new opensource lossy/lossless audio codec based on speech compression...

new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)

Originally committed as revision 3405 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/Makefile

@@ -20,7 +20,7 @@ OBJS= common.o utils.o mem.o allcodecs.o \
       roqvideo.o dpcm.o interplayvideo.o xan.o rpza.o cinepak.o msrle.o \
       msvideo1.o vqavideo.o idcinvideo.o adx.o rational.o faandct.o 8bps.o \
       smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o g726.o \
-      flac.o vp3dsp.o integer.o snow.o tscc.o
+      flac.o vp3dsp.o integer.o snow.o tscc.o sonic.o
 
 ifeq ($(AMR_NB),yes)
 ifeq ($(AMR_NB_FIXED),yes)

libavcodec/allcodecs.c

@@ -80,6 +80,8 @@ void avcodec_register_all(void)
     register_avcodec(&snow_encoder);
     register_avcodec(&zlib_encoder);
     register_avcodec(&dvvideo_encoder);
+    register_avcodec(&sonic_encoder);
+    register_avcodec(&sonic_ls_encoder);
 #endif /* CONFIG_ENCODERS */
     register_avcodec(&rawvideo_encoder);
     register_avcodec(&rawvideo_decoder);
@@ -153,6 +155,7 @@ void avcodec_register_all(void)
     register_avcodec(&vmdaudio_decoder);
     register_avcodec(&mszh_decoder);
     register_avcodec(&zlib_decoder);
+    register_avcodec(&sonic_decoder);
 #ifdef CONFIG_AC3
     register_avcodec(&ac3_decoder);
 #endif

libavcodec/avcodec.h

@@ -97,6 +97,8 @@ enum CodecID {
     CODEC_ID_VMDAUDIO,
     CODEC_ID_MSZH,
     CODEC_ID_ZLIB,
+    CODEC_ID_SONIC,
+    CODEC_ID_SONIC_LS,
     CODEC_ID_QTRLE,
     CODEC_ID_SNOW,
     CODEC_ID_TSCC,
@@ -1770,6 +1772,8 @@ extern AVCodec ffv1_encoder;
 extern AVCodec snow_encoder;
 extern AVCodec mdec_encoder;
 extern AVCodec zlib_encoder;
+extern AVCodec sonic_encoder;
+extern AVCodec sonic_ls_encoder;
 extern AVCodec svq1_encoder;
 
 extern AVCodec h263_decoder;
@@ -1843,6 +1847,7 @@ extern AVCodec ra_288_decoder;
 extern AVCodec roq_dpcm_decoder;
 extern AVCodec interplay_dpcm_decoder;
 extern AVCodec xan_dpcm_decoder;
+extern AVCodec sonic_decoder;
 extern AVCodec qtrle_decoder;
 extern AVCodec flac_decoder;
 extern AVCodec tscc_decoder;

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 "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
+
+typedef struct SonicContext {
+    int lossless, mid_side;
+    
+    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)
+        return -1; /* only stereo or mono for now */
+
+    if (avctx->channels == 2)
+	s->mid_side = 1;
+    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, 1, s->mid_side);
+    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 ms: %d taps: %d block: %d frame: %d downsamp: %d\n",
+	version, s->lossless, s->mid_side, 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++)
+    {
+	if (samples[i] < 0)
+	    s->int_samples[i] = samples[i]+32768;
+	else
+	    s->int_samples[i] = samples[i]-32768;
+    }
+
+    if (!s->lossless)
+	for (i = 0; i < s->frame_size; i++)
+	    s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
+
+    if (s->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);
+	}
+
+    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->mid_side = get_bits1(&gb);
+
+    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 ms: %d taps: %d block: %d frame: %d downsamp: %d\n",
+	version, s->lossless, s->mid_side, 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,
+			    int16_t *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];
+    }
+    
+    if (s->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];
+	}
+
+    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);
+
+//    if (buf_size != (get_bits_count(&gb)+7)/8)
+//	av_log(NULL, AV_LOG_INFO, "buf_size (%d) and used bytes (%d) differs\n", buf_size, (get_bits_count(&gb)+7)/8);
+
+    *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