Mercurial > libavcodec.hg
annotate alacenc.c @ 8247:e8cb66bdae99 libavcodec
Trivial rephrasing, avoids too long line
author | reynaldo |
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date | Tue, 02 Dec 2008 18:26:55 +0000 |
parents | 4525dcd81357 |
children | 5f48c58bfd51 |
rev | line source |
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7595 | 1 /** |
2 * ALAC audio encoder | |
3 * Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net> | |
4 * | |
5 * This file is part of FFmpeg. | |
6 * | |
7 * FFmpeg is free software; you can redistribute it and/or | |
8 * modify it under the terms of the GNU Lesser General Public | |
9 * License as published by the Free Software Foundation; either | |
10 * version 2.1 of the License, or (at your option) any later version. | |
11 * | |
12 * FFmpeg is distributed in the hope that it will be useful, | |
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
18 * License along with FFmpeg; if not, write to the Free Software | |
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
20 */ | |
21 | |
22 #include "avcodec.h" | |
23 #include "bitstream.h" | |
24 #include "dsputil.h" | |
25 #include "lpc.h" | |
26 | |
27 #define DEFAULT_FRAME_SIZE 4096 | |
28 #define DEFAULT_SAMPLE_SIZE 16 | |
29 #define MAX_CHANNELS 8 | |
30 #define ALAC_EXTRADATA_SIZE 36 | |
31 #define ALAC_FRAME_HEADER_SIZE 55 | |
32 #define ALAC_FRAME_FOOTER_SIZE 3 | |
33 | |
34 #define ALAC_ESCAPE_CODE 0x1FF | |
35 #define ALAC_MAX_LPC_ORDER 30 | |
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36 #define DEFAULT_MAX_PRED_ORDER 6 |
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37 #define DEFAULT_MIN_PRED_ORDER 4 |
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38 #define ALAC_MAX_LPC_PRECISION 9 |
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39 #define ALAC_MAX_LPC_SHIFT 9 |
7595 | 40 |
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41 #define ALAC_CHMODE_LEFT_RIGHT 0 |
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42 #define ALAC_CHMODE_LEFT_SIDE 1 |
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43 #define ALAC_CHMODE_RIGHT_SIDE 2 |
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44 #define ALAC_CHMODE_MID_SIDE 3 |
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45 |
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46 typedef struct RiceContext { |
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47 int history_mult; |
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48 int initial_history; |
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49 int k_modifier; |
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50 int rice_modifier; |
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51 } RiceContext; |
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52 |
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53 typedef struct LPCContext { |
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54 int lpc_order; |
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55 int lpc_coeff[ALAC_MAX_LPC_ORDER+1]; |
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56 int lpc_quant; |
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57 } LPCContext; |
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58 |
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59 typedef struct AlacEncodeContext { |
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60 int compression_level; |
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61 int min_prediction_order; |
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62 int max_prediction_order; |
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63 int max_coded_frame_size; |
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64 int write_sample_size; |
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65 int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE]; |
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66 int32_t predictor_buf[DEFAULT_FRAME_SIZE]; |
7595 | 67 int interlacing_shift; |
68 int interlacing_leftweight; | |
69 PutBitContext pbctx; | |
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70 RiceContext rc; |
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71 LPCContext lpc[MAX_CHANNELS]; |
7595 | 72 DSPContext dspctx; |
73 AVCodecContext *avctx; | |
74 } AlacEncodeContext; | |
75 | |
76 | |
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77 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples) |
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78 { |
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79 int ch, i; |
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80 |
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81 for(ch=0;ch<s->avctx->channels;ch++) { |
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82 int16_t *sptr = input_samples + ch; |
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83 for(i=0;i<s->avctx->frame_size;i++) { |
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84 s->sample_buf[ch][i] = *sptr; |
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85 sptr += s->avctx->channels; |
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86 } |
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87 } |
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88 } |
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89 |
7595 | 90 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size) |
91 { | |
92 int divisor, q, r; | |
93 | |
94 k = FFMIN(k, s->rc.k_modifier); | |
95 divisor = (1<<k) - 1; | |
96 q = x / divisor; | |
97 r = x % divisor; | |
98 | |
99 if(q > 8) { | |
100 // write escape code and sample value directly | |
101 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE); | |
102 put_bits(&s->pbctx, write_sample_size, x); | |
103 } else { | |
104 if(q) | |
105 put_bits(&s->pbctx, q, (1<<q) - 1); | |
106 put_bits(&s->pbctx, 1, 0); | |
107 | |
108 if(k != 1) { | |
109 if(r > 0) | |
110 put_bits(&s->pbctx, k, r+1); | |
111 else | |
112 put_bits(&s->pbctx, k-1, 0); | |
113 } | |
114 } | |
115 } | |
116 | |
117 static void write_frame_header(AlacEncodeContext *s, int is_verbatim) | |
118 { | |
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119 put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 |
7595 | 120 put_bits(&s->pbctx, 16, 0); // Seems to be zero |
121 put_bits(&s->pbctx, 1, 1); // Sample count is in the header | |
122 put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field | |
123 put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim | |
124 put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame | |
125 } | |
126 | |
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127 static void calc_predictor_params(AlacEncodeContext *s, int ch) |
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128 { |
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129 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER]; |
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130 int shift[MAX_LPC_ORDER]; |
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131 int opt_order; |
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132 |
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133 opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], s->avctx->frame_size, s->min_prediction_order, s->max_prediction_order, |
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134 ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1); |
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135 |
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136 s->lpc[ch].lpc_order = opt_order; |
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137 s->lpc[ch].lpc_quant = shift[opt_order-1]; |
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138 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int)); |
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139 } |
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140 |
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141 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) |
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142 { |
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143 int i, best; |
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144 int32_t lt, rt; |
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145 uint64_t sum[4]; |
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146 uint64_t score[4]; |
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147 |
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148 /* calculate sum of 2nd order residual for each channel */ |
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149 sum[0] = sum[1] = sum[2] = sum[3] = 0; |
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150 for(i=2; i<n; i++) { |
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151 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; |
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152 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; |
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153 sum[2] += FFABS((lt + rt) >> 1); |
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154 sum[3] += FFABS(lt - rt); |
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155 sum[0] += FFABS(lt); |
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156 sum[1] += FFABS(rt); |
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157 } |
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158 |
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159 /* calculate score for each mode */ |
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160 score[0] = sum[0] + sum[1]; |
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161 score[1] = sum[0] + sum[3]; |
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162 score[2] = sum[1] + sum[3]; |
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163 score[3] = sum[2] + sum[3]; |
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164 |
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165 /* return mode with lowest score */ |
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166 best = 0; |
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167 for(i=1; i<4; i++) { |
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168 if(score[i] < score[best]) { |
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169 best = i; |
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170 } |
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171 } |
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172 return best; |
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173 } |
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174 |
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175 static void alac_stereo_decorrelation(AlacEncodeContext *s) |
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176 { |
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177 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; |
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178 int i, mode, n = s->avctx->frame_size; |
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179 int32_t tmp; |
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180 |
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181 mode = estimate_stereo_mode(left, right, n); |
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182 |
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183 switch(mode) |
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184 { |
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185 case ALAC_CHMODE_LEFT_RIGHT: |
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186 s->interlacing_leftweight = 0; |
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187 s->interlacing_shift = 0; |
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188 break; |
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189 |
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190 case ALAC_CHMODE_LEFT_SIDE: |
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191 for(i=0; i<n; i++) { |
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192 right[i] = left[i] - right[i]; |
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193 } |
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194 s->interlacing_leftweight = 1; |
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195 s->interlacing_shift = 0; |
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196 break; |
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197 |
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198 case ALAC_CHMODE_RIGHT_SIDE: |
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199 for(i=0; i<n; i++) { |
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200 tmp = right[i]; |
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201 right[i] = left[i] - right[i]; |
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202 left[i] = tmp + (right[i] >> 31); |
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203 } |
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204 s->interlacing_leftweight = 1; |
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205 s->interlacing_shift = 31; |
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206 break; |
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207 |
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208 default: |
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209 for(i=0; i<n; i++) { |
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210 tmp = left[i]; |
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211 left[i] = (tmp + right[i]) >> 1; |
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212 right[i] = tmp - right[i]; |
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213 } |
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214 s->interlacing_leftweight = 1; |
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215 s->interlacing_shift = 1; |
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216 break; |
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217 } |
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218 } |
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219 |
7619 | 220 static void alac_linear_predictor(AlacEncodeContext *s, int ch) |
221 { | |
222 int i; | |
223 LPCContext lpc = s->lpc[ch]; | |
224 | |
225 if(lpc.lpc_order == 31) { | |
226 s->predictor_buf[0] = s->sample_buf[ch][0]; | |
227 | |
228 for(i=1; i<s->avctx->frame_size; i++) | |
229 s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; | |
230 | |
231 return; | |
232 } | |
233 | |
234 // generalised linear predictor | |
235 | |
236 if(lpc.lpc_order > 0) { | |
237 int32_t *samples = s->sample_buf[ch]; | |
238 int32_t *residual = s->predictor_buf; | |
239 | |
240 // generate warm-up samples | |
241 residual[0] = samples[0]; | |
242 for(i=1;i<=lpc.lpc_order;i++) | |
243 residual[i] = samples[i] - samples[i-1]; | |
244 | |
245 // perform lpc on remaining samples | |
246 for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { | |
247 int sum = 1 << (lpc.lpc_quant - 1), res_val, j; | |
248 | |
249 for (j = 0; j < lpc.lpc_order; j++) { | |
250 sum += (samples[lpc.lpc_order-j] - samples[0]) * | |
251 lpc.lpc_coeff[j]; | |
252 } | |
253 | |
254 sum >>= lpc.lpc_quant; | |
255 sum += samples[0]; | |
256 residual[i] = samples[lpc.lpc_order+1] - sum; | |
257 res_val = residual[i]; | |
258 | |
259 if(res_val) { | |
260 int index = lpc.lpc_order - 1; | |
261 int neg = (res_val < 0); | |
262 | |
263 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) { | |
264 int val = samples[0] - samples[lpc.lpc_order - index]; | |
265 int sign = (val ? FFSIGN(val) : 0); | |
266 | |
267 if(neg) | |
268 sign*=-1; | |
269 | |
270 lpc.lpc_coeff[index] -= sign; | |
271 val *= sign; | |
272 res_val -= ((val >> lpc.lpc_quant) * | |
273 (lpc.lpc_order - index)); | |
274 index--; | |
275 } | |
276 } | |
277 samples++; | |
278 } | |
279 } | |
280 } | |
281 | |
282 static void alac_entropy_coder(AlacEncodeContext *s) | |
283 { | |
284 unsigned int history = s->rc.initial_history; | |
285 int sign_modifier = 0, i, k; | |
286 int32_t *samples = s->predictor_buf; | |
287 | |
288 for(i=0;i < s->avctx->frame_size;) { | |
289 int x; | |
290 | |
291 k = av_log2((history >> 9) + 3); | |
292 | |
293 x = -2*(*samples)-1; | |
294 x ^= (x>>31); | |
295 | |
296 samples++; | |
297 i++; | |
298 | |
299 encode_scalar(s, x - sign_modifier, k, s->write_sample_size); | |
300 | |
301 history += x * s->rc.history_mult | |
302 - ((history * s->rc.history_mult) >> 9); | |
303 | |
304 sign_modifier = 0; | |
305 if(x > 0xFFFF) | |
306 history = 0xFFFF; | |
307 | |
308 if((history < 128) && (i < s->avctx->frame_size)) { | |
309 unsigned int block_size = 0; | |
310 | |
311 k = 7 - av_log2(history) + ((history + 16) >> 6); | |
312 | |
313 while((*samples == 0) && (i < s->avctx->frame_size)) { | |
314 samples++; | |
315 i++; | |
316 block_size++; | |
317 } | |
318 encode_scalar(s, block_size, k, 16); | |
319 | |
320 sign_modifier = (block_size <= 0xFFFF); | |
321 | |
322 history = 0; | |
323 } | |
324 | |
325 } | |
326 } | |
327 | |
7595 | 328 static void write_compressed_frame(AlacEncodeContext *s) |
329 { | |
330 int i, j; | |
331 | |
332 /* only simple mid/side decorrelation supported as of now */ | |
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333 if(s->avctx->channels == 2) |
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334 alac_stereo_decorrelation(s); |
7595 | 335 put_bits(&s->pbctx, 8, s->interlacing_shift); |
336 put_bits(&s->pbctx, 8, s->interlacing_leftweight); | |
337 | |
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338 for(i=0;i<s->avctx->channels;i++) { |
7595 | 339 |
340 calc_predictor_params(s, i); | |
341 | |
342 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd | |
343 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); | |
344 | |
345 put_bits(&s->pbctx, 3, s->rc.rice_modifier); | |
346 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); | |
347 // predictor coeff. table | |
348 for(j=0;j<s->lpc[i].lpc_order;j++) { | |
349 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); | |
350 } | |
351 } | |
352 | |
353 // apply lpc and entropy coding to audio samples | |
354 | |
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355 for(i=0;i<s->avctx->channels;i++) { |
7595 | 356 alac_linear_predictor(s, i); |
357 alac_entropy_coder(s); | |
358 } | |
359 } | |
360 | |
361 static av_cold int alac_encode_init(AVCodecContext *avctx) | |
362 { | |
363 AlacEncodeContext *s = avctx->priv_data; | |
364 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1); | |
365 | |
366 avctx->frame_size = DEFAULT_FRAME_SIZE; | |
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367 avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE; |
7595 | 368 |
369 if(avctx->sample_fmt != SAMPLE_FMT_S16) { | |
370 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); | |
371 return -1; | |
372 } | |
373 | |
374 // Set default compression level | |
375 if(avctx->compression_level == FF_COMPRESSION_DEFAULT) | |
376 s->compression_level = 1; | |
377 else | |
378 s->compression_level = av_clip(avctx->compression_level, 0, 1); | |
379 | |
380 // Initialize default Rice parameters | |
381 s->rc.history_mult = 40; | |
382 s->rc.initial_history = 10; | |
383 s->rc.k_modifier = 14; | |
384 s->rc.rice_modifier = 4; | |
385 | |
386 s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE + | |
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387 avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample)>>3; |
7595 | 388 |
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389 s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes |
7595 | 390 |
391 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE); | |
392 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c')); | |
393 AV_WB32(alac_extradata+12, avctx->frame_size); | |
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394 AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample); |
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395 AV_WB8 (alac_extradata+21, avctx->channels); |
7595 | 396 AV_WB32(alac_extradata+24, s->max_coded_frame_size); |
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397 AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate |
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398 AV_WB32(alac_extradata+32, avctx->sample_rate); |
7595 | 399 |
400 // Set relevant extradata fields | |
401 if(s->compression_level > 0) { | |
402 AV_WB8(alac_extradata+18, s->rc.history_mult); | |
403 AV_WB8(alac_extradata+19, s->rc.initial_history); | |
404 AV_WB8(alac_extradata+20, s->rc.k_modifier); | |
405 } | |
406 | |
7619 | 407 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER; |
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408 if(avctx->min_prediction_order >= 0) { |
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409 if(avctx->min_prediction_order < MIN_LPC_ORDER || |
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410 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { |
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411 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); |
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412 return -1; |
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413 } |
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414 |
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415 s->min_prediction_order = avctx->min_prediction_order; |
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416 } |
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417 |
7619 | 418 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER; |
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419 if(avctx->max_prediction_order >= 0) { |
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420 if(avctx->max_prediction_order < MIN_LPC_ORDER || |
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421 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { |
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422 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order); |
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423 return -1; |
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424 } |
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425 |
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426 s->max_prediction_order = avctx->max_prediction_order; |
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427 } |
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428 |
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429 if(s->max_prediction_order < s->min_prediction_order) { |
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430 av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", |
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431 s->min_prediction_order, s->max_prediction_order); |
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432 return -1; |
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433 } |
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434 |
7595 | 435 avctx->extradata = alac_extradata; |
436 avctx->extradata_size = ALAC_EXTRADATA_SIZE; | |
437 | |
438 avctx->coded_frame = avcodec_alloc_frame(); | |
439 avctx->coded_frame->key_frame = 1; | |
440 | |
441 s->avctx = avctx; | |
442 dsputil_init(&s->dspctx, avctx); | |
443 | |
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444 return 0; |
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445 } |
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446 |
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447 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, |
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448 int buf_size, void *data) |
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449 { |
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450 AlacEncodeContext *s = avctx->priv_data; |
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451 PutBitContext *pb = &s->pbctx; |
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452 int i, out_bytes, verbatim_flag = 0; |
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453 |
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454 if(avctx->frame_size > DEFAULT_FRAME_SIZE) { |
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455 av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n"); |
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456 return -1; |
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457 } |
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458 |
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459 if(buf_size < 2*s->max_coded_frame_size) { |
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460 av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); |
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461 return -1; |
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462 } |
7595 | 463 |
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464 verbatim: |
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465 init_put_bits(pb, frame, buf_size); |
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466 |
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467 if((s->compression_level == 0) || verbatim_flag) { |
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468 // Verbatim mode |
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469 int16_t *samples = data; |
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470 write_frame_header(s, 1); |
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471 for(i=0; i<avctx->frame_size*avctx->channels; i++) { |
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472 put_sbits(pb, 16, *samples++); |
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473 } |
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474 } else { |
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475 init_sample_buffers(s, data); |
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476 write_frame_header(s, 0); |
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477 write_compressed_frame(s); |
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478 } |
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479 |
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480 put_bits(pb, 3, 7); |
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481 flush_put_bits(pb); |
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482 out_bytes = put_bits_count(pb) >> 3; |
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483 |
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484 if(out_bytes > s->max_coded_frame_size) { |
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485 /* frame too large. use verbatim mode */ |
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486 if(verbatim_flag || (s->compression_level == 0)) { |
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487 /* still too large. must be an error. */ |
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488 av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); |
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489 return -1; |
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490 } |
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491 verbatim_flag = 1; |
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492 goto verbatim; |
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493 } |
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494 |
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495 return out_bytes; |
7595 | 496 } |
497 | |
498 static av_cold int alac_encode_close(AVCodecContext *avctx) | |
499 { | |
500 av_freep(&avctx->extradata); | |
501 avctx->extradata_size = 0; | |
502 av_freep(&avctx->coded_frame); | |
503 return 0; | |
504 } | |
505 | |
506 AVCodec alac_encoder = { | |
507 "alac", | |
508 CODEC_TYPE_AUDIO, | |
509 CODEC_ID_ALAC, | |
510 sizeof(AlacEncodeContext), | |
511 alac_encode_init, | |
512 alac_encode_frame, | |
513 alac_encode_close, | |
514 .capabilities = CODEC_CAP_SMALL_LAST_FRAME, | |
7598 | 515 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), |
7595 | 516 }; |