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* Simple free lossless/lossy audio codec
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* Copyright (c) 2004 Alex Beregszaszi
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* This file is part of FFmpeg.
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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#include "bitstream.h"
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* Simple free lossless/lossy audio codec
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* Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
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* Written and designed by Alex Beregszaszi
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* - CABAC put/get_symbol
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* - independent quantizer for channels
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* - >2 channels support
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* - more decorrelation types
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* - more tap_quant tests
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* - selectable intlist writers/readers (bonk-style, golomb, cabac)
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#define MAX_CHANNELS 2
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typedef struct SonicContext {
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int lossless, decorrelation;
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int num_taps, downsampling;
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int channels, samplerate, block_align, frame_size;
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int *coded_samples[MAX_CHANNELS];
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int *predictor_state[MAX_CHANNELS];
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#define LATTICE_SHIFT 10
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#define SAMPLE_SHIFT 4
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#define LATTICE_FACTOR (1 << LATTICE_SHIFT)
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#define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)
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#define BASE_QUANT 0.6
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#define RATE_VARIATION 3.0
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static inline int divide(int a, int b)
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return -( (-a + b/2)/b );
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static inline int shift(int a,int b)
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return (a+(1<<(b-1))) >> b;
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static inline int shift_down(int a,int b)
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return (a>>b)+((a<0)?1:0);
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static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
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for (i = 0; i < entries; i++)
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set_se_golomb(pb, buf[i]);
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static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
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for (i = 0; i < entries; i++)
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buf[i] = get_se_golomb(gb);
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#define ADAPT_LEVEL 8
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static int bits_to_store(uint64_t x)
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static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max)
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bits = bits_to_store(max);
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for (i = 0; i < bits-1; i++)
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put_bits(pb, 1, value & (1 << i));
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if ( (value | (1 << (bits-1))) <= max)
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put_bits(pb, 1, value & (1 << (bits-1)));
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static unsigned int read_uint_max(GetBitContext *gb, int max)
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int i, bits, value = 0;
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bits = bits_to_store(max);
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for (i = 0; i < bits-1; i++)
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if ( (value | (1<<(bits-1))) <= max)
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value += 1 << (bits-1);
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static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
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int i, j, x = 0, low_bits = 0, max = 0;
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int step = 256, pos = 0, dominant = 0, any = 0;
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copy = av_mallocz(4* entries);
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for (i = 0; i < entries; i++)
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energy += abs(buf[i]);
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low_bits = bits_to_store(energy / (entries * 2));
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put_bits(pb, 4, low_bits);
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for (i = 0; i < entries; i++)
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put_bits(pb, low_bits, abs(buf[i]));
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copy[i] = abs(buf[i]) >> low_bits;
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bits = av_mallocz(4* entries*max);
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for (i = 0; i <= max; i++)
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for (j = 0; j < entries; j++)
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bits[x++] = copy[j] > i;
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int steplet = step >> 8;
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if (pos + steplet > x)
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for (i = 0; i < steplet; i++)
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if (bits[i+pos] != dominant)
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put_bits(pb, 1, any);
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step += step / ADAPT_LEVEL;
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while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
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write_uint_max(pb, interloper, (step >> 8) - 1);
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pos += interloper + 1;
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step -= step / ADAPT_LEVEL;
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dominant = !dominant;
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for (i = 0; i < entries; i++)
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put_bits(pb, 1, buf[i] < 0);
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static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
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int i, low_bits = 0, x = 0;
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int n_zeros = 0, step = 256, dominant = 0;
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int pos = 0, level = 0;
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int *bits = av_mallocz(4* entries);
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low_bits = get_bits(gb, 4);
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for (i = 0; i < entries; i++)
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buf[i] = get_bits(gb, low_bits);
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// av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits);
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while (n_zeros < entries)
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int steplet = step >> 8;
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for (i = 0; i < steplet; i++)
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bits[x++] = dominant;
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step += step / ADAPT_LEVEL;
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int actual_run = read_uint_max(gb, steplet-1);
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// av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run);
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for (i = 0; i < actual_run; i++)
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bits[x++] = dominant;
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bits[x++] = !dominant;
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n_zeros += actual_run;
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step -= step / ADAPT_LEVEL;
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dominant = !dominant;
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// reconstruct unsigned values
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for (i = 0; n_zeros < entries; i++)
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level += 1 << low_bits;
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if (buf[pos] >= level)
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buf[pos] += 1 << low_bits;
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for (i = 0; i < entries; i++)
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if (buf[i] && get_bits1(gb))
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// av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos);
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static void predictor_init_state(int *k, int *state, int order)
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for (i = order-2; i >= 0; i--)
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int j, p, x = state[i];
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for (j = 0, p = i+1; p < order; j++,p++)
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int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);
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state[p] += shift_down(k[j]*x, LATTICE_SHIFT);
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static int predictor_calc_error(int *k, int *state, int order, int error)
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int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT);
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int *k_ptr = &(k[order-2]),
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*state_ptr = &(state[order-2]);
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for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--)
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int k_value = *k_ptr, state_value = *state_ptr;
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x -= shift_down(k_value * state_value, LATTICE_SHIFT);
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state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
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for (i = order-2; i >= 0; i--)
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x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
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state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT);
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// don't drift too far, to avoid overflows
403
if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16);
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if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16);
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#ifdef CONFIG_ENCODERS
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// Heavily modified Levinson-Durbin algorithm which
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// copes better with quantization, and calculates the
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// actual whitened result as it goes.
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static void modified_levinson_durbin(int *window, int window_entries,
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int *out, int out_entries, int channels, int *tap_quant)
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int *state = av_mallocz(4* window_entries);
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memcpy(state, window, 4* window_entries);
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for (i = 0; i < out_entries; i++)
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int step = (i+1)*channels, k, j;
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double xx = 0.0, xy = 0.0;
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int *x_ptr = &(window[step]), *state_ptr = &(state[0]);
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j = window_entries - step;
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for (;j>=0;j--,x_ptr++,state_ptr++)
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double x_value = *x_ptr, state_value = *state_ptr;
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xx += state_value*state_value;
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xy += x_value*state_value;
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for (j = 0; j <= (window_entries - step); j++);
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double stepval = window[step+j], stateval = window[j];
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// xx += (double)window[j]*(double)window[j];
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// xy += (double)window[step+j]*(double)window[j];
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xx += stateval*stateval;
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xy += stepval*stateval;
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k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));
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if (k > (LATTICE_FACTOR/tap_quant[i]))
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k = LATTICE_FACTOR/tap_quant[i];
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if (-k > (LATTICE_FACTOR/tap_quant[i]))
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k = -(LATTICE_FACTOR/tap_quant[i]);
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x_ptr = &(window[step]);
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state_ptr = &(state[0]);
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j = window_entries - step;
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for (;j>=0;j--,x_ptr++,state_ptr++)
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int x_value = *x_ptr, state_value = *state_ptr;
467
*x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
468
*state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
471
for (j=0; j <= (window_entries - step); j++)
473
int stepval = window[step+j], stateval=state[j];
474
window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
475
state[j] += shift_down(k * stepval, LATTICE_SHIFT);
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#endif /* CONFIG_ENCODERS */
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static int samplerate_table[] =
485
{ 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
487
#ifdef CONFIG_ENCODERS
489
static inline int code_samplerate(int samplerate)
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case 44100: return 0;
494
case 22050: return 1;
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case 11025: return 2;
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case 96000: return 3;
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case 48000: return 4;
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case 32000: return 5;
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case 24000: return 6;
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case 16000: return 7;
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static int sonic_encode_init(AVCodecContext *avctx)
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SonicContext *s = avctx->priv_data;
512
if (avctx->channels > MAX_CHANNELS)
514
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
515
return -1; /* only stereo or mono for now */
518
if (avctx->channels == 2)
519
s->decorrelation = MID_SIDE;
521
if (avctx->codec->id == CODEC_ID_SONIC_LS)
526
s->quantization = 0.0;
532
s->quantization = 1.0;
536
if ((s->num_taps < 32) || (s->num_taps > 1024) ||
537
((s->num_taps>>5)<<5 != s->num_taps))
539
av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
544
s->tap_quant = av_mallocz(4* s->num_taps);
545
for (i = 0; i < s->num_taps; i++)
546
s->tap_quant[i] = (int)(sqrt(i+1));
548
s->channels = avctx->channels;
549
s->samplerate = avctx->sample_rate;
551
s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling;
552
s->frame_size = s->channels*s->block_align*s->downsampling;
554
s->tail = av_mallocz(4* s->num_taps*s->channels);
557
s->tail_size = s->num_taps*s->channels;
559
s->predictor_k = av_mallocz(4 * s->num_taps);
563
for (i = 0; i < s->channels; i++)
565
s->coded_samples[i] = av_mallocz(4* s->block_align);
566
if (!s->coded_samples[i])
570
s->int_samples = av_mallocz(4* s->frame_size);
572
s->window_size = ((2*s->tail_size)+s->frame_size);
573
s->window = av_mallocz(4* s->window_size);
577
avctx->extradata = av_mallocz(16);
578
if (!avctx->extradata)
580
init_put_bits(&pb, avctx->extradata, 16*8);
582
put_bits(&pb, 2, version); // version
585
put_bits(&pb, 2, s->channels);
586
put_bits(&pb, 4, code_samplerate(s->samplerate));
588
put_bits(&pb, 1, s->lossless);
590
put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
591
put_bits(&pb, 2, s->decorrelation);
592
put_bits(&pb, 2, s->downsampling);
593
put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024
594
put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table
597
avctx->extradata_size = put_bits_count(&pb)/8;
599
av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
600
version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
602
avctx->coded_frame = avcodec_alloc_frame();
603
if (!avctx->coded_frame)
604
return AVERROR(ENOMEM);
605
avctx->coded_frame->key_frame = 1;
606
avctx->frame_size = s->block_align*s->downsampling;
611
static int sonic_encode_close(AVCodecContext *avctx)
613
SonicContext *s = avctx->priv_data;
616
av_freep(&avctx->coded_frame);
618
for (i = 0; i < s->channels; i++)
619
av_free(s->coded_samples[i]);
621
av_free(s->predictor_k);
623
av_free(s->tap_quant);
625
av_free(s->int_samples);
630
static int sonic_encode_frame(AVCodecContext *avctx,
631
uint8_t *buf, int buf_size, void *data)
633
SonicContext *s = avctx->priv_data;
635
int i, j, ch, quant = 0, x = 0;
636
short *samples = data;
638
init_put_bits(&pb, buf, buf_size*8);
641
for (i = 0; i < s->frame_size; i++)
642
s->int_samples[i] = samples[i];
645
for (i = 0; i < s->frame_size; i++)
646
s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
648
switch(s->decorrelation)
651
for (i = 0; i < s->frame_size; i += s->channels)
653
s->int_samples[i] += s->int_samples[i+1];
654
s->int_samples[i+1] -= shift(s->int_samples[i], 1);
658
for (i = 0; i < s->frame_size; i += s->channels)
659
s->int_samples[i+1] -= s->int_samples[i];
662
for (i = 0; i < s->frame_size; i += s->channels)
663
s->int_samples[i] -= s->int_samples[i+1];
667
memset(s->window, 0, 4* s->window_size);
669
for (i = 0; i < s->tail_size; i++)
670
s->window[x++] = s->tail[i];
672
for (i = 0; i < s->frame_size; i++)
673
s->window[x++] = s->int_samples[i];
675
for (i = 0; i < s->tail_size; i++)
678
for (i = 0; i < s->tail_size; i++)
679
s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];
682
modified_levinson_durbin(s->window, s->window_size,
683
s->predictor_k, s->num_taps, s->channels, s->tap_quant);
684
if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0)
687
for (ch = 0; ch < s->channels; ch++)
690
for (i = 0; i < s->block_align; i++)
693
for (j = 0; j < s->downsampling; j++, x += s->channels)
695
s->coded_samples[ch][i] = sum;
699
// simple rate control code
702
double energy1 = 0.0, energy2 = 0.0;
703
for (ch = 0; ch < s->channels; ch++)
705
for (i = 0; i < s->block_align; i++)
707
double sample = s->coded_samples[ch][i];
708
energy2 += sample*sample;
709
energy1 += fabs(sample);
713
energy2 = sqrt(energy2/(s->channels*s->block_align));
714
energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align);
716
// increase bitrate when samples are like a gaussian distribution
717
// reduce bitrate when samples are like a two-tailed exponential distribution
719
if (energy2 > energy1)
720
energy2 += (energy2-energy1)*RATE_VARIATION;
722
quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
723
// av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);
730
set_ue_golomb(&pb, quant);
732
quant *= SAMPLE_FACTOR;
735
// write out coded samples
736
for (ch = 0; ch < s->channels; ch++)
739
for (i = 0; i < s->block_align; i++)
740
s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant);
742
if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0)
746
// av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
749
return (put_bits_count(&pb)+7)/8;
751
#endif //CONFIG_ENCODERS
753
#ifdef CONFIG_DECODERS
754
static int sonic_decode_init(AVCodecContext *avctx)
756
SonicContext *s = avctx->priv_data;
760
s->channels = avctx->channels;
761
s->samplerate = avctx->sample_rate;
763
if (!avctx->extradata)
765
av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
769
init_get_bits(&gb, avctx->extradata, avctx->extradata_size);
771
version = get_bits(&gb, 2);
774
av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
780
s->channels = get_bits(&gb, 2);
781
s->samplerate = samplerate_table[get_bits(&gb, 4)];
782
av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
783
s->channels, s->samplerate);
786
if (s->channels > MAX_CHANNELS)
788
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
792
s->lossless = get_bits1(&gb);
794
skip_bits(&gb, 3); // XXX FIXME
795
s->decorrelation = get_bits(&gb, 2);
797
s->downsampling = get_bits(&gb, 2);
798
s->num_taps = (get_bits(&gb, 5)+1)<<5;
799
if (get_bits1(&gb)) // XXX FIXME
800
av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
802
s->block_align = (int)(2048.0*(s->samplerate/44100))/s->downsampling;
803
s->frame_size = s->channels*s->block_align*s->downsampling;
804
// avctx->frame_size = s->block_align;
806
av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
807
version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
810
s->tap_quant = av_mallocz(4* s->num_taps);
811
for (i = 0; i < s->num_taps; i++)
812
s->tap_quant[i] = (int)(sqrt(i+1));
814
s->predictor_k = av_mallocz(4* s->num_taps);
816
for (i = 0; i < s->channels; i++)
818
s->predictor_state[i] = av_mallocz(4* s->num_taps);
819
if (!s->predictor_state[i])
823
for (i = 0; i < s->channels; i++)
825
s->coded_samples[i] = av_mallocz(4* s->block_align);
826
if (!s->coded_samples[i])
829
s->int_samples = av_mallocz(4* s->frame_size);
834
static int sonic_decode_close(AVCodecContext *avctx)
836
SonicContext *s = avctx->priv_data;
839
av_free(s->int_samples);
840
av_free(s->tap_quant);
841
av_free(s->predictor_k);
843
for (i = 0; i < s->channels; i++)
845
av_free(s->predictor_state[i]);
846
av_free(s->coded_samples[i]);
852
static int sonic_decode_frame(AVCodecContext *avctx,
853
void *data, int *data_size,
854
uint8_t *buf, int buf_size)
856
SonicContext *s = avctx->priv_data;
859
short *samples = data;
861
if (buf_size == 0) return 0;
863
// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
865
init_get_bits(&gb, buf, buf_size*8);
867
intlist_read(&gb, s->predictor_k, s->num_taps, 0);
870
for (i = 0; i < s->num_taps; i++)
871
s->predictor_k[i] *= s->tap_quant[i];
876
quant = get_ue_golomb(&gb) * SAMPLE_FACTOR;
878
// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
880
for (ch = 0; ch < s->channels; ch++)
884
predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);
886
intlist_read(&gb, s->coded_samples[ch], s->block_align, 1);
888
for (i = 0; i < s->block_align; i++)
890
for (j = 0; j < s->downsampling - 1; j++)
892
s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
896
s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
900
for (i = 0; i < s->num_taps; i++)
901
s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
904
switch(s->decorrelation)
907
for (i = 0; i < s->frame_size; i += s->channels)
909
s->int_samples[i+1] += shift(s->int_samples[i], 1);
910
s->int_samples[i] -= s->int_samples[i+1];
914
for (i = 0; i < s->frame_size; i += s->channels)
915
s->int_samples[i+1] += s->int_samples[i];
918
for (i = 0; i < s->frame_size; i += s->channels)
919
s->int_samples[i] += s->int_samples[i+1];
924
for (i = 0; i < s->frame_size; i++)
925
s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);
928
for (i = 0; i < s->frame_size; i++)
930
if (s->int_samples[i] > 32767)
932
else if (s->int_samples[i] < -32768)
935
samples[i] = s->int_samples[i];
940
*data_size = s->frame_size * 2;
942
return (get_bits_count(&gb)+7)/8;
946
#ifdef CONFIG_ENCODERS
947
AVCodec sonic_encoder = {
951
sizeof(SonicContext),
958
AVCodec sonic_ls_encoder = {
962
sizeof(SonicContext),
970
#ifdef CONFIG_DECODERS
971
AVCodec sonic_decoder = {
975
sizeof(SonicContext),
added
removed
ffmpeg/libavcodec/sonic.c
