Mercurial > libavcodec.hg
annotate pcm.c @ 7456:211cca1e3f39 libavcodec
fix mpeg 4:2:2 vis_qp and vis_mb_type debug, fix issue #471
author | bcoudurier |
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date | Thu, 31 Jul 2008 19:51:19 +0000 |
parents | 85ab7655ad4d |
children | 2321e0384521 |
rev | line source |
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92 | 1 /* |
2 * PCM codecs | |
429 | 3 * Copyright (c) 2001 Fabrice Bellard. |
92 | 4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
429 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
92 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
92 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
429 | 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
15 * Lesser General Public License for more details. | |
92 | 16 * |
429 | 17 * You should have received a copy of the GNU Lesser General Public |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
92 | 20 */ |
2967 | 21 |
1108 | 22 /** |
23 * @file pcm.c | |
24 * PCM codecs | |
25 */ | |
2967 | 26 |
92 | 27 #include "avcodec.h" |
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28 #include "bitstream.h" // for ff_reverse |
4959 | 29 #include "bytestream.h" |
92 | 30 |
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31 #define MAX_CHANNELS 64 |
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32 |
92 | 33 /* from g711.c by SUN microsystems (unrestricted use) */ |
34 | |
2979 | 35 #define SIGN_BIT (0x80) /* Sign bit for a A-law byte. */ |
36 #define QUANT_MASK (0xf) /* Quantization field mask. */ | |
37 #define NSEGS (8) /* Number of A-law segments. */ | |
38 #define SEG_SHIFT (4) /* Left shift for segment number. */ | |
39 #define SEG_MASK (0x70) /* Segment field mask. */ | |
92 | 40 |
2979 | 41 #define BIAS (0x84) /* Bias for linear code. */ |
92 | 42 |
43 /* | |
44 * alaw2linear() - Convert an A-law value to 16-bit linear PCM | |
45 * | |
46 */ | |
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47 static av_cold int alaw2linear(unsigned char a_val) |
92 | 48 { |
2979 | 49 int t; |
50 int seg; | |
92 | 51 |
2979 | 52 a_val ^= 0x55; |
92 | 53 |
2979 | 54 t = a_val & QUANT_MASK; |
55 seg = ((unsigned)a_val & SEG_MASK) >> SEG_SHIFT; | |
56 if(seg) t= (t + t + 1 + 32) << (seg + 2); | |
57 else t= (t + t + 1 ) << 3; | |
1485 | 58 |
6750 | 59 return (a_val & SIGN_BIT) ? t : -t; |
92 | 60 } |
61 | |
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62 static av_cold int ulaw2linear(unsigned char u_val) |
92 | 63 { |
2979 | 64 int t; |
92 | 65 |
2979 | 66 /* Complement to obtain normal u-law value. */ |
67 u_val = ~u_val; | |
92 | 68 |
2979 | 69 /* |
70 * Extract and bias the quantization bits. Then | |
71 * shift up by the segment number and subtract out the bias. | |
72 */ | |
73 t = ((u_val & QUANT_MASK) << 3) + BIAS; | |
74 t <<= ((unsigned)u_val & SEG_MASK) >> SEG_SHIFT; | |
92 | 75 |
6750 | 76 return (u_val & SIGN_BIT) ? (BIAS - t) : (t - BIAS); |
92 | 77 } |
78 | |
79 /* 16384 entries per table */ | |
4660 | 80 static uint8_t linear_to_alaw[16384]; |
81 static uint8_t linear_to_ulaw[16384]; | |
92 | 82 |
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83 static av_cold void build_xlaw_table(uint8_t *linear_to_xlaw, |
92 | 84 int (*xlaw2linear)(unsigned char), |
2967 | 85 int mask) |
92 | 86 { |
87 int i, j, v, v1, v2; | |
88 | |
89 j = 0; | |
90 for(i=0;i<128;i++) { | |
91 if (i != 127) { | |
92 v1 = xlaw2linear(i ^ mask); | |
93 v2 = xlaw2linear((i + 1) ^ mask); | |
94 v = (v1 + v2 + 4) >> 3; | |
95 } else { | |
96 v = 8192; | |
97 } | |
98 for(;j<v;j++) { | |
99 linear_to_xlaw[8192 + j] = (i ^ mask); | |
100 if (j > 0) | |
101 linear_to_xlaw[8192 - j] = (i ^ (mask ^ 0x80)); | |
102 } | |
103 } | |
104 linear_to_xlaw[0] = linear_to_xlaw[1]; | |
105 } | |
106 | |
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107 static av_cold int pcm_encode_init(AVCodecContext *avctx) |
92 | 108 { |
109 avctx->frame_size = 1; | |
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110 if (avctx->codec->id==CODEC_ID_PCM_F32BE && avctx->sample_fmt!=SAMPLE_FMT_FLT) { |
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111 return -1; |
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112 } |
92 | 113 switch(avctx->codec->id) { |
114 case CODEC_ID_PCM_ALAW: | |
4660 | 115 build_xlaw_table(linear_to_alaw, alaw2linear, 0xd5); |
92 | 116 break; |
117 case CODEC_ID_PCM_MULAW: | |
4660 | 118 build_xlaw_table(linear_to_ulaw, ulaw2linear, 0xff); |
92 | 119 break; |
120 default: | |
121 break; | |
122 } | |
2967 | 123 |
2340 | 124 switch(avctx->codec->id) { |
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125 case CODEC_ID_PCM_S32LE: |
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126 case CODEC_ID_PCM_S32BE: |
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127 case CODEC_ID_PCM_U32LE: |
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128 case CODEC_ID_PCM_U32BE: |
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129 case CODEC_ID_PCM_F32BE: |
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130 avctx->block_align = 4 * avctx->channels; |
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131 break; |
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132 case CODEC_ID_PCM_S24LE: |
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133 case CODEC_ID_PCM_S24BE: |
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134 case CODEC_ID_PCM_U24LE: |
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135 case CODEC_ID_PCM_U24BE: |
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136 case CODEC_ID_PCM_S24DAUD: |
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137 avctx->block_align = 3 * avctx->channels; |
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138 break; |
2340 | 139 case CODEC_ID_PCM_S16LE: |
140 case CODEC_ID_PCM_S16BE: | |
141 case CODEC_ID_PCM_U16LE: | |
142 case CODEC_ID_PCM_U16BE: | |
143 avctx->block_align = 2 * avctx->channels; | |
144 break; | |
145 case CODEC_ID_PCM_S8: | |
146 case CODEC_ID_PCM_U8: | |
147 case CODEC_ID_PCM_MULAW: | |
148 case CODEC_ID_PCM_ALAW: | |
149 avctx->block_align = avctx->channels; | |
150 break; | |
151 default: | |
152 break; | |
153 } | |
154 | |
925 | 155 avctx->coded_frame= avcodec_alloc_frame(); |
156 avctx->coded_frame->key_frame= 1; | |
2967 | 157 |
92 | 158 return 0; |
159 } | |
160 | |
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161 static av_cold int pcm_encode_close(AVCodecContext *avctx) |
92 | 162 { |
925 | 163 av_freep(&avctx->coded_frame); |
164 | |
92 | 165 return 0; |
166 } | |
167 | |
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168 /** |
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169 * \brief convert samples from 16 bit |
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170 * \param bps byte per sample for the destination format, must be >= 2 |
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171 * \param le 0 for big-, 1 for little-endian |
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172 * \param us 0 for signed, 1 for unsigned output |
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173 * \param samples input samples |
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174 * \param dst output samples |
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175 * \param n number of samples in samples buffer. |
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176 */ |
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177 static inline void encode_from16(int bps, int le, int us, |
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178 short **samples, uint8_t **dst, int n) { |
4956 | 179 int usum = us ? 0x8000 : 0; |
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180 if (bps > 2) |
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181 memset(*dst, 0, n * bps); |
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182 if (le) *dst += bps - 2; |
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183 for(;n>0;n--) { |
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184 register int v = *(*samples)++; |
4956 | 185 v += usum; |
4973 | 186 if (le) AV_WL16(*dst, v); |
187 else AV_WB16(*dst, v); | |
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188 *dst += bps; |
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189 } |
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190 if (le) *dst -= bps - 2; |
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191 } |
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192 |
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193 static int pcm_encode_frame(AVCodecContext *avctx, |
2979 | 194 unsigned char *frame, int buf_size, void *data) |
92 | 195 { |
196 int n, sample_size, v; | |
197 short *samples; | |
198 unsigned char *dst; | |
199 | |
200 switch(avctx->codec->id) { | |
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201 case CODEC_ID_PCM_S32LE: |
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202 case CODEC_ID_PCM_S32BE: |
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203 case CODEC_ID_PCM_U32LE: |
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204 case CODEC_ID_PCM_U32BE: |
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205 case CODEC_ID_PCM_F32BE: |
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206 sample_size = 4; |
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207 break; |
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208 case CODEC_ID_PCM_S24LE: |
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209 case CODEC_ID_PCM_S24BE: |
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210 case CODEC_ID_PCM_U24LE: |
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211 case CODEC_ID_PCM_U24BE: |
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212 case CODEC_ID_PCM_S24DAUD: |
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213 sample_size = 3; |
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214 break; |
92 | 215 case CODEC_ID_PCM_S16LE: |
216 case CODEC_ID_PCM_S16BE: | |
217 case CODEC_ID_PCM_U16LE: | |
218 case CODEC_ID_PCM_U16BE: | |
219 sample_size = 2; | |
220 break; | |
221 default: | |
222 sample_size = 1; | |
223 break; | |
224 } | |
225 n = buf_size / sample_size; | |
226 samples = data; | |
227 dst = frame; | |
228 | |
229 switch(avctx->codec->id) { | |
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230 case CODEC_ID_PCM_F32BE: |
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231 { |
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232 float *fsamples = data; |
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233 for(;n>0;n--) { |
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234 float fv = *fsamples++; |
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235 bytestream_put_be32(&dst, av_flt2int(fv)); |
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236 } |
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237 samples = (void*)fsamples; |
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238 } |
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239 break; |
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240 case CODEC_ID_PCM_S32LE: |
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241 encode_from16(4, 1, 0, &samples, &dst, n); |
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242 break; |
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243 case CODEC_ID_PCM_S32BE: |
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244 encode_from16(4, 0, 0, &samples, &dst, n); |
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245 break; |
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246 case CODEC_ID_PCM_U32LE: |
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247 encode_from16(4, 1, 1, &samples, &dst, n); |
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248 break; |
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249 case CODEC_ID_PCM_U32BE: |
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250 encode_from16(4, 0, 1, &samples, &dst, n); |
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251 break; |
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252 case CODEC_ID_PCM_S24LE: |
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253 encode_from16(3, 1, 0, &samples, &dst, n); |
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254 break; |
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255 case CODEC_ID_PCM_S24BE: |
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256 encode_from16(3, 0, 0, &samples, &dst, n); |
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257 break; |
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258 case CODEC_ID_PCM_U24LE: |
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259 encode_from16(3, 1, 1, &samples, &dst, n); |
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260 break; |
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261 case CODEC_ID_PCM_U24BE: |
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262 encode_from16(3, 0, 1, &samples, &dst, n); |
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263 break; |
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264 case CODEC_ID_PCM_S24DAUD: |
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265 for(;n>0;n--) { |
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266 uint32_t tmp = ff_reverse[*samples >> 8] + |
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267 (ff_reverse[*samples & 0xff] << 8); |
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268 tmp <<= 4; // sync flags would go here |
4959 | 269 bytestream_put_be24(&dst, tmp); |
2852
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270 samples++; |
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271 } |
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272 break; |
92 | 273 case CODEC_ID_PCM_S16LE: |
274 for(;n>0;n--) { | |
275 v = *samples++; | |
4959 | 276 bytestream_put_le16(&dst, v); |
92 | 277 } |
278 break; | |
279 case CODEC_ID_PCM_S16BE: | |
280 for(;n>0;n--) { | |
281 v = *samples++; | |
4959 | 282 bytestream_put_be16(&dst, v); |
92 | 283 } |
284 break; | |
285 case CODEC_ID_PCM_U16LE: | |
286 for(;n>0;n--) { | |
287 v = *samples++; | |
288 v += 0x8000; | |
4959 | 289 bytestream_put_le16(&dst, v); |
92 | 290 } |
291 break; | |
292 case CODEC_ID_PCM_U16BE: | |
293 for(;n>0;n--) { | |
294 v = *samples++; | |
295 v += 0x8000; | |
4959 | 296 bytestream_put_be16(&dst, v); |
92 | 297 } |
298 break; | |
299 case CODEC_ID_PCM_S8: | |
300 for(;n>0;n--) { | |
301 v = *samples++; | |
4960 | 302 *dst++ = v >> 8; |
92 | 303 } |
304 break; | |
305 case CODEC_ID_PCM_U8: | |
306 for(;n>0;n--) { | |
307 v = *samples++; | |
4960 | 308 *dst++ = (v >> 8) + 128; |
92 | 309 } |
310 break; | |
5422 | 311 case CODEC_ID_PCM_ZORK: |
312 for(;n>0;n--) { | |
313 v= *samples++ >> 8; | |
314 if(v<0) v = -v; | |
315 else v+= 128; | |
316 *dst++ = v; | |
317 } | |
318 break; | |
92 | 319 case CODEC_ID_PCM_ALAW: |
320 for(;n>0;n--) { | |
321 v = *samples++; | |
4960 | 322 *dst++ = linear_to_alaw[(v + 32768) >> 2]; |
92 | 323 } |
324 break; | |
325 case CODEC_ID_PCM_MULAW: | |
326 for(;n>0;n--) { | |
327 v = *samples++; | |
4960 | 328 *dst++ = linear_to_ulaw[(v + 32768) >> 2]; |
92 | 329 } |
330 break; | |
331 default: | |
332 return -1; | |
333 } | |
381
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* Mea culpa: it seems that I broke encoding to 8-bit pcm files. This fixes it.
philipjsg
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diff
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334 //avctx->frame_size = (dst - frame) / (sample_size * avctx->channels); |
372 | 335 |
92 | 336 return dst - frame; |
337 } | |
338 | |
339 typedef struct PCMDecode { | |
340 short table[256]; | |
341 } PCMDecode; | |
342 | |
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343 static av_cold int pcm_decode_init(AVCodecContext * avctx) |
92 | 344 { |
345 PCMDecode *s = avctx->priv_data; | |
346 int i; | |
347 | |
348 switch(avctx->codec->id) { | |
349 case CODEC_ID_PCM_ALAW: | |
350 for(i=0;i<256;i++) | |
351 s->table[i] = alaw2linear(i); | |
352 break; | |
353 case CODEC_ID_PCM_MULAW: | |
354 for(i=0;i<256;i++) | |
355 s->table[i] = ulaw2linear(i); | |
356 break; | |
357 default: | |
358 break; | |
359 } | |
7409
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360 |
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361 switch(avctx->codec->id) { |
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362 case CODEC_ID_PCM_F32BE: |
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363 avctx->sample_fmt = SAMPLE_FMT_FLT; |
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364 break; |
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365 default: |
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366 avctx->sample_fmt = SAMPLE_FMT_S16; |
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367 break; |
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368 } |
92 | 369 return 0; |
370 } | |
371 | |
2852
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372 /** |
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373 * \brief convert samples to 16 bit |
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374 * \param bps byte per sample for the source format, must be >= 2 |
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375 * \param le 0 for big-, 1 for little-endian |
2853
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Document "us" parameter for PCM conversion functions.
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376 * \param us 0 for signed, 1 for unsigned input |
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377 * \param src input samples |
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378 * \param samples output samples |
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379 * \param src_len number of bytes in src |
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380 */ |
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381 static inline void decode_to16(int bps, int le, int us, |
6218 | 382 const uint8_t **src, short **samples, int src_len) |
2852
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383 { |
4956 | 384 int usum = us ? -0x8000 : 0; |
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385 register int n = src_len / bps; |
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386 if (le) *src += bps - 2; |
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387 for(;n>0;n--) { |
4958 | 388 register int v; |
389 if (le) v = AV_RL16(*src); | |
390 else v = AV_RB16(*src); | |
391 v += usum; | |
392 *(*samples)++ = v; | |
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393 *src += bps; |
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394 } |
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395 if (le) *src -= bps - 2; |
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396 } |
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|
397 |
440
000aeeac27a2
* started to cleanup name clashes for onetime compilation
kabi
parents:
429
diff
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|
398 static int pcm_decode_frame(AVCodecContext *avctx, |
2979 | 399 void *data, int *data_size, |
6218 | 400 const uint8_t *buf, int buf_size) |
92 | 401 { |
402 PCMDecode *s = avctx->priv_data; | |
5940
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add pcm_s16le_planar support for electronicarts files
aurel
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diff
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403 int c, n; |
92 | 404 short *samples; |
6218 | 405 const uint8_t *src, *src2[MAX_CHANNELS]; |
92 | 406 |
407 samples = data; | |
408 src = buf; | |
409 | |
6821
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LPCM 24 bits support, patch by Lars T¸«£uber, lars.taeuber gmx net
diego
parents:
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diff
changeset
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410 if(avctx->channels <= 0 || avctx->channels > MAX_CHANNELS){ |
1b9c458d6d60
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411 av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n"); |
1b9c458d6d60
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diego
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412 return -1; |
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diego
parents:
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|
413 } |
1b9c458d6d60
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diego
parents:
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|
414 |
1b9c458d6d60
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|
415 n = avctx->channels * av_get_bits_per_sample(avctx->codec_id)/8; |
1b9c458d6d60
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diego
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416 /* av_get_bits_per_sample returns 0 for CODEC_ID_PCM_DVD */ |
1b9c458d6d60
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417 if (CODEC_ID_PCM_DVD == avctx->codec_id) |
1b9c458d6d60
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418 /* 2 samples are interleaved per block in PCM_DVD */ |
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419 n = 2 * avctx->channels * avctx->bits_per_sample/8; |
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diego
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|
420 |
6033
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Fix crash in PCM decoder when number of channels is not set.
benoit
parents:
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diff
changeset
|
421 if(n && buf_size % n){ |
4506 | 422 av_log(avctx, AV_LOG_ERROR, "invalid PCM packet\n"); |
423 return -1; | |
424 } | |
425 | |
4351 | 426 buf_size= FFMIN(buf_size, *data_size/2); |
427 *data_size=0; | |
2506 | 428 |
5940
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|
429 n = buf_size/avctx->channels; |
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430 for(c=0;c<avctx->channels;c++) |
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431 src2[c] = &src[c*n]; |
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432 |
92 | 433 switch(avctx->codec->id) { |
7409
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434 case CODEC_ID_PCM_F32BE: |
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435 { |
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|
436 float *fsamples = data; |
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437 n = buf_size >> 2; |
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438 for(;n>0;n--) |
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439 *fsamples++ = av_int2flt(bytestream_get_be32(&src)); |
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440 samples = (void*)fsamples; |
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441 break; |
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add CODEC_ID_PCM_F32BE (32-bit floating point PCM big endian decoder)
pross
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|
442 } |
2852
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reimar
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|
443 case CODEC_ID_PCM_S32LE: |
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|
444 decode_to16(4, 1, 0, &src, &samples, buf_size); |
6f7428adc6ad
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reimar
parents:
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diff
changeset
|
445 break; |
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reimar
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|
446 case CODEC_ID_PCM_S32BE: |
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447 decode_to16(4, 0, 0, &src, &samples, buf_size); |
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reimar
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diff
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|
448 break; |
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|
449 case CODEC_ID_PCM_U32LE: |
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reimar
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changeset
|
450 decode_to16(4, 1, 1, &src, &samples, buf_size); |
6f7428adc6ad
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reimar
parents:
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diff
changeset
|
451 break; |
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reimar
parents:
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diff
changeset
|
452 case CODEC_ID_PCM_U32BE: |
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diff
changeset
|
453 decode_to16(4, 0, 1, &src, &samples, buf_size); |
6f7428adc6ad
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reimar
parents:
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diff
changeset
|
454 break; |
6f7428adc6ad
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reimar
parents:
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diff
changeset
|
455 case CODEC_ID_PCM_S24LE: |
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reimar
parents:
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diff
changeset
|
456 decode_to16(3, 1, 0, &src, &samples, buf_size); |
6f7428adc6ad
Support de-/encoding of 24 and 32 bit PCM (from and to internal 16 bit).
reimar
parents:
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diff
changeset
|
457 break; |
6f7428adc6ad
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reimar
parents:
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diff
changeset
|
458 case CODEC_ID_PCM_S24BE: |
6f7428adc6ad
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parents:
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diff
changeset
|
459 decode_to16(3, 0, 0, &src, &samples, buf_size); |
6f7428adc6ad
Support de-/encoding of 24 and 32 bit PCM (from and to internal 16 bit).
reimar
parents:
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diff
changeset
|
460 break; |
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461 case CODEC_ID_PCM_U24LE: |
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462 decode_to16(3, 1, 1, &src, &samples, buf_size); |
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463 break; |
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464 case CODEC_ID_PCM_U24BE: |
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465 decode_to16(3, 0, 1, &src, &samples, buf_size); |
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466 break; |
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467 case CODEC_ID_PCM_S24DAUD: |
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468 n = buf_size / 3; |
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469 for(;n>0;n--) { |
4959 | 470 uint32_t v = bytestream_get_be24(&src); |
2852
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471 v >>= 4; // sync flags are here |
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472 *samples++ = ff_reverse[(v >> 8) & 0xff] + |
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473 (ff_reverse[v & 0xff] << 8); |
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474 } |
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475 break; |
92 | 476 case CODEC_ID_PCM_S16LE: |
477 n = buf_size >> 1; | |
478 for(;n>0;n--) { | |
4959 | 479 *samples++ = bytestream_get_le16(&src); |
92 | 480 } |
481 break; | |
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482 case CODEC_ID_PCM_S16LE_PLANAR: |
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483 for(n>>=1;n>0;n--) |
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484 for(c=0;c<avctx->channels;c++) |
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485 *samples++ = bytestream_get_le16(&src2[c]); |
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486 src = src2[avctx->channels-1]; |
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487 break; |
92 | 488 case CODEC_ID_PCM_S16BE: |
489 n = buf_size >> 1; | |
490 for(;n>0;n--) { | |
4959 | 491 *samples++ = bytestream_get_be16(&src); |
92 | 492 } |
493 break; | |
494 case CODEC_ID_PCM_U16LE: | |
495 n = buf_size >> 1; | |
496 for(;n>0;n--) { | |
4959 | 497 *samples++ = bytestream_get_le16(&src) - 0x8000; |
92 | 498 } |
499 break; | |
500 case CODEC_ID_PCM_U16BE: | |
501 n = buf_size >> 1; | |
502 for(;n>0;n--) { | |
4959 | 503 *samples++ = bytestream_get_be16(&src) - 0x8000; |
92 | 504 } |
505 break; | |
506 case CODEC_ID_PCM_S8: | |
507 n = buf_size; | |
508 for(;n>0;n--) { | |
4960 | 509 *samples++ = *src++ << 8; |
92 | 510 } |
511 break; | |
512 case CODEC_ID_PCM_U8: | |
513 n = buf_size; | |
514 for(;n>0;n--) { | |
4960 | 515 *samples++ = ((int)*src++ - 128) << 8; |
92 | 516 } |
517 break; | |
5422 | 518 case CODEC_ID_PCM_ZORK: |
519 n = buf_size; | |
520 for(;n>0;n--) { | |
521 int x= *src++; | |
522 if(x&128) x-= 128; | |
523 else x = -x; | |
524 *samples++ = x << 8; | |
525 } | |
526 break; | |
92 | 527 case CODEC_ID_PCM_ALAW: |
528 case CODEC_ID_PCM_MULAW: | |
529 n = buf_size; | |
530 for(;n>0;n--) { | |
4960 | 531 *samples++ = s->table[*src++]; |
92 | 532 } |
533 break; | |
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534 case CODEC_ID_PCM_DVD: |
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535 if(avctx->bits_per_sample != 20 && avctx->bits_per_sample != 24) { |
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536 av_log(avctx, AV_LOG_ERROR, "PCM DVD unsupported sample depth\n"); |
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537 return -1; |
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538 } else { |
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539 int jump = avctx->channels * (avctx->bits_per_sample-16) / 4; |
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540 n = buf_size / (avctx->channels * 2 * avctx->bits_per_sample / 8); |
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541 while (n--) { |
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542 for (c=0; c < 2*avctx->channels; c++) |
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543 *samples++ = bytestream_get_be16(&src); |
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544 src += jump; |
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|
545 } |
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|
546 } |
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|
547 break; |
92 | 548 default: |
549 return -1; | |
550 } | |
1064 | 551 *data_size = (uint8_t *)samples - (uint8_t *)data; |
92 | 552 return src - buf; |
553 } | |
554 | |
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|
555 #ifdef CONFIG_ENCODERS |
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556 #define PCM_ENCODER(id,sample_fmt_,name,long_name_) \ |
92 | 557 AVCodec name ## _encoder = { \ |
558 #name, \ | |
559 CODEC_TYPE_AUDIO, \ | |
560 id, \ | |
561 0, \ | |
2979 | 562 pcm_encode_init, \ |
563 pcm_encode_frame, \ | |
564 pcm_encode_close, \ | |
92 | 565 NULL, \ |
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566 .sample_fmts = (enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \ |
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567 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ |
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568 }; |
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|
569 #else |
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570 #define PCM_ENCODER(id,sample_fmt_,name,long_name_) |
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571 #endif |
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|
572 |
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573 #ifdef CONFIG_DECODERS |
6815
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diff
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574 #define PCM_DECODER(id,name,long_name_) \ |
92 | 575 AVCodec name ## _decoder = { \ |
576 #name, \ | |
577 CODEC_TYPE_AUDIO, \ | |
578 id, \ | |
579 sizeof(PCMDecode), \ | |
2979 | 580 pcm_decode_init, \ |
92 | 581 NULL, \ |
582 NULL, \ | |
440
000aeeac27a2
* started to cleanup name clashes for onetime compilation
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diff
changeset
|
583 pcm_decode_frame, \ |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
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|
584 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ |
5880
6814207ffb27
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|
585 }; |
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diff
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|
586 #else |
6815
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Add PCM codec long names; by Stefano Sabatini, stefano.sabatini-lala poste it.
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|
587 #define PCM_DECODER(id,name,long_name_) |
5880
6814207ffb27
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diff
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|
588 #endif |
6814207ffb27
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aurel
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diff
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|
589 |
7451
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Modify all codecs to report their supported input and output sample format(s).
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590 #define PCM_CODEC(id, sample_fmt_, name, long_name_) \ |
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591 PCM_ENCODER(id,sample_fmt_,name,long_name_) PCM_DECODER(id,name,long_name_) |
92 | 592 |
7197
54f8d960f15b
Add a note to remind people that new PCM/ADPCM formats need to be added to
diego
parents:
7040
diff
changeset
|
593 /* Note: Do not forget to add new entries to the Makefile as well. */ |
7451
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diff
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|
594 PCM_CODEC (CODEC_ID_PCM_ALAW, SAMPLE_FMT_S16, pcm_alaw, "A-law PCM"); |
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595 PCM_CODEC (CODEC_ID_PCM_DVD, SAMPLE_FMT_S16, pcm_dvd, "signed 16|20|24-bit big-endian PCM"); |
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596 PCM_CODEC (CODEC_ID_PCM_F32BE, SAMPLE_FMT_FLT, pcm_f32be, "32-bit floating point big-endian PCM"); |
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597 PCM_CODEC (CODEC_ID_PCM_MULAW, SAMPLE_FMT_S16, pcm_mulaw, "mu-law PCM"); |
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598 PCM_CODEC (CODEC_ID_PCM_S8, SAMPLE_FMT_S16, pcm_s8, "signed 8-bit PCM"); |
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|
599 PCM_CODEC (CODEC_ID_PCM_S16BE, SAMPLE_FMT_S16, pcm_s16be, "signed 16-bit big-endian PCM"); |
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600 PCM_CODEC (CODEC_ID_PCM_S16LE, SAMPLE_FMT_S16, pcm_s16le, "signed 16-bit little-endian PCM"); |
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601 PCM_DECODER(CODEC_ID_PCM_S16LE_PLANAR, pcm_s16le_planar, "16-bit little-endian planar PCM"); |
7451
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602 PCM_CODEC (CODEC_ID_PCM_S24BE, SAMPLE_FMT_S16, pcm_s24be, "signed 24-bit big-endian PCM"); |
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|
603 PCM_CODEC (CODEC_ID_PCM_S24DAUD, SAMPLE_FMT_S16, pcm_s24daud, "D-Cinema audio signed 24-bit PCM"); |
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Modify all codecs to report their supported input and output sample format(s).
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|
604 PCM_CODEC (CODEC_ID_PCM_S24LE, SAMPLE_FMT_S16, pcm_s24le, "signed 24-bit little-endian PCM"); |
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Modify all codecs to report their supported input and output sample format(s).
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changeset
|
605 PCM_CODEC (CODEC_ID_PCM_S32BE, SAMPLE_FMT_S16, pcm_s32be, "signed 32-bit big-endian PCM"); |
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Modify all codecs to report their supported input and output sample format(s).
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diff
changeset
|
606 PCM_CODEC (CODEC_ID_PCM_S32LE, SAMPLE_FMT_S16, pcm_s32le, "signed 32-bit little-endian PCM"); |
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|
607 PCM_CODEC (CODEC_ID_PCM_U8, SAMPLE_FMT_S16, pcm_u8, "unsigned 8-bit PCM"); |
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|
608 PCM_CODEC (CODEC_ID_PCM_U16BE, SAMPLE_FMT_S16, pcm_u16be, "unsigned 16-bit big-endian PCM"); |
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|
609 PCM_CODEC (CODEC_ID_PCM_U16LE, SAMPLE_FMT_S16, pcm_u16le, "unsigned 16-bit little-endian PCM"); |
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|
610 PCM_CODEC (CODEC_ID_PCM_U24BE, SAMPLE_FMT_S16, pcm_u24be, "unsigned 24-bit big-endian PCM"); |
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changeset
|
611 PCM_CODEC (CODEC_ID_PCM_U24LE, SAMPLE_FMT_S16, pcm_u24le, "unsigned 24-bit little-endian PCM"); |
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|
612 PCM_CODEC (CODEC_ID_PCM_U32BE, SAMPLE_FMT_S16, pcm_u32be, "unsigned 32-bit big-endian PCM"); |
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
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diff
changeset
|
613 PCM_CODEC (CODEC_ID_PCM_U32LE, SAMPLE_FMT_S16, pcm_u32le, "unsigned 32-bit little-endian PCM"); |
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changeset
|
614 PCM_CODEC (CODEC_ID_PCM_ZORK, SAMPLE_FMT_S16, pcm_zork, "Zork PCM"); |