Mercurial > libavcodec.hg
annotate alacenc.c @ 9319:44de9c136513 libavcodec
Cosmetics: remove some pointless {}
author | reimar |
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date | Tue, 31 Mar 2009 19:39:47 +0000 |
parents | ac31a0265eb9 |
children | 4cb7c65fc775 |
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" | |
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 | |
125 put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame | |
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 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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135 ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1); |
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136 |
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137 s->lpc[ch].lpc_order = opt_order; |
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138 s->lpc[ch].lpc_quant = shift[opt_order-1]; |
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139 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int)); |
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140 } |
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141 |
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142 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) |
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143 { |
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144 int i, best; |
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145 int32_t lt, rt; |
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146 uint64_t sum[4]; |
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147 uint64_t score[4]; |
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148 |
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149 /* calculate sum of 2nd order residual for each channel */ |
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150 sum[0] = sum[1] = sum[2] = sum[3] = 0; |
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151 for(i=2; i<n; i++) { |
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152 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; |
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153 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; |
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154 sum[2] += FFABS((lt + rt) >> 1); |
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155 sum[3] += FFABS(lt - rt); |
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156 sum[0] += FFABS(lt); |
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157 sum[1] += FFABS(rt); |
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158 } |
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159 |
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160 /* calculate score for each mode */ |
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161 score[0] = sum[0] + sum[1]; |
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162 score[1] = sum[0] + sum[3]; |
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163 score[2] = sum[1] + sum[3]; |
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164 score[3] = sum[2] + sum[3]; |
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165 |
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166 /* return mode with lowest score */ |
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167 best = 0; |
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168 for(i=1; i<4; i++) { |
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169 if(score[i] < score[best]) { |
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170 best = i; |
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171 } |
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172 } |
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173 return best; |
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174 } |
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175 |
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176 static void alac_stereo_decorrelation(AlacEncodeContext *s) |
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177 { |
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178 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; |
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179 int i, mode, n = s->avctx->frame_size; |
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180 int32_t tmp; |
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181 |
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182 mode = estimate_stereo_mode(left, right, n); |
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183 |
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184 switch(mode) |
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185 { |
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186 case ALAC_CHMODE_LEFT_RIGHT: |
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187 s->interlacing_leftweight = 0; |
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188 s->interlacing_shift = 0; |
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189 break; |
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190 |
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191 case ALAC_CHMODE_LEFT_SIDE: |
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192 for(i=0; i<n; i++) { |
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193 right[i] = left[i] - right[i]; |
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194 } |
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195 s->interlacing_leftweight = 1; |
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196 s->interlacing_shift = 0; |
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197 break; |
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198 |
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199 case ALAC_CHMODE_RIGHT_SIDE: |
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200 for(i=0; i<n; i++) { |
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201 tmp = right[i]; |
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202 right[i] = left[i] - right[i]; |
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203 left[i] = tmp + (right[i] >> 31); |
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204 } |
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205 s->interlacing_leftweight = 1; |
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206 s->interlacing_shift = 31; |
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207 break; |
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208 |
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209 default: |
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210 for(i=0; i<n; i++) { |
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211 tmp = left[i]; |
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212 left[i] = (tmp + right[i]) >> 1; |
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213 right[i] = tmp - right[i]; |
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214 } |
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215 s->interlacing_leftweight = 1; |
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216 s->interlacing_shift = 1; |
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217 break; |
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218 } |
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219 } |
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220 |
7619 | 221 static void alac_linear_predictor(AlacEncodeContext *s, int ch) |
222 { | |
223 int i; | |
224 LPCContext lpc = s->lpc[ch]; | |
225 | |
226 if(lpc.lpc_order == 31) { | |
227 s->predictor_buf[0] = s->sample_buf[ch][0]; | |
228 | |
229 for(i=1; i<s->avctx->frame_size; i++) | |
230 s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; | |
231 | |
232 return; | |
233 } | |
234 | |
235 // generalised linear predictor | |
236 | |
237 if(lpc.lpc_order > 0) { | |
238 int32_t *samples = s->sample_buf[ch]; | |
239 int32_t *residual = s->predictor_buf; | |
240 | |
241 // generate warm-up samples | |
242 residual[0] = samples[0]; | |
243 for(i=1;i<=lpc.lpc_order;i++) | |
244 residual[i] = samples[i] - samples[i-1]; | |
245 | |
246 // perform lpc on remaining samples | |
247 for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { | |
248 int sum = 1 << (lpc.lpc_quant - 1), res_val, j; | |
249 | |
250 for (j = 0; j < lpc.lpc_order; j++) { | |
251 sum += (samples[lpc.lpc_order-j] - samples[0]) * | |
252 lpc.lpc_coeff[j]; | |
253 } | |
254 | |
255 sum >>= lpc.lpc_quant; | |
256 sum += samples[0]; | |
9110 | 257 residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum, |
258 s->write_sample_size); | |
7619 | 259 res_val = residual[i]; |
260 | |
261 if(res_val) { | |
262 int index = lpc.lpc_order - 1; | |
263 int neg = (res_val < 0); | |
264 | |
265 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) { | |
266 int val = samples[0] - samples[lpc.lpc_order - index]; | |
267 int sign = (val ? FFSIGN(val) : 0); | |
268 | |
269 if(neg) | |
270 sign*=-1; | |
271 | |
272 lpc.lpc_coeff[index] -= sign; | |
273 val *= sign; | |
274 res_val -= ((val >> lpc.lpc_quant) * | |
275 (lpc.lpc_order - index)); | |
276 index--; | |
277 } | |
278 } | |
279 samples++; | |
280 } | |
281 } | |
282 } | |
283 | |
284 static void alac_entropy_coder(AlacEncodeContext *s) | |
285 { | |
286 unsigned int history = s->rc.initial_history; | |
287 int sign_modifier = 0, i, k; | |
288 int32_t *samples = s->predictor_buf; | |
289 | |
290 for(i=0;i < s->avctx->frame_size;) { | |
291 int x; | |
292 | |
293 k = av_log2((history >> 9) + 3); | |
294 | |
295 x = -2*(*samples)-1; | |
296 x ^= (x>>31); | |
297 | |
298 samples++; | |
299 i++; | |
300 | |
301 encode_scalar(s, x - sign_modifier, k, s->write_sample_size); | |
302 | |
303 history += x * s->rc.history_mult | |
304 - ((history * s->rc.history_mult) >> 9); | |
305 | |
306 sign_modifier = 0; | |
307 if(x > 0xFFFF) | |
308 history = 0xFFFF; | |
309 | |
310 if((history < 128) && (i < s->avctx->frame_size)) { | |
311 unsigned int block_size = 0; | |
312 | |
313 k = 7 - av_log2(history) + ((history + 16) >> 6); | |
314 | |
315 while((*samples == 0) && (i < s->avctx->frame_size)) { | |
316 samples++; | |
317 i++; | |
318 block_size++; | |
319 } | |
320 encode_scalar(s, block_size, k, 16); | |
321 | |
322 sign_modifier = (block_size <= 0xFFFF); | |
323 | |
324 history = 0; | |
325 } | |
326 | |
327 } | |
328 } | |
329 | |
7595 | 330 static void write_compressed_frame(AlacEncodeContext *s) |
331 { | |
332 int i, j; | |
333 | |
334 /* only simple mid/side decorrelation supported as of now */ | |
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335 if(s->avctx->channels == 2) |
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336 alac_stereo_decorrelation(s); |
7595 | 337 put_bits(&s->pbctx, 8, s->interlacing_shift); |
338 put_bits(&s->pbctx, 8, s->interlacing_leftweight); | |
339 | |
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340 for(i=0;i<s->avctx->channels;i++) { |
7595 | 341 |
342 calc_predictor_params(s, i); | |
343 | |
344 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd | |
345 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); | |
346 | |
347 put_bits(&s->pbctx, 3, s->rc.rice_modifier); | |
348 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); | |
349 // predictor coeff. table | |
350 for(j=0;j<s->lpc[i].lpc_order;j++) { | |
351 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); | |
352 } | |
353 } | |
354 | |
355 // apply lpc and entropy coding to audio samples | |
356 | |
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357 for(i=0;i<s->avctx->channels;i++) { |
7595 | 358 alac_linear_predictor(s, i); |
359 alac_entropy_coder(s); | |
360 } | |
361 } | |
362 | |
363 static av_cold int alac_encode_init(AVCodecContext *avctx) | |
364 { | |
365 AlacEncodeContext *s = avctx->priv_data; | |
366 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1); | |
367 | |
368 avctx->frame_size = DEFAULT_FRAME_SIZE; | |
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369 avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE; |
7595 | 370 |
371 if(avctx->sample_fmt != SAMPLE_FMT_S16) { | |
372 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); | |
373 return -1; | |
374 } | |
375 | |
376 // Set default compression level | |
377 if(avctx->compression_level == FF_COMPRESSION_DEFAULT) | |
378 s->compression_level = 1; | |
379 else | |
380 s->compression_level = av_clip(avctx->compression_level, 0, 1); | |
381 | |
382 // Initialize default Rice parameters | |
383 s->rc.history_mult = 40; | |
384 s->rc.initial_history = 10; | |
385 s->rc.k_modifier = 14; | |
386 s->rc.rice_modifier = 4; | |
387 | |
388 s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE + | |
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389 avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample)>>3; |
7595 | 390 |
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391 s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes |
7595 | 392 |
393 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE); | |
394 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c')); | |
395 AV_WB32(alac_extradata+12, avctx->frame_size); | |
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396 AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample); |
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397 AV_WB8 (alac_extradata+21, avctx->channels); |
7595 | 398 AV_WB32(alac_extradata+24, s->max_coded_frame_size); |
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399 AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate |
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400 AV_WB32(alac_extradata+32, avctx->sample_rate); |
7595 | 401 |
402 // Set relevant extradata fields | |
403 if(s->compression_level > 0) { | |
404 AV_WB8(alac_extradata+18, s->rc.history_mult); | |
405 AV_WB8(alac_extradata+19, s->rc.initial_history); | |
406 AV_WB8(alac_extradata+20, s->rc.k_modifier); | |
407 } | |
408 | |
7619 | 409 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER; |
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410 if(avctx->min_prediction_order >= 0) { |
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411 if(avctx->min_prediction_order < MIN_LPC_ORDER || |
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412 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { |
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413 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); |
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414 return -1; |
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415 } |
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416 |
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417 s->min_prediction_order = avctx->min_prediction_order; |
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418 } |
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419 |
7619 | 420 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER; |
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421 if(avctx->max_prediction_order >= 0) { |
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422 if(avctx->max_prediction_order < MIN_LPC_ORDER || |
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423 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { |
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424 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order); |
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425 return -1; |
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426 } |
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427 |
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428 s->max_prediction_order = avctx->max_prediction_order; |
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429 } |
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430 |
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431 if(s->max_prediction_order < s->min_prediction_order) { |
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432 av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", |
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433 s->min_prediction_order, s->max_prediction_order); |
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434 return -1; |
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435 } |
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436 |
7595 | 437 avctx->extradata = alac_extradata; |
438 avctx->extradata_size = ALAC_EXTRADATA_SIZE; | |
439 | |
440 avctx->coded_frame = avcodec_alloc_frame(); | |
441 avctx->coded_frame->key_frame = 1; | |
442 | |
443 s->avctx = avctx; | |
444 dsputil_init(&s->dspctx, avctx); | |
445 | |
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446 return 0; |
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447 } |
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448 |
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449 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, |
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450 int buf_size, void *data) |
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451 { |
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452 AlacEncodeContext *s = avctx->priv_data; |
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453 PutBitContext *pb = &s->pbctx; |
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454 int i, out_bytes, verbatim_flag = 0; |
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455 |
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456 if(avctx->frame_size > DEFAULT_FRAME_SIZE) { |
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457 av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n"); |
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458 return -1; |
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459 } |
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460 |
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461 if(buf_size < 2*s->max_coded_frame_size) { |
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462 av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); |
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463 return -1; |
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464 } |
7595 | 465 |
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466 verbatim: |
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467 init_put_bits(pb, frame, buf_size); |
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468 |
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469 if((s->compression_level == 0) || verbatim_flag) { |
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470 // Verbatim mode |
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471 int16_t *samples = data; |
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472 write_frame_header(s, 1); |
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473 for(i=0; i<avctx->frame_size*avctx->channels; i++) { |
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474 put_sbits(pb, 16, *samples++); |
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475 } |
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476 } else { |
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477 init_sample_buffers(s, data); |
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478 write_frame_header(s, 0); |
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479 write_compressed_frame(s); |
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480 } |
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481 |
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482 put_bits(pb, 3, 7); |
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483 flush_put_bits(pb); |
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484 out_bytes = put_bits_count(pb) >> 3; |
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485 |
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486 if(out_bytes > s->max_coded_frame_size) { |
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487 /* frame too large. use verbatim mode */ |
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488 if(verbatim_flag || (s->compression_level == 0)) { |
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489 /* still too large. must be an error. */ |
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490 av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); |
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491 return -1; |
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492 } |
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493 verbatim_flag = 1; |
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494 goto verbatim; |
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495 } |
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496 |
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497 return out_bytes; |
7595 | 498 } |
499 | |
500 static av_cold int alac_encode_close(AVCodecContext *avctx) | |
501 { | |
502 av_freep(&avctx->extradata); | |
503 avctx->extradata_size = 0; | |
504 av_freep(&avctx->coded_frame); | |
505 return 0; | |
506 } | |
507 | |
508 AVCodec alac_encoder = { | |
509 "alac", | |
510 CODEC_TYPE_AUDIO, | |
511 CODEC_ID_ALAC, | |
512 sizeof(AlacEncodeContext), | |
513 alac_encode_init, | |
514 alac_encode_frame, | |
515 alac_encode_close, | |
516 .capabilities = CODEC_CAP_SMALL_LAST_FRAME, | |
7598 | 517 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), |
7595 | 518 }; |