Mercurial > libavcodec.hg
view pcm.c @ 3265:ba26b176dcf8 libavcodec
gcc-2.95 compile fix, patch by Wolfram Gloger <wmglo A dent PIS med PIS uni-muenchen PIS de>
author | gpoirier |
---|---|
date | Tue, 18 Apr 2006 03:48:30 +0000 |
parents | 0b546eab515d |
children | c8c591fe26f8 |
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/* * PCM codecs * Copyright (c) 2001 Fabrice Bellard. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file pcm.c * PCM codecs */ #include "avcodec.h" #include "bitstream.h" // for ff_reverse /* from g711.c by SUN microsystems (unrestricted use) */ #define SIGN_BIT (0x80) /* Sign bit for a A-law byte. */ #define QUANT_MASK (0xf) /* Quantization field mask. */ #define NSEGS (8) /* Number of A-law segments. */ #define SEG_SHIFT (4) /* Left shift for segment number. */ #define SEG_MASK (0x70) /* Segment field mask. */ #define BIAS (0x84) /* Bias for linear code. */ /* * alaw2linear() - Convert an A-law value to 16-bit linear PCM * */ static int alaw2linear(unsigned char a_val) { int t; int seg; a_val ^= 0x55; t = a_val & QUANT_MASK; seg = ((unsigned)a_val & SEG_MASK) >> SEG_SHIFT; if(seg) t= (t + t + 1 + 32) << (seg + 2); else t= (t + t + 1 ) << 3; return ((a_val & SIGN_BIT) ? t : -t); } static int ulaw2linear(unsigned char u_val) { int t; /* Complement to obtain normal u-law value. */ u_val = ~u_val; /* * Extract and bias the quantization bits. Then * shift up by the segment number and subtract out the bias. */ t = ((u_val & QUANT_MASK) << 3) + BIAS; t <<= ((unsigned)u_val & SEG_MASK) >> SEG_SHIFT; return ((u_val & SIGN_BIT) ? (BIAS - t) : (t - BIAS)); } /* 16384 entries per table */ static uint8_t *linear_to_alaw = NULL; static int linear_to_alaw_ref = 0; static uint8_t *linear_to_ulaw = NULL; static int linear_to_ulaw_ref = 0; static void build_xlaw_table(uint8_t *linear_to_xlaw, int (*xlaw2linear)(unsigned char), int mask) { int i, j, v, v1, v2; j = 0; for(i=0;i<128;i++) { if (i != 127) { v1 = xlaw2linear(i ^ mask); v2 = xlaw2linear((i + 1) ^ mask); v = (v1 + v2 + 4) >> 3; } else { v = 8192; } for(;j<v;j++) { linear_to_xlaw[8192 + j] = (i ^ mask); if (j > 0) linear_to_xlaw[8192 - j] = (i ^ (mask ^ 0x80)); } } linear_to_xlaw[0] = linear_to_xlaw[1]; } static int pcm_encode_init(AVCodecContext *avctx) { avctx->frame_size = 1; switch(avctx->codec->id) { case CODEC_ID_PCM_ALAW: if (linear_to_alaw_ref == 0) { linear_to_alaw = av_malloc(16384); if (!linear_to_alaw) return -1; build_xlaw_table(linear_to_alaw, alaw2linear, 0xd5); } linear_to_alaw_ref++; break; case CODEC_ID_PCM_MULAW: if (linear_to_ulaw_ref == 0) { linear_to_ulaw = av_malloc(16384); if (!linear_to_ulaw) return -1; build_xlaw_table(linear_to_ulaw, ulaw2linear, 0xff); } linear_to_ulaw_ref++; break; default: break; } switch(avctx->codec->id) { case CODEC_ID_PCM_S32LE: case CODEC_ID_PCM_S32BE: case CODEC_ID_PCM_U32LE: case CODEC_ID_PCM_U32BE: avctx->block_align = 4 * avctx->channels; break; case CODEC_ID_PCM_S24LE: case CODEC_ID_PCM_S24BE: case CODEC_ID_PCM_U24LE: case CODEC_ID_PCM_U24BE: case CODEC_ID_PCM_S24DAUD: avctx->block_align = 3 * avctx->channels; break; case CODEC_ID_PCM_S16LE: case CODEC_ID_PCM_S16BE: case CODEC_ID_PCM_U16LE: case CODEC_ID_PCM_U16BE: avctx->block_align = 2 * avctx->channels; break; case CODEC_ID_PCM_S8: case CODEC_ID_PCM_U8: case CODEC_ID_PCM_MULAW: case CODEC_ID_PCM_ALAW: avctx->block_align = avctx->channels; break; default: break; } avctx->coded_frame= avcodec_alloc_frame(); avctx->coded_frame->key_frame= 1; return 0; } static int pcm_encode_close(AVCodecContext *avctx) { av_freep(&avctx->coded_frame); switch(avctx->codec->id) { case CODEC_ID_PCM_ALAW: if (--linear_to_alaw_ref == 0) av_free(linear_to_alaw); break; case CODEC_ID_PCM_MULAW: if (--linear_to_ulaw_ref == 0) av_free(linear_to_ulaw); break; default: /* nothing to free */ break; } return 0; } /** * \brief convert samples from 16 bit * \param bps byte per sample for the destination format, must be >= 2 * \param le 0 for big-, 1 for little-endian * \param us 0 for signed, 1 for unsigned output * \param samples input samples * \param dst output samples * \param n number of samples in samples buffer. */ static inline void encode_from16(int bps, int le, int us, short **samples, uint8_t **dst, int n) { if (bps > 2) memset(*dst, 0, n * bps); if (le) *dst += bps - 2; for(;n>0;n--) { register int v = *(*samples)++; if (us) v += 0x8000; (*dst)[le] = v >> 8; (*dst)[1 - le] = v; *dst += bps; } if (le) *dst -= bps - 2; } static int pcm_encode_frame(AVCodecContext *avctx, unsigned char *frame, int buf_size, void *data) { int n, sample_size, v; short *samples; unsigned char *dst; switch(avctx->codec->id) { case CODEC_ID_PCM_S32LE: case CODEC_ID_PCM_S32BE: case CODEC_ID_PCM_U32LE: case CODEC_ID_PCM_U32BE: sample_size = 4; break; case CODEC_ID_PCM_S24LE: case CODEC_ID_PCM_S24BE: case CODEC_ID_PCM_U24LE: case CODEC_ID_PCM_U24BE: case CODEC_ID_PCM_S24DAUD: sample_size = 3; break; case CODEC_ID_PCM_S16LE: case CODEC_ID_PCM_S16BE: case CODEC_ID_PCM_U16LE: case CODEC_ID_PCM_U16BE: sample_size = 2; break; default: sample_size = 1; break; } n = buf_size / sample_size; samples = data; dst = frame; switch(avctx->codec->id) { case CODEC_ID_PCM_S32LE: encode_from16(4, 1, 0, &samples, &dst, n); break; case CODEC_ID_PCM_S32BE: encode_from16(4, 0, 0, &samples, &dst, n); break; case CODEC_ID_PCM_U32LE: encode_from16(4, 1, 1, &samples, &dst, n); break; case CODEC_ID_PCM_U32BE: encode_from16(4, 0, 1, &samples, &dst, n); break; case CODEC_ID_PCM_S24LE: encode_from16(3, 1, 0, &samples, &dst, n); break; case CODEC_ID_PCM_S24BE: encode_from16(3, 0, 0, &samples, &dst, n); break; case CODEC_ID_PCM_U24LE: encode_from16(3, 1, 1, &samples, &dst, n); break; case CODEC_ID_PCM_U24BE: encode_from16(3, 0, 1, &samples, &dst, n); break; case CODEC_ID_PCM_S24DAUD: for(;n>0;n--) { uint32_t tmp = ff_reverse[*samples >> 8] + (ff_reverse[*samples & 0xff] << 8); tmp <<= 4; // sync flags would go here dst[2] = tmp & 0xff; tmp >>= 8; dst[1] = tmp & 0xff; dst[0] = tmp >> 8; samples++; dst += 3; } break; case CODEC_ID_PCM_S16LE: for(;n>0;n--) { v = *samples++; dst[0] = v & 0xff; dst[1] = v >> 8; dst += 2; } break; case CODEC_ID_PCM_S16BE: for(;n>0;n--) { v = *samples++; dst[0] = v >> 8; dst[1] = v; dst += 2; } break; case CODEC_ID_PCM_U16LE: for(;n>0;n--) { v = *samples++; v += 0x8000; dst[0] = v & 0xff; dst[1] = v >> 8; dst += 2; } break; case CODEC_ID_PCM_U16BE: for(;n>0;n--) { v = *samples++; v += 0x8000; dst[0] = v >> 8; dst[1] = v; dst += 2; } break; case CODEC_ID_PCM_S8: for(;n>0;n--) { v = *samples++; dst[0] = v >> 8; dst++; } break; case CODEC_ID_PCM_U8: for(;n>0;n--) { v = *samples++; dst[0] = (v >> 8) + 128; dst++; } break; case CODEC_ID_PCM_ALAW: for(;n>0;n--) { v = *samples++; dst[0] = linear_to_alaw[(v + 32768) >> 2]; dst++; } break; case CODEC_ID_PCM_MULAW: for(;n>0;n--) { v = *samples++; dst[0] = linear_to_ulaw[(v + 32768) >> 2]; dst++; } break; default: return -1; } //avctx->frame_size = (dst - frame) / (sample_size * avctx->channels); return dst - frame; } typedef struct PCMDecode { short table[256]; } PCMDecode; static int pcm_decode_init(AVCodecContext * avctx) { PCMDecode *s = avctx->priv_data; int i; switch(avctx->codec->id) { case CODEC_ID_PCM_ALAW: for(i=0;i<256;i++) s->table[i] = alaw2linear(i); break; case CODEC_ID_PCM_MULAW: for(i=0;i<256;i++) s->table[i] = ulaw2linear(i); break; default: break; } return 0; } /** * \brief convert samples to 16 bit * \param bps byte per sample for the source format, must be >= 2 * \param le 0 for big-, 1 for little-endian * \param us 0 for signed, 1 for unsigned input * \param src input samples * \param samples output samples * \param src_len number of bytes in src */ static inline void decode_to16(int bps, int le, int us, uint8_t **src, short **samples, int src_len) { register int n = src_len / bps; if (le) *src += bps - 2; for(;n>0;n--) { *(*samples)++ = ((*src)[le] << 8 | (*src)[1 - le]) - (us?0x8000:0); *src += bps; } if (le) *src -= bps - 2; } static int pcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size, uint8_t *buf, int buf_size) { PCMDecode *s = avctx->priv_data; int n; short *samples; uint8_t *src; samples = data; src = buf; if(buf_size > AVCODEC_MAX_AUDIO_FRAME_SIZE/2) buf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE/2; switch(avctx->codec->id) { case CODEC_ID_PCM_S32LE: decode_to16(4, 1, 0, &src, &samples, buf_size); break; case CODEC_ID_PCM_S32BE: decode_to16(4, 0, 0, &src, &samples, buf_size); break; case CODEC_ID_PCM_U32LE: decode_to16(4, 1, 1, &src, &samples, buf_size); break; case CODEC_ID_PCM_U32BE: decode_to16(4, 0, 1, &src, &samples, buf_size); break; case CODEC_ID_PCM_S24LE: decode_to16(3, 1, 0, &src, &samples, buf_size); break; case CODEC_ID_PCM_S24BE: decode_to16(3, 0, 0, &src, &samples, buf_size); break; case CODEC_ID_PCM_U24LE: decode_to16(3, 1, 1, &src, &samples, buf_size); break; case CODEC_ID_PCM_U24BE: decode_to16(3, 0, 1, &src, &samples, buf_size); break; case CODEC_ID_PCM_S24DAUD: n = buf_size / 3; for(;n>0;n--) { uint32_t v = src[0] << 16 | src[1] << 8 | src[2]; v >>= 4; // sync flags are here *samples++ = ff_reverse[(v >> 8) & 0xff] + (ff_reverse[v & 0xff] << 8); src += 3; } break; case CODEC_ID_PCM_S16LE: n = buf_size >> 1; for(;n>0;n--) { *samples++ = src[0] | (src[1] << 8); src += 2; } break; case CODEC_ID_PCM_S16BE: n = buf_size >> 1; for(;n>0;n--) { *samples++ = (src[0] << 8) | src[1]; src += 2; } break; case CODEC_ID_PCM_U16LE: n = buf_size >> 1; for(;n>0;n--) { *samples++ = (src[0] | (src[1] << 8)) - 0x8000; src += 2; } break; case CODEC_ID_PCM_U16BE: n = buf_size >> 1; for(;n>0;n--) { *samples++ = ((src[0] << 8) | src[1]) - 0x8000; src += 2; } break; case CODEC_ID_PCM_S8: n = buf_size; for(;n>0;n--) { *samples++ = src[0] << 8; src++; } break; case CODEC_ID_PCM_U8: n = buf_size; for(;n>0;n--) { *samples++ = ((int)src[0] - 128) << 8; src++; } break; case CODEC_ID_PCM_ALAW: case CODEC_ID_PCM_MULAW: n = buf_size; for(;n>0;n--) { *samples++ = s->table[src[0]]; src++; } break; default: return -1; } *data_size = (uint8_t *)samples - (uint8_t *)data; return src - buf; } #define PCM_CODEC(id, name) \ AVCodec name ## _encoder = { \ #name, \ CODEC_TYPE_AUDIO, \ id, \ 0, \ pcm_encode_init, \ pcm_encode_frame, \ pcm_encode_close, \ NULL, \ }; \ AVCodec name ## _decoder = { \ #name, \ CODEC_TYPE_AUDIO, \ id, \ sizeof(PCMDecode), \ pcm_decode_init, \ NULL, \ NULL, \ pcm_decode_frame, \ } PCM_CODEC(CODEC_ID_PCM_S32LE, pcm_s32le); PCM_CODEC(CODEC_ID_PCM_S32BE, pcm_s32be); PCM_CODEC(CODEC_ID_PCM_U32LE, pcm_u32le); PCM_CODEC(CODEC_ID_PCM_U32BE, pcm_u32be); PCM_CODEC(CODEC_ID_PCM_S24LE, pcm_s24le); PCM_CODEC(CODEC_ID_PCM_S24BE, pcm_s24be); PCM_CODEC(CODEC_ID_PCM_U24LE, pcm_u24le); PCM_CODEC(CODEC_ID_PCM_U24BE, pcm_u24be); PCM_CODEC(CODEC_ID_PCM_S24DAUD, pcm_s24daud); PCM_CODEC(CODEC_ID_PCM_S16LE, pcm_s16le); PCM_CODEC(CODEC_ID_PCM_S16BE, pcm_s16be); PCM_CODEC(CODEC_ID_PCM_U16LE, pcm_u16le); PCM_CODEC(CODEC_ID_PCM_U16BE, pcm_u16be); PCM_CODEC(CODEC_ID_PCM_S8, pcm_s8); PCM_CODEC(CODEC_ID_PCM_U8, pcm_u8); PCM_CODEC(CODEC_ID_PCM_ALAW, pcm_alaw); PCM_CODEC(CODEC_ID_PCM_MULAW, pcm_mulaw); #undef PCM_CODEC