/*
  * 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

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

 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 = AV_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;
 }

 AVCodec ff_sonic_decoder = {

     "sonic",

     AVMEDIA_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 /* CONFIG_SONIC_DECODER */

 #if CONFIG_SONIC_ENCODER

 AVCodec ff_sonic_encoder = {

     "sonic",

     AVMEDIA_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 ff_sonic_ls_encoder = {

     "sonicls",

     AVMEDIA_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