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ac94b8bcc6
ffmpeg/libavcodec/adpcm.c
Justin Ruggles ac94b8bcc6 adpcm: simplify packet size bounds checking in the ADPCM IMA QT decoder. 
This is easier to understand. It also avoids returning existing samples mixed
with new samples when the packet is too small.
2011-09-29 16:54:00 -04:00

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/*
* Copyright (c) 2001-2003 The ffmpeg Project
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
/**
* @file
* ADPCM decoders
* First version by Francois Revol (revol@free.fr)
* Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
* by Mike Melanson (melanson@pcisys.net)
* CD-ROM XA ADPCM codec by BERO
* EA ADPCM decoder by Robin Kay (komadori@myrealbox.com)
* EA ADPCM R1/R2/R3 decoder by Peter Ross (pross@xvid.org)
* EA IMA EACS decoder by Peter Ross (pross@xvid.org)
* EA IMA SEAD decoder by Peter Ross (pross@xvid.org)
* EA ADPCM XAS decoder by Peter Ross (pross@xvid.org)
* MAXIS EA ADPCM decoder by Robert Marston (rmarston@gmail.com)
* THP ADPCM decoder by Marco Gerards (mgerards@xs4all.nl)
*
* Features and limitations:
*
* Reference documents:
* http://www.pcisys.net/~melanson/codecs/simpleaudio.html
* http://www.geocities.com/SiliconValley/8682/aud3.txt
* http://openquicktime.sourceforge.net/plugins.htm
* XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html
* http://www.cs.ucla.edu/~leec/mediabench/applications.html
* SoX source code http://home.sprynet.com/~cbagwell/sox.html
*
* CD-ROM XA:
* http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html
* vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html
* readstr http://www.geocities.co.jp/Playtown/2004/
*/
/* These are for CD-ROM XA ADPCM */
static const int xa_adpcm_table[5][2] = {
{ 0, 0 },
{ 60, 0 },
{ 115, -52 },
{ 98, -55 },
{ 122, -60 }
};
static const int ea_adpcm_table[] = {
0, 240, 460, 392,
0, 0, -208, -220,
0, 1, 3, 4,
7, 8, 10, 11,
0, -1, -3, -4
};
// padded to zero where table size is less then 16
static const int swf_index_tables[4][16] = {
/*2*/ { -1, 2 },
/*3*/ { -1, -1, 2, 4 },
/*4*/ { -1, -1, -1, -1, 2, 4, 6, 8 },
/*5*/ { -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 }
};
/* end of tables */
typedef struct ADPCMDecodeContext {
ADPCMChannelStatus status[6];
} ADPCMDecodeContext;
static av_cold int adpcm_decode_init(AVCodecContext * avctx)
{
ADPCMDecodeContext *c = avctx->priv_data;
unsigned int max_channels = 2;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_EA_R1:
case CODEC_ID_ADPCM_EA_R2:
case CODEC_ID_ADPCM_EA_R3:
case CODEC_ID_ADPCM_EA_XAS:
max_channels = 6;
break;
}
if(avctx->channels > max_channels){
return -1;
}
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_CT:
c->status[0].step = c->status[1].step = 511;
break;
case CODEC_ID_ADPCM_IMA_WAV:
if (avctx->bits_per_coded_sample != 4) {
av_log(avctx, AV_LOG_ERROR, "Only 4-bit ADPCM IMA WAV files are supported\n");
return -1;
}
break;
case CODEC_ID_ADPCM_IMA_WS:
if (avctx->extradata && avctx->extradata_size == 2 * 4) {
c->status[0].predictor = AV_RL32(avctx->extradata);
c->status[1].predictor = AV_RL32(avctx->extradata + 4);
}
break;
default:
break;
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble, int shift)
{
int step_index;
int predictor;
int sign, delta, diff, step;
step = ff_adpcm_step_table[c->step_index];
step_index = c->step_index + ff_adpcm_index_table[(unsigned)nibble];
if (step_index < 0) step_index = 0;
else if (step_index > 88) step_index = 88;
sign = nibble & 8;
delta = nibble & 7;
/* perform direct multiplication instead of series of jumps proposed by
* the reference ADPCM implementation since modern CPUs can do the mults
* quickly enough */
diff = ((2 * delta + 1) * step) >> shift;
predictor = c->predictor;
if (sign) predictor -= diff;
else predictor += diff;
c->predictor = av_clip_int16(predictor);
c->step_index = step_index;
return (short)c->predictor;
}
static inline int adpcm_ima_qt_expand_nibble(ADPCMChannelStatus *c, int nibble, int shift)
{
int step_index;
int predictor;
int diff, step;
step = ff_adpcm_step_table[c->step_index];
step_index = c->step_index + ff_adpcm_index_table[nibble];
step_index = av_clip(step_index, 0, 88);
diff = step >> 3;
if (nibble & 4) diff += step;
if (nibble & 2) diff += step >> 1;
if (nibble & 1) diff += step >> 2;
if (nibble & 8)
predictor = c->predictor - diff;
else
predictor = c->predictor + diff;
c->predictor = av_clip_int16(predictor);
c->step_index = step_index;
return c->predictor;
}
static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble)
{
int predictor;
predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
c->sample2 = c->sample1;
c->sample1 = av_clip_int16(predictor);
c->idelta = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8;
if (c->idelta < 16) c->idelta = 16;
return c->sample1;
}
static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
{
int sign, delta, diff;
int new_step;
sign = nibble & 8;
delta = nibble & 7;
/* perform direct multiplication instead of series of jumps proposed by
* the reference ADPCM implementation since modern CPUs can do the mults
* quickly enough */
diff = ((2 * delta + 1) * c->step) >> 3;
/* predictor update is not so trivial: predictor is multiplied on 254/256 before updating */
c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff);
c->predictor = av_clip_int16(c->predictor);
/* calculate new step and clamp it to range 511..32767 */
new_step = (ff_adpcm_AdaptationTable[nibble & 7] * c->step) >> 8;
c->step = av_clip(new_step, 511, 32767);
return (short)c->predictor;
}
static inline short adpcm_sbpro_expand_nibble(ADPCMChannelStatus *c, char nibble, int size, int shift)
{
int sign, delta, diff;
sign = nibble & (1<<(size-1));
delta = nibble & ((1<<(size-1))-1);
diff = delta << (7 + c->step + shift);
/* clamp result */
c->predictor = av_clip(c->predictor + (sign ? -diff : diff), -16384,16256);
/* calculate new step */
if (delta >= (2*size - 3) && c->step < 3)
c->step++;
else if (delta == 0 && c->step > 0)
c->step--;
return (short) c->predictor;
}
static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned char nibble)
{
if(!c->step) {
c->predictor = 0;
c->step = 127;
}
c->predictor += (c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8;
c->predictor = av_clip_int16(c->predictor);
c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
c->step = av_clip(c->step, 127, 24567);
return c->predictor;
}
static void xa_decode(short *out, const unsigned char *in,
ADPCMChannelStatus *left, ADPCMChannelStatus *right, int inc)
{
int i, j;
int shift,filter,f0,f1;
int s_1,s_2;
int d,s,t;
for(i=0;i<4;i++) {
shift = 12 - (in[4+i*2] & 15);
filter = in[4+i*2] >> 4;
f0 = xa_adpcm_table[filter][0];
f1 = xa_adpcm_table[filter][1];
s_1 = left->sample1;
s_2 = left->sample2;
for(j=0;j<28;j++) {
d = in[16+i+j*4];
t = (signed char)(d<<4)>>4;
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
}
if (inc==2) { /* stereo */
left->sample1 = s_1;
left->sample2 = s_2;
s_1 = right->sample1;
s_2 = right->sample2;
out = out + 1 - 28*2;
}
shift = 12 - (in[5+i*2] & 15);
filter = in[5+i*2] >> 4;
f0 = xa_adpcm_table[filter][0];
f1 = xa_adpcm_table[filter][1];
for(j=0;j<28;j++) {
d = in[16+i+j*4];
t = (signed char)d >> 4;
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
}
if (inc==2) { /* stereo */
right->sample1 = s_1;
right->sample2 = s_2;
out -= 1;
} else {
left->sample1 = s_1;
left->sample2 = s_2;
}
}
}
/* DK3 ADPCM support macro */
#define DK3_GET_NEXT_NIBBLE() \
if (decode_top_nibble_next) \
{ \
nibble = last_byte >> 4; \
decode_top_nibble_next = 0; \
} \
else \
{ \
last_byte = *src++; \
if (src >= buf + buf_size) break; \
nibble = last_byte & 0x0F; \
decode_top_nibble_next = 1; \
}
static int adpcm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ADPCMDecodeContext *c = avctx->priv_data;
ADPCMChannelStatus *cs;
int n, m, channel, i;
int block_predictor[2];
short *samples;
short *samples_end;
const uint8_t *src;
int st; /* stereo */
/* DK3 ADPCM accounting variables */
unsigned char last_byte = 0;
unsigned char nibble;
int decode_top_nibble_next = 0;
int diff_channel;
/* EA ADPCM state variables */
uint32_t samples_in_chunk;
int32_t previous_left_sample, previous_right_sample;
int32_t current_left_sample, current_right_sample;
int32_t next_left_sample, next_right_sample;
int32_t coeff1l, coeff2l, coeff1r, coeff2r;
uint8_t shift_left, shift_right;
int count1, count2;
int coeff[2][2], shift[2];//used in EA MAXIS ADPCM
if (!buf_size)
return 0;
//should protect all 4bit ADPCM variants
//8 is needed for CODEC_ID_ADPCM_IMA_WAV with 2 channels
//
if(*data_size/4 < buf_size + 8)
return -1;
samples = data;
samples_end= samples + *data_size/2;
*data_size= 0;
src = buf;
st = avctx->channels == 2 ? 1 : 0;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_QT:
/* In QuickTime, IMA is encoded by chunks of 34 bytes (=64 samples).
Channel data is interleaved per-chunk. */
if (buf_size / 34 < avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
return AVERROR(EINVAL);
}
for (channel = 0; channel < avctx->channels; channel++) {
int16_t predictor;
int step_index;
cs = &(c->status[channel]);
/* (pppppp) (piiiiiii) */
/* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */
predictor = AV_RB16(src);
step_index = predictor & 0x7F;
predictor &= 0xFF80;
src += 2;
if (cs->step_index == step_index) {
int diff = (int)predictor - cs->predictor;
if (diff < 0)
diff = - diff;
if (diff > 0x7f)
goto update;
} else {
update:
cs->step_index = step_index;
cs->predictor = predictor;
}
if (cs->step_index > 88){
av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
cs->step_index = 88;
}
samples = (short*)data + channel;
for (m = 0; m < 32; m++) {
*samples = adpcm_ima_qt_expand_nibble(cs, src[0] & 0x0F, 3);
samples += avctx->channels;
*samples = adpcm_ima_qt_expand_nibble(cs, src[0] >> 4 , 3);
samples += avctx->channels;
src ++;
}
}
if (st)
samples--;
break;
case CODEC_ID_ADPCM_IMA_WAV:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
// samples_per_block= (block_align-4*chanels)*8 / (bits_per_sample * chanels) + 1;
for(i=0; i<avctx->channels; i++){
cs = &(c->status[i]);
cs->predictor = *samples++ = (int16_t)bytestream_get_le16(&src);
cs->step_index = *src++;
if (cs->step_index > 88){
av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
cs->step_index = 88;
}
if (*src++) av_log(avctx, AV_LOG_ERROR, "unused byte should be null but is %d!!\n", src[-1]); /* unused */
}
while(src < buf + buf_size){
for(m=0; m<4; m++){
for(i=0; i<=st; i++)
*samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] & 0x0F, 3);
for(i=0; i<=st; i++)
*samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] >> 4 , 3);
src++;
}
src += 4*st;
}
break;
case CODEC_ID_ADPCM_4XM:
cs = &(c->status[0]);
c->status[0].predictor= (int16_t)bytestream_get_le16(&src);
if(st){
c->status[1].predictor= (int16_t)bytestream_get_le16(&src);
}
c->status[0].step_index= (int16_t)bytestream_get_le16(&src);
if(st){
c->status[1].step_index= (int16_t)bytestream_get_le16(&src);
}
if (cs->step_index < 0) cs->step_index = 0;
if (cs->step_index > 88) cs->step_index = 88;
m= (buf_size - (src - buf))>>st;
for(i=0; i<m; i++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[i] & 0x0F, 4);
if (st)
*samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] & 0x0F, 4);
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[i] >> 4, 4);
if (st)
*samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] >> 4, 4);
}
src += m<<st;
break;
case CODEC_ID_ADPCM_MS:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
n = buf_size - 7 * avctx->channels;
if (n < 0)
return -1;
block_predictor[0] = av_clip(*src++, 0, 6);
block_predictor[1] = 0;
if (st)
block_predictor[1] = av_clip(*src++, 0, 6);
c->status[0].idelta = (int16_t)bytestream_get_le16(&src);
if (st){
c->status[1].idelta = (int16_t)bytestream_get_le16(&src);
}
c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[0]];
c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[0]];
c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[1]];
c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[1]];
c->status[0].sample1 = bytestream_get_le16(&src);
if (st) c->status[1].sample1 = bytestream_get_le16(&src);
c->status[0].sample2 = bytestream_get_le16(&src);
if (st) c->status[1].sample2 = bytestream_get_le16(&src);
*samples++ = c->status[0].sample2;
if (st) *samples++ = c->status[1].sample2;
*samples++ = c->status[0].sample1;
if (st) *samples++ = c->status[1].sample1;
for(;n>0;n--) {
*samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 );
*samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F);
src ++;
}
break;
case CODEC_ID_ADPCM_IMA_DK4:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
c->status[0].predictor = (int16_t)bytestream_get_le16(&src);
c->status[0].step_index = *src++;
src++;
*samples++ = c->status[0].predictor;
if (st) {
c->status[1].predictor = (int16_t)bytestream_get_le16(&src);
c->status[1].step_index = *src++;
src++;
*samples++ = c->status[1].predictor;
}
while (src < buf + buf_size) {
uint8_t v = *src++;
*samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
}
break;
case CODEC_ID_ADPCM_IMA_DK3:
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
if(buf_size + 16 > (samples_end - samples)*3/8)
return -1;
c->status[0].predictor = (int16_t)AV_RL16(src + 10);
c->status[1].predictor = (int16_t)AV_RL16(src + 12);
c->status[0].step_index = src[14];
c->status[1].step_index = src[15];
/* sign extend the predictors */
src += 16;
diff_channel = c->status[1].predictor;
/* the DK3_GET_NEXT_NIBBLE macro issues the break statement when
* the buffer is consumed */
while (1) {
/* for this algorithm, c->status[0] is the sum channel and
* c->status[1] is the diff channel */
/* process the first predictor of the sum channel */
DK3_GET_NEXT_NIBBLE();
adpcm_ima_expand_nibble(&c->status[0], nibble, 3);
/* process the diff channel predictor */
DK3_GET_NEXT_NIBBLE();
adpcm_ima_expand_nibble(&c->status[1], nibble, 3);
/* process the first pair of stereo PCM samples */
diff_channel = (diff_channel + c->status[1].predictor) / 2;
*samples++ = c->status[0].predictor + c->status[1].predictor;
*samples++ = c->status[0].predictor - c->status[1].predictor;
/* process the second predictor of the sum channel */
DK3_GET_NEXT_NIBBLE();
adpcm_ima_expand_nibble(&c->status[0], nibble, 3);
/* process the second pair of stereo PCM samples */
diff_channel = (diff_channel + c->status[1].predictor) / 2;
*samples++ = c->status[0].predictor + c->status[1].predictor;
*samples++ = c->status[0].predictor - c->status[1].predictor;
}
break;
case CODEC_ID_ADPCM_IMA_ISS:
c->status[0].predictor = (int16_t)AV_RL16(src + 0);
c->status[0].step_index = src[2];
src += 4;
if(st) {
c->status[1].predictor = (int16_t)AV_RL16(src + 0);
c->status[1].step_index = src[2];
src += 4;
}
while (src < buf + buf_size) {
uint8_t v1, v2;
uint8_t v = *src++;
/* nibbles are swapped for mono */
if (st) {
v1 = v >> 4;
v2 = v & 0x0F;
} else {
v2 = v >> 4;
v1 = v & 0x0F;
}
*samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v1, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3);
}
break;
case CODEC_ID_ADPCM_IMA_WS:
while (src < buf + buf_size) {
uint8_t v = *src++;
*samples++ = adpcm_ima_expand_nibble(&c->status[0], v >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
}
break;
case CODEC_ID_ADPCM_XA:
while (buf_size >= 128) {
xa_decode(samples, src, &c->status[0], &c->status[1],
avctx->channels);
src += 128;
samples += 28 * 8;
buf_size -= 128;
}
break;
case CODEC_ID_ADPCM_IMA_EA_EACS:
samples_in_chunk = bytestream_get_le32(&src) >> (1-st);
if (samples_in_chunk > buf_size-4-(8<<st)) {
src += buf_size - 4;
break;
}
for (i=0; i<=st; i++)
c->status[i].step_index = bytestream_get_le32(&src);
for (i=0; i<=st; i++)
c->status[i].predictor = bytestream_get_le32(&src);
for (; samples_in_chunk; samples_in_chunk--, src++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], *src>>4, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[st], *src&0x0F, 3);
}
break;
case CODEC_ID_ADPCM_IMA_EA_SEAD:
for (; src < buf+buf_size; src++) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[0] >> 4, 6);
*samples++ = adpcm_ima_expand_nibble(&c->status[st],src[0]&0x0F, 6);
}
break;
case CODEC_ID_ADPCM_EA:
/* Each EA ADPCM frame has a 12-byte header followed by 30-byte pieces,
each coding 28 stereo samples. */
if (buf_size < 12) {
av_log(avctx, AV_LOG_ERROR, "frame too small\n");
return AVERROR(EINVAL);
}
samples_in_chunk = AV_RL32(src);
if (samples_in_chunk / 28 > (buf_size - 12) / 30) {
av_log(avctx, AV_LOG_ERROR, "invalid frame\n");
return AVERROR(EINVAL);
}
src += 4;
current_left_sample = (int16_t)bytestream_get_le16(&src);
previous_left_sample = (int16_t)bytestream_get_le16(&src);
current_right_sample = (int16_t)bytestream_get_le16(&src);
previous_right_sample = (int16_t)bytestream_get_le16(&src);
for (count1 = 0; count1 < samples_in_chunk/28;count1++) {
coeff1l = ea_adpcm_table[ *src >> 4 ];
coeff2l = ea_adpcm_table[(*src >> 4 ) + 4];
coeff1r = ea_adpcm_table[*src & 0x0F];
coeff2r = ea_adpcm_table[(*src & 0x0F) + 4];
src++;
shift_left = (*src >> 4 ) + 8;
shift_right = (*src & 0x0F) + 8;
src++;
for (count2 = 0; count2 < 28; count2++) {
next_left_sample = (int32_t)((*src & 0xF0) << 24) >> shift_left;
next_right_sample = (int32_t)((*src & 0x0F) << 28) >> shift_right;
src++;
next_left_sample = (next_left_sample +
(current_left_sample * coeff1l) +
(previous_left_sample * coeff2l) + 0x80) >> 8;
next_right_sample = (next_right_sample +
(current_right_sample * coeff1r) +
(previous_right_sample * coeff2r) + 0x80) >> 8;
previous_left_sample = current_left_sample;
current_left_sample = av_clip_int16(next_left_sample);
previous_right_sample = current_right_sample;
current_right_sample = av_clip_int16(next_right_sample);
*samples++ = (unsigned short)current_left_sample;
*samples++ = (unsigned short)current_right_sample;
}
}
if (src - buf == buf_size - 2)
src += 2; // Skip terminating 0x0000
break;
case CODEC_ID_ADPCM_EA_MAXIS_XA:
for(channel = 0; channel < avctx->channels; channel++) {
for (i=0; i<2; i++)
coeff[channel][i] = ea_adpcm_table[(*src >> 4) + 4*i];
shift[channel] = (*src & 0x0F) + 8;
src++;
}
for (count1 = 0; count1 < (buf_size - avctx->channels) / avctx->channels; count1++) {
for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */
for(channel = 0; channel < avctx->channels; channel++) {
int32_t sample = (int32_t)(((*(src+channel) >> i) & 0x0F) << 0x1C) >> shift[channel];
sample = (sample +
c->status[channel].sample1 * coeff[channel][0] +
c->status[channel].sample2 * coeff[channel][1] + 0x80) >> 8;
c->status[channel].sample2 = c->status[channel].sample1;
c->status[channel].sample1 = av_clip_int16(sample);
*samples++ = c->status[channel].sample1;
}
}
src+=avctx->channels;
}
break;
case CODEC_ID_ADPCM_EA_R1:
case CODEC_ID_ADPCM_EA_R2:
case CODEC_ID_ADPCM_EA_R3: {
/* channel numbering
2chan: 0=fl, 1=fr
4chan: 0=fl, 1=rl, 2=fr, 3=rr
6chan: 0=fl, 1=c, 2=fr, 3=rl, 4=rr, 5=sub */
const int big_endian = avctx->codec->id == CODEC_ID_ADPCM_EA_R3;
int32_t previous_sample, current_sample, next_sample;
int32_t coeff1, coeff2;
uint8_t shift;
unsigned int channel;
uint16_t *samplesC;
const uint8_t *srcC;
const uint8_t *src_end = buf + buf_size;
samples_in_chunk = (big_endian ? bytestream_get_be32(&src)
: bytestream_get_le32(&src)) / 28;
if (samples_in_chunk > UINT32_MAX/(28*avctx->channels) ||
28*samples_in_chunk*avctx->channels > samples_end-samples) {
src += buf_size - 4;
break;
}
for (channel=0; channel<avctx->channels; channel++) {
int32_t offset = (big_endian ? bytestream_get_be32(&src)
: bytestream_get_le32(&src))
+ (avctx->channels-channel-1) * 4;
if ((offset < 0) || (offset >= src_end - src - 4)) break;
srcC = src + offset;
samplesC = samples + channel;
if (avctx->codec->id == CODEC_ID_ADPCM_EA_R1) {
current_sample = (int16_t)bytestream_get_le16(&srcC);
previous_sample = (int16_t)bytestream_get_le16(&srcC);
} else {
current_sample = c->status[channel].predictor;
previous_sample = c->status[channel].prev_sample;
}
for (count1=0; count1<samples_in_chunk; count1++) {
if (*srcC == 0xEE) { /* only seen in R2 and R3 */
srcC++;
if (srcC > src_end - 30*2) break;
current_sample = (int16_t)bytestream_get_be16(&srcC);
previous_sample = (int16_t)bytestream_get_be16(&srcC);
for (count2=0; count2<28; count2++) {
*samplesC = (int16_t)bytestream_get_be16(&srcC);
samplesC += avctx->channels;
}
} else {
coeff1 = ea_adpcm_table[ *srcC>>4 ];
coeff2 = ea_adpcm_table[(*srcC>>4) + 4];
shift = (*srcC++ & 0x0F) + 8;
if (srcC > src_end - 14) break;
for (count2=0; count2<28; count2++) {
if (count2 & 1)
next_sample = (int32_t)((*srcC++ & 0x0F) << 28) >> shift;
else
next_sample = (int32_t)((*srcC & 0xF0) << 24) >> shift;
next_sample += (current_sample * coeff1) +
(previous_sample * coeff2);
next_sample = av_clip_int16(next_sample >> 8);
previous_sample = current_sample;
current_sample = next_sample;
*samplesC = current_sample;
samplesC += avctx->channels;
}
}
}
if (avctx->codec->id != CODEC_ID_ADPCM_EA_R1) {
c->status[channel].predictor = current_sample;
c->status[channel].prev_sample = previous_sample;
}
}
src = src + buf_size - (4 + 4*avctx->channels);
samples += 28 * samples_in_chunk * avctx->channels;
break;
}
case CODEC_ID_ADPCM_EA_XAS:
if (samples_end-samples < 32*4*avctx->channels
|| buf_size < (4+15)*4*avctx->channels) {
src += buf_size;
break;
}
for (channel=0; channel<avctx->channels; channel++) {
int coeff[2][4], shift[4];
short *s2, *s = &samples[channel];
for (n=0; n<4; n++, s+=32*avctx->channels) {
for (i=0; i<2; i++)
coeff[i][n] = ea_adpcm_table[(src[0]&0x0F)+4*i];
shift[n] = (src[2]&0x0F) + 8;
for (s2=s, i=0; i<2; i++, src+=2, s2+=avctx->channels)
s2[0] = (src[0]&0xF0) + (src[1]<<8);
}
for (m=2; m<32; m+=2) {
s = &samples[m*avctx->channels + channel];
for (n=0; n<4; n++, src++, s+=32*avctx->channels) {
for (s2=s, i=0; i<8; i+=4, s2+=avctx->channels) {
int level = (int32_t)((*src & (0xF0>>i)) << (24+i)) >> shift[n];
int pred = s2[-1*avctx->channels] * coeff[0][n]
+ s2[-2*avctx->channels] * coeff[1][n];
s2[0] = av_clip_int16((level + pred + 0x80) >> 8);
}
}
}
}
samples += 32*4*avctx->channels;
break;
case CODEC_ID_ADPCM_IMA_AMV:
case CODEC_ID_ADPCM_IMA_SMJPEG:
c->status[0].predictor = (int16_t)bytestream_get_le16(&src);
c->status[0].step_index = bytestream_get_le16(&src);
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
src+=4;
while (src < buf + buf_size) {
char hi, lo;
lo = *src & 0x0F;
hi = *src >> 4;
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
FFSWAP(char, hi, lo);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
lo, 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
hi, 3);
src++;
}
break;
case CODEC_ID_ADPCM_CT:
while (src < buf + buf_size) {
uint8_t v = *src++;
*samples++ = adpcm_ct_expand_nibble(&c->status[0 ], v >> 4 );
*samples++ = adpcm_ct_expand_nibble(&c->status[st], v & 0x0F);
}
break;
case CODEC_ID_ADPCM_SBPRO_4:
case CODEC_ID_ADPCM_SBPRO_3:
case CODEC_ID_ADPCM_SBPRO_2:
if (!c->status[0].step_index) {
/* the first byte is a raw sample */
*samples++ = 128 * (*src++ - 0x80);
if (st)
*samples++ = 128 * (*src++ - 0x80);
c->status[0].step_index = 1;
}
if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_4) {
while (src < buf + buf_size) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 4, 4, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
src[0] & 0x0F, 4, 0);
src++;
}
} else if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_3) {
while (src < buf + buf_size && samples + 2 < samples_end) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 5 , 3, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
(src[0] >> 2) & 0x07, 3, 0);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] & 0x03, 2, 0);
src++;
}
} else {
while (src < buf + buf_size && samples + 3 < samples_end) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
src[0] >> 6 , 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
(src[0] >> 4) & 0x03, 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
(src[0] >> 2) & 0x03, 2, 2);
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
src[0] & 0x03, 2, 2);
src++;
}
}
break;
case CODEC_ID_ADPCM_SWF:
{
GetBitContext gb;
const int *table;
int k0, signmask, nb_bits, count;
int size = buf_size*8;
init_get_bits(&gb, buf, size);
//read bits & initial values
nb_bits = get_bits(&gb, 2)+2;
//av_log(NULL,AV_LOG_INFO,"nb_bits: %d\n", nb_bits);
table = swf_index_tables[nb_bits-2];
k0 = 1 << (nb_bits-2);
signmask = 1 << (nb_bits-1);
while (get_bits_count(&gb) <= size - 22*avctx->channels) {
for (i = 0; i < avctx->channels; i++) {
*samples++ = c->status[i].predictor = get_sbits(&gb, 16);
c->status[i].step_index = get_bits(&gb, 6);
}
for (count = 0; get_bits_count(&gb) <= size - nb_bits*avctx->channels && count < 4095; count++) {
int i;
for (i = 0; i < avctx->channels; i++) {
// similar to IMA adpcm
int delta = get_bits(&gb, nb_bits);
int step = ff_adpcm_step_table[c->status[i].step_index];
long vpdiff = 0; // vpdiff = (delta+0.5)*step/4
int k = k0;
do {
if (delta & k)
vpdiff += step;
step >>= 1;
k >>= 1;
} while(k);
vpdiff += step;
if (delta & signmask)
c->status[i].predictor -= vpdiff;
else
c->status[i].predictor += vpdiff;
c->status[i].step_index += table[delta & (~signmask)];
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88);
c->status[i].predictor = av_clip_int16(c->status[i].predictor);
*samples++ = c->status[i].predictor;
if (samples >= samples_end) {
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
return -1;
}
}
}
}
src += buf_size;
break;
}
case CODEC_ID_ADPCM_YAMAHA:
while (src < buf + buf_size) {
uint8_t v = *src++;
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0 ], v & 0x0F);
*samples++ = adpcm_yamaha_expand_nibble(&c->status[st], v >> 4 );
}
break;
case CODEC_ID_ADPCM_THP:
{
int table[2][16];
unsigned int samplecnt;
int prev[2][2];
int ch;
if (buf_size < 80) {
av_log(avctx, AV_LOG_ERROR, "frame too small\n");
return -1;
}
src+=4;
samplecnt = bytestream_get_be32(&src);
for (i = 0; i < 32; i++)
table[0][i] = (int16_t)bytestream_get_be16(&src);
/* Initialize the previous sample. */
for (i = 0; i < 4; i++)
prev[0][i] = (int16_t)bytestream_get_be16(&src);
if (samplecnt >= (samples_end - samples) / (st + 1)) {
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
return -1;
}
for (ch = 0; ch <= st; ch++) {
samples = (unsigned short *) data + ch;
/* Read in every sample for this channel. */
for (i = 0; i < samplecnt / 14; i++) {
int index = (*src >> 4) & 7;
unsigned int exp = 28 - (*src++ & 15);
int factor1 = table[ch][index * 2];
int factor2 = table[ch][index * 2 + 1];
/* Decode 14 samples. */
for (n = 0; n < 14; n++) {
int32_t sampledat;
if(n&1) sampledat= *src++ <<28;
else sampledat= (*src&0xF0)<<24;
sampledat = ((prev[ch][0]*factor1
+ prev[ch][1]*factor2) >> 11) + (sampledat>>exp);
*samples = av_clip_int16(sampledat);
prev[ch][1] = prev[ch][0];
prev[ch][0] = *samples++;
/* In case of stereo, skip one sample, this sample
is for the other channel. */
samples += st;
}
}
}
/* In the previous loop, in case stereo is used, samples is
increased exactly one time too often. */
samples -= st;
break;
}
default:
return -1;
}
*data_size = (uint8_t *)samples - (uint8_t *)data;
return src - buf;
}
#define ADPCM_DECODER(id_, name_, long_name_) \
AVCodec ff_ ## name_ ## _decoder = { \
.name = #name_, \
.type = AVMEDIA_TYPE_AUDIO, \
.id = id_, \
.priv_data_size = sizeof(ADPCMDecodeContext), \
.init = adpcm_decode_init, \
.decode = adpcm_decode_frame, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
}
/* Note: Do not forget to add new entries to the Makefile as well. */
ADPCM_DECODER(CODEC_ID_ADPCM_4XM, adpcm_4xm, "ADPCM 4X Movie");
ADPCM_DECODER(CODEC_ID_ADPCM_CT, adpcm_ct, "ADPCM Creative Technology");
ADPCM_DECODER(CODEC_ID_ADPCM_EA, adpcm_ea, "ADPCM Electronic Arts");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_MAXIS_XA, adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R1, adpcm_ea_r1, "ADPCM Electronic Arts R1");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2, "ADPCM Electronic Arts R2");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3, "ADPCM Electronic Arts R3");
ADPCM_DECODER(CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv, "ADPCM IMA AMV");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_ISS, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws, "ADPCM IMA Westwood");
ADPCM_DECODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
ADPCM_DECODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP");
ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA");
ADPCM_DECODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
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