Mercurial > libavcodec.hg
annotate alacenc.c @ 10612:d6860312274c libavcodec
Support ASCII pnms.
Implements issue1452.
author | michael |
---|---|
date | Tue, 01 Dec 2009 16:56:13 +0000 |
parents | 1c33160b0722 |
children | 8a4984c5cacc |
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" | |
9428 | 23 #include "get_bits.h" |
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24 #include "put_bits.h" |
7595 | 25 #include "dsputil.h" |
26 #include "lpc.h" | |
9110 | 27 #include "mathops.h" |
7595 | 28 |
29 #define DEFAULT_FRAME_SIZE 4096 | |
30 #define DEFAULT_SAMPLE_SIZE 16 | |
31 #define MAX_CHANNELS 8 | |
32 #define ALAC_EXTRADATA_SIZE 36 | |
33 #define ALAC_FRAME_HEADER_SIZE 55 | |
34 #define ALAC_FRAME_FOOTER_SIZE 3 | |
35 | |
36 #define ALAC_ESCAPE_CODE 0x1FF | |
37 #define ALAC_MAX_LPC_ORDER 30 | |
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38 #define DEFAULT_MAX_PRED_ORDER 6 |
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39 #define DEFAULT_MIN_PRED_ORDER 4 |
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40 #define ALAC_MAX_LPC_PRECISION 9 |
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41 #define ALAC_MAX_LPC_SHIFT 9 |
7595 | 42 |
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43 #define ALAC_CHMODE_LEFT_RIGHT 0 |
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44 #define ALAC_CHMODE_LEFT_SIDE 1 |
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45 #define ALAC_CHMODE_RIGHT_SIDE 2 |
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46 #define ALAC_CHMODE_MID_SIDE 3 |
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47 |
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48 typedef struct RiceContext { |
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49 int history_mult; |
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50 int initial_history; |
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51 int k_modifier; |
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52 int rice_modifier; |
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53 } RiceContext; |
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54 |
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55 typedef struct LPCContext { |
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56 int lpc_order; |
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57 int lpc_coeff[ALAC_MAX_LPC_ORDER+1]; |
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58 int lpc_quant; |
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59 } LPCContext; |
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60 |
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61 typedef struct AlacEncodeContext { |
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62 int compression_level; |
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63 int min_prediction_order; |
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64 int max_prediction_order; |
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65 int max_coded_frame_size; |
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66 int write_sample_size; |
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67 int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE]; |
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68 int32_t predictor_buf[DEFAULT_FRAME_SIZE]; |
7595 | 69 int interlacing_shift; |
70 int interlacing_leftweight; | |
71 PutBitContext pbctx; | |
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72 RiceContext rc; |
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73 LPCContext lpc[MAX_CHANNELS]; |
7595 | 74 DSPContext dspctx; |
75 AVCodecContext *avctx; | |
76 } AlacEncodeContext; | |
77 | |
78 | |
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79 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples) |
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80 { |
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81 int ch, i; |
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82 |
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83 for(ch=0;ch<s->avctx->channels;ch++) { |
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84 int16_t *sptr = input_samples + ch; |
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85 for(i=0;i<s->avctx->frame_size;i++) { |
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86 s->sample_buf[ch][i] = *sptr; |
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87 sptr += s->avctx->channels; |
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88 } |
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89 } |
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90 } |
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91 |
7595 | 92 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size) |
93 { | |
94 int divisor, q, r; | |
95 | |
96 k = FFMIN(k, s->rc.k_modifier); | |
97 divisor = (1<<k) - 1; | |
98 q = x / divisor; | |
99 r = x % divisor; | |
100 | |
101 if(q > 8) { | |
102 // write escape code and sample value directly | |
103 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE); | |
104 put_bits(&s->pbctx, write_sample_size, x); | |
105 } else { | |
106 if(q) | |
107 put_bits(&s->pbctx, q, (1<<q) - 1); | |
108 put_bits(&s->pbctx, 1, 0); | |
109 | |
110 if(k != 1) { | |
111 if(r > 0) | |
112 put_bits(&s->pbctx, k, r+1); | |
113 else | |
114 put_bits(&s->pbctx, k-1, 0); | |
115 } | |
116 } | |
117 } | |
118 | |
119 static void write_frame_header(AlacEncodeContext *s, int is_verbatim) | |
120 { | |
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121 put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 |
7595 | 122 put_bits(&s->pbctx, 16, 0); // Seems to be zero |
123 put_bits(&s->pbctx, 1, 1); // Sample count is in the header | |
124 put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field | |
125 put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim | |
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126 put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame |
7595 | 127 } |
128 | |
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129 static void calc_predictor_params(AlacEncodeContext *s, int ch) |
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130 { |
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131 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER]; |
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132 int shift[MAX_LPC_ORDER]; |
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133 int opt_order; |
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134 |
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135 if (s->compression_level == 1) { |
10417 | 136 s->lpc[ch].lpc_order = 6; |
137 s->lpc[ch].lpc_quant = 6; | |
138 s->lpc[ch].lpc_coeff[0] = 160; | |
139 s->lpc[ch].lpc_coeff[1] = -190; | |
140 s->lpc[ch].lpc_coeff[2] = 170; | |
141 s->lpc[ch].lpc_coeff[3] = -130; | |
142 s->lpc[ch].lpc_coeff[4] = 80; | |
143 s->lpc[ch].lpc_coeff[5] = -25; | |
144 } else { | |
10366 | 145 opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], |
146 s->avctx->frame_size, | |
147 s->min_prediction_order, | |
148 s->max_prediction_order, | |
149 ALAC_MAX_LPC_PRECISION, coefs, shift, 1, | |
150 ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1); | |
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151 |
10366 | 152 s->lpc[ch].lpc_order = opt_order; |
153 s->lpc[ch].lpc_quant = shift[opt_order-1]; | |
154 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int)); | |
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155 } |
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156 } |
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157 |
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158 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) |
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159 { |
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160 int i, best; |
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161 int32_t lt, rt; |
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162 uint64_t sum[4]; |
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163 uint64_t score[4]; |
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164 |
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165 /* calculate sum of 2nd order residual for each channel */ |
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166 sum[0] = sum[1] = sum[2] = sum[3] = 0; |
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167 for(i=2; i<n; i++) { |
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168 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; |
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169 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; |
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170 sum[2] += FFABS((lt + rt) >> 1); |
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171 sum[3] += FFABS(lt - rt); |
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172 sum[0] += FFABS(lt); |
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173 sum[1] += FFABS(rt); |
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174 } |
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175 |
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176 /* calculate score for each mode */ |
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177 score[0] = sum[0] + sum[1]; |
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178 score[1] = sum[0] + sum[3]; |
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179 score[2] = sum[1] + sum[3]; |
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180 score[3] = sum[2] + sum[3]; |
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181 |
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182 /* return mode with lowest score */ |
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183 best = 0; |
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184 for(i=1; i<4; i++) { |
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185 if(score[i] < score[best]) { |
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186 best = i; |
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187 } |
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188 } |
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189 return best; |
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190 } |
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191 |
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192 static void alac_stereo_decorrelation(AlacEncodeContext *s) |
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193 { |
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194 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; |
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195 int i, mode, n = s->avctx->frame_size; |
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196 int32_t tmp; |
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197 |
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198 mode = estimate_stereo_mode(left, right, n); |
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199 |
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200 switch(mode) |
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201 { |
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202 case ALAC_CHMODE_LEFT_RIGHT: |
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203 s->interlacing_leftweight = 0; |
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204 s->interlacing_shift = 0; |
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205 break; |
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206 |
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207 case ALAC_CHMODE_LEFT_SIDE: |
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208 for(i=0; i<n; i++) { |
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209 right[i] = left[i] - right[i]; |
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210 } |
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211 s->interlacing_leftweight = 1; |
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212 s->interlacing_shift = 0; |
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213 break; |
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214 |
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215 case ALAC_CHMODE_RIGHT_SIDE: |
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216 for(i=0; i<n; i++) { |
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217 tmp = right[i]; |
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218 right[i] = left[i] - right[i]; |
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219 left[i] = tmp + (right[i] >> 31); |
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220 } |
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221 s->interlacing_leftweight = 1; |
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222 s->interlacing_shift = 31; |
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223 break; |
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224 |
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225 default: |
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226 for(i=0; i<n; i++) { |
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227 tmp = left[i]; |
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228 left[i] = (tmp + right[i]) >> 1; |
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229 right[i] = tmp - right[i]; |
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230 } |
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231 s->interlacing_leftweight = 1; |
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232 s->interlacing_shift = 1; |
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233 break; |
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234 } |
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235 } |
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236 |
7619 | 237 static void alac_linear_predictor(AlacEncodeContext *s, int ch) |
238 { | |
239 int i; | |
240 LPCContext lpc = s->lpc[ch]; | |
241 | |
242 if(lpc.lpc_order == 31) { | |
243 s->predictor_buf[0] = s->sample_buf[ch][0]; | |
244 | |
245 for(i=1; i<s->avctx->frame_size; i++) | |
246 s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; | |
247 | |
248 return; | |
249 } | |
250 | |
251 // generalised linear predictor | |
252 | |
253 if(lpc.lpc_order > 0) { | |
254 int32_t *samples = s->sample_buf[ch]; | |
255 int32_t *residual = s->predictor_buf; | |
256 | |
257 // generate warm-up samples | |
258 residual[0] = samples[0]; | |
259 for(i=1;i<=lpc.lpc_order;i++) | |
260 residual[i] = samples[i] - samples[i-1]; | |
261 | |
262 // perform lpc on remaining samples | |
263 for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { | |
264 int sum = 1 << (lpc.lpc_quant - 1), res_val, j; | |
265 | |
266 for (j = 0; j < lpc.lpc_order; j++) { | |
267 sum += (samples[lpc.lpc_order-j] - samples[0]) * | |
268 lpc.lpc_coeff[j]; | |
269 } | |
270 | |
271 sum >>= lpc.lpc_quant; | |
272 sum += samples[0]; | |
9110 | 273 residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum, |
274 s->write_sample_size); | |
7619 | 275 res_val = residual[i]; |
276 | |
277 if(res_val) { | |
278 int index = lpc.lpc_order - 1; | |
279 int neg = (res_val < 0); | |
280 | |
281 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) { | |
282 int val = samples[0] - samples[lpc.lpc_order - index]; | |
283 int sign = (val ? FFSIGN(val) : 0); | |
284 | |
285 if(neg) | |
286 sign*=-1; | |
287 | |
288 lpc.lpc_coeff[index] -= sign; | |
289 val *= sign; | |
290 res_val -= ((val >> lpc.lpc_quant) * | |
291 (lpc.lpc_order - index)); | |
292 index--; | |
293 } | |
294 } | |
295 samples++; | |
296 } | |
297 } | |
298 } | |
299 | |
300 static void alac_entropy_coder(AlacEncodeContext *s) | |
301 { | |
302 unsigned int history = s->rc.initial_history; | |
303 int sign_modifier = 0, i, k; | |
304 int32_t *samples = s->predictor_buf; | |
305 | |
306 for(i=0;i < s->avctx->frame_size;) { | |
307 int x; | |
308 | |
309 k = av_log2((history >> 9) + 3); | |
310 | |
311 x = -2*(*samples)-1; | |
312 x ^= (x>>31); | |
313 | |
314 samples++; | |
315 i++; | |
316 | |
317 encode_scalar(s, x - sign_modifier, k, s->write_sample_size); | |
318 | |
319 history += x * s->rc.history_mult | |
320 - ((history * s->rc.history_mult) >> 9); | |
321 | |
322 sign_modifier = 0; | |
323 if(x > 0xFFFF) | |
324 history = 0xFFFF; | |
325 | |
326 if((history < 128) && (i < s->avctx->frame_size)) { | |
327 unsigned int block_size = 0; | |
328 | |
329 k = 7 - av_log2(history) + ((history + 16) >> 6); | |
330 | |
331 while((*samples == 0) && (i < s->avctx->frame_size)) { | |
332 samples++; | |
333 i++; | |
334 block_size++; | |
335 } | |
336 encode_scalar(s, block_size, k, 16); | |
337 | |
338 sign_modifier = (block_size <= 0xFFFF); | |
339 | |
340 history = 0; | |
341 } | |
342 | |
343 } | |
344 } | |
345 | |
7595 | 346 static void write_compressed_frame(AlacEncodeContext *s) |
347 { | |
348 int i, j; | |
349 | |
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350 if(s->avctx->channels == 2) |
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351 alac_stereo_decorrelation(s); |
7595 | 352 put_bits(&s->pbctx, 8, s->interlacing_shift); |
353 put_bits(&s->pbctx, 8, s->interlacing_leftweight); | |
354 | |
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355 for(i=0;i<s->avctx->channels;i++) { |
7595 | 356 |
357 calc_predictor_params(s, i); | |
358 | |
359 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd | |
360 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); | |
361 | |
362 put_bits(&s->pbctx, 3, s->rc.rice_modifier); | |
363 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); | |
364 // predictor coeff. table | |
365 for(j=0;j<s->lpc[i].lpc_order;j++) { | |
366 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); | |
367 } | |
368 } | |
369 | |
370 // apply lpc and entropy coding to audio samples | |
371 | |
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372 for(i=0;i<s->avctx->channels;i++) { |
7595 | 373 alac_linear_predictor(s, i); |
374 alac_entropy_coder(s); | |
375 } | |
376 } | |
377 | |
378 static av_cold int alac_encode_init(AVCodecContext *avctx) | |
379 { | |
380 AlacEncodeContext *s = avctx->priv_data; | |
381 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1); | |
382 | |
383 avctx->frame_size = DEFAULT_FRAME_SIZE; | |
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384 avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE; |
7595 | 385 |
386 if(avctx->sample_fmt != SAMPLE_FMT_S16) { | |
387 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); | |
388 return -1; | |
389 } | |
390 | |
391 // Set default compression level | |
392 if(avctx->compression_level == FF_COMPRESSION_DEFAULT) | |
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393 s->compression_level = 2; |
7595 | 394 else |
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395 s->compression_level = av_clip(avctx->compression_level, 0, 2); |
7595 | 396 |
397 // Initialize default Rice parameters | |
398 s->rc.history_mult = 40; | |
399 s->rc.initial_history = 10; | |
400 s->rc.k_modifier = 14; | |
401 s->rc.rice_modifier = 4; | |
402 | |
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403 s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3); |
7595 | 404 |
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405 s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes |
7595 | 406 |
407 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE); | |
408 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c')); | |
409 AV_WB32(alac_extradata+12, avctx->frame_size); | |
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410 AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample); |
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411 AV_WB8 (alac_extradata+21, avctx->channels); |
7595 | 412 AV_WB32(alac_extradata+24, s->max_coded_frame_size); |
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413 AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate |
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414 AV_WB32(alac_extradata+32, avctx->sample_rate); |
7595 | 415 |
416 // Set relevant extradata fields | |
417 if(s->compression_level > 0) { | |
418 AV_WB8(alac_extradata+18, s->rc.history_mult); | |
419 AV_WB8(alac_extradata+19, s->rc.initial_history); | |
420 AV_WB8(alac_extradata+20, s->rc.k_modifier); | |
421 } | |
422 | |
7619 | 423 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER; |
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424 if(avctx->min_prediction_order >= 0) { |
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425 if(avctx->min_prediction_order < MIN_LPC_ORDER || |
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426 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { |
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427 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); |
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428 return -1; |
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429 } |
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430 |
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431 s->min_prediction_order = avctx->min_prediction_order; |
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432 } |
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433 |
7619 | 434 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER; |
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435 if(avctx->max_prediction_order >= 0) { |
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436 if(avctx->max_prediction_order < MIN_LPC_ORDER || |
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437 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { |
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438 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order); |
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439 return -1; |
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440 } |
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441 |
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442 s->max_prediction_order = avctx->max_prediction_order; |
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443 } |
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444 |
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445 if(s->max_prediction_order < s->min_prediction_order) { |
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446 av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", |
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447 s->min_prediction_order, s->max_prediction_order); |
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448 return -1; |
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449 } |
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450 |
7595 | 451 avctx->extradata = alac_extradata; |
452 avctx->extradata_size = ALAC_EXTRADATA_SIZE; | |
453 | |
454 avctx->coded_frame = avcodec_alloc_frame(); | |
455 avctx->coded_frame->key_frame = 1; | |
456 | |
457 s->avctx = avctx; | |
458 dsputil_init(&s->dspctx, avctx); | |
459 | |
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460 return 0; |
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461 } |
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462 |
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463 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, |
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464 int buf_size, void *data) |
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465 { |
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466 AlacEncodeContext *s = avctx->priv_data; |
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467 PutBitContext *pb = &s->pbctx; |
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468 int i, out_bytes, verbatim_flag = 0; |
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469 |
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470 if(avctx->frame_size > DEFAULT_FRAME_SIZE) { |
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471 av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n"); |
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472 return -1; |
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473 } |
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474 |
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475 if(buf_size < 2*s->max_coded_frame_size) { |
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476 av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); |
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477 return -1; |
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478 } |
7595 | 479 |
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480 verbatim: |
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481 init_put_bits(pb, frame, buf_size); |
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482 |
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483 if((s->compression_level == 0) || verbatim_flag) { |
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484 // Verbatim mode |
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485 int16_t *samples = data; |
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486 write_frame_header(s, 1); |
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487 for(i=0; i<avctx->frame_size*avctx->channels; i++) { |
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488 put_sbits(pb, 16, *samples++); |
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489 } |
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490 } else { |
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491 init_sample_buffers(s, data); |
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492 write_frame_header(s, 0); |
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493 write_compressed_frame(s); |
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494 } |
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495 |
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496 put_bits(pb, 3, 7); |
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497 flush_put_bits(pb); |
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498 out_bytes = put_bits_count(pb) >> 3; |
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499 |
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500 if(out_bytes > s->max_coded_frame_size) { |
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501 /* frame too large. use verbatim mode */ |
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502 if(verbatim_flag || (s->compression_level == 0)) { |
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503 /* still too large. must be an error. */ |
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504 av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); |
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505 return -1; |
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506 } |
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507 verbatim_flag = 1; |
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508 goto verbatim; |
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509 } |
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510 |
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511 return out_bytes; |
7595 | 512 } |
513 | |
514 static av_cold int alac_encode_close(AVCodecContext *avctx) | |
515 { | |
516 av_freep(&avctx->extradata); | |
517 avctx->extradata_size = 0; | |
518 av_freep(&avctx->coded_frame); | |
519 return 0; | |
520 } | |
521 | |
522 AVCodec alac_encoder = { | |
523 "alac", | |
524 CODEC_TYPE_AUDIO, | |
525 CODEC_ID_ALAC, | |
526 sizeof(AlacEncodeContext), | |
527 alac_encode_init, | |
528 alac_encode_frame, | |
529 alac_encode_close, | |
530 .capabilities = CODEC_CAP_SMALL_LAST_FRAME, | |
7598 | 531 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), |
7595 | 532 }; |