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25 #include "config_components.h"
43 #define CASE_0(codec_id, ...)
44 #define CASE_1(codec_id, ...) \
48 #define CASE_2(enabled, codec_id, ...) \
49 CASE_ ## enabled(codec_id, __VA_ARGS__)
50 #define CASE_3(config, codec_id, ...) \
51 CASE_2(config, codec_id, __VA_ARGS__)
52 #define CASE(codec, ...) \
53 CASE_3(CONFIG_ ## codec ## _ENCODER, AV_CODEC_ID_ ## codec, __VA_ARGS__)
79 #define FREEZE_INTERVAL 128
91 (
s->block_size & (
s->block_size - 1))) {
97 int frontier, max_paths;
99 if ((
unsigned)avctx->
trellis > 16
U) {
116 frontier = 1 << avctx->
trellis;
153 bytestream_put_le16(&extradata, avctx->
frame_size);
154 bytestream_put_le16(&extradata, 7);
155 for (
int i = 0;
i < 7;
i++) {
168 av_log(avctx, AV_LOG_ERROR,
"Sample rate must be 11025, "
170 return AVERROR(EINVAL);
182 av_log(avctx, AV_LOG_ERROR,
"Sample rate must be 22050\n");
183 return AVERROR(EINVAL);
247 const int sign = (
delta < 0) * 8;
254 nibble = sign | nibble;
256 c->prev_sample +=
diff;
267 int nibble = 8*(
delta < 0);
289 c->prev_sample -=
diff;
291 c->prev_sample +=
diff;
305 ((
c->sample2) * (
c->coeff2))) / 64;
309 bias =
c->idelta / 2;
311 bias = -
c->idelta / 2;
313 nibble = (nibble + bias) /
c->idelta;
316 predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) *
c->idelta;
318 c->sample2 =
c->sample1;
351 const int16_t *
samples, uint8_t *dst,
356 const int frontier = 1 << avctx->
trellis;
363 int pathn = 0, froze = -1,
i, j, k, generation = 0;
364 uint8_t *
hash =
s->trellis_hash;
365 memset(
hash, 0xff, 65536 *
sizeof(*
hash));
367 memset(nodep_buf, 0, 2 * frontier *
sizeof(*nodep_buf));
368 nodes[0] = node_buf + frontier;
371 nodes[0]->
step =
c->step_index;
380 nodes[0]->
step =
c->idelta;
383 nodes[0]->
step = 127;
386 nodes[0]->
step =
c->step;
391 for (
i = 0;
i < n;
i++) {
396 memset(nodes_next, 0, frontier *
sizeof(
TrellisNode*));
397 for (j = 0; j < frontier && nodes[j]; j++) {
400 const int range = (j < frontier / 2) ? 1 : 0;
401 const int step = nodes[j]->step;
404 const int predictor = ((nodes[j]->sample1 *
c->coeff1) +
405 (nodes[j]->sample2 *
c->coeff2)) / 64;
407 const int nmin =
av_clip(div-range, -8, 6);
408 const int nmax =
av_clip(div+range, -7, 7);
409 for (nidx = nmin; nidx <= nmax; nidx++) {
410 const int nibble = nidx & 0xf;
412 #define STORE_NODE(NAME, STEP_INDEX)\
418 dec_sample = av_clip_int16(dec_sample);\
419 d = sample - dec_sample;\
420 ssd = nodes[j]->ssd + d*(unsigned)d;\
425 if (ssd < nodes[j]->ssd)\
438 h = &hash[(uint16_t) dec_sample];\
439 if (*h == generation)\
441 if (heap_pos < frontier) {\
446 pos = (frontier >> 1) +\
447 (heap_pos & ((frontier >> 1) - 1));\
448 if (ssd > nodes_next[pos]->ssd)\
453 u = nodes_next[pos];\
455 av_assert1(pathn < FREEZE_INTERVAL << avctx->trellis);\
457 nodes_next[pos] = u;\
461 u->step = STEP_INDEX;\
462 u->sample2 = nodes[j]->sample1;\
463 u->sample1 = dec_sample;\
464 paths[u->path].nibble = nibble;\
465 paths[u->path].prev = nodes[j]->path;\
469 int parent = (pos - 1) >> 1;\
470 if (nodes_next[parent]->ssd <= ssd)\
472 FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
483 #define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
484 const int predictor = nodes[j]->sample1;\
485 const int div = (sample - predictor) * 4 / STEP_TABLE;\
486 int nmin = av_clip(div - range, -7, 6);\
487 int nmax = av_clip(div + range, -6, 7);\
492 for (nidx = nmin; nidx <= nmax; nidx++) {\
493 const int nibble = nidx < 0 ? 7 - nidx : nidx;\
494 int dec_sample = predictor +\
496 ff_adpcm_yamaha_difflookup[nibble]) / 8;\
497 STORE_NODE(NAME, STEP_INDEX);\
515 if (generation == 255) {
516 memset(
hash, 0xff, 65536 *
sizeof(*
hash));
521 if (nodes[0]->ssd > (1 << 28)) {
522 for (j = 1; j < frontier && nodes[j]; j++)
523 nodes[j]->ssd -= nodes[0]->ssd;
529 p = &paths[nodes[0]->path];
530 for (k =
i; k > froze; k--) {
539 memset(nodes + 1, 0, (frontier - 1) *
sizeof(
TrellisNode*));
543 p = &paths[nodes[0]->
path];
544 for (
i = n - 1;
i > froze;
i--) {
549 c->predictor = nodes[0]->sample1;
550 c->sample1 = nodes[0]->sample1;
551 c->sample2 = nodes[0]->sample2;
552 c->step_index = nodes[0]->step;
553 c->step = nodes[0]->step;
554 c->idelta = nodes[0]->step;
557 #if CONFIG_ADPCM_ARGO_ENCODER
568 return (nibble >>
shift) & 0x0F;
572 const int16_t *
samples,
int nsamples,
584 for (
int n = 0; n < nsamples; n++) {
602 int st, pkt_size,
ret;
610 samples_p = (int16_t **)
frame->extended_data;
626 int blocks = (
frame->nb_samples - 1) / 8;
630 status->prev_sample = samples_p[ch][0];
633 bytestream_put_le16(&dst,
status->prev_sample);
634 *dst++ =
status->step_index;
643 for (
int ch = 0; ch <
channels; ch++) {
645 buf + ch * blocks * 8, &
c->status[ch],
648 for (
int i = 0;
i < blocks;
i++) {
649 for (
int ch = 0; ch <
channels; ch++) {
650 uint8_t *buf1 = buf + ch * blocks * 8 +
i * 8;
651 for (
int j = 0; j < 8; j += 2)
652 *dst++ = buf1[j] | (buf1[j + 1] << 4);
657 for (
int i = 0;
i < blocks;
i++) {
658 for (
int ch = 0; ch <
channels; ch++) {
660 const int16_t *smp = &samples_p[ch][1 +
i * 8];
661 for (
int j = 0; j < 8; j += 2) {
674 for (
int ch = 0; ch <
channels; ch++) {
682 for (
int i = 0;
i < 64;
i++)
686 for (
int i = 0;
i < 64;
i += 2) {
704 for (
int i = 0;
i <
frame->nb_samples;
i++) {
705 for (
int ch = 0; ch <
channels; ch++) {
718 for (
int n =
frame->nb_samples / 2; n > 0; n--) {
719 for (
int ch = 0; ch <
channels; ch++) {
729 const int n =
frame->nb_samples - 1;
754 buf + n, &
c->status[1], n,
756 for (
int i = 0;
i < n;
i++) {
762 for (
int i = 1;
i <
frame->nb_samples;
i++) {
780 if (
c->status[
i].idelta < 16)
781 c->status[
i].idelta = 16;
782 bytestream_put_le16(&dst,
c->status[
i].idelta);
788 bytestream_put_le16(&dst,
c->status[
i].sample1);
791 bytestream_put_le16(&dst,
c->status[
i].sample2);
801 for (
int i = 0;
i < n;
i += 2)
802 *dst++ = (buf[
i] << 4) | buf[
i + 1];
808 for (
int i = 0;
i < n;
i++)
809 *dst++ = (buf[
i] << 4) | buf[n +
i];
822 int n =
frame->nb_samples / 2;
831 for (
int i = 0;
i < n;
i += 2)
832 *dst++ = buf[
i] | (buf[
i + 1] << 4);
838 for (
int i = 0;
i < n;
i++)
839 *dst++ = buf[
i] | (buf[n +
i] << 4);
856 for (
int n =
frame->nb_samples / 2; n > 0; n--) {
857 for (
int ch = 0; ch <
channels; ch++) {
869 c->status[0].prev_sample = *
samples;
870 bytestream_put_le16(&dst,
c->status[0].prev_sample);
871 bytestream_put_byte(&dst,
c->status[0].step_index);
872 bytestream_put_byte(&dst, 0);
876 const int n =
frame->nb_samples >> 1;
883 for (
int i = 0;
i < n;
i++)
884 bytestream_put_byte(&dst, (buf[2 *
i] << 4) | buf[2 *
i + 1]);
888 }
else for (
int n =
frame->nb_samples >> 1; n > 0; n--) {
892 bytestream_put_byte(&dst, nibble);
897 bytestream_put_byte(&dst, nibble);
906 for (
int ch = 0; ch <
channels; ch++) {
907 int64_t
error = INT64_MAX, tmperr = INT64_MAX;
909 int saved1 =
c->status[ch].sample1;
910 int saved2 =
c->status[ch].sample2;
913 for (
int s = 2;
s < 18 && tmperr != 0;
s++) {
914 for (
int f = 0;
f < 2 && tmperr != 0;
f++) {
915 c->status[ch].sample1 = saved1;
916 c->status[ch].sample2 = saved2;
917 tmperr = adpcm_argo_compress_block(
c->status + ch,
NULL, samples_p[ch],
919 if (tmperr <
error) {
928 c->status[ch].sample1 = saved1;
929 c->status[ch].sample2 = saved2;
930 adpcm_argo_compress_block(
c->status + ch, &pb, samples_p[ch],
941 for (
int n =
frame->nb_samples / 2; n > 0; n--) {
943 for (
int ch = 0; ch <
channels; ch++) {
978 .
name =
"block_size",
979 .help =
"set the block size",
982 .default_val = {.i64 = 1024},
997 #define ADPCM_ENCODER_0(id_, name_, sample_fmts_, capabilities_, long_name_)
998 #define ADPCM_ENCODER_1(id_, name_, sample_fmts_, capabilities_, long_name_) \
999 const FFCodec ff_ ## name_ ## _encoder = { \
1001 .p.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
1002 .p.type = AVMEDIA_TYPE_AUDIO, \
1004 .p.sample_fmts = sample_fmts_, \
1005 .p.ch_layouts = ch_layouts, \
1006 .p.capabilities = capabilities_ | AV_CODEC_CAP_DR1, \
1007 .p.priv_class = &adpcm_encoder_class, \
1008 .priv_data_size = sizeof(ADPCMEncodeContext), \
1009 .init = adpcm_encode_init, \
1010 FF_CODEC_ENCODE_CB(adpcm_encode_frame), \
1011 .close = adpcm_encode_close, \
1012 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP | FF_CODEC_CAP_INIT_THREADSAFE, \
1014 #define ADPCM_ENCODER_2(enabled, codec_id, name, sample_fmts, capabilities, long_name) \
1015 ADPCM_ENCODER_ ## enabled(codec_id, name, sample_fmts, capabilities, long_name)
1016 #define ADPCM_ENCODER_3(config, codec_id, name, sample_fmts, capabilities, long_name) \
1017 ADPCM_ENCODER_2(config, codec_id, name, sample_fmts, capabilities, long_name)
1018 #define ADPCM_ENCODER(codec, name, sample_fmts, capabilities, long_name) \
1019 ADPCM_ENCODER_3(CONFIG_ ## codec ## _ENCODER, AV_CODEC_ID_ ## codec, \
1020 name, sample_fmts, capabilities, long_name)
static void error(const char *err)
int frame_size
Number of samples per channel in an audio frame.
static uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, int16_t sample)
@ AV_CODEC_ID_ADPCM_IMA_QT
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)
int sample_rate
samples per second
#define u(width, name, range_min, range_max)
static enum AVSampleFormat sample_fmts[]
#define ADPCM_ENCODER(codec, name, sample_fmts, capabilities, long_name)
const int16_t ff_adpcm_AdaptationTable[]
static const AVClass adpcm_encoder_class
static void put_sbits(PutBitContext *pb, int n, int32_t value)
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
This structure describes decoded (raw) audio or video data.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about which is also called distortion Distortion can be quantified by almost any quality measurement one chooses the sum of squared differences is used but more complex methods that consider psychovisual effects can be used as well It makes no difference in this discussion First step
#define AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_STEREO
int nb_channels
Number of channels in this layout.
const struct AVCodec * codec
#define STORE_NODE(NAME, STEP_INDEX)
AVChannelLayout ch_layout
Audio channel layout.
#define FF_ALLOC_TYPED_ARRAY(p, nelem)
ADPCMChannelStatus status[6]
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const AVOption options[]
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static void adpcm_compress_trellis(AVCodecContext *avctx, const int16_t *samples, uint8_t *dst, ADPCMChannelStatus *c, int n, int stride)
#define AV_OPT_FLAG_AUDIO_PARAM
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
@ AV_CODEC_ID_ADPCM_YAMAHA
@ AV_CODEC_ID_ADPCM_IMA_WS
const char * av_default_item_name(void *ptr)
Return the context name.
@ AV_CODEC_ID_ADPCM_IMA_AMV
int trellis
trellis RD quantization
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
static uint8_t adpcm_ima_alp_compress_sample(ADPCMChannelStatus *c, int16_t sample)
const int8_t ff_adpcm_yamaha_difflookup[]
An AVChannelLayout holds information about the channel layout of audio data.
@ AV_CODEC_ID_ADPCM_IMA_ALP
const int16_t ff_adpcm_step_table[89]
This is the step table.
static void predictor(uint8_t *src, ptrdiff_t size)
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
const uint8_t ff_adpcm_AdaptCoeff1[]
Divided by 4 to fit in 8-bit integers.
#define i(width, name, range_min, range_max)
const int8_t ff_adpcm_AdaptCoeff2[]
Divided by 4 to fit in 8-bit integers.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
AVSampleFormat
Audio sample formats.
static uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c, int16_t sample)
static const AVChannelLayout ch_layouts[]
@ AV_CODEC_ID_ADPCM_IMA_APM
@ AV_SAMPLE_FMT_S16
signed 16 bits
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
int16_t ff_adpcm_argo_expand_nibble(ADPCMChannelStatus *cs, int nibble, int shift, int flag)
const int8_t ff_adpcm_index_table[16]
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static enum AVSampleFormat sample_fmts_p[]
#define AV_INPUT_BUFFER_PADDING_SIZE
main external API structure.
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
const int16_t ff_adpcm_yamaha_indexscale[]
Filter the word “frame” indicates either a video frame or a group of audio samples
static int shift(int a, int b)
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
@ AV_CODEC_ID_ADPCM_IMA_SSI
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
static av_always_inline int diff(const uint32_t a, const uint32_t b)
This structure stores compressed data.
@ AV_CODEC_ID_ADPCM_IMA_WAV
static uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c, int16_t sample)
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
static uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c, int16_t sample)
static av_cold int adpcm_encode_close(AVCodecContext *avctx)