* ADPCM codecs
* Copyright (c) 2001-2003 The ffmpeg Project
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * 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 "bitstream.h"
+#include "get_bits.h"
+#include "put_bits.h"
#include "bytestream.h"
/**
- * @file adpcm.c
+ * @file
* ADPCM codecs.
* First version by Francois Revol (revol@free.fr)
* Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
* 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:
768, 614, 512, 409, 307, 230, 230, 230
};
-static const int AdaptCoeff1[] = {
- 256, 512, 0, 192, 240, 460, 392
+/** Divided by 4 to fit in 8-bit integers */
+static const uint8_t AdaptCoeff1[] = {
+ 64, 128, 0, 48, 60, 115, 98
};
-static const int AdaptCoeff2[] = {
- 0, -256, 0, 64, 0, -208, -232
+/** Divided by 4 to fit in 8-bit integers */
+static const int8_t AdaptCoeff2[] = {
+ 0, -64, 0, 16, 0, -52, -58
};
/* These are for CD-ROM XA ADPCM */
3, 4, 7, 8, 10, 11, 0, -1, -3, -4
};
-static const int ct_adpcm_table[8] = {
- 0x00E6, 0x00E6, 0x00E6, 0x00E6,
- 0x0133, 0x0199, 0x0200, 0x0266
-};
-
// padded to zero where table size is less then 16
static const int swf_index_tables[4][16] = {
/*2*/ { -1, 2 },
int idelta;
} ADPCMChannelStatus;
+typedef struct TrellisPath {
+ int nibble;
+ int prev;
+} TrellisPath;
+
+typedef struct TrellisNode {
+ uint32_t ssd;
+ int path;
+ int sample1;
+ int sample2;
+ int step;
+} TrellisNode;
+
typedef struct ADPCMContext {
- int channel; /* for stereo MOVs, decode left, then decode right, then tell it's decoded */
ADPCMChannelStatus status[6];
+ TrellisPath *paths;
+ TrellisNode *node_buf;
+ TrellisNode **nodep_buf;
+ uint8_t *trellis_hash;
} ADPCMContext;
+#define FREEZE_INTERVAL 128
+
/* XXX: implement encoding */
-#ifdef CONFIG_ENCODERS
-static int adpcm_encode_init(AVCodecContext *avctx)
+#if CONFIG_ENCODERS
+static av_cold int adpcm_encode_init(AVCodecContext *avctx)
{
+ ADPCMContext *s = avctx->priv_data;
+ uint8_t *extradata;
+ int i;
if (avctx->channels > 2)
return -1; /* only stereo or mono =) */
- switch(avctx->codec->id) {
- case CODEC_ID_ADPCM_IMA_QT:
- av_log(avctx, AV_LOG_ERROR, "ADPCM: codec adpcm_ima_qt unsupported for encoding !\n");
- avctx->frame_size = 64; /* XXX: can multiple of avctx->channels * 64 (left and right blocks are interleaved) */
+
+ if(avctx->trellis && (unsigned)avctx->trellis > 16U){
+ av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
return -1;
- break;
+ }
+
+ if (avctx->trellis) {
+ int frontier = 1 << avctx->trellis;
+ int max_paths = frontier * FREEZE_INTERVAL;
+ FF_ALLOC_OR_GOTO(avctx, s->paths, max_paths * sizeof(*s->paths), error);
+ FF_ALLOC_OR_GOTO(avctx, s->node_buf, 2 * frontier * sizeof(*s->node_buf), error);
+ FF_ALLOC_OR_GOTO(avctx, s->nodep_buf, 2 * frontier * sizeof(*s->nodep_buf), error);
+ FF_ALLOC_OR_GOTO(avctx, s->trellis_hash, 65536 * sizeof(*s->trellis_hash), error);
+ }
+
+ switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_WAV:
avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 / (4 * avctx->channels) + 1; /* each 16 bits sample gives one nibble */
/* and we have 4 bytes per channel overhead */
avctx->block_align = BLKSIZE;
/* seems frame_size isn't taken into account... have to buffer the samples :-( */
break;
+ case CODEC_ID_ADPCM_IMA_QT:
+ avctx->frame_size = 64;
+ avctx->block_align = 34 * avctx->channels;
+ break;
case CODEC_ID_ADPCM_MS:
avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2; /* each 16 bits sample gives one nibble */
/* and we have 7 bytes per channel overhead */
avctx->block_align = BLKSIZE;
+ avctx->extradata_size = 32;
+ extradata = avctx->extradata = av_malloc(avctx->extradata_size);
+ if (!extradata)
+ return AVERROR(ENOMEM);
+ bytestream_put_le16(&extradata, avctx->frame_size);
+ bytestream_put_le16(&extradata, 7); /* wNumCoef */
+ for (i = 0; i < 7; i++) {
+ bytestream_put_le16(&extradata, AdaptCoeff1[i] * 4);
+ bytestream_put_le16(&extradata, AdaptCoeff2[i] * 4);
+ }
break;
case CODEC_ID_ADPCM_YAMAHA:
avctx->frame_size = BLKSIZE * avctx->channels;
avctx->sample_rate != 22050 &&
avctx->sample_rate != 44100) {
av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, 22050 or 44100\n");
- return -1;
+ goto error;
}
avctx->frame_size = 512 * (avctx->sample_rate / 11025);
break;
default:
- return -1;
- break;
+ goto error;
}
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
+error:
+ av_freep(&s->paths);
+ av_freep(&s->node_buf);
+ av_freep(&s->nodep_buf);
+ av_freep(&s->trellis_hash);
+ return -1;
}
-static int adpcm_encode_close(AVCodecContext *avctx)
+static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
+ ADPCMContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
+ av_freep(&s->paths);
+ av_freep(&s->node_buf);
+ av_freep(&s->nodep_buf);
+ av_freep(&s->trellis_hash);
return 0;
}
{
int predictor, nibble, bias;
- predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256;
+ predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
nibble= sample - predictor;
if(nibble>=0) bias= c->idelta/2;
return nibble;
}
-typedef struct TrellisPath {
- int nibble;
- int prev;
-} TrellisPath;
-
-typedef struct TrellisNode {
- uint32_t ssd;
- int path;
- int sample1;
- int sample2;
- int step;
-} TrellisNode;
-
static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
uint8_t *dst, ADPCMChannelStatus *c, int n)
{
-#define FREEZE_INTERVAL 128
//FIXME 6% faster if frontier is a compile-time constant
+ ADPCMContext *s = avctx->priv_data;
const int frontier = 1 << avctx->trellis;
const int stride = avctx->channels;
const int version = avctx->codec->id;
- const int max_paths = frontier*FREEZE_INTERVAL;
- TrellisPath paths[max_paths], *p;
- TrellisNode node_buf[2][frontier];
- TrellisNode *nodep_buf[2][frontier];
- TrellisNode **nodes = nodep_buf[0]; // nodes[] is always sorted by .ssd
- TrellisNode **nodes_next = nodep_buf[1];
- int pathn = 0, froze = -1, i, j, k;
-
- assert(!(max_paths&(max_paths-1)));
-
- memset(nodep_buf, 0, sizeof(nodep_buf));
- nodes[0] = &node_buf[1][0];
+ TrellisPath *paths = s->paths, *p;
+ TrellisNode *node_buf = s->node_buf;
+ TrellisNode **nodep_buf = s->nodep_buf;
+ TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
+ TrellisNode **nodes_next = nodep_buf + frontier;
+ int pathn = 0, froze = -1, i, j, k, generation = 0;
+ uint8_t *hash = s->trellis_hash;
+ memset(hash, 0xff, 65536 * sizeof(*hash));
+
+ memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
+ nodes[0] = node_buf + frontier;
nodes[0]->ssd = 0;
nodes[0]->path = 0;
nodes[0]->step = c->step_index;
nodes[0]->sample1 = c->sample1;
nodes[0]->sample2 = c->sample2;
- if((version == CODEC_ID_ADPCM_IMA_WAV) || (version == CODEC_ID_ADPCM_SWF))
+ if((version == CODEC_ID_ADPCM_IMA_WAV) || (version == CODEC_ID_ADPCM_IMA_QT) || (version == CODEC_ID_ADPCM_SWF))
nodes[0]->sample1 = c->prev_sample;
if(version == CODEC_ID_ADPCM_MS)
nodes[0]->step = c->idelta;
}
for(i=0; i<n; i++) {
- TrellisNode *t = node_buf[i&1];
+ TrellisNode *t = node_buf + frontier*(i&1);
TrellisNode **u;
int sample = samples[i*stride];
+ int heap_pos = 0;
memset(nodes_next, 0, frontier*sizeof(TrellisNode*));
for(j=0; j<frontier && nodes[j]; j++) {
- // higher j have higher ssd already, so they're unlikely to use a suboptimal next sample too
+ // higher j have higher ssd already, so they're likely to yield a suboptimal next sample too
const int range = (j < frontier/2) ? 1 : 0;
const int step = nodes[j]->step;
int nidx;
if(version == CODEC_ID_ADPCM_MS) {
- const int predictor = ((nodes[j]->sample1 * c->coeff1) + (nodes[j]->sample2 * c->coeff2)) / 256;
+ const int predictor = ((nodes[j]->sample1 * c->coeff1) + (nodes[j]->sample2 * c->coeff2)) / 64;
const int div = (sample - predictor) / step;
const int nmin = av_clip(div-range, -8, 6);
const int nmax = av_clip(div+range, -7, 7);
#define STORE_NODE(NAME, STEP_INDEX)\
int d;\
uint32_t ssd;\
+ int pos;\
+ TrellisNode *u;\
+ uint8_t *h;\
dec_sample = av_clip_int16(dec_sample);\
d = sample - dec_sample;\
ssd = nodes[j]->ssd + d*d;\
- if(nodes_next[frontier-1] && ssd >= nodes_next[frontier-1]->ssd)\
- continue;\
+ /* Check for wraparound, skip such samples completely. \
+ * Note, changing ssd to a 64 bit variable would be \
+ * simpler, avoiding this check, but it's slower on \
+ * x86 32 bit at the moment. */\
+ if (ssd < nodes[j]->ssd)\
+ goto next_##NAME;\
/* Collapse any two states with the same previous sample value. \
* One could also distinguish states by step and by 2nd to last
- * sample, but the effects of that are negligible. */\
- for(k=0; k<frontier && nodes_next[k]; k++) {\
- if(dec_sample == nodes_next[k]->sample1) {\
- assert(ssd >= nodes_next[k]->ssd);\
+ * sample, but the effects of that are negligible.
+ * Since nodes in the previous generation are iterated
+ * through a heap, they're roughly ordered from better to
+ * worse, but not strictly ordered. Therefore, an earlier
+ * node with the same sample value is better in most cases
+ * (and thus the current is skipped), but not strictly
+ * in all cases. Only skipping samples where ssd >=
+ * ssd of the earlier node with the same sample gives
+ * slightly worse quality, though, for some reason. */ \
+ h = &hash[(uint16_t) dec_sample];\
+ if (*h == generation)\
+ goto next_##NAME;\
+ if (heap_pos < frontier) {\
+ pos = heap_pos++;\
+ } else {\
+ /* Try to replace one of the leaf nodes with the new \
+ * one, but try a different slot each time. */\
+ pos = (frontier >> 1) + (heap_pos & ((frontier >> 1) - 1));\
+ if (ssd > nodes_next[pos]->ssd)\
goto next_##NAME;\
- }\
+ heap_pos++;\
+ }\
+ *h = generation;\
+ u = nodes_next[pos];\
+ if(!u) {\
+ assert(pathn < FREEZE_INTERVAL<<avctx->trellis);\
+ u = t++;\
+ nodes_next[pos] = u;\
+ u->path = pathn++;\
}\
- for(k=0; k<frontier; k++) {\
- if(!nodes_next[k] || ssd < nodes_next[k]->ssd) {\
- TrellisNode *u = nodes_next[frontier-1];\
- if(!u) {\
- assert(pathn < max_paths);\
- u = t++;\
- u->path = pathn++;\
- }\
- u->ssd = ssd;\
- u->step = STEP_INDEX;\
- u->sample2 = nodes[j]->sample1;\
- u->sample1 = dec_sample;\
- paths[u->path].nibble = nibble;\
- paths[u->path].prev = nodes[j]->path;\
- memmove(&nodes_next[k+1], &nodes_next[k], (frontier-k-1)*sizeof(TrellisNode*));\
- nodes_next[k] = u;\
+ u->ssd = ssd;\
+ u->step = STEP_INDEX;\
+ u->sample2 = nodes[j]->sample1;\
+ u->sample1 = dec_sample;\
+ paths[u->path].nibble = nibble;\
+ paths[u->path].prev = nodes[j]->path;\
+ /* Sift the newly inserted node up in the heap to \
+ * restore the heap property. */\
+ while (pos > 0) {\
+ int parent = (pos - 1) >> 1;\
+ if (nodes_next[parent]->ssd <= ssd)\
break;\
- }\
+ FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
+ pos = parent;\
}\
next_##NAME:;
STORE_NODE(ms, FFMAX(16, (AdaptationTable[nibble] * step) >> 8));
}
- } else if((version == CODEC_ID_ADPCM_IMA_WAV)|| (version == CODEC_ID_ADPCM_SWF)) {
+ } else if((version == CODEC_ID_ADPCM_IMA_WAV)|| (version == CODEC_ID_ADPCM_IMA_QT)|| (version == CODEC_ID_ADPCM_SWF)) {
#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
const int predictor = nodes[j]->sample1;\
const int div = (sample - predictor) * 4 / STEP_TABLE;\
nodes = nodes_next;
nodes_next = u;
+ generation++;
+ if (generation == 255) {
+ memset(hash, 0xff, 65536 * sizeof(*hash));
+ generation = 0;
+ }
+
// prevent overflow
if(nodes[0]->ssd > (1<<28)) {
for(j=1; j<frontier && nodes[j]; j++)
short *samples;
unsigned char *dst;
ADPCMContext *c = avctx->priv_data;
+ uint8_t *buf;
dst = frame;
samples = (short *)data;
/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
switch(avctx->codec->id) {
- case CODEC_ID_ADPCM_IMA_QT: /* XXX: can't test until we get .mov writer */
- break;
case CODEC_ID_ADPCM_IMA_WAV:
n = avctx->frame_size / 8;
c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
*dst++ = 0; /* unknown */
samples++;
if (avctx->channels == 2) {
- c->status[1].prev_sample = (signed short)samples[1];
+ c->status[1].prev_sample = (signed short)samples[0];
/* c->status[1].step_index = 0; */
bytestream_put_le16(&dst, c->status[1].prev_sample);
*dst++ = (unsigned char)c->status[1].step_index;
/* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */
if(avctx->trellis > 0) {
- uint8_t buf[2][n*8];
- adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n*8);
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n*8, error);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n*8);
if(avctx->channels == 2)
- adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n*8);
+ adpcm_compress_trellis(avctx, samples+1, buf + n*8, &c->status[1], n*8);
for(i=0; i<n; i++) {
- *dst++ = buf[0][8*i+0] | (buf[0][8*i+1] << 4);
- *dst++ = buf[0][8*i+2] | (buf[0][8*i+3] << 4);
- *dst++ = buf[0][8*i+4] | (buf[0][8*i+5] << 4);
- *dst++ = buf[0][8*i+6] | (buf[0][8*i+7] << 4);
+ *dst++ = buf[8*i+0] | (buf[8*i+1] << 4);
+ *dst++ = buf[8*i+2] | (buf[8*i+3] << 4);
+ *dst++ = buf[8*i+4] | (buf[8*i+5] << 4);
+ *dst++ = buf[8*i+6] | (buf[8*i+7] << 4);
if (avctx->channels == 2) {
- *dst++ = buf[1][8*i+0] | (buf[1][8*i+1] << 4);
- *dst++ = buf[1][8*i+2] | (buf[1][8*i+3] << 4);
- *dst++ = buf[1][8*i+4] | (buf[1][8*i+5] << 4);
- *dst++ = buf[1][8*i+6] | (buf[1][8*i+7] << 4);
+ uint8_t *buf1 = buf + n*8;
+ *dst++ = buf1[8*i+0] | (buf1[8*i+1] << 4);
+ *dst++ = buf1[8*i+2] | (buf1[8*i+3] << 4);
+ *dst++ = buf1[8*i+4] | (buf1[8*i+5] << 4);
+ *dst++ = buf1[8*i+6] | (buf1[8*i+7] << 4);
}
}
+ av_free(buf);
} else
for (; n>0; n--) {
*dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
samples += 8 * avctx->channels;
}
break;
+ case CODEC_ID_ADPCM_IMA_QT:
+ {
+ int ch, i;
+ PutBitContext pb;
+ init_put_bits(&pb, dst, buf_size*8);
+
+ for(ch=0; ch<avctx->channels; ch++){
+ put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
+ put_bits(&pb, 7, c->status[ch].step_index);
+ if(avctx->trellis > 0) {
+ uint8_t buf[64];
+ adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
+ for(i=0; i<64; i++)
+ put_bits(&pb, 4, buf[i^1]);
+ c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F;
+ } else {
+ for (i=0; i<64; i+=2){
+ int t1, t2;
+ t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]);
+ t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]);
+ put_bits(&pb, 4, t2);
+ put_bits(&pb, 4, t1);
+ }
+ c->status[ch].prev_sample &= ~0x7F;
+ }
+ }
+
+ flush_put_bits(&pb);
+ dst += put_bits_count(&pb)>>3;
+ break;
+ }
case CODEC_ID_ADPCM_SWF:
{
int i;
//Init the encoder state
for(i=0; i<avctx->channels; i++){
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits
- put_bits(&pb, 16, samples[i] & 0xFFFF);
+ put_sbits(&pb, 16, samples[i]);
put_bits(&pb, 6, c->status[i].step_index);
c->status[i].prev_sample = (signed short)samples[i];
}
if(avctx->trellis > 0) {
- uint8_t buf[2][n];
- adpcm_compress_trellis(avctx, samples+2, buf[0], &c->status[0], n);
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
+ adpcm_compress_trellis(avctx, samples+2, buf, &c->status[0], n);
if (avctx->channels == 2)
- adpcm_compress_trellis(avctx, samples+3, buf[1], &c->status[1], n);
+ adpcm_compress_trellis(avctx, samples+3, buf+n, &c->status[1], n);
for(i=0; i<n; i++) {
- put_bits(&pb, 4, buf[0][i]);
+ put_bits(&pb, 4, buf[i]);
if (avctx->channels == 2)
- put_bits(&pb, 4, buf[1][i]);
+ put_bits(&pb, 4, buf[n+i]);
}
+ av_free(buf);
} else {
for (i=1; i<avctx->frame_size; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i]));
bytestream_put_le16(&dst, c->status[i].idelta);
}
+ for(i=0; i<avctx->channels; i++){
+ c->status[i].sample2= *samples++;
+ }
for(i=0; i<avctx->channels; i++){
c->status[i].sample1= *samples++;
bytestream_put_le16(&dst, c->status[i].sample1);
}
- for(i=0; i<avctx->channels; i++){
- c->status[i].sample2= *samples++;
-
+ for(i=0; i<avctx->channels; i++)
bytestream_put_le16(&dst, c->status[i].sample2);
- }
if(avctx->trellis > 0) {
int n = avctx->block_align - 7*avctx->channels;
- uint8_t buf[2][n];
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
if(avctx->channels == 1) {
- n *= 2;
- adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
for(i=0; i<n; i+=2)
- *dst++ = (buf[0][i] << 4) | buf[0][i+1];
+ *dst++ = (buf[i] << 4) | buf[i+1];
} else {
- adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
- adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
for(i=0; i<n; i++)
- *dst++ = (buf[0][i] << 4) | buf[1][i];
+ *dst++ = (buf[i] << 4) | buf[n+i];
}
+ av_free(buf);
} else
for(i=7*avctx->channels; i<avctx->block_align; i++) {
int nibble;
case CODEC_ID_ADPCM_YAMAHA:
n = avctx->frame_size / 2;
if(avctx->trellis > 0) {
- uint8_t buf[2][n*2];
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n*2, error);
n *= 2;
if(avctx->channels == 1) {
- adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
for(i=0; i<n; i+=2)
- *dst++ = buf[0][i] | (buf[0][i+1] << 4);
+ *dst++ = buf[i] | (buf[i+1] << 4);
} else {
- adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n);
- adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
for(i=0; i<n; i++)
- *dst++ = buf[0][i] | (buf[1][i] << 4);
+ *dst++ = buf[i] | (buf[n+i] << 4);
}
+ av_free(buf);
} else
- for (; n>0; n--) {
- for(i = 0; i < avctx->channels; i++) {
+ for (n *= avctx->channels; n>0; n--) {
int nibble;
- nibble = adpcm_yamaha_compress_sample(&c->status[i], samples[i]);
- nibble |= adpcm_yamaha_compress_sample(&c->status[i], samples[i+avctx->channels]) << 4;
+ nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
+ nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
*dst++ = nibble;
}
- samples += 2 * avctx->channels;
- }
break;
default:
+ error:
return -1;
}
return dst - frame;
}
#endif //CONFIG_ENCODERS
-static int adpcm_decode_init(AVCodecContext * avctx)
+static av_cold int adpcm_decode_init(AVCodecContext * avctx)
{
ADPCMContext *c = avctx->priv_data;
unsigned int max_channels = 2;
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);
default:
break;
}
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
{
int predictor;
- predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256;
+ predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
c->sample2 = c->sample1;
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 = (ct_adpcm_table[nibble & 7] * c->step) >> 8;
+ new_step = (AdaptationTable[nibble & 7] * c->step) >> 8;
c->step = av_clip(new_step, 511, 32767);
return (short)c->predictor;
#define DK3_GET_NEXT_NIBBLE() \
if (decode_top_nibble_next) \
{ \
- nibble = (last_byte >> 4) & 0x0F; \
+ nibble = last_byte >> 4; \
decode_top_nibble_next = 0; \
} \
else \
static int adpcm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
- uint8_t *buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
ADPCMContext *c = avctx->priv_data;
ADPCMChannelStatus *cs;
int n, m, channel, i;
int block_predictor[2];
short *samples;
short *samples_end;
- uint8_t *src;
+ const uint8_t *src;
int st; /* stereo */
/* DK3 ADPCM accounting variables */
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;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_QT:
- n = (buf_size - 2);/* >> 2*avctx->channels;*/
- channel = c->channel;
- cs = &(c->status[channel]);
- /* (pppppp) (piiiiiii) */
+ n = buf_size - 2*avctx->channels;
+ for (channel = 0; channel < avctx->channels; channel++) {
+ cs = &(c->status[channel]);
+ /* (pppppp) (piiiiiii) */
- /* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */
- cs->predictor = (*src++) << 8;
- cs->predictor |= (*src & 0x80);
- cs->predictor &= 0xFF80;
+ /* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */
+ cs->predictor = (*src++) << 8;
+ cs->predictor |= (*src & 0x80);
+ cs->predictor &= 0xFF80;
- /* sign extension */
- if(cs->predictor & 0x8000)
- cs->predictor -= 0x10000;
+ /* sign extension */
+ if(cs->predictor & 0x8000)
+ cs->predictor -= 0x10000;
- cs->predictor = av_clip_int16(cs->predictor);
+ cs->predictor = av_clip_int16(cs->predictor);
- cs->step_index = (*src++) & 0x7F;
+ cs->step_index = (*src++) & 0x7F;
- if (cs->step_index > 88){
- av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
- cs->step_index = 88;
- }
-
- cs->step = step_table[cs->step_index];
+ if (cs->step_index > 88){
+ av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
+ cs->step_index = 88;
+ }
- if (st && channel)
- samples++;
+ cs->step = step_table[cs->step_index];
- for(m=32; n>0 && m>0; n--, m--) { /* in QuickTime, IMA is encoded by chuncks of 34 bytes (=64 samples) */
- *samples = adpcm_ima_expand_nibble(cs, src[0] & 0x0F, 3);
- samples += avctx->channels;
- *samples = adpcm_ima_expand_nibble(cs, (src[0] >> 4) & 0x0F, 3);
- samples += avctx->channels;
- src ++;
- }
+ samples = (short*)data + channel;
- if(st) { /* handle stereo interlacing */
- c->channel = (channel + 1) % 2; /* we get one packet for left, then one for right data */
- if(channel == 1) { /* wait for the other packet before outputing anything */
- return src - buf;
+ for(m=32; n>0 && m>0; n--, m--) { /* in QuickTime, IMA is encoded by chuncks of 34 bytes (=64 samples) */
+ *samples = adpcm_ima_expand_nibble(cs, src[0] & 0x0F, 3);
+ samples += avctx->channels;
+ *samples = adpcm_ima_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)
for(i=0; i<avctx->channels; i++){
cs = &(c->status[i]);
- cs->predictor = (int16_t)(src[0] + (src[1]<<8));
- src+=2;
-
- // XXX: is this correct ??: *samples++ = cs->predictor;
+ cs->predictor = *samples++ = (int16_t)bytestream_get_le16(&src);
cs->step_index = *src++;
if (cs->step_index > 88){
break;
case CODEC_ID_ADPCM_4XM:
cs = &(c->status[0]);
- c->status[0].predictor= (int16_t)(src[0] + (src[1]<<8)); src+=2;
+ c->status[0].predictor= (int16_t)bytestream_get_le16(&src);
if(st){
- c->status[1].predictor= (int16_t)(src[0] + (src[1]<<8)); src+=2;
+ c->status[1].predictor= (int16_t)bytestream_get_le16(&src);
}
- c->status[0].step_index= (int16_t)(src[0] + (src[1]<<8)); src+=2;
+ c->status[0].step_index= (int16_t)bytestream_get_le16(&src);
if(st){
- c->status[1].step_index= (int16_t)(src[0] + (src[1]<<8)); src+=2;
+ 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;
n = buf_size - 7 * avctx->channels;
if (n < 0)
return -1;
- block_predictor[0] = av_clip(*src++, 0, 7);
+ block_predictor[0] = av_clip(*src++, 0, 6);
block_predictor[1] = 0;
if (st)
- block_predictor[1] = av_clip(*src++, 0, 7);
- c->status[0].idelta = (int16_t)((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
- src+=2;
+ 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)((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
- src+=2;
+ c->status[1].idelta = (int16_t)bytestream_get_le16(&src);
}
c->status[0].coeff1 = AdaptCoeff1[block_predictor[0]];
c->status[0].coeff2 = AdaptCoeff2[block_predictor[0]];
c->status[1].coeff1 = AdaptCoeff1[block_predictor[1]];
c->status[1].coeff2 = AdaptCoeff2[block_predictor[1]];
- c->status[0].sample1 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
- src+=2;
- if (st) c->status[1].sample1 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
- if (st) src+=2;
- c->status[0].sample2 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
- src+=2;
- if (st) c->status[1].sample2 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00));
- if (st) src+=2;
+ 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].sample1;
- if (st) *samples++ = c->status[1].sample1;
*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) & 0x0F);
+ *samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 );
*samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F);
src ++;
}
if (avctx->block_align != 0 && buf_size > avctx->block_align)
buf_size = avctx->block_align;
- c->status[0].predictor = (int16_t)(src[0] | (src[1] << 8));
- c->status[0].step_index = src[2];
- src += 4;
+ 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)(src[0] | (src[1] << 8));
- c->status[1].step_index = src[2];
- src += 4;
+ 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) {
/* take care of the top nibble (always left or mono channel) */
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
- (src[0] >> 4) & 0x0F, 3);
+ src[0] >> 4, 3);
/* take care of the bottom nibble, which is right sample for
* stereo, or another mono sample */
if(buf_size + 16 > (samples_end - samples)*3/8)
return -1;
- c->status[0].predictor = (int16_t)(src[10] | (src[11] << 8));
- c->status[1].predictor = (int16_t)(src[12] | (src[13] << 8));
+ 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 */
*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) {
+
+ if (st) {
+ *samples++ = adpcm_ima_expand_nibble(&c->status[0],
+ src[0] >> 4 , 3);
+ *samples++ = adpcm_ima_expand_nibble(&c->status[1],
+ src[0] & 0x0F, 3);
+ } else {
+ *samples++ = adpcm_ima_expand_nibble(&c->status[0],
+ src[0] & 0x0F, 3);
+ *samples++ = adpcm_ima_expand_nibble(&c->status[0],
+ src[0] >> 4 , 3);
+ }
+
+ src++;
+ }
+ break;
case CODEC_ID_ADPCM_IMA_WS:
/* no per-block initialization; just start decoding the data */
while (src < buf + buf_size) {
if (st) {
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
- (src[0] >> 4) & 0x0F, 3);
+ src[0] >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[1],
src[0] & 0x0F, 3);
} else {
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
- (src[0] >> 4) & 0x0F, 3);
+ src[0] >> 4 , 3);
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
src[0] & 0x0F, 3);
}
}
break;
case CODEC_ID_ADPCM_XA:
- c->status[0].sample1 = c->status[0].sample2 =
- c->status[1].sample1 = c->status[1].sample2 = 0;
while (buf_size >= 128) {
xa_decode(samples, src, &c->status[0], &c->status[1],
avctx->channels);
}
break;
case CODEC_ID_ADPCM_EA:
- samples_in_chunk = AV_RL32(src);
- if (samples_in_chunk >= ((buf_size - 12) * 2)) {
+ if (buf_size < 4 || AV_RL32(src) >= ((buf_size - 12) * 2)) {
src += buf_size;
break;
}
+ samples_in_chunk = AV_RL32(src);
src += 4;
- current_left_sample = (int16_t)AV_RL16(src);
- src += 2;
- previous_left_sample = (int16_t)AV_RL16(src);
- src += 2;
- current_right_sample = (int16_t)AV_RL16(src);
- src += 2;
- previous_right_sample = (int16_t)AV_RL16(src);
- src += 2;
+ 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) & 0x0F];
- coeff2l = ea_adpcm_table[((*src >> 4) & 0x0F) + 4];
+ 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) & 0x0F) + 8;
+ shift_left = (*src >> 4 ) + 8;
shift_right = (*src & 0x0F) + 8;
src++;
for (count2 = 0; count2 < 28; count2++) {
- next_left_sample = (((*src & 0xF0) << 24) >> shift_left);
- next_right_sample = (((*src & 0x0F) << 28) >> shift_right);
+ 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 +
*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:
uint8_t shift;
unsigned int channel;
uint16_t *samplesC;
- uint8_t *srcC;
+ 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;
}
for (channel=0; channel<avctx->channels; channel++) {
- srcC = src + (big_endian ? bytestream_get_be32(&src)
- : bytestream_get_le32(&src))
- + (avctx->channels-channel-1) * 4;
+ 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) {
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);
samplesC += avctx->channels;
}
} else {
- coeff1 = ea_adpcm_table[ (*srcC>>4) & 0x0F ];
- coeff2 = ea_adpcm_table[((*srcC>>4) & 0x0F) + 4];
+ 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 = ((*srcC++ & 0x0F) << 28) >> shift;
+ next_sample = (int32_t)((*srcC++ & 0x0F) << 28) >> shift;
else
- next_sample = ((*srcC & 0xF0) << 24) >> shift;
+ next_sample = (int32_t)((*srcC & 0xF0) << 24) >> shift;
next_sample += (current_sample * coeff1) +
(previous_sample * coeff2);
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);
while (src < buf + buf_size) {
char hi, lo;
lo = *src & 0x0F;
- hi = (*src >> 4) & 0x0F;
+ hi = *src >> 4;
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
FFSWAP(char, hi, lo);
while (src < buf + buf_size) {
if (st) {
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
- (src[0] >> 4) & 0x0F);
+ src[0] >> 4);
*samples++ = adpcm_ct_expand_nibble(&c->status[1],
src[0] & 0x0F);
} else {
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
- (src[0] >> 4) & 0x0F);
+ src[0] >> 4);
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
src[0] & 0x0F);
}
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) & 0x0F, 4, 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) & 0x07, 3, 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],
} else {
while (src < buf + buf_size && samples + 3 < samples_end) {
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
- (src[0] >> 6) & 0x03, 2, 2);
+ 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],
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
src[0] & 0x0F);
*samples++ = adpcm_yamaha_expand_nibble(&c->status[1],
- (src[0] >> 4) & 0x0F);
+ src[0] >> 4 );
} else {
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
src[0] & 0x0F);
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
- (src[0] >> 4) & 0x0F);
+ src[0] >> 4 );
}
src++;
}
-#ifdef CONFIG_ENCODERS
-#define ADPCM_ENCODER(id,name) \
-AVCodec name ## _encoder = { \
+#if CONFIG_ENCODERS
+#define ADPCM_ENCODER(id,name,long_name_) \
+AVCodec ff_ ## name ## _encoder = { \
#name, \
- CODEC_TYPE_AUDIO, \
+ AVMEDIA_TYPE_AUDIO, \
id, \
sizeof(ADPCMContext), \
adpcm_encode_init, \
adpcm_encode_frame, \
adpcm_encode_close, \
NULL, \
-};
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \
+ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
+}
#else
-#define ADPCM_ENCODER(id,name)
+#define ADPCM_ENCODER(id,name,long_name_)
#endif
-#ifdef CONFIG_DECODERS
-#define ADPCM_DECODER(id,name) \
-AVCodec name ## _decoder = { \
+#if CONFIG_DECODERS
+#define ADPCM_DECODER(id,name,long_name_) \
+AVCodec ff_ ## name ## _decoder = { \
#name, \
- CODEC_TYPE_AUDIO, \
+ AVMEDIA_TYPE_AUDIO, \
id, \
sizeof(ADPCMContext), \
adpcm_decode_init, \
NULL, \
NULL, \
adpcm_decode_frame, \
-};
+ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
+}
#else
-#define ADPCM_DECODER(id,name)
+#define ADPCM_DECODER(id,name,long_name_)
#endif
-#define ADPCM_CODEC(id, name) \
-ADPCM_ENCODER(id,name) ADPCM_DECODER(id,name)
-
-ADPCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm);
-ADPCM_CODEC(CODEC_ID_ADPCM_CT, adpcm_ct);
-ADPCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav);
-ADPCM_CODEC(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws);
-ADPCM_CODEC(CODEC_ID_ADPCM_MS, adpcm_ms);
-ADPCM_CODEC(CODEC_ID_ADPCM_EA_R1, adpcm_ea_r1);
-ADPCM_CODEC(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2);
-ADPCM_CODEC(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3);
-ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4);
-ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3);
-ADPCM_CODEC(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2);
-ADPCM_CODEC(CODEC_ID_ADPCM_SWF, adpcm_swf);
-ADPCM_CODEC(CODEC_ID_ADPCM_THP, adpcm_thp);
-ADPCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa);
-ADPCM_CODEC(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha);
-
-#undef ADPCM_CODEC
+#define ADPCM_CODEC(id,name,long_name_) \
+ ADPCM_ENCODER(id,name,long_name_); ADPCM_DECODER(id,name,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_CODEC (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_CODEC (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_CODEC (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_CODEC (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_CODEC (CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
projects / ffmpeg / blobdiff