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/* |
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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 "config_components.h" |
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#include "avcodec.h" |
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#include "codec_internal.h" |
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#include "encode.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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#include "put_golomb.h" |
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#include "rangecoder.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 version; |
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int minor_version; |
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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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static av_always_inline av_flatten void put_symbol(RangeCoder *c, uint8_t *state, int v, int is_signed, uint64_t rc_stat[256][2], uint64_t rc_stat2[32][2]){ |
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int i; |
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#define put_rac(C,S,B) \ |
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do{\ |
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if(rc_stat){\ |
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rc_stat[*(S)][B]++;\ |
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rc_stat2[(S)-state][B]++;\ |
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}\ |
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put_rac(C,S,B);\ |
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}while(0) |
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if(v){ |
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const int a= FFABS(v); |
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const int e= av_log2(a); |
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put_rac(c, state+0, 0); |
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if(e<=9){ |
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for(i=0; i<e; i++){ |
116 |
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put_rac(c, state+1+i, 1); //1..10 |
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} |
118 |
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put_rac(c, state+1+i, 0); |
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120 |
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for(i=e-1; i>=0; i--){ |
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put_rac(c, state+22+i, (a>>i)&1); //22..31 |
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} |
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if(is_signed) |
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put_rac(c, state+11 + e, v < 0); //11..21 |
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}else{ |
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for(i=0; i<e; i++){ |
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put_rac(c, state+1+FFMIN(i,9), 1); //1..10 |
129 |
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} |
130 |
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put_rac(c, state+1+9, 0); |
131 |
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132 |
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for(i=e-1; i>=0; i--){ |
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put_rac(c, state+22+FFMIN(i,9), (a>>i)&1); //22..31 |
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} |
135 |
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136 |
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if(is_signed) |
137 |
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put_rac(c, state+11 + 10, v < 0); //11..21 |
138 |
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} |
139 |
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}else{ |
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put_rac(c, state+0, 1); |
141 |
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} |
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#undef put_rac |
143 |
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} |
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static inline av_flatten int get_symbol(RangeCoder *c, uint8_t *state, int is_signed){ |
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if(get_rac(c, state+0)) |
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return 0; |
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else{ |
149 |
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int i, e; |
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unsigned a; |
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e= 0; |
152 |
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while(get_rac(c, state+1 + FFMIN(e,9))){ //1..10 |
153 |
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e++; |
154 |
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if (e > 31) |
155 |
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return AVERROR_INVALIDDATA; |
156 |
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} |
157 |
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158 |
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a= 1; |
159 |
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for(i=e-1; i>=0; i--){ |
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a += a + get_rac(c, state+22 + FFMIN(i,9)); //22..31 |
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} |
162 |
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163 |
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e= -(is_signed && get_rac(c, state+11 + FFMIN(e, 10))); //11..21 |
164 |
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return (a^e)-e; |
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} |
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} |
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#if 1 |
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static inline int intlist_write(RangeCoder *c, uint8_t *state, 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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put_symbol(c, state, buf[i], 1, NULL, NULL); |
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return 1; |
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} |
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static inline int intlist_read(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part) |
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{ |
181 |
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int i; |
182 |
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183 |
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for (i = 0; i < entries; i++) |
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buf[i] = get_symbol(c, state, 1); |
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186 |
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return 1; |
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} |
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#elif 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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193 |
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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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199 |
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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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203 |
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for (i = 0; i < entries; i++) |
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buf[i] = get_se_golomb(gb); |
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206 |
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return 1; |
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} |
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209 |
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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; |
216 |
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217 |
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while(x) |
218 |
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{ |
219 |
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res++; |
220 |
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x >>= 1; |
221 |
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} |
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return res; |
223 |
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} |
224 |
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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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229 |
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if (!max) |
230 |
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return; |
231 |
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232 |
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bits = bits_to_store(max); |
233 |
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234 |
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for (i = 0; i < bits-1; i++) |
235 |
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put_bits(pb, 1, value & (1 << i)); |
236 |
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237 |
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if ( (value | (1 << (bits-1))) <= max) |
238 |
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put_bits(pb, 1, value & (1 << (bits-1))); |
239 |
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} |
240 |
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241 |
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static unsigned int read_uint_max(GetBitContext *gb, int max) |
242 |
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{ |
243 |
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int i, bits, value = 0; |
244 |
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245 |
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if (!max) |
246 |
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return 0; |
247 |
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248 |
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bits = bits_to_store(max); |
249 |
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250 |
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for (i = 0; i < bits-1; i++) |
251 |
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if (get_bits1(gb)) |
252 |
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value += 1 << i; |
253 |
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254 |
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if ( (value | (1<<(bits-1))) <= max) |
255 |
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if (get_bits1(gb)) |
256 |
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value += 1 << (bits-1); |
257 |
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258 |
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return value; |
259 |
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} |
260 |
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261 |
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static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part) |
262 |
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{ |
263 |
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int i, j, x = 0, low_bits = 0, max = 0; |
264 |
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int step = 256, pos = 0, dominant = 0, any = 0; |
265 |
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int *copy, *bits; |
266 |
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267 |
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copy = av_calloc(entries, sizeof(*copy)); |
268 |
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if (!copy) |
269 |
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return AVERROR(ENOMEM); |
270 |
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271 |
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if (base_2_part) |
272 |
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{ |
273 |
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int energy = 0; |
274 |
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275 |
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for (i = 0; i < entries; i++) |
276 |
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energy += abs(buf[i]); |
277 |
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278 |
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low_bits = bits_to_store(energy / (entries * 2)); |
279 |
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if (low_bits > 15) |
280 |
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low_bits = 15; |
281 |
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282 |
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put_bits(pb, 4, low_bits); |
283 |
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} |
284 |
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285 |
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for (i = 0; i < entries; i++) |
286 |
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{ |
287 |
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put_bits(pb, low_bits, abs(buf[i])); |
288 |
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copy[i] = abs(buf[i]) >> low_bits; |
289 |
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if (copy[i] > max) |
290 |
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max = abs(copy[i]); |
291 |
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} |
292 |
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293 |
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bits = av_calloc(entries*max, sizeof(*bits)); |
294 |
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if (!bits) |
295 |
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{ |
296 |
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av_free(copy); |
297 |
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return AVERROR(ENOMEM); |
298 |
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} |
299 |
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300 |
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for (i = 0; i <= max; i++) |
301 |
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{ |
302 |
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for (j = 0; j < entries; j++) |
303 |
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if (copy[j] >= i) |
304 |
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bits[x++] = copy[j] > i; |
305 |
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} |
306 |
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307 |
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// store bitstream |
308 |
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while (pos < x) |
309 |
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{ |
310 |
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int steplet = step >> 8; |
311 |
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312 |
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if (pos + steplet > x) |
313 |
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steplet = x - pos; |
314 |
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315 |
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for (i = 0; i < steplet; i++) |
316 |
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if (bits[i+pos] != dominant) |
317 |
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any = 1; |
318 |
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319 |
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put_bits(pb, 1, any); |
320 |
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321 |
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if (!any) |
322 |
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{ |
323 |
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pos += steplet; |
324 |
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step += step / ADAPT_LEVEL; |
325 |
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} |
326 |
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else |
327 |
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{ |
328 |
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int interloper = 0; |
329 |
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330 |
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while (((pos + interloper) < x) && (bits[pos + interloper] == dominant)) |
331 |
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interloper++; |
332 |
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333 |
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// note change |
334 |
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write_uint_max(pb, interloper, (step >> 8) - 1); |
335 |
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336 |
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pos += interloper + 1; |
337 |
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step -= step / ADAPT_LEVEL; |
338 |
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} |
339 |
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340 |
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if (step < 256) |
341 |
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{ |
342 |
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step = 65536 / step; |
343 |
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dominant = !dominant; |
344 |
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} |
345 |
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} |
346 |
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347 |
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// store signs |
348 |
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for (i = 0; i < entries; i++) |
349 |
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if (buf[i]) |
350 |
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put_bits(pb, 1, buf[i] < 0); |
351 |
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352 |
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av_free(bits); |
353 |
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av_free(copy); |
354 |
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355 |
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return 0; |
356 |
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} |
357 |
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358 |
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static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part) |
359 |
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{ |
360 |
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int i, low_bits = 0, x = 0; |
361 |
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int n_zeros = 0, step = 256, dominant = 0; |
362 |
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int pos = 0, level = 0; |
363 |
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int *bits = av_calloc(entries, sizeof(*bits)); |
364 |
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365 |
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if (!bits) |
366 |
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return AVERROR(ENOMEM); |
367 |
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368 |
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if (base_2_part) |
369 |
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{ |
370 |
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low_bits = get_bits(gb, 4); |
371 |
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|
372 |
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if (low_bits) |
373 |
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for (i = 0; i < entries; i++) |
374 |
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buf[i] = get_bits(gb, low_bits); |
375 |
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} |
376 |
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|
377 |
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// av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits); |
378 |
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379 |
|
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while (n_zeros < entries) |
380 |
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{ |
381 |
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int steplet = step >> 8; |
382 |
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|
383 |
|
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if (!get_bits1(gb)) |
384 |
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{ |
385 |
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for (i = 0; i < steplet; i++) |
386 |
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bits[x++] = dominant; |
387 |
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|
388 |
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if (!dominant) |
389 |
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n_zeros += steplet; |
390 |
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|
391 |
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step += step / ADAPT_LEVEL; |
392 |
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} |
393 |
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else |
394 |
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{ |
395 |
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int actual_run = read_uint_max(gb, steplet-1); |
396 |
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|
397 |
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// av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run); |
398 |
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399 |
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for (i = 0; i < actual_run; i++) |
400 |
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bits[x++] = dominant; |
401 |
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|
402 |
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bits[x++] = !dominant; |
403 |
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404 |
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if (!dominant) |
405 |
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n_zeros += actual_run; |
406 |
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else |
407 |
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n_zeros++; |
408 |
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|
409 |
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|
step -= step / ADAPT_LEVEL; |
410 |
|
|
} |
411 |
|
|
|
412 |
|
|
if (step < 256) |
413 |
|
|
{ |
414 |
|
|
step = 65536 / step; |
415 |
|
|
dominant = !dominant; |
416 |
|
|
} |
417 |
|
|
} |
418 |
|
|
|
419 |
|
|
// reconstruct unsigned values |
420 |
|
|
n_zeros = 0; |
421 |
|
|
for (i = 0; n_zeros < entries; i++) |
422 |
|
|
{ |
423 |
|
|
while(1) |
424 |
|
|
{ |
425 |
|
|
if (pos >= entries) |
426 |
|
|
{ |
427 |
|
|
pos = 0; |
428 |
|
|
level += 1 << low_bits; |
429 |
|
|
} |
430 |
|
|
|
431 |
|
|
if (buf[pos] >= level) |
432 |
|
|
break; |
433 |
|
|
|
434 |
|
|
pos++; |
435 |
|
|
} |
436 |
|
|
|
437 |
|
|
if (bits[i]) |
438 |
|
|
buf[pos] += 1 << low_bits; |
439 |
|
|
else |
440 |
|
|
n_zeros++; |
441 |
|
|
|
442 |
|
|
pos++; |
443 |
|
|
} |
444 |
|
|
av_free(bits); |
445 |
|
|
|
446 |
|
|
// read signs |
447 |
|
|
for (i = 0; i < entries; i++) |
448 |
|
|
if (buf[i] && get_bits1(gb)) |
449 |
|
|
buf[i] = -buf[i]; |
450 |
|
|
|
451 |
|
|
// av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos); |
452 |
|
|
|
453 |
|
|
return 0; |
454 |
|
|
} |
455 |
|
|
#endif |
456 |
|
|
|
457 |
|
✗ |
static void predictor_init_state(int *k, int *state, int order) |
458 |
|
|
{ |
459 |
|
|
int i; |
460 |
|
|
|
461 |
|
✗ |
for (i = order-2; i >= 0; i--) |
462 |
|
|
{ |
463 |
|
✗ |
int j, p, x = state[i]; |
464 |
|
|
|
465 |
|
✗ |
for (j = 0, p = i+1; p < order; j++,p++) |
466 |
|
|
{ |
467 |
|
✗ |
int tmp = x + shift_down(k[j] * (unsigned)state[p], LATTICE_SHIFT); |
468 |
|
✗ |
state[p] += shift_down(k[j]* (unsigned)x, LATTICE_SHIFT); |
469 |
|
✗ |
x = tmp; |
470 |
|
|
} |
471 |
|
|
} |
472 |
|
|
} |
473 |
|
|
|
474 |
|
✗ |
static int predictor_calc_error(int *k, int *state, int order, int error) |
475 |
|
|
{ |
476 |
|
✗ |
int i, x = error - shift_down(k[order-1] * (unsigned)state[order-1], LATTICE_SHIFT); |
477 |
|
|
|
478 |
|
|
#if 1 |
479 |
|
✗ |
int *k_ptr = &(k[order-2]), |
480 |
|
✗ |
*state_ptr = &(state[order-2]); |
481 |
|
✗ |
for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--) |
482 |
|
|
{ |
483 |
|
✗ |
int k_value = *k_ptr, state_value = *state_ptr; |
484 |
|
✗ |
x -= (unsigned)shift_down(k_value * (unsigned)state_value, LATTICE_SHIFT); |
485 |
|
✗ |
state_ptr[1] = state_value + shift_down(k_value * (unsigned)x, LATTICE_SHIFT); |
486 |
|
|
} |
487 |
|
|
#else |
488 |
|
|
for (i = order-2; i >= 0; i--) |
489 |
|
|
{ |
490 |
|
|
x -= (unsigned)shift_down(k[i] * state[i], LATTICE_SHIFT); |
491 |
|
|
state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT); |
492 |
|
|
} |
493 |
|
|
#endif |
494 |
|
|
|
495 |
|
|
// don't drift too far, to avoid overflows |
496 |
|
✗ |
if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16); |
497 |
|
✗ |
if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16); |
498 |
|
|
|
499 |
|
✗ |
state[0] = x; |
500 |
|
|
|
501 |
|
✗ |
return x; |
502 |
|
|
} |
503 |
|
|
|
504 |
|
|
#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER |
505 |
|
|
// Heavily modified Levinson-Durbin algorithm which |
506 |
|
|
// copes better with quantization, and calculates the |
507 |
|
|
// actual whitened result as it goes. |
508 |
|
|
|
509 |
|
✗ |
static void modified_levinson_durbin(int *window, int window_entries, |
510 |
|
|
int *out, int out_entries, int channels, int *tap_quant) |
511 |
|
|
{ |
512 |
|
|
int i; |
513 |
|
✗ |
int *state = window + window_entries; |
514 |
|
|
|
515 |
|
✗ |
memcpy(state, window, window_entries * sizeof(*state)); |
516 |
|
|
|
517 |
|
✗ |
for (i = 0; i < out_entries; i++) |
518 |
|
|
{ |
519 |
|
✗ |
int step = (i+1)*channels, k, j; |
520 |
|
✗ |
double xx = 0.0, xy = 0.0; |
521 |
|
|
#if 1 |
522 |
|
✗ |
int *x_ptr = &(window[step]); |
523 |
|
✗ |
int *state_ptr = &(state[0]); |
524 |
|
✗ |
j = window_entries - step; |
525 |
|
✗ |
for (;j>0;j--,x_ptr++,state_ptr++) |
526 |
|
|
{ |
527 |
|
✗ |
double x_value = *x_ptr; |
528 |
|
✗ |
double state_value = *state_ptr; |
529 |
|
✗ |
xx += state_value*state_value; |
530 |
|
✗ |
xy += x_value*state_value; |
531 |
|
|
} |
532 |
|
|
#else |
533 |
|
|
for (j = 0; j <= (window_entries - step); j++); |
534 |
|
|
{ |
535 |
|
|
double stepval = window[step+j]; |
536 |
|
|
double stateval = window[j]; |
537 |
|
|
// xx += (double)window[j]*(double)window[j]; |
538 |
|
|
// xy += (double)window[step+j]*(double)window[j]; |
539 |
|
|
xx += stateval*stateval; |
540 |
|
|
xy += stepval*stateval; |
541 |
|
|
} |
542 |
|
|
#endif |
543 |
|
✗ |
if (xx == 0.0) |
544 |
|
✗ |
k = 0; |
545 |
|
|
else |
546 |
|
✗ |
k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5)); |
547 |
|
|
|
548 |
|
✗ |
if (k > (LATTICE_FACTOR/tap_quant[i])) |
549 |
|
✗ |
k = LATTICE_FACTOR/tap_quant[i]; |
550 |
|
✗ |
if (-k > (LATTICE_FACTOR/tap_quant[i])) |
551 |
|
✗ |
k = -(LATTICE_FACTOR/tap_quant[i]); |
552 |
|
|
|
553 |
|
✗ |
out[i] = k; |
554 |
|
✗ |
k *= tap_quant[i]; |
555 |
|
|
|
556 |
|
|
#if 1 |
557 |
|
✗ |
x_ptr = &(window[step]); |
558 |
|
✗ |
state_ptr = &(state[0]); |
559 |
|
✗ |
j = window_entries - step; |
560 |
|
✗ |
for (;j>0;j--,x_ptr++,state_ptr++) |
561 |
|
|
{ |
562 |
|
✗ |
int x_value = *x_ptr; |
563 |
|
✗ |
int state_value = *state_ptr; |
564 |
|
✗ |
*x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT); |
565 |
|
✗ |
*state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT); |
566 |
|
|
} |
567 |
|
|
#else |
568 |
|
|
for (j=0; j <= (window_entries - step); j++) |
569 |
|
|
{ |
570 |
|
|
int stepval = window[step+j]; |
571 |
|
|
int stateval=state[j]; |
572 |
|
|
window[step+j] += shift_down(k * stateval, LATTICE_SHIFT); |
573 |
|
|
state[j] += shift_down(k * stepval, LATTICE_SHIFT); |
574 |
|
|
} |
575 |
|
|
#endif |
576 |
|
|
} |
577 |
|
|
} |
578 |
|
|
|
579 |
|
✗ |
static inline int code_samplerate(int samplerate) |
580 |
|
|
{ |
581 |
|
✗ |
switch (samplerate) |
582 |
|
|
{ |
583 |
|
✗ |
case 44100: return 0; |
584 |
|
✗ |
case 22050: return 1; |
585 |
|
✗ |
case 11025: return 2; |
586 |
|
✗ |
case 96000: return 3; |
587 |
|
✗ |
case 48000: return 4; |
588 |
|
✗ |
case 32000: return 5; |
589 |
|
✗ |
case 24000: return 6; |
590 |
|
✗ |
case 16000: return 7; |
591 |
|
✗ |
case 8000: return 8; |
592 |
|
|
} |
593 |
|
✗ |
return AVERROR(EINVAL); |
594 |
|
|
} |
595 |
|
|
|
596 |
|
✗ |
static av_cold int sonic_encode_init(AVCodecContext *avctx) |
597 |
|
|
{ |
598 |
|
✗ |
SonicContext *s = avctx->priv_data; |
599 |
|
|
int *coded_samples; |
600 |
|
|
PutBitContext pb; |
601 |
|
|
int i; |
602 |
|
|
|
603 |
|
✗ |
s->version = 2; |
604 |
|
|
|
605 |
|
✗ |
if (avctx->ch_layout.nb_channels > MAX_CHANNELS) |
606 |
|
|
{ |
607 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); |
608 |
|
✗ |
return AVERROR(EINVAL); /* only stereo or mono for now */ |
609 |
|
|
} |
610 |
|
|
|
611 |
|
✗ |
if (avctx->ch_layout.nb_channels == 2) |
612 |
|
✗ |
s->decorrelation = MID_SIDE; |
613 |
|
|
else |
614 |
|
✗ |
s->decorrelation = 3; |
615 |
|
|
|
616 |
|
✗ |
if (avctx->codec->id == AV_CODEC_ID_SONIC_LS) |
617 |
|
|
{ |
618 |
|
✗ |
s->lossless = 1; |
619 |
|
✗ |
s->num_taps = 32; |
620 |
|
✗ |
s->downsampling = 1; |
621 |
|
✗ |
s->quantization = 0.0; |
622 |
|
|
} |
623 |
|
|
else |
624 |
|
|
{ |
625 |
|
✗ |
s->num_taps = 128; |
626 |
|
✗ |
s->downsampling = 2; |
627 |
|
✗ |
s->quantization = 1.0; |
628 |
|
|
} |
629 |
|
|
|
630 |
|
|
// max tap 2048 |
631 |
|
✗ |
if (s->num_taps < 32 || s->num_taps > 1024 || s->num_taps % 32) { |
632 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n"); |
633 |
|
✗ |
return AVERROR_INVALIDDATA; |
634 |
|
|
} |
635 |
|
|
|
636 |
|
|
// generate taps |
637 |
|
✗ |
s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant)); |
638 |
|
✗ |
if (!s->tap_quant) |
639 |
|
✗ |
return AVERROR(ENOMEM); |
640 |
|
|
|
641 |
|
✗ |
for (i = 0; i < s->num_taps; i++) |
642 |
|
✗ |
s->tap_quant[i] = ff_sqrt(i+1); |
643 |
|
|
|
644 |
|
✗ |
s->channels = avctx->ch_layout.nb_channels; |
645 |
|
✗ |
s->samplerate = avctx->sample_rate; |
646 |
|
|
|
647 |
|
✗ |
s->block_align = 2048LL*s->samplerate/(44100*s->downsampling); |
648 |
|
✗ |
s->frame_size = s->channels*s->block_align*s->downsampling; |
649 |
|
|
|
650 |
|
✗ |
s->tail_size = s->num_taps*s->channels; |
651 |
|
✗ |
s->tail = av_calloc(s->tail_size, sizeof(*s->tail)); |
652 |
|
✗ |
if (!s->tail) |
653 |
|
✗ |
return AVERROR(ENOMEM); |
654 |
|
|
|
655 |
|
✗ |
s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) ); |
656 |
|
✗ |
if (!s->predictor_k) |
657 |
|
✗ |
return AVERROR(ENOMEM); |
658 |
|
|
|
659 |
|
✗ |
coded_samples = av_calloc(s->block_align, s->channels * sizeof(**s->coded_samples)); |
660 |
|
✗ |
if (!coded_samples) |
661 |
|
✗ |
return AVERROR(ENOMEM); |
662 |
|
✗ |
for (i = 0; i < s->channels; i++, coded_samples += s->block_align) |
663 |
|
✗ |
s->coded_samples[i] = coded_samples; |
664 |
|
|
|
665 |
|
✗ |
s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples)); |
666 |
|
|
|
667 |
|
✗ |
s->window_size = ((2*s->tail_size)+s->frame_size); |
668 |
|
✗ |
s->window = av_calloc(s->window_size, 2 * sizeof(*s->window)); |
669 |
|
✗ |
if (!s->window || !s->int_samples) |
670 |
|
✗ |
return AVERROR(ENOMEM); |
671 |
|
|
|
672 |
|
✗ |
avctx->extradata = av_mallocz(16); |
673 |
|
✗ |
if (!avctx->extradata) |
674 |
|
✗ |
return AVERROR(ENOMEM); |
675 |
|
✗ |
init_put_bits(&pb, avctx->extradata, 16*8); |
676 |
|
|
|
677 |
|
✗ |
put_bits(&pb, 2, s->version); // version |
678 |
|
✗ |
if (s->version >= 1) |
679 |
|
|
{ |
680 |
|
✗ |
if (s->version >= 2) { |
681 |
|
✗ |
put_bits(&pb, 8, s->version); |
682 |
|
✗ |
put_bits(&pb, 8, s->minor_version); |
683 |
|
|
} |
684 |
|
✗ |
put_bits(&pb, 2, s->channels); |
685 |
|
✗ |
put_bits(&pb, 4, code_samplerate(s->samplerate)); |
686 |
|
|
} |
687 |
|
✗ |
put_bits(&pb, 1, s->lossless); |
688 |
|
✗ |
if (!s->lossless) |
689 |
|
✗ |
put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision |
690 |
|
✗ |
put_bits(&pb, 2, s->decorrelation); |
691 |
|
✗ |
put_bits(&pb, 2, s->downsampling); |
692 |
|
✗ |
put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024 |
693 |
|
✗ |
put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table |
694 |
|
|
|
695 |
|
✗ |
flush_put_bits(&pb); |
696 |
|
✗ |
avctx->extradata_size = put_bytes_output(&pb); |
697 |
|
|
|
698 |
|
✗ |
av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
699 |
|
|
s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); |
700 |
|
|
|
701 |
|
✗ |
avctx->frame_size = s->block_align*s->downsampling; |
702 |
|
|
|
703 |
|
✗ |
return 0; |
704 |
|
|
} |
705 |
|
|
|
706 |
|
✗ |
static av_cold int sonic_encode_close(AVCodecContext *avctx) |
707 |
|
|
{ |
708 |
|
✗ |
SonicContext *s = avctx->priv_data; |
709 |
|
|
|
710 |
|
✗ |
av_freep(&s->coded_samples[0]); |
711 |
|
✗ |
av_freep(&s->predictor_k); |
712 |
|
✗ |
av_freep(&s->tail); |
713 |
|
✗ |
av_freep(&s->tap_quant); |
714 |
|
✗ |
av_freep(&s->window); |
715 |
|
✗ |
av_freep(&s->int_samples); |
716 |
|
|
|
717 |
|
✗ |
return 0; |
718 |
|
|
} |
719 |
|
|
|
720 |
|
✗ |
static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, |
721 |
|
|
const AVFrame *frame, int *got_packet_ptr) |
722 |
|
|
{ |
723 |
|
✗ |
SonicContext *s = avctx->priv_data; |
724 |
|
|
RangeCoder c; |
725 |
|
✗ |
int i, j, ch, quant = 0, x = 0; |
726 |
|
|
int ret; |
727 |
|
✗ |
const short *samples = (const int16_t*)frame->data[0]; |
728 |
|
|
uint8_t state[32]; |
729 |
|
|
|
730 |
|
✗ |
if ((ret = ff_alloc_packet(avctx, avpkt, s->frame_size * 5 + 1000)) < 0) |
731 |
|
✗ |
return ret; |
732 |
|
|
|
733 |
|
✗ |
ff_init_range_encoder(&c, avpkt->data, avpkt->size); |
734 |
|
✗ |
ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8); |
735 |
|
✗ |
memset(state, 128, sizeof(state)); |
736 |
|
|
|
737 |
|
|
// short -> internal |
738 |
|
✗ |
for (i = 0; i < s->frame_size; i++) |
739 |
|
✗ |
s->int_samples[i] = samples[i]; |
740 |
|
|
|
741 |
|
✗ |
if (!s->lossless) |
742 |
|
✗ |
for (i = 0; i < s->frame_size; i++) |
743 |
|
✗ |
s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT; |
744 |
|
|
|
745 |
|
✗ |
switch(s->decorrelation) |
746 |
|
|
{ |
747 |
|
✗ |
case MID_SIDE: |
748 |
|
✗ |
for (i = 0; i < s->frame_size; i += s->channels) |
749 |
|
|
{ |
750 |
|
✗ |
s->int_samples[i] += s->int_samples[i+1]; |
751 |
|
✗ |
s->int_samples[i+1] -= shift(s->int_samples[i], 1); |
752 |
|
|
} |
753 |
|
✗ |
break; |
754 |
|
✗ |
case LEFT_SIDE: |
755 |
|
✗ |
for (i = 0; i < s->frame_size; i += s->channels) |
756 |
|
✗ |
s->int_samples[i+1] -= s->int_samples[i]; |
757 |
|
✗ |
break; |
758 |
|
✗ |
case RIGHT_SIDE: |
759 |
|
✗ |
for (i = 0; i < s->frame_size; i += s->channels) |
760 |
|
✗ |
s->int_samples[i] -= s->int_samples[i+1]; |
761 |
|
✗ |
break; |
762 |
|
|
} |
763 |
|
|
|
764 |
|
✗ |
memset(s->window, 0, s->window_size * sizeof(*s->window)); |
765 |
|
|
|
766 |
|
✗ |
for (i = 0; i < s->tail_size; i++) |
767 |
|
✗ |
s->window[x++] = s->tail[i]; |
768 |
|
|
|
769 |
|
✗ |
for (i = 0; i < s->frame_size; i++) |
770 |
|
✗ |
s->window[x++] = s->int_samples[i]; |
771 |
|
|
|
772 |
|
✗ |
for (i = 0; i < s->tail_size; i++) |
773 |
|
✗ |
s->window[x++] = 0; |
774 |
|
|
|
775 |
|
✗ |
for (i = 0; i < s->tail_size; i++) |
776 |
|
✗ |
s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i]; |
777 |
|
|
|
778 |
|
|
// generate taps |
779 |
|
✗ |
modified_levinson_durbin(s->window, s->window_size, |
780 |
|
|
s->predictor_k, s->num_taps, s->channels, s->tap_quant); |
781 |
|
|
|
782 |
|
✗ |
if ((ret = intlist_write(&c, state, s->predictor_k, s->num_taps, 0)) < 0) |
783 |
|
✗ |
return ret; |
784 |
|
|
|
785 |
|
✗ |
for (ch = 0; ch < s->channels; ch++) |
786 |
|
|
{ |
787 |
|
✗ |
x = s->tail_size+ch; |
788 |
|
✗ |
for (i = 0; i < s->block_align; i++) |
789 |
|
|
{ |
790 |
|
✗ |
int sum = 0; |
791 |
|
✗ |
for (j = 0; j < s->downsampling; j++, x += s->channels) |
792 |
|
✗ |
sum += s->window[x]; |
793 |
|
✗ |
s->coded_samples[ch][i] = sum; |
794 |
|
|
} |
795 |
|
|
} |
796 |
|
|
|
797 |
|
|
// simple rate control code |
798 |
|
✗ |
if (!s->lossless) |
799 |
|
|
{ |
800 |
|
✗ |
double energy1 = 0.0, energy2 = 0.0; |
801 |
|
✗ |
for (ch = 0; ch < s->channels; ch++) |
802 |
|
|
{ |
803 |
|
✗ |
for (i = 0; i < s->block_align; i++) |
804 |
|
|
{ |
805 |
|
✗ |
double sample = s->coded_samples[ch][i]; |
806 |
|
✗ |
energy2 += sample*sample; |
807 |
|
✗ |
energy1 += fabs(sample); |
808 |
|
|
} |
809 |
|
|
} |
810 |
|
|
|
811 |
|
✗ |
energy2 = sqrt(energy2/(s->channels*s->block_align)); |
812 |
|
✗ |
energy1 = M_SQRT2*energy1/(s->channels*s->block_align); |
813 |
|
|
|
814 |
|
|
// increase bitrate when samples are like a gaussian distribution |
815 |
|
|
// reduce bitrate when samples are like a two-tailed exponential distribution |
816 |
|
|
|
817 |
|
✗ |
if (energy2 > energy1) |
818 |
|
✗ |
energy2 += (energy2-energy1)*RATE_VARIATION; |
819 |
|
|
|
820 |
|
✗ |
quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR); |
821 |
|
|
// av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2); |
822 |
|
|
|
823 |
|
✗ |
quant = av_clip(quant, 1, 65534); |
824 |
|
|
|
825 |
|
✗ |
put_symbol(&c, state, quant, 0, NULL, NULL); |
826 |
|
|
|
827 |
|
✗ |
quant *= SAMPLE_FACTOR; |
828 |
|
|
} |
829 |
|
|
|
830 |
|
|
// write out coded samples |
831 |
|
✗ |
for (ch = 0; ch < s->channels; ch++) |
832 |
|
|
{ |
833 |
|
✗ |
if (!s->lossless) |
834 |
|
✗ |
for (i = 0; i < s->block_align; i++) |
835 |
|
✗ |
s->coded_samples[ch][i] = ROUNDED_DIV(s->coded_samples[ch][i], quant); |
836 |
|
|
|
837 |
|
✗ |
if ((ret = intlist_write(&c, state, s->coded_samples[ch], s->block_align, 1)) < 0) |
838 |
|
✗ |
return ret; |
839 |
|
|
} |
840 |
|
|
|
841 |
|
✗ |
avpkt->size = ff_rac_terminate(&c, 0); |
842 |
|
✗ |
*got_packet_ptr = 1; |
843 |
|
✗ |
return 0; |
844 |
|
|
|
845 |
|
|
} |
846 |
|
|
#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */ |
847 |
|
|
|
848 |
|
|
#if CONFIG_SONIC_DECODER |
849 |
|
|
static const int samplerate_table[] = |
850 |
|
|
{ 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 }; |
851 |
|
|
|
852 |
|
✗ |
static av_cold int sonic_decode_init(AVCodecContext *avctx) |
853 |
|
|
{ |
854 |
|
✗ |
SonicContext *s = avctx->priv_data; |
855 |
|
|
int *tmp; |
856 |
|
|
GetBitContext gb; |
857 |
|
|
int i; |
858 |
|
|
int ret; |
859 |
|
|
|
860 |
|
✗ |
s->channels = avctx->ch_layout.nb_channels; |
861 |
|
✗ |
s->samplerate = avctx->sample_rate; |
862 |
|
|
|
863 |
|
✗ |
if (!avctx->extradata) |
864 |
|
|
{ |
865 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n"); |
866 |
|
✗ |
return AVERROR_INVALIDDATA; |
867 |
|
|
} |
868 |
|
|
|
869 |
|
✗ |
ret = init_get_bits8(&gb, avctx->extradata, avctx->extradata_size); |
870 |
|
✗ |
if (ret < 0) |
871 |
|
✗ |
return ret; |
872 |
|
|
|
873 |
|
✗ |
s->version = get_bits(&gb, 2); |
874 |
|
✗ |
if (s->version >= 2) { |
875 |
|
✗ |
s->version = get_bits(&gb, 8); |
876 |
|
✗ |
s->minor_version = get_bits(&gb, 8); |
877 |
|
|
} |
878 |
|
✗ |
if (s->version != 2) |
879 |
|
|
{ |
880 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n"); |
881 |
|
✗ |
return AVERROR_INVALIDDATA; |
882 |
|
|
} |
883 |
|
|
|
884 |
|
✗ |
if (s->version >= 1) |
885 |
|
|
{ |
886 |
|
|
int sample_rate_index; |
887 |
|
✗ |
s->channels = get_bits(&gb, 2); |
888 |
|
✗ |
sample_rate_index = get_bits(&gb, 4); |
889 |
|
✗ |
if (sample_rate_index >= FF_ARRAY_ELEMS(samplerate_table)) { |
890 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "Invalid sample_rate_index %d\n", sample_rate_index); |
891 |
|
✗ |
return AVERROR_INVALIDDATA; |
892 |
|
|
} |
893 |
|
✗ |
s->samplerate = samplerate_table[sample_rate_index]; |
894 |
|
✗ |
av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n", |
895 |
|
|
s->channels, s->samplerate); |
896 |
|
|
} |
897 |
|
|
|
898 |
|
✗ |
if (s->channels > MAX_CHANNELS || s->channels < 1) |
899 |
|
|
{ |
900 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); |
901 |
|
✗ |
return AVERROR_INVALIDDATA; |
902 |
|
|
} |
903 |
|
✗ |
av_channel_layout_uninit(&avctx->ch_layout); |
904 |
|
✗ |
avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC; |
905 |
|
✗ |
avctx->ch_layout.nb_channels = s->channels; |
906 |
|
|
|
907 |
|
✗ |
s->lossless = get_bits1(&gb); |
908 |
|
✗ |
if (!s->lossless) |
909 |
|
✗ |
skip_bits(&gb, 3); // XXX FIXME |
910 |
|
✗ |
s->decorrelation = get_bits(&gb, 2); |
911 |
|
✗ |
if (s->decorrelation != 3 && s->channels != 2) { |
912 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation); |
913 |
|
✗ |
return AVERROR_INVALIDDATA; |
914 |
|
|
} |
915 |
|
|
|
916 |
|
✗ |
s->downsampling = get_bits(&gb, 2); |
917 |
|
✗ |
if (!s->downsampling) { |
918 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n"); |
919 |
|
✗ |
return AVERROR_INVALIDDATA; |
920 |
|
|
} |
921 |
|
|
|
922 |
|
✗ |
s->num_taps = (get_bits(&gb, 5)+1)<<5; |
923 |
|
✗ |
if (get_bits1(&gb)) // XXX FIXME |
924 |
|
✗ |
av_log(avctx, AV_LOG_INFO, "Custom quant table\n"); |
925 |
|
|
|
926 |
|
✗ |
s->block_align = 2048LL*s->samplerate/(44100*s->downsampling); |
927 |
|
✗ |
s->frame_size = s->channels*s->block_align*s->downsampling; |
928 |
|
|
// avctx->frame_size = s->block_align; |
929 |
|
|
|
930 |
|
✗ |
if (s->num_taps * s->channels > s->frame_size) { |
931 |
|
✗ |
av_log(avctx, AV_LOG_ERROR, |
932 |
|
|
"number of taps times channels (%d * %d) larger than frame size %d\n", |
933 |
|
|
s->num_taps, s->channels, s->frame_size); |
934 |
|
✗ |
return AVERROR_INVALIDDATA; |
935 |
|
|
} |
936 |
|
|
|
937 |
|
✗ |
av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
938 |
|
|
s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); |
939 |
|
|
|
940 |
|
|
// generate taps |
941 |
|
✗ |
s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant)); |
942 |
|
✗ |
if (!s->tap_quant) |
943 |
|
✗ |
return AVERROR(ENOMEM); |
944 |
|
|
|
945 |
|
✗ |
for (i = 0; i < s->num_taps; i++) |
946 |
|
✗ |
s->tap_quant[i] = ff_sqrt(i+1); |
947 |
|
|
|
948 |
|
✗ |
s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k)); |
949 |
|
|
|
950 |
|
✗ |
tmp = av_calloc(s->num_taps, s->channels * sizeof(**s->predictor_state)); |
951 |
|
✗ |
if (!tmp) |
952 |
|
✗ |
return AVERROR(ENOMEM); |
953 |
|
✗ |
for (i = 0; i < s->channels; i++, tmp += s->num_taps) |
954 |
|
✗ |
s->predictor_state[i] = tmp; |
955 |
|
|
|
956 |
|
✗ |
tmp = av_calloc(s->block_align, s->channels * sizeof(**s->coded_samples)); |
957 |
|
✗ |
if (!tmp) |
958 |
|
✗ |
return AVERROR(ENOMEM); |
959 |
|
✗ |
for (i = 0; i < s->channels; i++, tmp += s->block_align) |
960 |
|
✗ |
s->coded_samples[i] = tmp; |
961 |
|
|
|
962 |
|
✗ |
s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples)); |
963 |
|
✗ |
if (!s->int_samples) |
964 |
|
✗ |
return AVERROR(ENOMEM); |
965 |
|
|
|
966 |
|
✗ |
avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
967 |
|
✗ |
return 0; |
968 |
|
|
} |
969 |
|
|
|
970 |
|
✗ |
static av_cold int sonic_decode_close(AVCodecContext *avctx) |
971 |
|
|
{ |
972 |
|
✗ |
SonicContext *s = avctx->priv_data; |
973 |
|
|
|
974 |
|
✗ |
av_freep(&s->int_samples); |
975 |
|
✗ |
av_freep(&s->tap_quant); |
976 |
|
✗ |
av_freep(&s->predictor_k); |
977 |
|
✗ |
av_freep(&s->predictor_state[0]); |
978 |
|
✗ |
av_freep(&s->coded_samples[0]); |
979 |
|
|
|
980 |
|
✗ |
return 0; |
981 |
|
|
} |
982 |
|
|
|
983 |
|
✗ |
static int sonic_decode_frame(AVCodecContext *avctx, AVFrame *frame, |
984 |
|
|
int *got_frame_ptr, AVPacket *avpkt) |
985 |
|
|
{ |
986 |
|
✗ |
const uint8_t *buf = avpkt->data; |
987 |
|
✗ |
int buf_size = avpkt->size; |
988 |
|
✗ |
SonicContext *s = avctx->priv_data; |
989 |
|
|
RangeCoder c; |
990 |
|
|
uint8_t state[32]; |
991 |
|
|
int i, quant, ch, j, ret; |
992 |
|
|
int16_t *samples; |
993 |
|
|
|
994 |
|
✗ |
if (buf_size == 0) return 0; |
995 |
|
|
|
996 |
|
✗ |
frame->nb_samples = s->frame_size / avctx->ch_layout.nb_channels; |
997 |
|
✗ |
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
998 |
|
✗ |
return ret; |
999 |
|
✗ |
samples = (int16_t *)frame->data[0]; |
1000 |
|
|
|
1001 |
|
|
// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size); |
1002 |
|
|
|
1003 |
|
✗ |
memset(state, 128, sizeof(state)); |
1004 |
|
✗ |
ff_init_range_decoder(&c, buf, buf_size); |
1005 |
|
✗ |
ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8); |
1006 |
|
|
|
1007 |
|
✗ |
intlist_read(&c, state, s->predictor_k, s->num_taps, 0); |
1008 |
|
|
|
1009 |
|
|
// dequantize |
1010 |
|
✗ |
for (i = 0; i < s->num_taps; i++) |
1011 |
|
✗ |
s->predictor_k[i] *= (unsigned) s->tap_quant[i]; |
1012 |
|
|
|
1013 |
|
✗ |
if (s->lossless) |
1014 |
|
✗ |
quant = 1; |
1015 |
|
|
else |
1016 |
|
✗ |
quant = get_symbol(&c, state, 0) * SAMPLE_FACTOR; |
1017 |
|
|
|
1018 |
|
|
// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant); |
1019 |
|
|
|
1020 |
|
✗ |
for (ch = 0; ch < s->channels; ch++) |
1021 |
|
|
{ |
1022 |
|
✗ |
int x = ch; |
1023 |
|
|
|
1024 |
|
✗ |
if (c.overread > MAX_OVERREAD) |
1025 |
|
✗ |
return AVERROR_INVALIDDATA; |
1026 |
|
|
|
1027 |
|
✗ |
predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps); |
1028 |
|
|
|
1029 |
|
✗ |
intlist_read(&c, state, s->coded_samples[ch], s->block_align, 1); |
1030 |
|
|
|
1031 |
|
✗ |
for (i = 0; i < s->block_align; i++) |
1032 |
|
|
{ |
1033 |
|
✗ |
for (j = 0; j < s->downsampling - 1; j++) |
1034 |
|
|
{ |
1035 |
|
✗ |
s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0); |
1036 |
|
✗ |
x += s->channels; |
1037 |
|
|
} |
1038 |
|
|
|
1039 |
|
✗ |
s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * (unsigned)quant); |
1040 |
|
✗ |
x += s->channels; |
1041 |
|
|
} |
1042 |
|
|
|
1043 |
|
✗ |
for (i = 0; i < s->num_taps; i++) |
1044 |
|
✗ |
s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels]; |
1045 |
|
|
} |
1046 |
|
|
|
1047 |
|
✗ |
switch(s->decorrelation) |
1048 |
|
|
{ |
1049 |
|
✗ |
case MID_SIDE: |
1050 |
|
✗ |
for (i = 0; i < s->frame_size; i += s->channels) |
1051 |
|
|
{ |
1052 |
|
✗ |
s->int_samples[i+1] += shift(s->int_samples[i], 1); |
1053 |
|
✗ |
s->int_samples[i] -= s->int_samples[i+1]; |
1054 |
|
|
} |
1055 |
|
✗ |
break; |
1056 |
|
✗ |
case LEFT_SIDE: |
1057 |
|
✗ |
for (i = 0; i < s->frame_size; i += s->channels) |
1058 |
|
✗ |
s->int_samples[i+1] += s->int_samples[i]; |
1059 |
|
✗ |
break; |
1060 |
|
✗ |
case RIGHT_SIDE: |
1061 |
|
✗ |
for (i = 0; i < s->frame_size; i += s->channels) |
1062 |
|
✗ |
s->int_samples[i] += s->int_samples[i+1]; |
1063 |
|
✗ |
break; |
1064 |
|
|
} |
1065 |
|
|
|
1066 |
|
✗ |
if (!s->lossless) |
1067 |
|
✗ |
for (i = 0; i < s->frame_size; i++) |
1068 |
|
✗ |
s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT); |
1069 |
|
|
|
1070 |
|
|
// internal -> short |
1071 |
|
✗ |
for (i = 0; i < s->frame_size; i++) |
1072 |
|
✗ |
samples[i] = av_clip_int16(s->int_samples[i]); |
1073 |
|
|
|
1074 |
|
✗ |
*got_frame_ptr = 1; |
1075 |
|
|
|
1076 |
|
✗ |
return buf_size; |
1077 |
|
|
} |
1078 |
|
|
|
1079 |
|
|
const FFCodec ff_sonic_decoder = { |
1080 |
|
|
.p.name = "sonic", |
1081 |
|
|
.p.long_name = NULL_IF_CONFIG_SMALL("Sonic"), |
1082 |
|
|
.p.type = AVMEDIA_TYPE_AUDIO, |
1083 |
|
|
.p.id = AV_CODEC_ID_SONIC, |
1084 |
|
|
.priv_data_size = sizeof(SonicContext), |
1085 |
|
|
.init = sonic_decode_init, |
1086 |
|
|
.close = sonic_decode_close, |
1087 |
|
|
FF_CODEC_DECODE_CB(sonic_decode_frame), |
1088 |
|
|
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL | AV_CODEC_CAP_CHANNEL_CONF, |
1089 |
|
|
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, |
1090 |
|
|
}; |
1091 |
|
|
#endif /* CONFIG_SONIC_DECODER */ |
1092 |
|
|
|
1093 |
|
|
#if CONFIG_SONIC_ENCODER |
1094 |
|
|
const FFCodec ff_sonic_encoder = { |
1095 |
|
|
.p.name = "sonic", |
1096 |
|
|
.p.long_name = NULL_IF_CONFIG_SMALL("Sonic"), |
1097 |
|
|
.p.type = AVMEDIA_TYPE_AUDIO, |
1098 |
|
|
.p.id = AV_CODEC_ID_SONIC, |
1099 |
|
|
.priv_data_size = sizeof(SonicContext), |
1100 |
|
|
.init = sonic_encode_init, |
1101 |
|
|
FF_CODEC_ENCODE_CB(sonic_encode_frame), |
1102 |
|
|
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, |
1103 |
|
|
.p.capabilities = AV_CODEC_CAP_EXPERIMENTAL, |
1104 |
|
|
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, |
1105 |
|
|
.close = sonic_encode_close, |
1106 |
|
|
}; |
1107 |
|
|
#endif |
1108 |
|
|
|
1109 |
|
|
#if CONFIG_SONIC_LS_ENCODER |
1110 |
|
|
const FFCodec ff_sonic_ls_encoder = { |
1111 |
|
|
.p.name = "sonicls", |
1112 |
|
|
.p.long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"), |
1113 |
|
|
.p.type = AVMEDIA_TYPE_AUDIO, |
1114 |
|
|
.p.id = AV_CODEC_ID_SONIC_LS, |
1115 |
|
|
.priv_data_size = sizeof(SonicContext), |
1116 |
|
|
.init = sonic_encode_init, |
1117 |
|
|
FF_CODEC_ENCODE_CB(sonic_encode_frame), |
1118 |
|
|
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, |
1119 |
|
|
.p.capabilities = AV_CODEC_CAP_EXPERIMENTAL, |
1120 |
|
|
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, |
1121 |
|
|
.close = sonic_encode_close, |
1122 |
|
|
}; |
1123 |
|
|
#endif |
1124 |
|
|
|