diff --git a/configure b/configure
index 1494994d253ebe646885f7db2529fd24d777461d..7b89564b25ed3b24261ffa4df266e1ee0b804b76 100755
--- a/configure
+++ b/configure
@@ -1324,9 +1324,6 @@ shorten_decoder_select="golomb"
 sipr_decoder_select="lsp"
 snow_decoder_select="dwt"
 snow_encoder_select="aandct dwt"
-sonic_decoder_select="golomb"
-sonic_encoder_select="golomb"
-sonic_ls_encoder_select="golomb"
 svq1_encoder_select="aandct"
 svq3_decoder_select="golomb h264dsp h264pred"
 svq3_decoder_suggest="zlib"
diff --git a/doc/general.texi b/doc/general.texi
index 3ef4d678ebe7b91023e19b7b8041fbeb4fe4e17d..f6c61a23426d8a4329dc0ebf6864aa690dc0fa33 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -665,10 +665,6 @@ following image formats are supported:
 @item Sierra VMD audio       @tab     @tab  X
     @tab Used in Sierra VMD files.
 @item Smacker audio          @tab     @tab  X
-@item Sonic                  @tab  X  @tab  X
-    @tab experimental codec
-@item Sonic lossless         @tab  X  @tab  X
-    @tab experimental codec
 @item Speex                  @tab     @tab  E
     @tab supported through external library libspeex
 @item True Audio (TTA)       @tab     @tab  X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 21bdbf42d1dd5de863a89099ffcf73cb7caf1593..d72a34062e471ac26a780f0af2df13a160fabae9 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -337,9 +337,6 @@ OBJS-$(CONFIG_SNOW_ENCODER)            += snow.o rangecoder.o motion_est.o \
                                           ituh263enc.o mpegvideo_enc.o     \
                                           mpeg12data.o
 OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
-OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
-OBJS-$(CONFIG_SONIC_ENCODER)           += sonic.o
-OBJS-$(CONFIG_SONIC_LS_ENCODER)        += sonic.o
 OBJS-$(CONFIG_SP5X_DECODER)            += sp5xdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_SRT_DECODER)             += srtdec.o ass.o
 OBJS-$(CONFIG_SUNRAST_DECODER)         += sunrast.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 767a50283c994f8f2aa6eb8e1ed20445ee184809..8de6ad842944846febd62529a944660ad0235e6c 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -266,8 +266,6 @@ void avcodec_register_all(void)
     REGISTER_DECODER (SHORTEN, shorten);
     REGISTER_DECODER (SIPR, sipr);
     REGISTER_DECODER (SMACKAUD, smackaud);
-    REGISTER_ENCDEC  (SONIC, sonic);
-    REGISTER_ENCODER (SONIC_LS, sonic_ls);
     REGISTER_DECODER (TRUEHD, truehd);
     REGISTER_DECODER (TRUESPEECH, truespeech);
     REGISTER_DECODER (TTA, tta);
diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c
deleted file mode 100644
index bd6691f61b2c1cbfe88273d5c709384a5232f0a8..0000000000000000000000000000000000000000
--- a/libavcodec/sonic.c
+++ /dev/null
@@ -1,977 +0,0 @@
-/*
- * Simple free lossless/lossy audio codec
- * Copyright (c) 2004 Alex Beregszaszi
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; 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