Mercurial > libavcodec.hg
annotate sonic.c @ 9501:880d9562d218 libavcodec
Avoid code duplication in xan_unpack for the final memcpy.
author | reimar |
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date | Sat, 18 Apr 2009 17:26:00 +0000 |
parents | 0dce4fe6e6f3 |
children | 4ebcb6c121e4 |
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 libavcodec/sonic.c |
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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 } |
8590 | 482 #endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */ |
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483 |
7129 | 484 static const int samplerate_table[] = |
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485 { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 }; |
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486 |
8590 | 487 #if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER |
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488 static inline int code_samplerate(int samplerate) |
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489 { |
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490 switch (samplerate) |
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491 { |
2979 | 492 case 44100: return 0; |
493 case 22050: return 1; | |
494 case 11025: return 2; | |
495 case 96000: return 3; | |
496 case 48000: return 4; | |
497 case 32000: return 5; | |
498 case 24000: return 6; | |
499 case 16000: return 7; | |
500 case 8000: return 8; | |
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501 } |
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502 return -1; |
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503 } |
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504 |
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505 static av_cold int sonic_encode_init(AVCodecContext *avctx) |
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506 { |
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507 SonicContext *s = avctx->priv_data; |
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508 PutBitContext pb; |
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509 int i, version = 0; |
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510 |
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511 if (avctx->channels > MAX_CHANNELS) |
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512 { |
2979 | 513 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); |
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514 return -1; /* only stereo or mono for now */ |
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515 } |
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516 |
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517 if (avctx->channels == 2) |
2979 | 518 s->decorrelation = MID_SIDE; |
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519 |
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520 if (avctx->codec->id == CODEC_ID_SONIC_LS) |
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521 { |
2979 | 522 s->lossless = 1; |
523 s->num_taps = 32; | |
524 s->downsampling = 1; | |
525 s->quantization = 0.0; | |
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526 } |
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527 else |
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528 { |
2979 | 529 s->num_taps = 128; |
530 s->downsampling = 2; | |
531 s->quantization = 1.0; | |
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532 } |
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533 |
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534 // max tap 2048 |
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535 if ((s->num_taps < 32) || (s->num_taps > 1024) || |
2979 | 536 ((s->num_taps>>5)<<5 != s->num_taps)) |
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537 { |
2979 | 538 av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n"); |
539 return -1; | |
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540 } |
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541 |
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542 // generate taps |
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543 s->tap_quant = av_mallocz(4* s->num_taps); |
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544 for (i = 0; i < s->num_taps; i++) |
2979 | 545 s->tap_quant[i] = (int)(sqrt(i+1)); |
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546 |
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547 s->channels = avctx->channels; |
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548 s->samplerate = avctx->sample_rate; |
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549 |
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550 s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling; |
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551 s->frame_size = s->channels*s->block_align*s->downsampling; |
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552 |
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553 s->tail = av_mallocz(4* s->num_taps*s->channels); |
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554 if (!s->tail) |
2979 | 555 return -1; |
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556 s->tail_size = s->num_taps*s->channels; |
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557 |
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558 s->predictor_k = av_mallocz(4 * s->num_taps); |
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559 if (!s->predictor_k) |
2979 | 560 return -1; |
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561 |
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562 for (i = 0; i < s->channels; i++) |
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563 { |
2979 | 564 s->coded_samples[i] = av_mallocz(4* s->block_align); |
565 if (!s->coded_samples[i]) | |
566 return -1; | |
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567 } |
2967 | 568 |
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569 s->int_samples = av_mallocz(4* s->frame_size); |
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570 |
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571 s->window_size = ((2*s->tail_size)+s->frame_size); |
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572 s->window = av_mallocz(4* s->window_size); |
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573 if (!s->window) |
2979 | 574 return -1; |
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575 |
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576 avctx->extradata = av_mallocz(16); |
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577 if (!avctx->extradata) |
2979 | 578 return -1; |
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579 init_put_bits(&pb, avctx->extradata, 16*8); |
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580 |
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581 put_bits(&pb, 2, version); // version |
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582 if (version == 1) |
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583 { |
2979 | 584 put_bits(&pb, 2, s->channels); |
585 put_bits(&pb, 4, code_samplerate(s->samplerate)); | |
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586 } |
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587 put_bits(&pb, 1, s->lossless); |
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588 if (!s->lossless) |
2979 | 589 put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision |
2205 | 590 put_bits(&pb, 2, s->decorrelation); |
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591 put_bits(&pb, 2, s->downsampling); |
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592 put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024 |
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593 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table |
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594 |
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595 flush_put_bits(&pb); |
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596 avctx->extradata_size = put_bits_count(&pb)/8; |
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597 |
2205 | 598 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
2979 | 599 version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); |
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600 |
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601 avctx->coded_frame = avcodec_alloc_frame(); |
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602 if (!avctx->coded_frame) |
4520
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603 return AVERROR(ENOMEM); |
2182
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604 avctx->coded_frame->key_frame = 1; |
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605 avctx->frame_size = s->block_align*s->downsampling; |
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606 |
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607 return 0; |
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|
608 } |
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|
609 |
6517
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Apply 'cold' attribute to init/uninit functions in libavcodec
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610 static av_cold int sonic_encode_close(AVCodecContext *avctx) |
2182
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611 { |
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612 SonicContext *s = avctx->priv_data; |
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613 int i; |
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614 |
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615 av_freep(&avctx->coded_frame); |
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616 |
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617 for (i = 0; i < s->channels; i++) |
2979 | 618 av_free(s->coded_samples[i]); |
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619 |
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620 av_free(s->predictor_k); |
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621 av_free(s->tail); |
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622 av_free(s->tap_quant); |
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623 av_free(s->window); |
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624 av_free(s->int_samples); |
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|
625 |
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626 return 0; |
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627 } |
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628 |
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629 static int sonic_encode_frame(AVCodecContext *avctx, |
2979 | 630 uint8_t *buf, int buf_size, void *data) |
2182
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631 { |
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|
632 SonicContext *s = avctx->priv_data; |
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parents:
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633 PutBitContext pb; |
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634 int i, j, ch, quant = 0, x = 0; |
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635 short *samples = data; |
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parents:
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636 |
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637 init_put_bits(&pb, buf, buf_size*8); |
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alex
parents:
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638 |
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alex
parents:
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|
639 // short -> internal |
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parents:
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|
640 for (i = 0; i < s->frame_size; i++) |
2979 | 641 s->int_samples[i] = samples[i]; |
2182
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642 |
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643 if (!s->lossless) |
2979 | 644 for (i = 0; i < s->frame_size; i++) |
645 s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT; | |
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646 |
2205 | 647 switch(s->decorrelation) |
648 { | |
2979 | 649 case MID_SIDE: |
650 for (i = 0; i < s->frame_size; i += s->channels) | |
651 { | |
652 s->int_samples[i] += s->int_samples[i+1]; | |
653 s->int_samples[i+1] -= shift(s->int_samples[i], 1); | |
654 } | |
655 break; | |
656 case LEFT_SIDE: | |
657 for (i = 0; i < s->frame_size; i += s->channels) | |
658 s->int_samples[i+1] -= s->int_samples[i]; | |
659 break; | |
660 case RIGHT_SIDE: | |
661 for (i = 0; i < s->frame_size; i += s->channels) | |
662 s->int_samples[i] -= s->int_samples[i+1]; | |
663 break; | |
2205 | 664 } |
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parents:
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665 |
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parents:
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|
666 memset(s->window, 0, 4* s->window_size); |
2967 | 667 |
2182
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alex
parents:
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668 for (i = 0; i < s->tail_size; i++) |
2979 | 669 s->window[x++] = s->tail[i]; |
2182
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alex
parents:
diff
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|
670 |
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alex
parents:
diff
changeset
|
671 for (i = 0; i < s->frame_size; i++) |
2979 | 672 s->window[x++] = s->int_samples[i]; |
2967 | 673 |
2182
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parents:
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|
674 for (i = 0; i < s->tail_size; i++) |
2979 | 675 s->window[x++] = 0; |
2182
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alex
parents:
diff
changeset
|
676 |
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alex
parents:
diff
changeset
|
677 for (i = 0; i < s->tail_size; i++) |
2979 | 678 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
|
679 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
680 // generate taps |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
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parents:
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|
681 modified_levinson_durbin(s->window, s->window_size, |
2979 | 682 s->predictor_k, s->num_taps, s->channels, s->tap_quant); |
2182
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683 if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0) |
2979 | 684 return -1; |
2182
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alex
parents:
diff
changeset
|
685 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
686 for (ch = 0; ch < s->channels; ch++) |
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alex
parents:
diff
changeset
|
687 { |
2979 | 688 x = s->tail_size+ch; |
689 for (i = 0; i < s->block_align; i++) | |
690 { | |
691 int sum = 0; | |
692 for (j = 0; j < s->downsampling; j++, x += s->channels) | |
693 sum += s->window[x]; | |
694 s->coded_samples[ch][i] = sum; | |
695 } | |
2182
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alex
parents:
diff
changeset
|
696 } |
2967 | 697 |
698 // simple rate control code | |
2182
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alex
parents:
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|
699 if (!s->lossless) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
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|
700 { |
2979 | 701 double energy1 = 0.0, energy2 = 0.0; |
702 for (ch = 0; ch < s->channels; ch++) | |
703 { | |
704 for (i = 0; i < s->block_align; i++) | |
705 { | |
706 double sample = s->coded_samples[ch][i]; | |
707 energy2 += sample*sample; | |
708 energy1 += fabs(sample); | |
709 } | |
710 } | |
2967 | 711 |
2979 | 712 energy2 = sqrt(energy2/(s->channels*s->block_align)); |
713 energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align); | |
2967 | 714 |
2979 | 715 // increase bitrate when samples are like a gaussian distribution |
716 // reduce bitrate when samples are like a two-tailed exponential distribution | |
2967 | 717 |
2979 | 718 if (energy2 > energy1) |
719 energy2 += (energy2-energy1)*RATE_VARIATION; | |
2967 | 720 |
2979 | 721 quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR); |
722 // av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2); | |
2182
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alex
parents:
diff
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|
723 |
2979 | 724 if (quant < 1) |
725 quant = 1; | |
726 if (quant > 65535) | |
727 quant = 65535; | |
2967 | 728 |
2979 | 729 set_ue_golomb(&pb, quant); |
2967 | 730 |
2979 | 731 quant *= 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
|
732 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
733 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
734 // write out coded samples |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
735 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
|
736 { |
2979 | 737 if (!s->lossless) |
738 for (i = 0; i < s->block_align; i++) | |
739 s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant); | |
2182
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alex
parents:
diff
changeset
|
740 |
2979 | 741 if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0) |
742 return -1; | |
2182
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alex
parents:
diff
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|
743 } |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
744 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
745 // av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8); |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
746 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
747 flush_put_bits(&pb); |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
748 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
|
749 } |
8590 | 750 #endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */ |
2182
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alex
parents:
diff
changeset
|
751 |
8590 | 752 #if CONFIG_SONIC_DECODER |
6517
48759bfbd073
Apply 'cold' attribute to init/uninit functions in libavcodec
zuxy
parents:
6253
diff
changeset
|
753 static av_cold int sonic_decode_init(AVCodecContext *avctx) |
2182
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alex
parents:
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|
754 { |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
755 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
|
756 GetBitContext gb; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
757 int i, version; |
2967 | 758 |
2182
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alex
parents:
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changeset
|
759 s->channels = avctx->channels; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
760 s->samplerate = avctx->sample_rate; |
2967 | 761 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
762 if (!avctx->extradata) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
763 { |
2979 | 764 av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n"); |
765 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
|
766 } |
2967 | 767 |
2182
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alex
parents:
diff
changeset
|
768 init_get_bits(&gb, avctx->extradata, avctx->extradata_size); |
2967 | 769 |
2182
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alex
parents:
diff
changeset
|
770 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
|
771 if (version > 1) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
772 { |
2979 | 773 av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n"); |
774 return -1; | |
2182
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alex
parents:
diff
changeset
|
775 } |
113732773bda
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alex
parents:
diff
changeset
|
776 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
777 if (version == 1) |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
778 { |
2979 | 779 s->channels = get_bits(&gb, 2); |
780 s->samplerate = samplerate_table[get_bits(&gb, 4)]; | |
781 av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n", | |
782 s->channels, s->samplerate); | |
2182
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alex
parents:
diff
changeset
|
783 } |
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alex
parents:
diff
changeset
|
784 |
113732773bda
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alex
parents:
diff
changeset
|
785 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
|
786 { |
2979 | 787 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); |
788 return -1; | |
2182
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|
789 } |
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alex
parents:
diff
changeset
|
790 |
113732773bda
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alex
parents:
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changeset
|
791 s->lossless = get_bits1(&gb); |
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alex
parents:
diff
changeset
|
792 if (!s->lossless) |
2979 | 793 skip_bits(&gb, 3); // XXX FIXME |
2205 | 794 s->decorrelation = get_bits(&gb, 2); |
2182
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parents:
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|
795 |
113732773bda
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alex
parents:
diff
changeset
|
796 s->downsampling = get_bits(&gb, 2); |
113732773bda
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alex
parents:
diff
changeset
|
797 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
|
798 if (get_bits1(&gb)) // XXX FIXME |
2979 | 799 av_log(avctx, AV_LOG_INFO, "Custom quant table\n"); |
2967 | 800 |
2182
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alex
parents:
diff
changeset
|
801 s->block_align = (int)(2048.0*(s->samplerate/44100))/s->downsampling; |
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alex
parents:
diff
changeset
|
802 s->frame_size = s->channels*s->block_align*s->downsampling; |
113732773bda
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alex
parents:
diff
changeset
|
803 // avctx->frame_size = s->block_align; |
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alex
parents:
diff
changeset
|
804 |
2205 | 805 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
2979 | 806 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
|
807 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
808 // 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
|
809 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
|
810 for (i = 0; i < s->num_taps; i++) |
2979 | 811 s->tap_quant[i] = (int)(sqrt(i+1)); |
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 s->predictor_k = av_mallocz(4* s->num_taps); |
2967 | 814 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
815 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
|
816 { |
2979 | 817 s->predictor_state[i] = av_mallocz(4* s->num_taps); |
818 if (!s->predictor_state[i]) | |
819 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
|
820 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
821 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
822 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
|
823 { |
2979 | 824 s->coded_samples[i] = av_mallocz(4* s->block_align); |
825 if (!s->coded_samples[i]) | |
826 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
|
827 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
828 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
|
829 |
7451
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
pross
parents:
7129
diff
changeset
|
830 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
|
831 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
|
832 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
833 |
6517
48759bfbd073
Apply 'cold' attribute to init/uninit functions in libavcodec
zuxy
parents:
6253
diff
changeset
|
834 static av_cold int sonic_decode_close(AVCodecContext *avctx) |
2182
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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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|
835 { |
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alex
parents:
diff
changeset
|
836 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
|
837 int i; |
2967 | 838 |
2182
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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->int_samples); |
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alex
parents:
diff
changeset
|
840 av_free(s->tap_quant); |
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alex
parents:
diff
changeset
|
841 av_free(s->predictor_k); |
2967 | 842 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
843 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
|
844 { |
2979 | 845 av_free(s->predictor_state[i]); |
846 av_free(s->coded_samples[i]); | |
2182
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alex
parents:
diff
changeset
|
847 } |
2967 | 848 |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
849 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
|
850 } |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
851 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
852 static int sonic_decode_frame(AVCodecContext *avctx, |
2979 | 853 void *data, int *data_size, |
9355
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
8718
diff
changeset
|
854 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
|
855 { |
9355
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
8718
diff
changeset
|
856 const uint8_t *buf = avpkt->data; |
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
8718
diff
changeset
|
857 int buf_size = avpkt->size; |
2182
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
858 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
|
859 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
|
860 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
|
861 short *samples = data; |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
862 |
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new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
863 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
|
864 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
865 // av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size); |
2967 | 866 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
867 init_get_bits(&gb, buf, buf_size*8); |
2967 | 868 |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
869 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
|
870 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
871 // dequantize |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
872 for (i = 0; i < s->num_taps; i++) |
2979 | 873 s->predictor_k[i] *= s->tap_quant[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
|
874 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
875 if (s->lossless) |
2979 | 876 quant = 1; |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
877 else |
2979 | 878 quant = get_ue_golomb(&gb) * SAMPLE_FACTOR; |
2182
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 // 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
|
881 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
882 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
|
883 { |
2979 | 884 int x = ch; |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
885 |
2979 | 886 predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps); |
2967 | 887 |
2979 | 888 intlist_read(&gb, s->coded_samples[ch], s->block_align, 1); |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
889 |
2979 | 890 for (i = 0; i < s->block_align; i++) |
891 { | |
892 for (j = 0; j < s->downsampling - 1; j++) | |
893 { | |
894 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0); | |
895 x += s->channels; | |
896 } | |
2967 | 897 |
2979 | 898 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant); |
899 x += s->channels; | |
900 } | |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
901 |
2979 | 902 for (i = 0; i < s->num_taps; i++) |
903 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
|
904 } |
2967 | 905 |
2205 | 906 switch(s->decorrelation) |
907 { | |
2979 | 908 case MID_SIDE: |
909 for (i = 0; i < s->frame_size; i += s->channels) | |
910 { | |
911 s->int_samples[i+1] += shift(s->int_samples[i], 1); | |
912 s->int_samples[i] -= s->int_samples[i+1]; | |
913 } | |
914 break; | |
915 case LEFT_SIDE: | |
916 for (i = 0; i < s->frame_size; i += s->channels) | |
917 s->int_samples[i+1] += s->int_samples[i]; | |
918 break; | |
919 case RIGHT_SIDE: | |
920 for (i = 0; i < s->frame_size; i += s->channels) | |
921 s->int_samples[i] += s->int_samples[i+1]; | |
922 break; | |
2205 | 923 } |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
924 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
925 if (!s->lossless) |
2979 | 926 for (i = 0; i < s->frame_size; i++) |
927 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
|
928 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
929 // internal -> short |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
930 for (i = 0; i < s->frame_size; i++) |
5523 | 931 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
|
932 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
933 align_get_bits(&gb); |
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 *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
|
936 |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
937 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
|
938 } |
7776
dbcdd0165e55
Replace generic CONFIG_DECODERS preprocessor conditionals by more specific
diego
parents:
7451
diff
changeset
|
939 #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
|
940 |
8590 | 941 #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
|
942 AVCodec sonic_encoder = { |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
943 "sonic", |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
944 CODEC_TYPE_AUDIO, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
945 CODEC_ID_SONIC, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
946 sizeof(SonicContext), |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
947 sonic_encode_init, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
948 sonic_encode_frame, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
949 sonic_encode_close, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
950 NULL, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6713
diff
changeset
|
951 .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
|
952 }; |
7782
6efb15a24e91
Replace generic CONFIG_ENCODERS preprocessor conditionals by more specific
diego
parents:
7776
diff
changeset
|
953 #endif |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
954 |
8590 | 955 #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
|
956 AVCodec sonic_ls_encoder = { |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
957 "sonicls", |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
958 CODEC_TYPE_AUDIO, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
959 CODEC_ID_SONIC_LS, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
960 sizeof(SonicContext), |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
961 sonic_encode_init, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
962 sonic_encode_frame, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
963 sonic_encode_close, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
964 NULL, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6713
diff
changeset
|
965 .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
|
966 }; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
967 #endif |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
968 |
8590 | 969 #if CONFIG_SONIC_DECODER |
2182
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
970 AVCodec sonic_decoder = { |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
971 "sonic", |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
972 CODEC_TYPE_AUDIO, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
973 CODEC_ID_SONIC, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
974 sizeof(SonicContext), |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
975 sonic_decode_init, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
976 NULL, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
977 sonic_decode_close, |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
978 sonic_decode_frame, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6713
diff
changeset
|
979 .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
|
980 }; |
113732773bda
new opensource lossy/lossless audio codec based on speech compression techniques (actually based on bonk)
alex
parents:
diff
changeset
|
981 #endif |