60 #define FREEZE_INTERVAL 128
80 int frontier = 1 << avctx->
trellis;
83 max_paths *
sizeof(*s->
paths), error);
85 2 * frontier *
sizeof(*s->
node_buf), error);
87 2 * frontier *
sizeof(*s->
nodep_buf), error);
118 bytestream_put_le16(&extradata, avctx->
frame_size);
119 bytestream_put_le16(&extradata, 7);
120 for (i = 0; i < 7; i++) {
170 int nibble =
FFMIN(7, abs(delta) * 4 /
184 int diff = step >> 3;
192 for (mask = 4;
mask;) {
227 nibble = (nibble + bias) / c->
idelta;
228 nibble = av_clip(nibble, -8, 7) & 0x0F;
230 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->
idelta;
233 c->
sample1 = av_clip_int16(predictor);
254 nibble =
FFMIN(7, abs(delta) * 4 / c->
step) + (delta < 0) * 8;
259 c->
step = av_clip(c->
step, 127, 24567);
265 const int16_t *samples,
uint8_t *dst,
270 const int frontier = 1 << avctx->
trellis;
277 int pathn = 0, froze = -1, i, j, k, generation = 0;
279 memset(hash, 0xff, 65536 *
sizeof(*hash));
281 memset(nodep_buf, 0, 2 * frontier *
sizeof(*nodep_buf));
282 nodes[0] = node_buf + frontier;
296 nodes[0]->
step = 127;
304 for (i = 0; i < n; i++) {
309 memset(nodes_next, 0, frontier *
sizeof(
TrellisNode*));
310 for (j = 0; j < frontier && nodes[j]; j++) {
313 const int range = (j < frontier / 2) ? 1 : 0;
314 const int step = nodes[j]->step;
318 (nodes[j]->sample2 * c->
coeff2)) / 64;
319 const int div = (sample -
predictor) / step;
320 const int nmin = av_clip(div-range, -8, 6);
321 const int nmax = av_clip(div+range, -7, 7);
322 for (nidx = nmin; nidx <= nmax; nidx++) {
323 const int nibble = nidx & 0xf;
324 int dec_sample = predictor + nidx *
step;
325 #define STORE_NODE(NAME, STEP_INDEX)\
331 dec_sample = av_clip_int16(dec_sample);\
332 d = sample - dec_sample;\
333 ssd = nodes[j]->ssd + d*d;\
338 if (ssd < nodes[j]->ssd)\
351 h = &hash[(uint16_t) dec_sample];\
352 if (*h == generation)\
354 if (heap_pos < frontier) {\
359 pos = (frontier >> 1) +\
360 (heap_pos & ((frontier >> 1) - 1));\
361 if (ssd > nodes_next[pos]->ssd)\
366 u = nodes_next[pos];\
368 assert(pathn < FREEZE_INTERVAL << avctx->trellis);\
370 nodes_next[pos] = u;\
374 u->step = STEP_INDEX;\
375 u->sample2 = nodes[j]->sample1;\
376 u->sample1 = dec_sample;\
377 paths[u->path].nibble = nibble;\
378 paths[u->path].prev = nodes[j]->path;\
382 int parent = (pos - 1) >> 1;\
383 if (nodes_next[parent]->ssd <= ssd)\
385 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
395 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
396 const int predictor = nodes[j]->sample1;\
397 const int div = (sample - predictor) * 4 / STEP_TABLE;\
398 int nmin = av_clip(div - range, -7, 6);\
399 int nmax = av_clip(div + range, -6, 7);\
404 for (nidx = nmin; nidx <= nmax; nidx++) {\
405 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
406 int dec_sample = predictor +\
408 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
409 STORE_NODE(NAME, STEP_INDEX);\
427 if (generation == 255) {
428 memset(hash, 0xff, 65536 *
sizeof(*hash));
433 if (nodes[0]->ssd > (1 << 28)) {
434 for (j = 1; j < frontier && nodes[j]; j++)
435 nodes[j]->ssd -= nodes[0]->ssd;
441 p = &paths[nodes[0]->path];
442 for (k = i; k > froze; k--) {
451 memset(nodes + 1, 0, (frontier - 1) *
sizeof(
TrellisNode*));
455 p = &paths[nodes[0]->
path];
456 for (i = n - 1; i > froze; i--) {
462 c->
sample1 = nodes[0]->sample1;
463 c->
sample2 = nodes[0]->sample2;
465 c->
step = nodes[0]->step;
466 c->
idelta = nodes[0]->step;
470 const AVFrame *frame,
int *got_packet_ptr)
472 int n, i, ch, st, pkt_size, ret;
473 const int16_t *samples;
479 samples = (
const int16_t *)frame->
data[0];
500 for (ch = 0; ch < avctx->
channels; ch++) {
513 for (ch = 0; ch < avctx->
channels; ch++) {
515 buf + ch * blocks * 8, &c->
status[ch],
518 for (i = 0; i < blocks; i++) {
519 for (ch = 0; ch < avctx->
channels; ch++) {
520 uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
521 for (j = 0; j < 8; j += 2)
522 *dst++ = buf1[j] | (buf1[j + 1] << 4);
527 for (i = 0; i < blocks; i++) {
528 for (ch = 0; ch < avctx->
channels; ch++) {
530 const int16_t *smp = &samples_p[ch][1 + i * 8];
531 for (j = 0; j < 8; j += 2) {
546 for (ch = 0; ch < avctx->
channels; ch++) {
554 for (i = 0; i < 64; i++)
558 for (i = 0; i < 64; i += 2) {
582 for (i = 0; i < avctx->
channels; i++) {
596 buf + n, &c->
status[1], n,
598 for (i = 0; i < n; i++) {
610 samples[2 * i + 1]));
617 for (i = 0; i < avctx->
channels; i++) {
623 for (i = 0; i < avctx->
channels; i++) {
628 for (i = 0; i < avctx->
channels; i++)
634 for (i = 0; i < avctx->
channels; i++)
643 for (i = 0; i < n; i += 2)
644 *dst++ = (buf[i] << 4) | buf[i + 1];
650 for (i = 0; i < n; i++)
651 *dst++ = (buf[i] << 4) | buf[n + i];
655 for (i = 7 * avctx->
channels; i < avctx->block_align; i++) {
671 for (i = 0; i < n; i += 2)
672 *dst++ = buf[i] | (buf[i + 1] << 4);
678 for (i = 0; i < n; i++)
679 *dst++ = buf[i] | (buf[n + i] << 4);
683 for (n *= avctx->
channels; n > 0; n--) {
694 avpkt->
size = pkt_size;
709 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
710 AVCodec ff_ ## name_ ## _encoder = { \
712 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
713 .type = AVMEDIA_TYPE_AUDIO, \
715 .priv_data_size = sizeof(ADPCMEncodeContext), \
716 .init = adpcm_encode_init, \
717 .encode2 = adpcm_encode_frame, \
718 .close = adpcm_encode_close, \
719 .sample_fmts = sample_fmts_, \
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
This structure describes decoded (raw) audio or video data.
static void put_sbits(PutBitContext *pb, int n, int32_t value)
static uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c, int16_t sample)
static int hash(int head, const int add)
Hash function adding character.
static uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c, int16_t sample)
uint8_t ** extended_data
pointers to the data planes/channels.
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
void av_freep(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
static uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, int16_t sample)
const uint8_t ff_adpcm_AdaptCoeff1[]
Divided by 4 to fit in 8-bit integers.
bitstream reader API header.
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
static void predictor(uint8_t *src, int size)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
ADPCM encoder/decoder common header.
static const uint16_t mask[17]
#define STORE_NODE(NAME, STEP_INDEX)
const int16_t ff_adpcm_step_table[89]
This is the step table.
static void put_bits(PutBitContext *s, int n, unsigned int value)
Write up to 31 bits into a bitstream.
Libavcodec external API header.
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
const int8_t ff_adpcm_index_table[16]
static uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c, int16_t sample)
const int8_t ff_adpcm_AdaptCoeff2[]
Divided by 4 to fit in 8-bit integers.
static void adpcm_compress_trellis(AVCodecContext *avctx, const int16_t *samples, uint8_t *dst, ADPCMChannelStatus *c, int n, int stride)
void * av_malloc(size_t size) av_malloc_attrib 1(1)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
int ff_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
if(ac->has_optimized_func)
int frame_size
Number of samples per channel in an audio frame.
const int16_t ff_adpcm_AdaptationTable[]
AVSampleFormat
Audio Sample Formats.
int sample_rate
samples per second
main external API structure.
#define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_)
const int8_t ff_adpcm_yamaha_difflookup[]
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
const int16_t ff_adpcm_yamaha_indexscale[]
#define FF_ALLOC_OR_GOTO(ctx, p, size, label)
#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
const struct AVCodec * codec
int trellis
trellis RD quantization
int channels
number of audio channels
static enum AVSampleFormat sample_fmts[]
ADPCMChannelStatus status[6]
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static enum AVSampleFormat sample_fmts_p[]