny number of bits in the header

/*
 * 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