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FFmpeg/libavcodec/pcm.c
Diego Biurrun b78e7197a8 Change license headers to say 'FFmpeg' instead of 'this program/this library' 
and fix GPL/LGPL version mismatches.

Originally committed as revision 6577 to svn://svn.ffmpeg.org/ffmpeg/trunk
2006-10-07 15:30:46 +00:00

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/*
* PCM codecs
* Copyright (c) 2001 Fabrice Bellard.
*
* This file is part of FFmpeg.
*
* FFmpeg 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 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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 FFmpeg; 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
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