forked from KolibriOS/kolibrios
990 lines
26 KiB
C
990 lines
26 KiB
C
/*
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* Simple free lossless/lossy audio codec
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* Copyright (c) 2004 Alex Beregszaszi
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*
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* This file is part of FFmpeg.
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*
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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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*
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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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*
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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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*/
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#include "avcodec.h"
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#include "get_bits.h"
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#include "golomb.h"
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#include "internal.h"
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/**
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* @file
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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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*
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* TODO:
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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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*/
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#define MAX_CHANNELS 2
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#define MID_SIDE 0
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#define LEFT_SIDE 1
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#define RIGHT_SIDE 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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double quantization;
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int channels, samplerate, block_align, frame_size;
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int *tap_quant;
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int *int_samples;
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int *coded_samples[MAX_CHANNELS];
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// for encoding
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int *tail;
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int tail_size;
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int *window;
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int window_size;
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// for decoding
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int *predictor_k;
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int *predictor_state[MAX_CHANNELS];
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} SonicContext;
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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 shift(int a,int b)
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{
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return (a+(1<<(b-1))) >> b;
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}
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static inline int shift_down(int a,int b)
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{
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return (a>>b)+(a<0);
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}
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#if 1
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static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
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{
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int i;
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for (i = 0; i < entries; i++)
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set_se_golomb(pb, buf[i]);
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return 1;
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}
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static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
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{
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int i;
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for (i = 0; i < entries; i++)
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buf[i] = get_se_golomb(gb);
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return 1;
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}
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#else
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#define ADAPT_LEVEL 8
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static int bits_to_store(uint64_t x)
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{
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int res = 0;
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while(x)
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{
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res++;
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x >>= 1;
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}
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return res;
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}
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static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max)
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{
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int i, bits;
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if (!max)
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return;
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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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}
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static unsigned int read_uint_max(GetBitContext *gb, int max)
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{
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int i, bits, value = 0;
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if (!max)
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return 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 (get_bits1(gb))
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value += 1 << i;
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if ( (value | (1<<(bits-1))) <= max)
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if (get_bits1(gb))
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value += 1 << (bits-1);
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return value;
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}
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static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
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{
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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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int *copy, *bits;
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copy = av_calloc(entries, sizeof(*copy));
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if (!copy)
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return AVERROR(ENOMEM);
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if (base_2_part)
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{
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int energy = 0;
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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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if (low_bits > 15)
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low_bits = 15;
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put_bits(pb, 4, low_bits);
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}
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for (i = 0; i < entries; i++)
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{
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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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if (copy[i] > max)
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max = abs(copy[i]);
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}
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bits = av_calloc(entries*max, sizeof(*bits));
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if (!bits)
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{
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// av_free(copy);
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return AVERROR(ENOMEM);
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}
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for (i = 0; i <= max; i++)
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{
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for (j = 0; j < entries; j++)
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if (copy[j] >= i)
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bits[x++] = copy[j] > i;
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}
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// store bitstream
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while (pos < x)
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{
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int steplet = step >> 8;
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if (pos + steplet > x)
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steplet = x - pos;
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for (i = 0; i < steplet; i++)
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if (bits[i+pos] != dominant)
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any = 1;
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put_bits(pb, 1, any);
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if (!any)
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{
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pos += steplet;
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step += step / ADAPT_LEVEL;
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}
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else
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{
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int interloper = 0;
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while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
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interloper++;
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// note change
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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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}
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if (step < 256)
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{
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step = 65536 / step;
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dominant = !dominant;
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}
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}
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// store signs
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for (i = 0; i < entries; i++)
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if (buf[i])
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put_bits(pb, 1, buf[i] < 0);
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// av_free(bits);
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// av_free(copy);
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return 0;
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}
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static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
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{
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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_calloc(entries, sizeof(*bits));
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if (!bits)
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return AVERROR(ENOMEM);
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if (base_2_part)
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{
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low_bits = get_bits(gb, 4);
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if (low_bits)
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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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}
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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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{
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int steplet = step >> 8;
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if (!get_bits1(gb))
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{
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for (i = 0; i < steplet; i++)
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bits[x++] = dominant;
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if (!dominant)
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n_zeros += steplet;
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step += step / ADAPT_LEVEL;
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}
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else
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{
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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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if (!dominant)
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n_zeros += actual_run;
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else
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n_zeros++;
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step -= step / ADAPT_LEVEL;
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}
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if (step < 256)
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{
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step = 65536 / step;
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dominant = !dominant;
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}
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}
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// reconstruct unsigned values
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n_zeros = 0;
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for (i = 0; n_zeros < entries; i++)
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{
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while(1)
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{
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if (pos >= entries)
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{
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pos = 0;
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level += 1 << low_bits;
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}
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if (buf[pos] >= level)
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break;
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pos++;
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}
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if (bits[i])
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buf[pos] += 1 << low_bits;
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else
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n_zeros++;
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pos++;
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}
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// av_free(bits);
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// read signs
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for (i = 0; i < entries; i++)
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if (buf[i] && get_bits1(gb))
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buf[i] = -buf[i];
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// av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos);
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return 0;
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}
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#endif
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static void predictor_init_state(int *k, int *state, int order)
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{
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int i;
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for (i = order-2; i >= 0; i--)
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{
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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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{
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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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x = tmp;
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}
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}
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}
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static int predictor_calc_error(int *k, int *state, int order, int error)
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{
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int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT);
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#if 1
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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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{
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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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}
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#else
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for (i = order-2; i >= 0; i--)
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{
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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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}
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#endif
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// don't drift too far, to avoid overflows
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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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state[0] = x;
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return x;
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}
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#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
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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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{
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int i;
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int *state = av_calloc(window_entries, sizeof(*state));
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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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{
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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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#if 1
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int *x_ptr = &(window[step]);
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int *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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{
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double x_value = *x_ptr;
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double 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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}
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#else
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for (j = 0; j <= (window_entries - step); j++);
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{
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double stepval = window[step+j];
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double 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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}
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#endif
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if (xx == 0.0)
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k = 0;
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else
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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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out[i] = k;
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k *= tap_quant[i];
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#if 1
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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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{
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int x_value = *x_ptr;
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int state_value = *state_ptr;
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*x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
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*state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
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}
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#else
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for (j=0; j <= (window_entries - step); j++)
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{
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int stepval = window[step+j];
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int stateval=state[j];
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window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
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state[j] += shift_down(k * stepval, LATTICE_SHIFT);
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}
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#endif
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}
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av_free(state);
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}
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static inline int code_samplerate(int samplerate)
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{
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switch (samplerate)
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{
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case 44100: return 0;
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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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case 8000: return 8;
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}
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return AVERROR(EINVAL);
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}
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static av_cold int sonic_encode_init(AVCodecContext *avctx)
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{
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SonicContext *s = avctx->priv_data;
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PutBitContext pb;
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int i, version = 0;
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if (avctx->channels > MAX_CHANNELS)
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{
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av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
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return AVERROR(EINVAL); /* only stereo or mono for now */
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}
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if (avctx->channels == 2)
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s->decorrelation = MID_SIDE;
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else
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s->decorrelation = 3;
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if (avctx->codec->id == AV_CODEC_ID_SONIC_LS)
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{
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s->lossless = 1;
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s->num_taps = 32;
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s->downsampling = 1;
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s->quantization = 0.0;
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}
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else
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{
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s->num_taps = 128;
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s->downsampling = 2;
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s->quantization = 1.0;
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}
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// max tap 2048
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if (s->num_taps < 32 || s->num_taps > 1024 || s->num_taps % 32) {
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av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
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return AVERROR_INVALIDDATA;
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}
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// generate taps
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s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
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for (i = 0; i < s->num_taps; i++)
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s->tap_quant[i] = ff_sqrt(i+1);
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s->channels = avctx->channels;
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s->samplerate = avctx->sample_rate;
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s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
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s->frame_size = s->channels*s->block_align*s->downsampling;
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s->tail_size = s->num_taps*s->channels;
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s->tail = av_calloc(s->tail_size, sizeof(*s->tail));
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if (!s->tail)
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return AVERROR(ENOMEM);
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s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) );
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if (!s->predictor_k)
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return AVERROR(ENOMEM);
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for (i = 0; i < s->channels; i++)
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{
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s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
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if (!s->coded_samples[i])
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return AVERROR(ENOMEM);
|
|
}
|
|
|
|
s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
|
|
|
|
s->window_size = ((2*s->tail_size)+s->frame_size);
|
|
s->window = av_calloc(s->window_size, sizeof(*s->window));
|
|
if (!s->window)
|
|
return AVERROR(ENOMEM);
|
|
|
|
avctx->extradata = av_mallocz(16);
|
|
if (!avctx->extradata)
|
|
return AVERROR(ENOMEM);
|
|
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->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;
|
|
|
|
for (i = 0; i < s->channels; i++)
|
|
av_freep(&s->coded_samples[i]);
|
|
|
|
av_freep(&s->predictor_k);
|
|
av_freep(&s->tail);
|
|
av_freep(&s->tap_quant);
|
|
av_freep(&s->window);
|
|
av_freep(&s->int_samples);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
|
const AVFrame *frame, int *got_packet_ptr)
|
|
{
|
|
SonicContext *s = avctx->priv_data;
|
|
PutBitContext pb;
|
|
int i, j, ch, quant = 0, x = 0;
|
|
int ret;
|
|
const short *samples = (const int16_t*)frame->data[0];
|
|
|
|
if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0)
|
|
return ret;
|
|
|
|
init_put_bits(&pb, avpkt->data, avpkt->size);
|
|
|
|
// 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 ((ret = intlist_write(&pb, s->predictor_k, s->num_taps, 0)) < 0)
|
|
return ret;
|
|
|
|
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);
|
|
|
|
quant = av_clip(quant, 1, 65534);
|
|
|
|
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] = ROUNDED_DIV(s->coded_samples[ch][i], quant);
|
|
|
|
if ((ret = intlist_write(&pb, s->coded_samples[ch], s->block_align, 1)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
// av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
|
|
|
|
flush_put_bits(&pb);
|
|
avpkt->size = (put_bits_count(&pb)+7)/8;
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
#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 AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
init_get_bits8(&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 AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
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 AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
s->lossless = get_bits1(&gb);
|
|
if (!s->lossless)
|
|
skip_bits(&gb, 3); // XXX FIXME
|
|
s->decorrelation = get_bits(&gb, 2);
|
|
if (s->decorrelation != 3 && s->channels != 2) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
s->downsampling = get_bits(&gb, 2);
|
|
if (!s->downsampling) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
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 = 2048LL*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_calloc(s->num_taps, sizeof(*s->tap_quant));
|
|
for (i = 0; i < s->num_taps; i++)
|
|
s->tap_quant[i] = ff_sqrt(i+1);
|
|
|
|
s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k));
|
|
|
|
for (i = 0; i < s->channels; i++)
|
|
{
|
|
s->predictor_state[i] = av_calloc(s->num_taps, sizeof(**s->predictor_state));
|
|
if (!s->predictor_state[i])
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
for (i = 0; i < s->channels; i++)
|
|
{
|
|
s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
|
|
if (!s->coded_samples[i])
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
|
|
|
|
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_freep(&s->int_samples);
|
|
av_freep(&s->tap_quant);
|
|
av_freep(&s->predictor_k);
|
|
|
|
for (i = 0; i < s->channels; i++)
|
|
{
|
|
av_freep(&s->predictor_state[i]);
|
|
av_freep(&s->coded_samples[i]);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int sonic_decode_frame(AVCodecContext *avctx,
|
|
void *data, int *got_frame_ptr,
|
|
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, ret;
|
|
int16_t *samples;
|
|
AVFrame *frame = data;
|
|
|
|
if (buf_size == 0) return 0;
|
|
|
|
frame->nb_samples = s->frame_size / avctx->channels;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
samples = (int16_t *)frame->data[0];
|
|
|
|
// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
|
|
|
|
init_get_bits8(&gb, buf, buf_size);
|
|
|
|
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);
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return (get_bits_count(&gb)+7)/8;
|
|
}
|
|
|
|
AVCodec ff_sonic_decoder = {
|
|
.name = "sonic",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Sonic"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_SONIC,
|
|
.priv_data_size = sizeof(SonicContext),
|
|
.init = sonic_decode_init,
|
|
.close = sonic_decode_close,
|
|
.decode = sonic_decode_frame,
|
|
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_EXPERIMENTAL,
|
|
};
|
|
#endif /* CONFIG_SONIC_DECODER */
|
|
|
|
#if CONFIG_SONIC_ENCODER
|
|
AVCodec ff_sonic_encoder = {
|
|
.name = "sonic",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Sonic"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_SONIC,
|
|
.priv_data_size = sizeof(SonicContext),
|
|
.init = sonic_encode_init,
|
|
.encode2 = sonic_encode_frame,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
|
|
.capabilities = CODEC_CAP_EXPERIMENTAL,
|
|
.close = sonic_encode_close,
|
|
};
|
|
#endif
|
|
|
|
#if CONFIG_SONIC_LS_ENCODER
|
|
AVCodec ff_sonic_ls_encoder = {
|
|
.name = "sonicls",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_SONIC_LS,
|
|
.priv_data_size = sizeof(SonicContext),
|
|
.init = sonic_encode_init,
|
|
.encode2 = sonic_encode_frame,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
|
|
.capabilities = CODEC_CAP_EXPERIMENTAL,
|
|
.close = sonic_encode_close,
|
|
};
|
|
#endif
|