Mercurial > libavcodec.hg
annotate pcm.c @ 7518:a40c76e98909 libavcodec
Simplify PCM codec; change 'n' in pcm_decode_frame() to equal "total number of samples".
| author | pross |
|---|---|
| date | Thu, 07 Aug 2008 09:23:56 +0000 |
| parents | d4e19b465fcb |
| children | 4b6f7cdce8c5 |
| rev | line source |
|---|---|
| 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 |
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124 avctx->block_align = avctx->channels * av_get_bits_per_sample(avctx->codec->id)/8; |
| 925 | 125 avctx->coded_frame= avcodec_alloc_frame(); |
| 126 avctx->coded_frame->key_frame= 1; | |
| 2967 | 127 |
| 92 | 128 return 0; |
| 129 } | |
| 130 | |
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131 static av_cold int pcm_encode_close(AVCodecContext *avctx) |
| 92 | 132 { |
| 925 | 133 av_freep(&avctx->coded_frame); |
| 134 | |
| 92 | 135 return 0; |
| 136 } | |
| 137 | |
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138 /** |
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139 * \brief convert samples from 16 bit |
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140 * \param bps byte per sample for the destination format, must be >= 2 |
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141 * \param le 0 for big-, 1 for little-endian |
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142 * \param us 0 for signed, 1 for unsigned output |
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143 * \param samples input samples |
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144 * \param dst output samples |
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145 * \param n number of samples in samples buffer. |
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146 */ |
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147 static inline void encode_from16(int bps, int le, int us, |
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148 short **samples, uint8_t **dst, int n) { |
| 4956 | 149 int usum = us ? 0x8000 : 0; |
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150 if (bps > 2) |
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151 memset(*dst, 0, n * bps); |
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152 if (le) *dst += bps - 2; |
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153 for(;n>0;n--) { |
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154 register int v = *(*samples)++; |
| 4956 | 155 v += usum; |
| 4973 | 156 if (le) AV_WL16(*dst, v); |
| 157 else AV_WB16(*dst, v); | |
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158 *dst += bps; |
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159 } |
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160 if (le) *dst -= bps - 2; |
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161 } |
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162 |
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163 static int pcm_encode_frame(AVCodecContext *avctx, |
| 2979 | 164 unsigned char *frame, int buf_size, void *data) |
| 92 | 165 { |
| 166 int n, sample_size, v; | |
| 167 short *samples; | |
| 168 unsigned char *dst; | |
| 169 | |
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170 sample_size = av_get_bits_per_sample(avctx->codec->id)/8; |
| 92 | 171 n = buf_size / sample_size; |
| 172 samples = data; | |
| 173 dst = frame; | |
| 174 | |
| 175 switch(avctx->codec->id) { | |
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176 case CODEC_ID_PCM_F32BE: |
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177 { |
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178 float *fsamples = data; |
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179 for(;n>0;n--) { |
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180 float fv = *fsamples++; |
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181 bytestream_put_be32(&dst, av_flt2int(fv)); |
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182 } |
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183 samples = (void*)fsamples; |
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184 } |
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185 break; |
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186 case CODEC_ID_PCM_S32LE: |
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187 encode_from16(4, 1, 0, &samples, &dst, n); |
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188 break; |
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189 case CODEC_ID_PCM_S32BE: |
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190 encode_from16(4, 0, 0, &samples, &dst, n); |
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191 break; |
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192 case CODEC_ID_PCM_U32LE: |
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193 encode_from16(4, 1, 1, &samples, &dst, n); |
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194 break; |
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195 case CODEC_ID_PCM_U32BE: |
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196 encode_from16(4, 0, 1, &samples, &dst, n); |
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197 break; |
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198 case CODEC_ID_PCM_S24LE: |
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199 encode_from16(3, 1, 0, &samples, &dst, n); |
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200 break; |
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201 case CODEC_ID_PCM_S24BE: |
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202 encode_from16(3, 0, 0, &samples, &dst, n); |
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203 break; |
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204 case CODEC_ID_PCM_U24LE: |
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205 encode_from16(3, 1, 1, &samples, &dst, n); |
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206 break; |
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207 case CODEC_ID_PCM_U24BE: |
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208 encode_from16(3, 0, 1, &samples, &dst, n); |
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209 break; |
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210 case CODEC_ID_PCM_S24DAUD: |
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211 for(;n>0;n--) { |
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212 uint32_t tmp = ff_reverse[(*samples >> 8) & 0xff] + |
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213 (ff_reverse[*samples & 0xff] << 8); |
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214 tmp <<= 4; // sync flags would go here |
| 4959 | 215 bytestream_put_be24(&dst, tmp); |
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216 samples++; |
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217 } |
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218 break; |
| 92 | 219 case CODEC_ID_PCM_S16LE: |
| 220 for(;n>0;n--) { | |
| 221 v = *samples++; | |
| 4959 | 222 bytestream_put_le16(&dst, v); |
| 92 | 223 } |
| 224 break; | |
| 225 case CODEC_ID_PCM_S16BE: | |
| 226 for(;n>0;n--) { | |
| 227 v = *samples++; | |
| 4959 | 228 bytestream_put_be16(&dst, v); |
| 92 | 229 } |
| 230 break; | |
| 231 case CODEC_ID_PCM_U16LE: | |
| 232 for(;n>0;n--) { | |
| 233 v = *samples++; | |
| 234 v += 0x8000; | |
| 4959 | 235 bytestream_put_le16(&dst, v); |
| 92 | 236 } |
| 237 break; | |
| 238 case CODEC_ID_PCM_U16BE: | |
| 239 for(;n>0;n--) { | |
| 240 v = *samples++; | |
| 241 v += 0x8000; | |
| 4959 | 242 bytestream_put_be16(&dst, v); |
| 92 | 243 } |
| 244 break; | |
| 245 case CODEC_ID_PCM_S8: | |
| 246 for(;n>0;n--) { | |
| 247 v = *samples++; | |
| 4960 | 248 *dst++ = v >> 8; |
| 92 | 249 } |
| 250 break; | |
| 251 case CODEC_ID_PCM_U8: | |
| 252 for(;n>0;n--) { | |
| 253 v = *samples++; | |
| 4960 | 254 *dst++ = (v >> 8) + 128; |
| 92 | 255 } |
| 256 break; | |
| 5422 | 257 case CODEC_ID_PCM_ZORK: |
| 258 for(;n>0;n--) { | |
| 259 v= *samples++ >> 8; | |
| 260 if(v<0) v = -v; | |
| 261 else v+= 128; | |
| 262 *dst++ = v; | |
| 263 } | |
| 264 break; | |
| 92 | 265 case CODEC_ID_PCM_ALAW: |
| 266 for(;n>0;n--) { | |
| 267 v = *samples++; | |
| 4960 | 268 *dst++ = linear_to_alaw[(v + 32768) >> 2]; |
| 92 | 269 } |
| 270 break; | |
| 271 case CODEC_ID_PCM_MULAW: | |
| 272 for(;n>0;n--) { | |
| 273 v = *samples++; | |
| 4960 | 274 *dst++ = linear_to_ulaw[(v + 32768) >> 2]; |
| 92 | 275 } |
| 276 break; | |
| 277 default: | |
| 278 return -1; | |
| 279 } | |
|
381
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* Mea culpa: it seems that I broke encoding to 8-bit pcm files. This fixes it.
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280 //avctx->frame_size = (dst - frame) / (sample_size * avctx->channels); |
| 372 | 281 |
| 92 | 282 return dst - frame; |
| 283 } | |
| 284 | |
| 285 typedef struct PCMDecode { | |
| 286 short table[256]; | |
| 287 } PCMDecode; | |
| 288 | |
|
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|
289 static av_cold int pcm_decode_init(AVCodecContext * avctx) |
| 92 | 290 { |
| 291 PCMDecode *s = avctx->priv_data; | |
| 292 int i; | |
| 293 | |
| 294 switch(avctx->codec->id) { | |
| 295 case CODEC_ID_PCM_ALAW: | |
| 296 for(i=0;i<256;i++) | |
| 297 s->table[i] = alaw2linear(i); | |
| 298 break; | |
| 299 case CODEC_ID_PCM_MULAW: | |
| 300 for(i=0;i<256;i++) | |
| 301 s->table[i] = ulaw2linear(i); | |
| 302 break; | |
| 303 default: | |
| 304 break; | |
| 305 } | |
|
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306 |
|
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307 avctx->sample_fmt = avctx->codec->sample_fmts[0]; |
| 92 | 308 return 0; |
| 309 } | |
| 310 | |
|
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311 /** |
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312 * \brief convert samples to 16 bit |
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313 * \param bps byte per sample for the source format, must be >= 2 |
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314 * \param le 0 for big-, 1 for little-endian |
|
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315 * \param us 0 for signed, 1 for unsigned input |
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316 * \param src input samples |
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317 * \param samples output samples |
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318 * \param src_len number of bytes in src |
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319 */ |
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320 static inline void decode_to16(int bps, int le, int us, |
| 6218 | 321 const uint8_t **src, short **samples, int src_len) |
|
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322 { |
| 4956 | 323 int usum = us ? -0x8000 : 0; |
|
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324 register int n = src_len / bps; |
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325 if (le) *src += bps - 2; |
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326 for(;n>0;n--) { |
| 4958 | 327 register int v; |
| 328 if (le) v = AV_RL16(*src); | |
| 329 else v = AV_RB16(*src); | |
| 330 v += usum; | |
| 331 *(*samples)++ = v; | |
|
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332 *src += bps; |
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333 } |
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334 if (le) *src -= bps - 2; |
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335 } |
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336 |
|
440
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* started to cleanup name clashes for onetime compilation
kabi
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337 static int pcm_decode_frame(AVCodecContext *avctx, |
| 2979 | 338 void *data, int *data_size, |
| 6218 | 339 const uint8_t *buf, int buf_size) |
| 92 | 340 { |
| 341 PCMDecode *s = avctx->priv_data; | |
|
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342 int sample_size, c, n; |
| 92 | 343 short *samples; |
| 6218 | 344 const uint8_t *src, *src2[MAX_CHANNELS]; |
| 92 | 345 |
| 346 samples = data; | |
| 347 src = buf; | |
| 348 | |
|
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349 if(avctx->channels <= 0 || avctx->channels > MAX_CHANNELS){ |
|
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350 av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n"); |
|
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351 return -1; |
|
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352 } |
|
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353 |
|
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354 sample_size = av_get_bits_per_sample(avctx->codec_id)/8; |
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355 |
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356 n = avctx->channels * sample_size; |
|
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357 /* av_get_bits_per_sample returns 0 for CODEC_ID_PCM_DVD */ |
|
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358 if (CODEC_ID_PCM_DVD == avctx->codec_id) |
|
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359 /* 2 samples are interleaved per block in PCM_DVD */ |
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360 n = 2 * avctx->channels * avctx->bits_per_sample/8; |
|
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361 |
|
6033
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Fix crash in PCM decoder when number of channels is not set.
benoit
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362 if(n && buf_size % n){ |
| 4506 | 363 av_log(avctx, AV_LOG_ERROR, "invalid PCM packet\n"); |
| 364 return -1; | |
| 365 } | |
| 366 | |
| 4351 | 367 buf_size= FFMIN(buf_size, *data_size/2); |
| 368 *data_size=0; | |
| 2506 | 369 |
|
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370 n = buf_size/sample_size; |
|
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371 |
| 92 | 372 switch(avctx->codec->id) { |
|
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373 case CODEC_ID_PCM_F32BE: |
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374 { |
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375 float *fsamples = data; |
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376 for(;n>0;n--) |
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377 *fsamples++ = av_int2flt(bytestream_get_be32(&src)); |
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378 samples = (void*)fsamples; |
|
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379 break; |
|
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380 } |
|
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381 case CODEC_ID_PCM_S32LE: |
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382 decode_to16(4, 1, 0, &src, &samples, buf_size); |
|
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383 break; |
|
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|
384 case CODEC_ID_PCM_S32BE: |
|
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385 decode_to16(4, 0, 0, &src, &samples, buf_size); |
|
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reimar
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|
386 break; |
|
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387 case CODEC_ID_PCM_U32LE: |
|
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388 decode_to16(4, 1, 1, &src, &samples, buf_size); |
|
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reimar
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|
389 break; |
|
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|
390 case CODEC_ID_PCM_U32BE: |
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391 decode_to16(4, 0, 1, &src, &samples, buf_size); |
|
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392 break; |
|
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|
393 case CODEC_ID_PCM_S24LE: |
|
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394 decode_to16(3, 1, 0, &src, &samples, buf_size); |
|
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reimar
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|
395 break; |
|
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|
396 case CODEC_ID_PCM_S24BE: |
|
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|
397 decode_to16(3, 0, 0, &src, &samples, buf_size); |
|
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|
398 break; |
|
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|
399 case CODEC_ID_PCM_U24LE: |
|
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|
400 decode_to16(3, 1, 1, &src, &samples, buf_size); |
|
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reimar
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diff
changeset
|
401 break; |
|
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|
402 case CODEC_ID_PCM_U24BE: |
|
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|
403 decode_to16(3, 0, 1, &src, &samples, buf_size); |
|
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reimar
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|
404 break; |
|
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reimar
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diff
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|
405 case CODEC_ID_PCM_S24DAUD: |
|
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|
406 for(;n>0;n--) { |
| 4959 | 407 uint32_t v = bytestream_get_be24(&src); |
|
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|
408 v >>= 4; // sync flags are here |
|
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|
409 *samples++ = ff_reverse[(v >> 8) & 0xff] + |
|
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|
410 (ff_reverse[v & 0xff] << 8); |
|
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|
411 } |
|
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|
412 break; |
| 92 | 413 case CODEC_ID_PCM_S16LE: |
| 414 for(;n>0;n--) { | |
| 4959 | 415 *samples++ = bytestream_get_le16(&src); |
| 92 | 416 } |
| 417 break; | |
|
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|
418 case CODEC_ID_PCM_S16LE_PLANAR: |
|
7518
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Simplify PCM codec; change 'n' in pcm_decode_frame() to equal "total number of samples".
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changeset
|
419 n /= avctx->channels; |
|
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Simplify PCM codec; change 'n' in pcm_decode_frame() to equal "total number of samples".
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changeset
|
420 for(c=0;c<avctx->channels;c++) |
|
a40c76e98909
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|
421 src2[c] = &src[c*n]; |
|
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|
422 for(n>>=1;n>0;n--) |
|
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|
423 for(c=0;c<avctx->channels;c++) |
|
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424 *samples++ = bytestream_get_le16(&src2[c]); |
|
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|
425 src = src2[avctx->channels-1]; |
|
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|
426 break; |
| 92 | 427 case CODEC_ID_PCM_S16BE: |
| 428 for(;n>0;n--) { | |
| 4959 | 429 *samples++ = bytestream_get_be16(&src); |
| 92 | 430 } |
| 431 break; | |
| 432 case CODEC_ID_PCM_U16LE: | |
| 433 for(;n>0;n--) { | |
| 4959 | 434 *samples++ = bytestream_get_le16(&src) - 0x8000; |
| 92 | 435 } |
| 436 break; | |
| 437 case CODEC_ID_PCM_U16BE: | |
| 438 for(;n>0;n--) { | |
| 4959 | 439 *samples++ = bytestream_get_be16(&src) - 0x8000; |
| 92 | 440 } |
| 441 break; | |
| 442 case CODEC_ID_PCM_S8: | |
| 443 for(;n>0;n--) { | |
| 4960 | 444 *samples++ = *src++ << 8; |
| 92 | 445 } |
| 446 break; | |
| 447 case CODEC_ID_PCM_U8: | |
| 448 for(;n>0;n--) { | |
| 4960 | 449 *samples++ = ((int)*src++ - 128) << 8; |
| 92 | 450 } |
| 451 break; | |
| 5422 | 452 case CODEC_ID_PCM_ZORK: |
| 453 for(;n>0;n--) { | |
| 454 int x= *src++; | |
| 455 if(x&128) x-= 128; | |
| 456 else x = -x; | |
| 457 *samples++ = x << 8; | |
| 458 } | |
| 459 break; | |
| 92 | 460 case CODEC_ID_PCM_ALAW: |
| 461 case CODEC_ID_PCM_MULAW: | |
| 462 for(;n>0;n--) { | |
| 4960 | 463 *samples++ = s->table[*src++]; |
| 92 | 464 } |
| 465 break; | |
|
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466 case CODEC_ID_PCM_DVD: |
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467 if(avctx->bits_per_sample != 20 && avctx->bits_per_sample != 24) { |
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468 av_log(avctx, AV_LOG_ERROR, "PCM DVD unsupported sample depth\n"); |
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469 return -1; |
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470 } else { |
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471 int jump = avctx->channels * (avctx->bits_per_sample-16) / 4; |
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472 n = buf_size / (avctx->channels * 2 * avctx->bits_per_sample / 8); |
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473 while (n--) { |
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474 for (c=0; c < 2*avctx->channels; c++) |
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475 *samples++ = bytestream_get_be16(&src); |
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476 src += jump; |
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477 } |
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478 } |
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479 break; |
| 92 | 480 default: |
| 481 return -1; | |
| 482 } | |
| 1064 | 483 *data_size = (uint8_t *)samples - (uint8_t *)data; |
| 92 | 484 return src - buf; |
| 485 } | |
| 486 | |
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487 #ifdef CONFIG_ENCODERS |
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488 #define PCM_ENCODER(id,sample_fmt_,name,long_name_) \ |
| 92 | 489 AVCodec name ## _encoder = { \ |
| 490 #name, \ | |
| 491 CODEC_TYPE_AUDIO, \ | |
| 492 id, \ | |
| 493 0, \ | |
| 2979 | 494 pcm_encode_init, \ |
| 495 pcm_encode_frame, \ | |
| 496 pcm_encode_close, \ | |
| 92 | 497 NULL, \ |
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498 .sample_fmts = (enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \ |
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499 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ |
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500 }; |
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501 #else |
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502 #define PCM_ENCODER(id,sample_fmt_,name,long_name_) |
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503 #endif |
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504 |
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505 #ifdef CONFIG_DECODERS |
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506 #define PCM_DECODER(id,sample_fmt_,name,long_name_) \ |
| 92 | 507 AVCodec name ## _decoder = { \ |
| 508 #name, \ | |
| 509 CODEC_TYPE_AUDIO, \ | |
| 510 id, \ | |
| 511 sizeof(PCMDecode), \ | |
| 2979 | 512 pcm_decode_init, \ |
| 92 | 513 NULL, \ |
| 514 NULL, \ | |
|
440
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515 pcm_decode_frame, \ |
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516 .sample_fmts = (enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \ |
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517 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ |
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518 }; |
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519 #else |
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520 #define PCM_DECODER(id,sample_fmt_,name,long_name_) |
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521 #endif |
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522 |
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523 #define PCM_CODEC(id, sample_fmt_, name, long_name_) \ |
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524 PCM_ENCODER(id,sample_fmt_,name,long_name_) PCM_DECODER(id,sample_fmt_,name,long_name_) |
| 92 | 525 |
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526 /* Note: Do not forget to add new entries to the Makefile as well. */ |
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527 PCM_CODEC (CODEC_ID_PCM_ALAW, SAMPLE_FMT_S16, pcm_alaw, "A-law PCM"); |
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528 PCM_CODEC (CODEC_ID_PCM_DVD, SAMPLE_FMT_S16, pcm_dvd, "signed 16|20|24-bit big-endian PCM"); |
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529 PCM_CODEC (CODEC_ID_PCM_F32BE, SAMPLE_FMT_FLT, pcm_f32be, "32-bit floating point big-endian PCM"); |
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530 PCM_CODEC (CODEC_ID_PCM_MULAW, SAMPLE_FMT_S16, pcm_mulaw, "mu-law PCM"); |
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531 PCM_CODEC (CODEC_ID_PCM_S8, SAMPLE_FMT_S16, pcm_s8, "signed 8-bit PCM"); |
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532 PCM_CODEC (CODEC_ID_PCM_S16BE, SAMPLE_FMT_S16, pcm_s16be, "signed 16-bit big-endian PCM"); |
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533 PCM_CODEC (CODEC_ID_PCM_S16LE, SAMPLE_FMT_S16, pcm_s16le, "signed 16-bit little-endian PCM"); |
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534 PCM_DECODER(CODEC_ID_PCM_S16LE_PLANAR, SAMPLE_FMT_S16, pcm_s16le_planar, "16-bit little-endian planar PCM"); |
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535 PCM_CODEC (CODEC_ID_PCM_S24BE, SAMPLE_FMT_S16, pcm_s24be, "signed 24-bit big-endian PCM"); |
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536 PCM_CODEC (CODEC_ID_PCM_S24DAUD, SAMPLE_FMT_S16, pcm_s24daud, "D-Cinema audio signed 24-bit PCM"); |
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537 PCM_CODEC (CODEC_ID_PCM_S24LE, SAMPLE_FMT_S16, pcm_s24le, "signed 24-bit little-endian PCM"); |
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538 PCM_CODEC (CODEC_ID_PCM_S32BE, SAMPLE_FMT_S16, pcm_s32be, "signed 32-bit big-endian PCM"); |
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539 PCM_CODEC (CODEC_ID_PCM_S32LE, SAMPLE_FMT_S16, pcm_s32le, "signed 32-bit little-endian PCM"); |
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540 PCM_CODEC (CODEC_ID_PCM_U8, SAMPLE_FMT_S16, pcm_u8, "unsigned 8-bit PCM"); |
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541 PCM_CODEC (CODEC_ID_PCM_U16BE, SAMPLE_FMT_S16, pcm_u16be, "unsigned 16-bit big-endian PCM"); |
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542 PCM_CODEC (CODEC_ID_PCM_U16LE, SAMPLE_FMT_S16, pcm_u16le, "unsigned 16-bit little-endian PCM"); |
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543 PCM_CODEC (CODEC_ID_PCM_U24BE, SAMPLE_FMT_S16, pcm_u24be, "unsigned 24-bit big-endian PCM"); |
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544 PCM_CODEC (CODEC_ID_PCM_U24LE, SAMPLE_FMT_S16, pcm_u24le, "unsigned 24-bit little-endian PCM"); |
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545 PCM_CODEC (CODEC_ID_PCM_U32BE, SAMPLE_FMT_S16, pcm_u32be, "unsigned 32-bit big-endian PCM"); |
|
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|
546 PCM_CODEC (CODEC_ID_PCM_U32LE, SAMPLE_FMT_S16, pcm_u32le, "unsigned 32-bit little-endian PCM"); |
|
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547 PCM_CODEC (CODEC_ID_PCM_ZORK, SAMPLE_FMT_S16, pcm_zork, "Zork PCM"); |
