Mercurial > libavcodec.hg
annotate sonic.c @ 12266:48d6738904a9 libavcodec
Fix SPLATB_REG mess. Used to be a if/elseif/elseif/elseif spaghetti, so this
splits it into small optimization-specific macros which are selected for each
DSP function. The advantage of this approach is that the sse4 functions now
use the ssse3 codepath also without needing an explicit sse4 codepath.
author | rbultje |
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date | Sat, 24 Jul 2010 19:33:05 +0000 |
parents | 7dd2a45249a9 |
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rev | line source |
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1 /* |
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2 * Simple free lossless/lossy audio codec |
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3 * Copyright (c) 2004 Alex Beregszaszi |
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4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
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8 * modify it under the terms of the GNU Lesser General Public |
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9 * License as published by the Free Software Foundation; either |
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10 * version 2.1 of the License, or (at your option) any later version. |
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11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 * Lesser General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU Lesser General Public |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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20 */ |
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21 #include "avcodec.h" |
9428 | 22 #include "get_bits.h" |
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23 #include "golomb.h" |
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24 |
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25 /** |
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26 * @file |
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27 * Simple free lossless/lossy audio codec |
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28 * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk) |
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29 * Written and designed by Alex Beregszaszi |
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30 * |
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31 * TODO: |
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32 * - CABAC put/get_symbol |
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33 * - independent quantizer for channels |
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34 * - >2 channels support |
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35 * - more decorrelation types |
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36 * - more tap_quant tests |
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37 * - selectable intlist writers/readers (bonk-style, golomb, cabac) |
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38 */ |
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39 |
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40 #define MAX_CHANNELS 2 |
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41 |
2205 | 42 #define MID_SIDE 0 |
43 #define LEFT_SIDE 1 | |
44 #define RIGHT_SIDE 2 | |
45 | |
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46 typedef struct SonicContext { |
2205 | 47 int lossless, decorrelation; |
2967 | 48 |
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49 int num_taps, downsampling; |
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50 double quantization; |
2967 | 51 |
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52 int channels, samplerate, block_align, frame_size; |
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53 |
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54 int *tap_quant; |
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55 int *int_samples; |
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56 int *coded_samples[MAX_CHANNELS]; |
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57 |
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58 // for encoding |
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59 int *tail; |
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60 int tail_size; |
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61 int *window; |
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62 int window_size; |
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63 |
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64 // for decoding |
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65 int *predictor_k; |
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66 int *predictor_state[MAX_CHANNELS]; |
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67 } SonicContext; |
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68 |
2979 | 69 #define LATTICE_SHIFT 10 |
70 #define SAMPLE_SHIFT 4 | |
71 #define LATTICE_FACTOR (1 << LATTICE_SHIFT) | |
72 #define SAMPLE_FACTOR (1 << SAMPLE_SHIFT) | |
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73 |
2979 | 74 #define BASE_QUANT 0.6 |
75 #define RATE_VARIATION 3.0 | |
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76 |
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77 static inline int divide(int a, int b) |
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78 { |
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79 if (a < 0) |
2979 | 80 return -( (-a + b/2)/b ); |
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81 else |
2979 | 82 return (a + b/2)/b; |
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83 } |
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84 |
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85 static inline int shift(int a,int b) |
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86 { |
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87 return (a+(1<<(b-1))) >> b; |
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88 } |
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89 |
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90 static inline int shift_down(int a,int b) |
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91 { |
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92 return (a>>b)+((a<0)?1:0); |
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93 } |
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94 |
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95 #if 1 |
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96 static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part) |
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97 { |
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98 int i; |
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99 |
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100 for (i = 0; i < entries; i++) |
2979 | 101 set_se_golomb(pb, buf[i]); |
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102 |
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103 return 1; |
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104 } |
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105 |
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106 static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part) |
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107 { |
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108 int i; |
2967 | 109 |
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110 for (i = 0; i < entries; i++) |
2979 | 111 buf[i] = get_se_golomb(gb); |
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112 |
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113 return 1; |
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114 } |
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115 |
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116 #else |
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117 |
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118 #define ADAPT_LEVEL 8 |
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119 |
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120 static int bits_to_store(uint64_t x) |
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121 { |
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122 int res = 0; |
2967 | 123 |
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124 while(x) |
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125 { |
2979 | 126 res++; |
127 x >>= 1; | |
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128 } |
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129 return res; |
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130 } |
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131 |
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132 static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max) |
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133 { |
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134 int i, bits; |
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135 |
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136 if (!max) |
2979 | 137 return; |
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138 |
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139 bits = bits_to_store(max); |
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140 |
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141 for (i = 0; i < bits-1; i++) |
2979 | 142 put_bits(pb, 1, value & (1 << i)); |
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143 |
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144 if ( (value | (1 << (bits-1))) <= max) |
2979 | 145 put_bits(pb, 1, value & (1 << (bits-1))); |
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146 } |
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147 |
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148 static unsigned int read_uint_max(GetBitContext *gb, int max) |
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149 { |
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150 int i, bits, value = 0; |
2967 | 151 |
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152 if (!max) |
2979 | 153 return 0; |
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154 |
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155 bits = bits_to_store(max); |
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156 |
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157 for (i = 0; i < bits-1; i++) |
2979 | 158 if (get_bits1(gb)) |
159 value += 1 << i; | |
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160 |
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161 if ( (value | (1<<(bits-1))) <= max) |
2979 | 162 if (get_bits1(gb)) |
163 value += 1 << (bits-1); | |
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164 |
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165 return value; |
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166 } |
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167 |
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168 static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part) |
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169 { |
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170 int i, j, x = 0, low_bits = 0, max = 0; |
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171 int step = 256, pos = 0, dominant = 0, any = 0; |
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172 int *copy, *bits; |
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173 |
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174 copy = av_mallocz(4* entries); |
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175 if (!copy) |
2979 | 176 return -1; |
2967 | 177 |
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178 if (base_2_part) |
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179 { |
2979 | 180 int energy = 0; |
2967 | 181 |
2979 | 182 for (i = 0; i < entries; i++) |
183 energy += abs(buf[i]); | |
2967 | 184 |
2979 | 185 low_bits = bits_to_store(energy / (entries * 2)); |
186 if (low_bits > 15) | |
187 low_bits = 15; | |
2967 | 188 |
2979 | 189 put_bits(pb, 4, low_bits); |
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190 } |
2967 | 191 |
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192 for (i = 0; i < entries; i++) |
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193 { |
2979 | 194 put_bits(pb, low_bits, abs(buf[i])); |
195 copy[i] = abs(buf[i]) >> low_bits; | |
196 if (copy[i] > max) | |
197 max = abs(copy[i]); | |
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198 } |
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199 |
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200 bits = av_mallocz(4* entries*max); |
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201 if (!bits) |
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202 { |
2979 | 203 // av_free(copy); |
204 return -1; | |
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205 } |
2967 | 206 |
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207 for (i = 0; i <= max; i++) |
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208 { |
2979 | 209 for (j = 0; j < entries; j++) |
210 if (copy[j] >= i) | |
211 bits[x++] = copy[j] > i; | |
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212 } |
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213 |
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214 // store bitstream |
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215 while (pos < x) |
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216 { |
2979 | 217 int steplet = step >> 8; |
2967 | 218 |
2979 | 219 if (pos + steplet > x) |
220 steplet = x - pos; | |
2967 | 221 |
2979 | 222 for (i = 0; i < steplet; i++) |
223 if (bits[i+pos] != dominant) | |
224 any = 1; | |
2967 | 225 |
2979 | 226 put_bits(pb, 1, any); |
2967 | 227 |
2979 | 228 if (!any) |
229 { | |
230 pos += steplet; | |
231 step += step / ADAPT_LEVEL; | |
232 } | |
233 else | |
234 { | |
235 int interloper = 0; | |
2967 | 236 |
2979 | 237 while (((pos + interloper) < x) && (bits[pos + interloper] == dominant)) |
238 interloper++; | |
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239 |
2979 | 240 // note change |
241 write_uint_max(pb, interloper, (step >> 8) - 1); | |
2967 | 242 |
2979 | 243 pos += interloper + 1; |
244 step -= step / ADAPT_LEVEL; | |
245 } | |
2967 | 246 |
2979 | 247 if (step < 256) |
248 { | |
249 step = 65536 / step; | |
250 dominant = !dominant; | |
251 } | |
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252 } |
2967 | 253 |
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254 // store signs |
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255 for (i = 0; i < entries; i++) |
2979 | 256 if (buf[i]) |
257 put_bits(pb, 1, buf[i] < 0); | |
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258 |
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259 // av_free(bits); |
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260 // av_free(copy); |
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261 |
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262 return 0; |
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263 } |
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264 |
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265 static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part) |
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266 { |
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267 int i, low_bits = 0, x = 0; |
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268 int n_zeros = 0, step = 256, dominant = 0; |
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269 int pos = 0, level = 0; |
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270 int *bits = av_mallocz(4* entries); |
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271 |
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272 if (!bits) |
2979 | 273 return -1; |
2967 | 274 |
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275 if (base_2_part) |
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276 { |
2979 | 277 low_bits = get_bits(gb, 4); |
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278 |
2979 | 279 if (low_bits) |
280 for (i = 0; i < entries; i++) | |
281 buf[i] = get_bits(gb, low_bits); | |
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282 } |
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283 |
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284 // av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits); |
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285 |
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286 while (n_zeros < entries) |
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287 { |
2979 | 288 int steplet = step >> 8; |
2967 | 289 |
2979 | 290 if (!get_bits1(gb)) |
291 { | |
292 for (i = 0; i < steplet; i++) | |
293 bits[x++] = dominant; | |
2967 | 294 |
2979 | 295 if (!dominant) |
296 n_zeros += steplet; | |
2967 | 297 |
2979 | 298 step += step / ADAPT_LEVEL; |
299 } | |
300 else | |
301 { | |
302 int actual_run = read_uint_max(gb, steplet-1); | |
2967 | 303 |
2979 | 304 // av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run); |
2967 | 305 |
2979 | 306 for (i = 0; i < actual_run; i++) |
307 bits[x++] = dominant; | |
2967 | 308 |
2979 | 309 bits[x++] = !dominant; |
2967 | 310 |
2979 | 311 if (!dominant) |
312 n_zeros += actual_run; | |
313 else | |
314 n_zeros++; | |
2967 | 315 |
2979 | 316 step -= step / ADAPT_LEVEL; |
317 } | |
2967 | 318 |
2979 | 319 if (step < 256) |
320 { | |
321 step = 65536 / step; | |
322 dominant = !dominant; | |
323 } | |
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324 } |
2967 | 325 |
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326 // reconstruct unsigned values |
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327 n_zeros = 0; |
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328 for (i = 0; n_zeros < entries; i++) |
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329 { |
2979 | 330 while(1) |
331 { | |
332 if (pos >= entries) | |
333 { | |
334 pos = 0; | |
335 level += 1 << low_bits; | |
336 } | |
2967 | 337 |
2979 | 338 if (buf[pos] >= level) |
339 break; | |
2967 | 340 |
2979 | 341 pos++; |
342 } | |
2967 | 343 |
2979 | 344 if (bits[i]) |
345 buf[pos] += 1 << low_bits; | |
346 else | |
347 n_zeros++; | |
2967 | 348 |
2979 | 349 pos++; |
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350 } |
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351 // av_free(bits); |
2967 | 352 |
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353 // read signs |
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354 for (i = 0; i < entries; i++) |
2979 | 355 if (buf[i] && get_bits1(gb)) |
356 buf[i] = -buf[i]; | |
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357 |
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358 // av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos); |
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359 |
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360 return 0; |
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361 } |
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362 #endif |
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363 |
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364 static void predictor_init_state(int *k, int *state, int order) |
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365 { |
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366 int i; |
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367 |
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368 for (i = order-2; i >= 0; i--) |
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369 { |
2979 | 370 int j, p, x = state[i]; |
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371 |
2979 | 372 for (j = 0, p = i+1; p < order; j++,p++) |
373 { | |
374 int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT); | |
375 state[p] += shift_down(k[j]*x, LATTICE_SHIFT); | |
376 x = tmp; | |
377 } | |
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378 } |
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379 } |
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380 |
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381 static int predictor_calc_error(int *k, int *state, int order, int error) |
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382 { |
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383 int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT); |
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384 |
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385 #if 1 |
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386 int *k_ptr = &(k[order-2]), |
2979 | 387 *state_ptr = &(state[order-2]); |
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388 for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--) |
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389 { |
2979 | 390 int k_value = *k_ptr, state_value = *state_ptr; |
391 x -= shift_down(k_value * state_value, LATTICE_SHIFT); | |
392 state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT); | |
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393 } |
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394 #else |
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395 for (i = order-2; i >= 0; i--) |
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396 { |
2979 | 397 x -= shift_down(k[i] * state[i], LATTICE_SHIFT); |
398 state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT); | |
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399 } |
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400 #endif |
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401 |
2967 | 402 // don't drift too far, to avoid overflows |
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403 if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16); |
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404 if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16); |
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405 |
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406 state[0] = x; |
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407 |
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408 return x; |
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409 } |
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410 |
8590 | 411 #if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER |
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412 // Heavily modified Levinson-Durbin algorithm which |
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413 // copes better with quantization, and calculates the |
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414 // actual whitened result as it goes. |
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415 |
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416 static void modified_levinson_durbin(int *window, int window_entries, |
2979 | 417 int *out, int out_entries, int channels, int *tap_quant) |
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418 { |
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419 int i; |
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420 int *state = av_mallocz(4* window_entries); |
2967 | 421 |
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422 memcpy(state, window, 4* window_entries); |
2967 | 423 |
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424 for (i = 0; i < out_entries; i++) |
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425 { |
2979 | 426 int step = (i+1)*channels, k, j; |
427 double xx = 0.0, xy = 0.0; | |
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428 #if 1 |
2979 | 429 int *x_ptr = &(window[step]), *state_ptr = &(state[0]); |
430 j = window_entries - step; | |
431 for (;j>=0;j--,x_ptr++,state_ptr++) | |
432 { | |
433 double x_value = *x_ptr, state_value = *state_ptr; | |
434 xx += state_value*state_value; | |
435 xy += x_value*state_value; | |
436 } | |
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437 #else |
2979 | 438 for (j = 0; j <= (window_entries - step); j++); |
439 { | |
440 double stepval = window[step+j], stateval = window[j]; | |
441 // xx += (double)window[j]*(double)window[j]; | |
442 // xy += (double)window[step+j]*(double)window[j]; | |
443 xx += stateval*stateval; | |
444 xy += stepval*stateval; | |
445 } | |
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446 #endif |
2979 | 447 if (xx == 0.0) |
448 k = 0; | |
449 else | |
450 k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5)); | |
2967 | 451 |
2979 | 452 if (k > (LATTICE_FACTOR/tap_quant[i])) |
453 k = LATTICE_FACTOR/tap_quant[i]; | |
454 if (-k > (LATTICE_FACTOR/tap_quant[i])) | |
455 k = -(LATTICE_FACTOR/tap_quant[i]); | |
2967 | 456 |
2979 | 457 out[i] = k; |
458 k *= tap_quant[i]; | |
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459 |
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460 #if 1 |
2979 | 461 x_ptr = &(window[step]); |
462 state_ptr = &(state[0]); | |
463 j = window_entries - step; | |
464 for (;j>=0;j--,x_ptr++,state_ptr++) | |
465 { | |
466 int x_value = *x_ptr, state_value = *state_ptr; | |
467 *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT); | |
468 *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT); | |
469 } | |
2967 | 470 #else |
2979 | 471 for (j=0; j <= (window_entries - step); j++) |
472 { | |
473 int stepval = window[step+j], stateval=state[j]; | |
474 window[step+j] += shift_down(k * stateval, LATTICE_SHIFT); | |
475 state[j] += shift_down(k * stepval, LATTICE_SHIFT); | |
476 } | |
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477 #endif |
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478 } |
2967 | 479 |
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480 av_free(state); |
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481 } |
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482 |
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483 static inline int code_samplerate(int samplerate) |
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484 { |
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485 switch (samplerate) |
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486 { |
2979 | 487 case 44100: return 0; |
488 case 22050: return 1; | |
489 case 11025: return 2; | |
490 case 96000: return 3; | |
491 case 48000: return 4; | |
492 case 32000: return 5; | |
493 case 24000: return 6; | |
494 case 16000: return 7; | |
495 case 8000: return 8; | |
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496 } |
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497 return -1; |
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498 } |
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499 |
6517
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500 static av_cold int sonic_encode_init(AVCodecContext *avctx) |
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501 { |
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502 SonicContext *s = avctx->priv_data; |
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503 PutBitContext pb; |
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504 int i, version = 0; |
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505 |
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506 if (avctx->channels > MAX_CHANNELS) |
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507 { |
2979 | 508 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); |
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509 return -1; /* only stereo or mono for now */ |
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510 } |
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511 |
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512 if (avctx->channels == 2) |
2979 | 513 s->decorrelation = MID_SIDE; |
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514 |
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515 if (avctx->codec->id == CODEC_ID_SONIC_LS) |
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516 { |
2979 | 517 s->lossless = 1; |
518 s->num_taps = 32; | |
519 s->downsampling = 1; | |
520 s->quantization = 0.0; | |
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521 } |
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522 else |
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523 { |
2979 | 524 s->num_taps = 128; |
525 s->downsampling = 2; | |
526 s->quantization = 1.0; | |
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527 } |
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528 |
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529 // max tap 2048 |
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530 if ((s->num_taps < 32) || (s->num_taps > 1024) || |
2979 | 531 ((s->num_taps>>5)<<5 != s->num_taps)) |
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532 { |
2979 | 533 av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n"); |
534 return -1; | |
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535 } |
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536 |
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537 // generate taps |
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538 s->tap_quant = av_mallocz(4* s->num_taps); |
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539 for (i = 0; i < s->num_taps; i++) |
2979 | 540 s->tap_quant[i] = (int)(sqrt(i+1)); |
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541 |
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542 s->channels = avctx->channels; |
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543 s->samplerate = avctx->sample_rate; |
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544 |
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545 s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling; |
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546 s->frame_size = s->channels*s->block_align*s->downsampling; |
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547 |
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548 s->tail = av_mallocz(4* s->num_taps*s->channels); |
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549 if (!s->tail) |
2979 | 550 return -1; |
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551 s->tail_size = s->num_taps*s->channels; |
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552 |
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553 s->predictor_k = av_mallocz(4 * s->num_taps); |
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554 if (!s->predictor_k) |
2979 | 555 return -1; |
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556 |
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557 for (i = 0; i < s->channels; i++) |
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558 { |
2979 | 559 s->coded_samples[i] = av_mallocz(4* s->block_align); |
560 if (!s->coded_samples[i]) | |
561 return -1; | |
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562 } |
2967 | 563 |
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564 s->int_samples = av_mallocz(4* s->frame_size); |
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565 |
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566 s->window_size = ((2*s->tail_size)+s->frame_size); |
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567 s->window = av_mallocz(4* s->window_size); |
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568 if (!s->window) |
2979 | 569 return -1; |
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570 |
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571 avctx->extradata = av_mallocz(16); |
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572 if (!avctx->extradata) |
2979 | 573 return -1; |
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574 init_put_bits(&pb, avctx->extradata, 16*8); |
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575 |
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576 put_bits(&pb, 2, version); // version |
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577 if (version == 1) |
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578 { |
2979 | 579 put_bits(&pb, 2, s->channels); |
580 put_bits(&pb, 4, code_samplerate(s->samplerate)); | |
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581 } |
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582 put_bits(&pb, 1, s->lossless); |
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583 if (!s->lossless) |
2979 | 584 put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision |
2205 | 585 put_bits(&pb, 2, s->decorrelation); |
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586 put_bits(&pb, 2, s->downsampling); |
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587 put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024 |
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588 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table |
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589 |
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590 flush_put_bits(&pb); |
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591 avctx->extradata_size = put_bits_count(&pb)/8; |
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592 |
2205 | 593 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
2979 | 594 version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); |
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595 |
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596 avctx->coded_frame = avcodec_alloc_frame(); |
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597 if (!avctx->coded_frame) |
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598 return AVERROR(ENOMEM); |
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599 avctx->coded_frame->key_frame = 1; |
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600 avctx->frame_size = s->block_align*s->downsampling; |
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alex
parents:
diff
changeset
|
601 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
602 return 0; |
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alex
parents:
diff
changeset
|
603 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
604 |
6517
48759bfbd073
Apply 'cold' attribute to init/uninit functions in libavcodec
zuxy
parents:
6253
diff
changeset
|
605 static av_cold int sonic_encode_close(AVCodecContext *avctx) |
2182
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alex
parents:
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|
606 { |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
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parents:
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|
607 SonicContext *s = avctx->priv_data; |
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alex
parents:
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changeset
|
608 int i; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
609 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
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|
610 av_freep(&avctx->coded_frame); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
611 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
612 for (i = 0; i < s->channels; i++) |
2979 | 613 av_free(s->coded_samples[i]); |
2182
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alex
parents:
diff
changeset
|
614 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
615 av_free(s->predictor_k); |
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alex
parents:
diff
changeset
|
616 av_free(s->tail); |
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alex
parents:
diff
changeset
|
617 av_free(s->tap_quant); |
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alex
parents:
diff
changeset
|
618 av_free(s->window); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
619 av_free(s->int_samples); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
620 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
621 return 0; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
622 } |
113732773bda
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alex
parents:
diff
changeset
|
623 |
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alex
parents:
diff
changeset
|
624 static int sonic_encode_frame(AVCodecContext *avctx, |
2979 | 625 uint8_t *buf, int buf_size, void *data) |
2182
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alex
parents:
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|
626 { |
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alex
parents:
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|
627 SonicContext *s = avctx->priv_data; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
628 PutBitContext pb; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
629 int i, j, ch, quant = 0, x = 0; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
630 short *samples = data; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
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|
631 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
632 init_put_bits(&pb, buf, buf_size*8); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
633 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
634 // short -> internal |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
635 for (i = 0; i < s->frame_size; i++) |
2979 | 636 s->int_samples[i] = samples[i]; |
2182
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alex
parents:
diff
changeset
|
637 |
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alex
parents:
diff
changeset
|
638 if (!s->lossless) |
2979 | 639 for (i = 0; i < s->frame_size; i++) |
640 s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT; | |
2182
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alex
parents:
diff
changeset
|
641 |
2205 | 642 switch(s->decorrelation) |
643 { | |
2979 | 644 case MID_SIDE: |
645 for (i = 0; i < s->frame_size; i += s->channels) | |
646 { | |
647 s->int_samples[i] += s->int_samples[i+1]; | |
648 s->int_samples[i+1] -= shift(s->int_samples[i], 1); | |
649 } | |
650 break; | |
651 case LEFT_SIDE: | |
652 for (i = 0; i < s->frame_size; i += s->channels) | |
653 s->int_samples[i+1] -= s->int_samples[i]; | |
654 break; | |
655 case RIGHT_SIDE: | |
656 for (i = 0; i < s->frame_size; i += s->channels) | |
657 s->int_samples[i] -= s->int_samples[i+1]; | |
658 break; | |
2205 | 659 } |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
660 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
661 memset(s->window, 0, 4* s->window_size); |
2967 | 662 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
663 for (i = 0; i < s->tail_size; i++) |
2979 | 664 s->window[x++] = s->tail[i]; |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
665 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
666 for (i = 0; i < s->frame_size; i++) |
2979 | 667 s->window[x++] = s->int_samples[i]; |
2967 | 668 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
669 for (i = 0; i < s->tail_size; i++) |
2979 | 670 s->window[x++] = 0; |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
671 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
672 for (i = 0; i < s->tail_size; i++) |
2979 | 673 s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i]; |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
674 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
675 // generate taps |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
676 modified_levinson_durbin(s->window, s->window_size, |
2979 | 677 s->predictor_k, s->num_taps, s->channels, s->tap_quant); |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
678 if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0) |
2979 | 679 return -1; |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
680 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
681 for (ch = 0; ch < s->channels; ch++) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
682 { |
2979 | 683 x = s->tail_size+ch; |
684 for (i = 0; i < s->block_align; i++) | |
685 { | |
686 int sum = 0; | |
687 for (j = 0; j < s->downsampling; j++, x += s->channels) | |
688 sum += s->window[x]; | |
689 s->coded_samples[ch][i] = sum; | |
690 } | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
691 } |
2967 | 692 |
693 // simple rate control code | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
694 if (!s->lossless) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
695 { |
2979 | 696 double energy1 = 0.0, energy2 = 0.0; |
697 for (ch = 0; ch < s->channels; ch++) | |
698 { | |
699 for (i = 0; i < s->block_align; i++) | |
700 { | |
701 double sample = s->coded_samples[ch][i]; | |
702 energy2 += sample*sample; | |
703 energy1 += fabs(sample); | |
704 } | |
705 } | |
2967 | 706 |
2979 | 707 energy2 = sqrt(energy2/(s->channels*s->block_align)); |
708 energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align); | |
2967 | 709 |
2979 | 710 // increase bitrate when samples are like a gaussian distribution |
711 // reduce bitrate when samples are like a two-tailed exponential distribution | |
2967 | 712 |
2979 | 713 if (energy2 > energy1) |
714 energy2 += (energy2-energy1)*RATE_VARIATION; | |
2967 | 715 |
2979 | 716 quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR); |
717 // av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2); | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
718 |
2979 | 719 if (quant < 1) |
720 quant = 1; | |
721 if (quant > 65535) | |
722 quant = 65535; | |
2967 | 723 |
2979 | 724 set_ue_golomb(&pb, quant); |
2967 | 725 |
2979 | 726 quant *= SAMPLE_FACTOR; |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
727 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
728 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
729 // write out coded samples |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
730 for (ch = 0; ch < s->channels; ch++) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
731 { |
2979 | 732 if (!s->lossless) |
733 for (i = 0; i < s->block_align; i++) | |
734 s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant); | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
735 |
2979 | 736 if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0) |
737 return -1; | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
738 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
739 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
740 // av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
741 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
742 flush_put_bits(&pb); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
743 return (put_bits_count(&pb)+7)/8; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
744 } |
8590 | 745 #endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */ |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
746 |
8590 | 747 #if CONFIG_SONIC_DECODER |
9955
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
748 static const int samplerate_table[] = |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
749 { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 }; |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
750 |
6517
48759bfbd073
Apply 'cold' attribute to init/uninit functions in libavcodec
zuxy
parents:
6253
diff
changeset
|
751 static av_cold int sonic_decode_init(AVCodecContext *avctx) |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
752 { |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
753 SonicContext *s = avctx->priv_data; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
754 GetBitContext gb; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
755 int i, version; |
2967 | 756 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
757 s->channels = avctx->channels; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
758 s->samplerate = avctx->sample_rate; |
2967 | 759 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
760 if (!avctx->extradata) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
761 { |
2979 | 762 av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n"); |
763 return -1; | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
764 } |
2967 | 765 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
766 init_get_bits(&gb, avctx->extradata, avctx->extradata_size); |
2967 | 767 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
768 version = get_bits(&gb, 2); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
769 if (version > 1) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
770 { |
2979 | 771 av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n"); |
772 return -1; | |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
773 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
774 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
775 if (version == 1) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
776 { |
2979 | 777 s->channels = get_bits(&gb, 2); |
778 s->samplerate = samplerate_table[get_bits(&gb, 4)]; | |
779 av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n", | |
780 s->channels, s->samplerate); | |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
781 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
782 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
783 if (s->channels > MAX_CHANNELS) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
784 { |
2979 | 785 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); |
786 return -1; | |
2182
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alex
parents:
diff
changeset
|
787 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
788 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
789 s->lossless = get_bits1(&gb); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
790 if (!s->lossless) |
2979 | 791 skip_bits(&gb, 3); // XXX FIXME |
2205 | 792 s->decorrelation = get_bits(&gb, 2); |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
793 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
794 s->downsampling = get_bits(&gb, 2); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
795 s->num_taps = (get_bits(&gb, 5)+1)<<5; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
796 if (get_bits1(&gb)) // XXX FIXME |
2979 | 797 av_log(avctx, AV_LOG_INFO, "Custom quant table\n"); |
2967 | 798 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
799 s->block_align = (int)(2048.0*(s->samplerate/44100))/s->downsampling; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
800 s->frame_size = s->channels*s->block_align*s->downsampling; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
801 // avctx->frame_size = s->block_align; |
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alex
parents:
diff
changeset
|
802 |
2205 | 803 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
2979 | 804 version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
805 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
806 // generate taps |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
807 s->tap_quant = av_mallocz(4* s->num_taps); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
808 for (i = 0; i < s->num_taps; i++) |
2979 | 809 s->tap_quant[i] = (int)(sqrt(i+1)); |
2967 | 810 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
811 s->predictor_k = av_mallocz(4* s->num_taps); |
2967 | 812 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
813 for (i = 0; i < s->channels; i++) |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
814 { |
2979 | 815 s->predictor_state[i] = av_mallocz(4* s->num_taps); |
816 if (!s->predictor_state[i]) | |
817 return -1; | |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
818 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
819 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
820 for (i = 0; i < s->channels; i++) |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
821 { |
2979 | 822 s->coded_samples[i] = av_mallocz(4* s->block_align); |
823 if (!s->coded_samples[i]) | |
824 return -1; | |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
825 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
826 s->int_samples = av_mallocz(4* s->frame_size); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
827 |
7451
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
pross
parents:
7129
diff
changeset
|
828 avctx->sample_fmt = SAMPLE_FMT_S16; |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
829 return 0; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
830 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
831 |
6517
48759bfbd073
Apply 'cold' attribute to init/uninit functions in libavcodec
zuxy
parents:
6253
diff
changeset
|
832 static av_cold int sonic_decode_close(AVCodecContext *avctx) |
2182
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alex
parents:
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|
833 { |
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alex
parents:
diff
changeset
|
834 SonicContext *s = avctx->priv_data; |
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alex
parents:
diff
changeset
|
835 int i; |
2967 | 836 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
837 av_free(s->int_samples); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
838 av_free(s->tap_quant); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
839 av_free(s->predictor_k); |
2967 | 840 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
841 for (i = 0; i < s->channels; i++) |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
842 { |
2979 | 843 av_free(s->predictor_state[i]); |
844 av_free(s->coded_samples[i]); | |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
845 } |
2967 | 846 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
847 return 0; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
848 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
849 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
850 static int sonic_decode_frame(AVCodecContext *avctx, |
2979 | 851 void *data, int *data_size, |
9355
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
8718
diff
changeset
|
852 AVPacket *avpkt) |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
853 { |
9355
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
8718
diff
changeset
|
854 const uint8_t *buf = avpkt->data; |
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
8718
diff
changeset
|
855 int buf_size = avpkt->size; |
2182
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alex
parents:
diff
changeset
|
856 SonicContext *s = avctx->priv_data; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
857 GetBitContext gb; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
858 int i, quant, ch, j; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
859 short *samples = data; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
860 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
861 if (buf_size == 0) return 0; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
862 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
863 // av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size); |
2967 | 864 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
865 init_get_bits(&gb, buf, buf_size*8); |
2967 | 866 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
867 intlist_read(&gb, s->predictor_k, s->num_taps, 0); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
868 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
869 // dequantize |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
870 for (i = 0; i < s->num_taps; i++) |
2979 | 871 s->predictor_k[i] *= s->tap_quant[i]; |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
872 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
873 if (s->lossless) |
2979 | 874 quant = 1; |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
875 else |
2979 | 876 quant = get_ue_golomb(&gb) * SAMPLE_FACTOR; |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
877 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
878 // av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
879 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
880 for (ch = 0; ch < s->channels; ch++) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
881 { |
2979 | 882 int x = ch; |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
883 |
2979 | 884 predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps); |
2967 | 885 |
2979 | 886 intlist_read(&gb, s->coded_samples[ch], s->block_align, 1); |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
887 |
2979 | 888 for (i = 0; i < s->block_align; i++) |
889 { | |
890 for (j = 0; j < s->downsampling - 1; j++) | |
891 { | |
892 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0); | |
893 x += s->channels; | |
894 } | |
2967 | 895 |
2979 | 896 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant); |
897 x += s->channels; | |
898 } | |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
899 |
2979 | 900 for (i = 0; i < s->num_taps; i++) |
901 s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels]; | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
902 } |
2967 | 903 |
2205 | 904 switch(s->decorrelation) |
905 { | |
2979 | 906 case MID_SIDE: |
907 for (i = 0; i < s->frame_size; i += s->channels) | |
908 { | |
909 s->int_samples[i+1] += shift(s->int_samples[i], 1); | |
910 s->int_samples[i] -= s->int_samples[i+1]; | |
911 } | |
912 break; | |
913 case LEFT_SIDE: | |
914 for (i = 0; i < s->frame_size; i += s->channels) | |
915 s->int_samples[i+1] += s->int_samples[i]; | |
916 break; | |
917 case RIGHT_SIDE: | |
918 for (i = 0; i < s->frame_size; i += s->channels) | |
919 s->int_samples[i] += s->int_samples[i+1]; | |
920 break; | |
2205 | 921 } |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
922 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
923 if (!s->lossless) |
2979 | 924 for (i = 0; i < s->frame_size; i++) |
925 s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT); | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
926 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
927 // internal -> short |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
928 for (i = 0; i < s->frame_size; i++) |
5523 | 929 samples[i] = av_clip_int16(s->int_samples[i]); |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
930 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
931 align_get_bits(&gb); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
932 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
933 *data_size = s->frame_size * 2; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
934 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
935 return (get_bits_count(&gb)+7)/8; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
936 } |
9955
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
937 |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
938 AVCodec sonic_decoder = { |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
939 "sonic", |
11560
8a4984c5cacc
Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
parents:
9955
diff
changeset
|
940 AVMEDIA_TYPE_AUDIO, |
9955
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
941 CODEC_ID_SONIC, |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
942 sizeof(SonicContext), |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
943 sonic_decode_init, |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
944 NULL, |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
945 sonic_decode_close, |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
946 sonic_decode_frame, |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
947 .long_name = NULL_IF_CONFIG_SMALL("Sonic"), |
4ebcb6c121e4
cosmetics: Move some code blocks to reduce the number of #ifdefs.
diego
parents:
9428
diff
changeset
|
948 }; |
7776
dbcdd0165e55
Replace generic CONFIG_DECODERS preprocessor conditionals by more specific
diego
parents:
7451
diff
changeset
|
949 #endif /* CONFIG_SONIC_DECODER */ |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
950 |
8590 | 951 #if CONFIG_SONIC_ENCODER |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
952 AVCodec sonic_encoder = { |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
953 "sonic", |
11560
8a4984c5cacc
Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
parents:
9955
diff
changeset
|
954 AVMEDIA_TYPE_AUDIO, |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
955 CODEC_ID_SONIC, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
956 sizeof(SonicContext), |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
957 sonic_encode_init, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
958 sonic_encode_frame, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
959 sonic_encode_close, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
960 NULL, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6713
diff
changeset
|
961 .long_name = NULL_IF_CONFIG_SMALL("Sonic"), |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
962 }; |
7782
6efb15a24e91
Replace generic CONFIG_ENCODERS preprocessor conditionals by more specific
diego
parents:
7776
diff
changeset
|
963 #endif |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
964 |
8590 | 965 #if CONFIG_SONIC_LS_ENCODER |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
966 AVCodec sonic_ls_encoder = { |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
967 "sonicls", |
11560
8a4984c5cacc
Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
parents:
9955
diff
changeset
|
968 AVMEDIA_TYPE_AUDIO, |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
969 CODEC_ID_SONIC_LS, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
970 sizeof(SonicContext), |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
971 sonic_encode_init, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
972 sonic_encode_frame, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
973 sonic_encode_close, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
974 NULL, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6713
diff
changeset
|
975 .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"), |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
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976 }; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
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977 #endif |