Mercurial > libavcodec.hg
annotate alacenc.c @ 12104:2a6873ee2fc9 libavcodec
ARM: hide a .size directive on non-ELF targets
author | mru |
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date | Wed, 07 Jul 2010 20:09:37 +0000 |
parents | 01fe2ac0c8ac |
children | e59926e2c50c |
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" | |
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23 #include "put_bits.h" |
7595 | 24 #include "dsputil.h" |
25 #include "lpc.h" | |
9110 | 26 #include "mathops.h" |
7595 | 27 |
28 #define DEFAULT_FRAME_SIZE 4096 | |
29 #define DEFAULT_SAMPLE_SIZE 16 | |
30 #define MAX_CHANNELS 8 | |
31 #define ALAC_EXTRADATA_SIZE 36 | |
32 #define ALAC_FRAME_HEADER_SIZE 55 | |
33 #define ALAC_FRAME_FOOTER_SIZE 3 | |
34 | |
35 #define ALAC_ESCAPE_CODE 0x1FF | |
36 #define ALAC_MAX_LPC_ORDER 30 | |
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37 #define DEFAULT_MAX_PRED_ORDER 6 |
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38 #define DEFAULT_MIN_PRED_ORDER 4 |
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39 #define ALAC_MAX_LPC_PRECISION 9 |
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40 #define ALAC_MAX_LPC_SHIFT 9 |
7595 | 41 |
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42 #define ALAC_CHMODE_LEFT_RIGHT 0 |
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43 #define ALAC_CHMODE_LEFT_SIDE 1 |
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44 #define ALAC_CHMODE_RIGHT_SIDE 2 |
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45 #define ALAC_CHMODE_MID_SIDE 3 |
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46 |
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47 typedef struct RiceContext { |
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48 int history_mult; |
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49 int initial_history; |
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50 int k_modifier; |
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51 int rice_modifier; |
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52 } RiceContext; |
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53 |
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54 typedef struct LPCContext { |
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55 int lpc_order; |
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56 int lpc_coeff[ALAC_MAX_LPC_ORDER+1]; |
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57 int lpc_quant; |
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58 } LPCContext; |
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59 |
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60 typedef struct AlacEncodeContext { |
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61 int compression_level; |
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62 int min_prediction_order; |
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63 int max_prediction_order; |
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64 int max_coded_frame_size; |
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65 int write_sample_size; |
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66 int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE]; |
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67 int32_t predictor_buf[DEFAULT_FRAME_SIZE]; |
7595 | 68 int interlacing_shift; |
69 int interlacing_leftweight; | |
70 PutBitContext pbctx; | |
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71 RiceContext rc; |
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72 LPCContext lpc[MAX_CHANNELS]; |
7595 | 73 DSPContext dspctx; |
74 AVCodecContext *avctx; | |
75 } AlacEncodeContext; | |
76 | |
77 | |
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78 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples) |
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79 { |
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80 int ch, i; |
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81 |
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82 for(ch=0;ch<s->avctx->channels;ch++) { |
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83 int16_t *sptr = input_samples + ch; |
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84 for(i=0;i<s->avctx->frame_size;i++) { |
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85 s->sample_buf[ch][i] = *sptr; |
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86 sptr += s->avctx->channels; |
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87 } |
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88 } |
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89 } |
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90 |
7595 | 91 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size) |
92 { | |
93 int divisor, q, r; | |
94 | |
95 k = FFMIN(k, s->rc.k_modifier); | |
96 divisor = (1<<k) - 1; | |
97 q = x / divisor; | |
98 r = x % divisor; | |
99 | |
100 if(q > 8) { | |
101 // write escape code and sample value directly | |
102 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE); | |
103 put_bits(&s->pbctx, write_sample_size, x); | |
104 } else { | |
105 if(q) | |
106 put_bits(&s->pbctx, q, (1<<q) - 1); | |
107 put_bits(&s->pbctx, 1, 0); | |
108 | |
109 if(k != 1) { | |
110 if(r > 0) | |
111 put_bits(&s->pbctx, k, r+1); | |
112 else | |
113 put_bits(&s->pbctx, k-1, 0); | |
114 } | |
115 } | |
116 } | |
117 | |
118 static void write_frame_header(AlacEncodeContext *s, int is_verbatim) | |
119 { | |
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120 put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 |
7595 | 121 put_bits(&s->pbctx, 16, 0); // Seems to be zero |
122 put_bits(&s->pbctx, 1, 1); // Sample count is in the header | |
123 put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field | |
124 put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim | |
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125 put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame |
7595 | 126 } |
127 | |
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128 static void calc_predictor_params(AlacEncodeContext *s, int ch) |
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129 { |
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130 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER]; |
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131 int shift[MAX_LPC_ORDER]; |
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132 int opt_order; |
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133 |
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134 if (s->compression_level == 1) { |
10417 | 135 s->lpc[ch].lpc_order = 6; |
136 s->lpc[ch].lpc_quant = 6; | |
137 s->lpc[ch].lpc_coeff[0] = 160; | |
138 s->lpc[ch].lpc_coeff[1] = -190; | |
139 s->lpc[ch].lpc_coeff[2] = 170; | |
140 s->lpc[ch].lpc_coeff[3] = -130; | |
141 s->lpc[ch].lpc_coeff[4] = 80; | |
142 s->lpc[ch].lpc_coeff[5] = -25; | |
143 } else { | |
10366 | 144 opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], |
145 s->avctx->frame_size, | |
146 s->min_prediction_order, | |
147 s->max_prediction_order, | |
148 ALAC_MAX_LPC_PRECISION, coefs, shift, 1, | |
149 ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1); | |
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150 |
10366 | 151 s->lpc[ch].lpc_order = opt_order; |
152 s->lpc[ch].lpc_quant = shift[opt_order-1]; | |
153 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int)); | |
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154 } |
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155 } |
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156 |
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157 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) |
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158 { |
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159 int i, best; |
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160 int32_t lt, rt; |
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161 uint64_t sum[4]; |
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162 uint64_t score[4]; |
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163 |
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164 /* calculate sum of 2nd order residual for each channel */ |
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165 sum[0] = sum[1] = sum[2] = sum[3] = 0; |
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166 for(i=2; i<n; i++) { |
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167 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; |
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168 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; |
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169 sum[2] += FFABS((lt + rt) >> 1); |
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170 sum[3] += FFABS(lt - rt); |
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171 sum[0] += FFABS(lt); |
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172 sum[1] += FFABS(rt); |
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173 } |
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174 |
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175 /* calculate score for each mode */ |
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176 score[0] = sum[0] + sum[1]; |
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177 score[1] = sum[0] + sum[3]; |
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178 score[2] = sum[1] + sum[3]; |
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179 score[3] = sum[2] + sum[3]; |
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180 |
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181 /* return mode with lowest score */ |
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182 best = 0; |
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183 for(i=1; i<4; i++) { |
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184 if(score[i] < score[best]) { |
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185 best = i; |
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186 } |
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187 } |
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188 return best; |
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189 } |
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190 |
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191 static void alac_stereo_decorrelation(AlacEncodeContext *s) |
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192 { |
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193 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; |
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194 int i, mode, n = s->avctx->frame_size; |
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195 int32_t tmp; |
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196 |
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197 mode = estimate_stereo_mode(left, right, n); |
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198 |
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199 switch(mode) |
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200 { |
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201 case ALAC_CHMODE_LEFT_RIGHT: |
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202 s->interlacing_leftweight = 0; |
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203 s->interlacing_shift = 0; |
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204 break; |
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205 |
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206 case ALAC_CHMODE_LEFT_SIDE: |
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207 for(i=0; i<n; i++) { |
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208 right[i] = left[i] - right[i]; |
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209 } |
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210 s->interlacing_leftweight = 1; |
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211 s->interlacing_shift = 0; |
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212 break; |
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213 |
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214 case ALAC_CHMODE_RIGHT_SIDE: |
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215 for(i=0; i<n; i++) { |
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216 tmp = right[i]; |
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217 right[i] = left[i] - right[i]; |
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218 left[i] = tmp + (right[i] >> 31); |
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219 } |
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220 s->interlacing_leftweight = 1; |
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221 s->interlacing_shift = 31; |
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222 break; |
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223 |
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224 default: |
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225 for(i=0; i<n; i++) { |
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226 tmp = left[i]; |
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227 left[i] = (tmp + right[i]) >> 1; |
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228 right[i] = tmp - right[i]; |
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229 } |
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230 s->interlacing_leftweight = 1; |
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231 s->interlacing_shift = 1; |
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232 break; |
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233 } |
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234 } |
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235 |
7619 | 236 static void alac_linear_predictor(AlacEncodeContext *s, int ch) |
237 { | |
238 int i; | |
239 LPCContext lpc = s->lpc[ch]; | |
240 | |
241 if(lpc.lpc_order == 31) { | |
242 s->predictor_buf[0] = s->sample_buf[ch][0]; | |
243 | |
244 for(i=1; i<s->avctx->frame_size; i++) | |
245 s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; | |
246 | |
247 return; | |
248 } | |
249 | |
250 // generalised linear predictor | |
251 | |
252 if(lpc.lpc_order > 0) { | |
253 int32_t *samples = s->sample_buf[ch]; | |
254 int32_t *residual = s->predictor_buf; | |
255 | |
256 // generate warm-up samples | |
257 residual[0] = samples[0]; | |
258 for(i=1;i<=lpc.lpc_order;i++) | |
259 residual[i] = samples[i] - samples[i-1]; | |
260 | |
261 // perform lpc on remaining samples | |
262 for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { | |
263 int sum = 1 << (lpc.lpc_quant - 1), res_val, j; | |
264 | |
265 for (j = 0; j < lpc.lpc_order; j++) { | |
266 sum += (samples[lpc.lpc_order-j] - samples[0]) * | |
267 lpc.lpc_coeff[j]; | |
268 } | |
269 | |
270 sum >>= lpc.lpc_quant; | |
271 sum += samples[0]; | |
9110 | 272 residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum, |
273 s->write_sample_size); | |
7619 | 274 res_val = residual[i]; |
275 | |
276 if(res_val) { | |
277 int index = lpc.lpc_order - 1; | |
278 int neg = (res_val < 0); | |
279 | |
280 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) { | |
281 int val = samples[0] - samples[lpc.lpc_order - index]; | |
282 int sign = (val ? FFSIGN(val) : 0); | |
283 | |
284 if(neg) | |
285 sign*=-1; | |
286 | |
287 lpc.lpc_coeff[index] -= sign; | |
288 val *= sign; | |
289 res_val -= ((val >> lpc.lpc_quant) * | |
290 (lpc.lpc_order - index)); | |
291 index--; | |
292 } | |
293 } | |
294 samples++; | |
295 } | |
296 } | |
297 } | |
298 | |
299 static void alac_entropy_coder(AlacEncodeContext *s) | |
300 { | |
301 unsigned int history = s->rc.initial_history; | |
302 int sign_modifier = 0, i, k; | |
303 int32_t *samples = s->predictor_buf; | |
304 | |
305 for(i=0;i < s->avctx->frame_size;) { | |
306 int x; | |
307 | |
308 k = av_log2((history >> 9) + 3); | |
309 | |
310 x = -2*(*samples)-1; | |
311 x ^= (x>>31); | |
312 | |
313 samples++; | |
314 i++; | |
315 | |
316 encode_scalar(s, x - sign_modifier, k, s->write_sample_size); | |
317 | |
318 history += x * s->rc.history_mult | |
319 - ((history * s->rc.history_mult) >> 9); | |
320 | |
321 sign_modifier = 0; | |
322 if(x > 0xFFFF) | |
323 history = 0xFFFF; | |
324 | |
325 if((history < 128) && (i < s->avctx->frame_size)) { | |
326 unsigned int block_size = 0; | |
327 | |
328 k = 7 - av_log2(history) + ((history + 16) >> 6); | |
329 | |
330 while((*samples == 0) && (i < s->avctx->frame_size)) { | |
331 samples++; | |
332 i++; | |
333 block_size++; | |
334 } | |
335 encode_scalar(s, block_size, k, 16); | |
336 | |
337 sign_modifier = (block_size <= 0xFFFF); | |
338 | |
339 history = 0; | |
340 } | |
341 | |
342 } | |
343 } | |
344 | |
7595 | 345 static void write_compressed_frame(AlacEncodeContext *s) |
346 { | |
347 int i, j; | |
348 | |
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349 if(s->avctx->channels == 2) |
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350 alac_stereo_decorrelation(s); |
7595 | 351 put_bits(&s->pbctx, 8, s->interlacing_shift); |
352 put_bits(&s->pbctx, 8, s->interlacing_leftweight); | |
353 | |
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354 for(i=0;i<s->avctx->channels;i++) { |
7595 | 355 |
356 calc_predictor_params(s, i); | |
357 | |
358 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd | |
359 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); | |
360 | |
361 put_bits(&s->pbctx, 3, s->rc.rice_modifier); | |
362 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); | |
363 // predictor coeff. table | |
364 for(j=0;j<s->lpc[i].lpc_order;j++) { | |
365 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); | |
366 } | |
367 } | |
368 | |
369 // apply lpc and entropy coding to audio samples | |
370 | |
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371 for(i=0;i<s->avctx->channels;i++) { |
7595 | 372 alac_linear_predictor(s, i); |
373 alac_entropy_coder(s); | |
374 } | |
375 } | |
376 | |
377 static av_cold int alac_encode_init(AVCodecContext *avctx) | |
378 { | |
379 AlacEncodeContext *s = avctx->priv_data; | |
380 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1); | |
381 | |
382 avctx->frame_size = DEFAULT_FRAME_SIZE; | |
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383 avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE; |
7595 | 384 |
385 if(avctx->sample_fmt != SAMPLE_FMT_S16) { | |
386 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); | |
387 return -1; | |
388 } | |
389 | |
390 // Set default compression level | |
391 if(avctx->compression_level == FF_COMPRESSION_DEFAULT) | |
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392 s->compression_level = 2; |
7595 | 393 else |
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394 s->compression_level = av_clip(avctx->compression_level, 0, 2); |
7595 | 395 |
396 // Initialize default Rice parameters | |
397 s->rc.history_mult = 40; | |
398 s->rc.initial_history = 10; | |
399 s->rc.k_modifier = 14; | |
400 s->rc.rice_modifier = 4; | |
401 | |
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402 s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3); |
7595 | 403 |
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404 s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes |
7595 | 405 |
406 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE); | |
407 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c')); | |
408 AV_WB32(alac_extradata+12, avctx->frame_size); | |
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409 AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample); |
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410 AV_WB8 (alac_extradata+21, avctx->channels); |
7595 | 411 AV_WB32(alac_extradata+24, s->max_coded_frame_size); |
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412 AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate |
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413 AV_WB32(alac_extradata+32, avctx->sample_rate); |
7595 | 414 |
415 // Set relevant extradata fields | |
416 if(s->compression_level > 0) { | |
417 AV_WB8(alac_extradata+18, s->rc.history_mult); | |
418 AV_WB8(alac_extradata+19, s->rc.initial_history); | |
419 AV_WB8(alac_extradata+20, s->rc.k_modifier); | |
420 } | |
421 | |
7619 | 422 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER; |
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423 if(avctx->min_prediction_order >= 0) { |
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424 if(avctx->min_prediction_order < MIN_LPC_ORDER || |
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425 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { |
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426 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); |
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427 return -1; |
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428 } |
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429 |
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430 s->min_prediction_order = avctx->min_prediction_order; |
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431 } |
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432 |
7619 | 433 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER; |
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434 if(avctx->max_prediction_order >= 0) { |
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435 if(avctx->max_prediction_order < MIN_LPC_ORDER || |
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436 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { |
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437 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order); |
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438 return -1; |
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439 } |
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440 |
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441 s->max_prediction_order = avctx->max_prediction_order; |
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442 } |
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443 |
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444 if(s->max_prediction_order < s->min_prediction_order) { |
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445 av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", |
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446 s->min_prediction_order, s->max_prediction_order); |
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447 return -1; |
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448 } |
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449 |
7595 | 450 avctx->extradata = alac_extradata; |
451 avctx->extradata_size = ALAC_EXTRADATA_SIZE; | |
452 | |
453 avctx->coded_frame = avcodec_alloc_frame(); | |
454 avctx->coded_frame->key_frame = 1; | |
455 | |
456 s->avctx = avctx; | |
457 dsputil_init(&s->dspctx, avctx); | |
458 | |
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459 return 0; |
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460 } |
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461 |
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462 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, |
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463 int buf_size, void *data) |
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464 { |
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465 AlacEncodeContext *s = avctx->priv_data; |
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466 PutBitContext *pb = &s->pbctx; |
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467 int i, out_bytes, verbatim_flag = 0; |
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468 |
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469 if(avctx->frame_size > DEFAULT_FRAME_SIZE) { |
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470 av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n"); |
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471 return -1; |
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472 } |
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473 |
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474 if(buf_size < 2*s->max_coded_frame_size) { |
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475 av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); |
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476 return -1; |
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477 } |
7595 | 478 |
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479 verbatim: |
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480 init_put_bits(pb, frame, buf_size); |
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481 |
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482 if((s->compression_level == 0) || verbatim_flag) { |
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483 // Verbatim mode |
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484 int16_t *samples = data; |
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485 write_frame_header(s, 1); |
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486 for(i=0; i<avctx->frame_size*avctx->channels; i++) { |
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487 put_sbits(pb, 16, *samples++); |
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488 } |
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489 } else { |
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490 init_sample_buffers(s, data); |
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491 write_frame_header(s, 0); |
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492 write_compressed_frame(s); |
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493 } |
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494 |
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495 put_bits(pb, 3, 7); |
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496 flush_put_bits(pb); |
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497 out_bytes = put_bits_count(pb) >> 3; |
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498 |
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499 if(out_bytes > s->max_coded_frame_size) { |
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500 /* frame too large. use verbatim mode */ |
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501 if(verbatim_flag || (s->compression_level == 0)) { |
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502 /* still too large. must be an error. */ |
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503 av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); |
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504 return -1; |
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505 } |
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506 verbatim_flag = 1; |
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507 goto verbatim; |
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508 } |
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509 |
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510 return out_bytes; |
7595 | 511 } |
512 | |
513 static av_cold int alac_encode_close(AVCodecContext *avctx) | |
514 { | |
515 av_freep(&avctx->extradata); | |
516 avctx->extradata_size = 0; | |
517 av_freep(&avctx->coded_frame); | |
518 return 0; | |
519 } | |
520 | |
521 AVCodec alac_encoder = { | |
522 "alac", | |
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523 AVMEDIA_TYPE_AUDIO, |
7595 | 524 CODEC_ID_ALAC, |
525 sizeof(AlacEncodeContext), | |
526 alac_encode_init, | |
527 alac_encode_frame, | |
528 alac_encode_close, | |
529 .capabilities = CODEC_CAP_SMALL_LAST_FRAME, | |
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530 .sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE}, |
7598 | 531 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), |
7595 | 532 }; |