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https://github.com/xenia-project/FFmpeg.git
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7993df6527
I have underestimated this a little, and these are just some ... Originally committed as revision 11708 to svn://svn.ffmpeg.org/ffmpeg/trunk
556 lines
15 KiB
C
556 lines
15 KiB
C
/*
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* PCM codecs
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* Copyright (c) 2001 Fabrice Bellard.
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file pcm.c
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* PCM codecs
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*/
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#include "avcodec.h"
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#include "bitstream.h" // for ff_reverse
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#include "bytestream.h"
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#define MAX_CHANNELS 64
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/* from g711.c by SUN microsystems (unrestricted use) */
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#define SIGN_BIT (0x80) /* Sign bit for a A-law byte. */
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#define QUANT_MASK (0xf) /* Quantization field mask. */
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#define NSEGS (8) /* Number of A-law segments. */
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#define SEG_SHIFT (4) /* Left shift for segment number. */
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#define SEG_MASK (0x70) /* Segment field mask. */
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#define BIAS (0x84) /* Bias for linear code. */
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/*
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* alaw2linear() - Convert an A-law value to 16-bit linear PCM
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*
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*/
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static int alaw2linear(unsigned char a_val)
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{
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int t;
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int seg;
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a_val ^= 0x55;
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t = a_val & QUANT_MASK;
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seg = ((unsigned)a_val & SEG_MASK) >> SEG_SHIFT;
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if(seg) t= (t + t + 1 + 32) << (seg + 2);
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else t= (t + t + 1 ) << 3;
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return ((a_val & SIGN_BIT) ? t : -t);
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}
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static int ulaw2linear(unsigned char u_val)
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{
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int t;
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/* Complement to obtain normal u-law value. */
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u_val = ~u_val;
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/*
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* Extract and bias the quantization bits. Then
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* shift up by the segment number and subtract out the bias.
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*/
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t = ((u_val & QUANT_MASK) << 3) + BIAS;
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t <<= ((unsigned)u_val & SEG_MASK) >> SEG_SHIFT;
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return ((u_val & SIGN_BIT) ? (BIAS - t) : (t - BIAS));
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}
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/* 16384 entries per table */
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static uint8_t linear_to_alaw[16384];
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static uint8_t linear_to_ulaw[16384];
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static void build_xlaw_table(uint8_t *linear_to_xlaw,
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int (*xlaw2linear)(unsigned char),
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int mask)
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{
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int i, j, v, v1, v2;
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j = 0;
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for(i=0;i<128;i++) {
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if (i != 127) {
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v1 = xlaw2linear(i ^ mask);
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v2 = xlaw2linear((i + 1) ^ mask);
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v = (v1 + v2 + 4) >> 3;
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} else {
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v = 8192;
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}
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for(;j<v;j++) {
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linear_to_xlaw[8192 + j] = (i ^ mask);
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if (j > 0)
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linear_to_xlaw[8192 - j] = (i ^ (mask ^ 0x80));
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}
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}
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linear_to_xlaw[0] = linear_to_xlaw[1];
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}
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static int pcm_encode_init(AVCodecContext *avctx)
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{
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avctx->frame_size = 1;
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switch(avctx->codec->id) {
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case CODEC_ID_PCM_ALAW:
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build_xlaw_table(linear_to_alaw, alaw2linear, 0xd5);
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break;
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case CODEC_ID_PCM_MULAW:
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build_xlaw_table(linear_to_ulaw, ulaw2linear, 0xff);
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break;
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default:
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break;
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}
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switch(avctx->codec->id) {
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case CODEC_ID_PCM_S32LE:
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case CODEC_ID_PCM_S32BE:
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case CODEC_ID_PCM_U32LE:
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case CODEC_ID_PCM_U32BE:
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avctx->block_align = 4 * avctx->channels;
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break;
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case CODEC_ID_PCM_S24LE:
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case CODEC_ID_PCM_S24BE:
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case CODEC_ID_PCM_U24LE:
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case CODEC_ID_PCM_U24BE:
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case CODEC_ID_PCM_S24DAUD:
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avctx->block_align = 3 * avctx->channels;
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break;
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case CODEC_ID_PCM_S16LE:
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case CODEC_ID_PCM_S16BE:
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case CODEC_ID_PCM_U16LE:
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case CODEC_ID_PCM_U16BE:
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avctx->block_align = 2 * avctx->channels;
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break;
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case CODEC_ID_PCM_S8:
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case CODEC_ID_PCM_U8:
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case CODEC_ID_PCM_MULAW:
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case CODEC_ID_PCM_ALAW:
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avctx->block_align = avctx->channels;
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break;
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default:
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break;
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}
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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return 0;
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}
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static int pcm_encode_close(AVCodecContext *avctx)
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{
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av_freep(&avctx->coded_frame);
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return 0;
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}
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/**
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* \brief convert samples from 16 bit
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* \param bps byte per sample for the destination format, must be >= 2
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* \param le 0 for big-, 1 for little-endian
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* \param us 0 for signed, 1 for unsigned output
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* \param samples input samples
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* \param dst output samples
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* \param n number of samples in samples buffer.
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*/
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static inline void encode_from16(int bps, int le, int us,
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short **samples, uint8_t **dst, int n) {
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int usum = us ? 0x8000 : 0;
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if (bps > 2)
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memset(*dst, 0, n * bps);
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if (le) *dst += bps - 2;
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for(;n>0;n--) {
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register int v = *(*samples)++;
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v += usum;
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if (le) AV_WL16(*dst, v);
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else AV_WB16(*dst, v);
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*dst += bps;
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}
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if (le) *dst -= bps - 2;
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}
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static int pcm_encode_frame(AVCodecContext *avctx,
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unsigned char *frame, int buf_size, void *data)
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{
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int n, sample_size, v;
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short *samples;
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unsigned char *dst;
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switch(avctx->codec->id) {
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case CODEC_ID_PCM_S32LE:
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case CODEC_ID_PCM_S32BE:
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case CODEC_ID_PCM_U32LE:
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case CODEC_ID_PCM_U32BE:
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sample_size = 4;
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break;
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case CODEC_ID_PCM_S24LE:
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case CODEC_ID_PCM_S24BE:
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case CODEC_ID_PCM_U24LE:
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case CODEC_ID_PCM_U24BE:
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case CODEC_ID_PCM_S24DAUD:
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sample_size = 3;
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break;
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case CODEC_ID_PCM_S16LE:
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case CODEC_ID_PCM_S16BE:
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case CODEC_ID_PCM_U16LE:
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case CODEC_ID_PCM_U16BE:
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sample_size = 2;
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break;
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default:
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sample_size = 1;
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break;
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}
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n = buf_size / sample_size;
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samples = data;
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dst = frame;
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switch(avctx->codec->id) {
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case CODEC_ID_PCM_S32LE:
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encode_from16(4, 1, 0, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_S32BE:
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encode_from16(4, 0, 0, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_U32LE:
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encode_from16(4, 1, 1, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_U32BE:
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encode_from16(4, 0, 1, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_S24LE:
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encode_from16(3, 1, 0, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_S24BE:
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encode_from16(3, 0, 0, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_U24LE:
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encode_from16(3, 1, 1, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_U24BE:
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encode_from16(3, 0, 1, &samples, &dst, n);
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break;
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case CODEC_ID_PCM_S24DAUD:
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for(;n>0;n--) {
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uint32_t tmp = ff_reverse[*samples >> 8] +
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(ff_reverse[*samples & 0xff] << 8);
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tmp <<= 4; // sync flags would go here
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bytestream_put_be24(&dst, tmp);
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samples++;
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}
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break;
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case CODEC_ID_PCM_S16LE:
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for(;n>0;n--) {
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v = *samples++;
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bytestream_put_le16(&dst, v);
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}
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break;
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case CODEC_ID_PCM_S16BE:
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for(;n>0;n--) {
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v = *samples++;
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bytestream_put_be16(&dst, v);
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}
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break;
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case CODEC_ID_PCM_U16LE:
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for(;n>0;n--) {
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v = *samples++;
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v += 0x8000;
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bytestream_put_le16(&dst, v);
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}
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break;
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case CODEC_ID_PCM_U16BE:
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for(;n>0;n--) {
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v = *samples++;
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v += 0x8000;
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bytestream_put_be16(&dst, v);
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}
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break;
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case CODEC_ID_PCM_S8:
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for(;n>0;n--) {
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v = *samples++;
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*dst++ = v >> 8;
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}
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break;
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case CODEC_ID_PCM_U8:
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for(;n>0;n--) {
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v = *samples++;
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*dst++ = (v >> 8) + 128;
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}
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break;
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case CODEC_ID_PCM_ZORK:
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for(;n>0;n--) {
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v= *samples++ >> 8;
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if(v<0) v = -v;
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else v+= 128;
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*dst++ = v;
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}
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break;
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case CODEC_ID_PCM_ALAW:
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for(;n>0;n--) {
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v = *samples++;
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*dst++ = linear_to_alaw[(v + 32768) >> 2];
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}
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break;
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case CODEC_ID_PCM_MULAW:
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for(;n>0;n--) {
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v = *samples++;
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*dst++ = linear_to_ulaw[(v + 32768) >> 2];
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}
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break;
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default:
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return -1;
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}
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//avctx->frame_size = (dst - frame) / (sample_size * avctx->channels);
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return dst - frame;
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}
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typedef struct PCMDecode {
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short table[256];
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} PCMDecode;
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static int pcm_decode_init(AVCodecContext * avctx)
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{
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PCMDecode *s = avctx->priv_data;
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int i;
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switch(avctx->codec->id) {
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case CODEC_ID_PCM_ALAW:
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for(i=0;i<256;i++)
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s->table[i] = alaw2linear(i);
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break;
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case CODEC_ID_PCM_MULAW:
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for(i=0;i<256;i++)
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s->table[i] = ulaw2linear(i);
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break;
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default:
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break;
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}
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return 0;
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}
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/**
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* \brief convert samples to 16 bit
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* \param bps byte per sample for the source format, must be >= 2
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* \param le 0 for big-, 1 for little-endian
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* \param us 0 for signed, 1 for unsigned input
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* \param src input samples
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* \param samples output samples
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* \param src_len number of bytes in src
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*/
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static inline void decode_to16(int bps, int le, int us,
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const uint8_t **src, short **samples, int src_len)
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{
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int usum = us ? -0x8000 : 0;
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register int n = src_len / bps;
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if (le) *src += bps - 2;
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for(;n>0;n--) {
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register int v;
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if (le) v = AV_RL16(*src);
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else v = AV_RB16(*src);
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v += usum;
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*(*samples)++ = v;
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*src += bps;
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}
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if (le) *src -= bps - 2;
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}
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static int pcm_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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const uint8_t *buf, int buf_size)
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{
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PCMDecode *s = avctx->priv_data;
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int c, n;
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short *samples;
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const uint8_t *src, *src2[MAX_CHANNELS];
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samples = data;
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src = buf;
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n= av_get_bits_per_sample(avctx->codec_id)/8;
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if(n && buf_size % n){
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av_log(avctx, AV_LOG_ERROR, "invalid PCM packet\n");
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return -1;
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}
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if(avctx->channels <= 0 || avctx->channels > MAX_CHANNELS){
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av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
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return -1;
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}
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buf_size= FFMIN(buf_size, *data_size/2);
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*data_size=0;
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n = buf_size/avctx->channels;
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for(c=0;c<avctx->channels;c++)
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src2[c] = &src[c*n];
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switch(avctx->codec->id) {
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case CODEC_ID_PCM_S32LE:
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decode_to16(4, 1, 0, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_S32BE:
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decode_to16(4, 0, 0, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_U32LE:
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decode_to16(4, 1, 1, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_U32BE:
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decode_to16(4, 0, 1, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_S24LE:
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decode_to16(3, 1, 0, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_S24BE:
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decode_to16(3, 0, 0, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_U24LE:
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decode_to16(3, 1, 1, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_U24BE:
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decode_to16(3, 0, 1, &src, &samples, buf_size);
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break;
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case CODEC_ID_PCM_S24DAUD:
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n = buf_size / 3;
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for(;n>0;n--) {
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uint32_t v = bytestream_get_be24(&src);
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v >>= 4; // sync flags are here
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*samples++ = ff_reverse[(v >> 8) & 0xff] +
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(ff_reverse[v & 0xff] << 8);
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}
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break;
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case CODEC_ID_PCM_S16LE:
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n = buf_size >> 1;
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for(;n>0;n--) {
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*samples++ = bytestream_get_le16(&src);
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}
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break;
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case CODEC_ID_PCM_S16LE_PLANAR:
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for(n>>=1;n>0;n--)
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for(c=0;c<avctx->channels;c++)
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*samples++ = bytestream_get_le16(&src2[c]);
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src = src2[avctx->channels-1];
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break;
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case CODEC_ID_PCM_S16BE:
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n = buf_size >> 1;
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for(;n>0;n--) {
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*samples++ = bytestream_get_be16(&src);
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}
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break;
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case CODEC_ID_PCM_U16LE:
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n = buf_size >> 1;
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for(;n>0;n--) {
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*samples++ = bytestream_get_le16(&src) - 0x8000;
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}
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break;
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case CODEC_ID_PCM_U16BE:
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n = buf_size >> 1;
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for(;n>0;n--) {
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*samples++ = bytestream_get_be16(&src) - 0x8000;
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}
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break;
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case CODEC_ID_PCM_S8:
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n = buf_size;
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for(;n>0;n--) {
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*samples++ = *src++ << 8;
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}
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break;
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case CODEC_ID_PCM_U8:
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n = buf_size;
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for(;n>0;n--) {
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*samples++ = ((int)*src++ - 128) << 8;
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}
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break;
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case CODEC_ID_PCM_ZORK:
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n = buf_size;
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for(;n>0;n--) {
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int x= *src++;
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if(x&128) x-= 128;
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else x = -x;
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*samples++ = x << 8;
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}
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break;
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case CODEC_ID_PCM_ALAW:
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case CODEC_ID_PCM_MULAW:
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n = buf_size;
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for(;n>0;n--) {
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*samples++ = s->table[*src++];
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}
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break;
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default:
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return -1;
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}
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*data_size = (uint8_t *)samples - (uint8_t *)data;
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return src - buf;
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}
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#ifdef CONFIG_ENCODERS
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#define PCM_ENCODER(id,name) \
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AVCodec name ## _encoder = { \
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#name, \
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CODEC_TYPE_AUDIO, \
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id, \
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0, \
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pcm_encode_init, \
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pcm_encode_frame, \
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pcm_encode_close, \
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NULL, \
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};
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#else
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#define PCM_ENCODER(id,name)
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#endif
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#ifdef CONFIG_DECODERS
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#define PCM_DECODER(id,name) \
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AVCodec name ## _decoder = { \
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#name, \
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CODEC_TYPE_AUDIO, \
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id, \
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sizeof(PCMDecode), \
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pcm_decode_init, \
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NULL, \
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NULL, \
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pcm_decode_frame, \
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};
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#else
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#define PCM_DECODER(id,name)
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#endif
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#define PCM_CODEC(id, name) \
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PCM_ENCODER(id,name) PCM_DECODER(id,name)
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PCM_CODEC (CODEC_ID_PCM_S32LE, pcm_s32le);
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PCM_CODEC (CODEC_ID_PCM_S32BE, pcm_s32be);
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PCM_CODEC (CODEC_ID_PCM_U32LE, pcm_u32le);
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PCM_CODEC (CODEC_ID_PCM_U32BE, pcm_u32be);
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PCM_CODEC (CODEC_ID_PCM_S24LE, pcm_s24le);
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PCM_CODEC (CODEC_ID_PCM_S24BE, pcm_s24be);
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PCM_CODEC (CODEC_ID_PCM_U24LE, pcm_u24le);
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PCM_CODEC (CODEC_ID_PCM_U24BE, pcm_u24be);
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PCM_CODEC (CODEC_ID_PCM_S24DAUD, pcm_s24daud);
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PCM_CODEC (CODEC_ID_PCM_S16LE, pcm_s16le);
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PCM_DECODER(CODEC_ID_PCM_S16LE_PLANAR, pcm_s16le_planar);
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PCM_CODEC (CODEC_ID_PCM_S16BE, pcm_s16be);
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PCM_CODEC (CODEC_ID_PCM_U16LE, pcm_u16le);
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PCM_CODEC (CODEC_ID_PCM_U16BE, pcm_u16be);
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PCM_CODEC (CODEC_ID_PCM_S8, pcm_s8);
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PCM_CODEC (CODEC_ID_PCM_U8, pcm_u8);
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PCM_CODEC (CODEC_ID_PCM_ALAW, pcm_alaw);
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PCM_CODEC (CODEC_ID_PCM_MULAW, pcm_mulaw);
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PCM_CODEC (CODEC_ID_PCM_ZORK, pcm_zork);
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