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38 #define FLAC_SUBFRAME_CONSTANT 0
39 #define FLAC_SUBFRAME_VERBATIM 1
40 #define FLAC_SUBFRAME_FIXED 8
41 #define FLAC_SUBFRAME_LPC 32
43 #define MAX_FIXED_ORDER 4
44 #define MAX_PARTITION_ORDER 8
45 #define MAX_PARTITIONS (1 << MAX_PARTITION_ORDER)
46 #define MAX_LPC_PRECISION 15
47 #define MIN_LPC_SHIFT 0
48 #define MAX_LPC_SHIFT 15
149 put_bits(&pb, 5,
s->avctx->bits_per_raw_sample - 1);
151 put_bits(&pb, 24, (
s->sample_count & 0xFFFFFF000LL) >> 12);
152 put_bits(&pb, 12,
s->sample_count & 0x000000FFFLL);
154 memcpy(&
header[18],
s->md5sum, 16);
170 target = (samplerate * block_time_ms) / 1000;
171 for (
i = 0;
i < 16;
i++) {
196 av_log(avctx,
AV_LOG_DEBUG,
" lpc type: Levinson-Durbin recursion with Welch window\n");
272 for (
i = 4;
i < 12;
i++) {
282 if (freq % 1000 == 0 && freq < 255000) {
284 s->sr_code[1] = freq / 1000;
285 }
else if (freq % 10 == 0 && freq < 655350) {
287 s->sr_code[1] = freq / 10;
288 }
else if (freq < 65535) {
290 s->sr_code[1] = freq;
295 s->samplerate = freq;
300 s->options.compression_level = 5;
304 level =
s->options.compression_level;
307 s->options.compression_level);
311 s->options.block_time_ms = ((
int[]){ 27, 27, 27,105,105,105,105,105,105,105,105,105,105})[
level];
320 if (
s->options.min_prediction_order < 0)
321 s->options.min_prediction_order = ((
int[]){ 2, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1})[
level];
322 if (
s->options.max_prediction_order < 0)
323 s->options.max_prediction_order = ((
int[]){ 3, 4, 4, 6, 8, 8, 8, 8, 12, 12, 12, 32, 32})[
level];
325 if (
s->options.prediction_order_method < 0)
332 if (
s->options.min_partition_order >
s->options.max_partition_order) {
334 s->options.min_partition_order,
s->options.max_partition_order);
337 if (
s->options.min_partition_order < 0)
338 s->options.min_partition_order = ((
int[]){ 2, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0})[
level];
339 if (
s->options.max_partition_order < 0)
340 s->options.max_partition_order = ((
int[]){ 2, 2, 3, 3, 3, 8, 8, 8, 8, 8, 8, 8, 8})[
level];
342 #if FF_API_PRIVATE_OPT
348 "invalid min prediction order %d, clamped to %d\n",
364 "invalid max prediction order %d, clamped to %d\n",
379 s->options.min_prediction_order = 0;
380 s->options.max_prediction_order = 0;
384 "invalid min prediction order %d, clamped to %d\n",
390 "invalid max prediction order %d, clamped to %d\n",
396 if (
s->options.max_prediction_order <
s->options.min_prediction_order) {
398 s->options.min_prediction_order,
s->options.max_prediction_order);
412 s->max_blocksize =
s->avctx->frame_size;
417 s->avctx->bits_per_raw_sample);
433 s->min_framesize =
s->max_framesize;
448 "output stream will have incorrect "
449 "channel layout.\n");
452 "will use Flac channel layout for "
477 for (
i = 0;
i < 16;
i++) {
481 frame->bs_code[1] = 0;
486 frame->blocksize = nb_samples;
487 if (
frame->blocksize <= 256) {
488 frame->bs_code[0] = 6;
491 frame->bs_code[0] = 7;
496 for (
ch = 0;
ch <
s->channels;
ch++) {
500 sub->
obits =
s->avctx->bits_per_raw_sample;
508 frame->verbatim_only = 0;
520 s->avctx->bits_per_raw_sample;
522 #define COPY_SAMPLES(bits) do { \
523 const int ## bits ## _t *samples0 = samples; \
525 for (i = 0, j = 0; i < frame->blocksize; i++) \
526 for (ch = 0; ch < s->channels; ch++, j++) \
527 frame->subframes[ch].samples[i] = samples0[j] >> shift; \
542 for (
i = 0;
i <
n;
i++) {
545 count += (v >> k) + 1 + k;
554 int p, porder, psize;
575 count += 4 + 5 + pred_order *
s->options.lpc_coeff_precision;
582 psize =
s->frame.blocksize >> porder;
588 for (p = 0; p < 1 << porder; p++) {
593 part_end =
FFMIN(
s->frame.blocksize, part_end + psize);
601 #define rice_encode_count(sum, n, k) (((n)*((k)+1))+((sum-(n>>1))>>(k)))
613 sum2 = sum - (
n >> 1);
614 k =
av_log2(av_clipl_int32(sum2 /
n));
615 return FFMIN(k, max_param);
621 int64_t bestbits = INT64_MAX;
624 for (k = 0; k <= max_param; k++) {
625 int64_t
bits = sums[k][
i];
626 if (
bits < bestbits) {
637 int n,
int pred_order,
int max_param,
int exact)
643 part = (1 << porder);
646 cnt = (
n >> porder) - pred_order;
647 for (
i = 0;
i < part;
i++) {
650 all_bits += sums[k][
i];
670 const uint32_t *res, *res_end;
675 for (k = 0; k <= kmax; k++) {
676 res = &
data[pred_order];
677 res_end = &
data[
n >> pmax];
678 for (
i = 0;
i < parts;
i++) {
680 uint64_t sum = (1LL + k) * (res_end - res);
681 while (res < res_end)
682 sum += *(res++) >> k;
686 while (res < res_end)
690 res_end +=
n >> pmax;
698 int parts = (1 <<
level);
699 for (
i = 0;
i < parts;
i++) {
700 for (k=0; k<=kmax; k++)
701 sums[k][
i] = sums[k][2*
i] + sums[k][2*
i+1];
723 for (
i = 0;
i <
n;
i++)
726 calc_sum_top(pmax, exact ? kmax : 0, udata,
n, pred_order, sums);
729 bits[pmin] = UINT32_MAX;
732 if (
bits[
i] <
bits[opt_porder] || pmax == pmin) {
741 return bits[opt_porder];
758 s->frame.blocksize, pred_order);
760 s->frame.blocksize, pred_order);
764 bits += 4 + 5 + pred_order *
s->options.lpc_coeff_precision;
766 s->frame.blocksize, pred_order,
s->options.exact_rice_parameters);
776 for (
i = 0;
i < order;
i++)
780 for (
i = order;
i <
n;
i++)
782 }
else if (order == 1) {
783 for (
i = order;
i <
n;
i++)
784 res[
i] = smp[
i] - smp[
i-1];
785 }
else if (order == 2) {
786 int a = smp[order-1] - smp[order-2];
787 for (
i = order;
i <
n;
i += 2) {
788 int b = smp[
i ] - smp[
i-1];
790 a = smp[
i+1] - smp[
i ];
793 }
else if (order == 3) {
794 int a = smp[order-1] - smp[order-2];
795 int c = smp[order-1] - 2*smp[order-2] + smp[order-3];
796 for (
i = order;
i <
n;
i += 2) {
797 int b = smp[
i ] - smp[
i-1];
800 a = smp[
i+1] - smp[
i ];
805 int a = smp[order-1] - smp[order-2];
806 int c = smp[order-1] - 2*smp[order-2] + smp[order-3];
807 int e = smp[order-1] - 3*smp[order-2] + 3*smp[order-3] - smp[order-4];
808 for (
i = order;
i <
n;
i += 2) {
809 int b = smp[
i ] - smp[
i-1];
813 a = smp[
i+1] - smp[
i ];
825 int min_order, max_order, opt_order, omethod;
839 for (
i = 1;
i <
n;
i++)
849 if (
frame->verbatim_only ||
n < 5) {
851 memcpy(res, smp,
n *
sizeof(
int32_t));
855 min_order =
s->options.min_prediction_order;
856 max_order =
s->options.max_prediction_order;
857 omethod =
s->options.prediction_order_method;
867 bits[0] = UINT32_MAX;
868 for (
i = min_order;
i <= max_order;
i++) {
874 sub->
order = opt_order;
876 if (sub->
order != max_order) {
886 s->options.lpc_coeff_precision, coefs,
shift,
s->options.lpc_type,
887 s->options.lpc_passes, omethod,
893 int levels = 1 << omethod;
896 int opt_index = levels-1;
897 opt_order = max_order-1;
898 bits[opt_index] = UINT32_MAX;
899 for (
i = levels-1;
i >= 0;
i--) {
900 int last_order = order;
901 order = min_order + (((max_order-min_order+1) * (
i+1)) / levels)-1;
902 order = av_clip(order, min_order - 1, max_order - 1);
903 if (order == last_order)
905 if (
s->bps_code * 4 +
s->options.lpc_coeff_precision +
av_log2(order) <= 32) {
906 s->flac_dsp.lpc16_encode(res, smp,
n, order+1, coefs[order],
909 s->flac_dsp.lpc32_encode(res, smp,
n, order+1, coefs[order],
923 bits[0] = UINT32_MAX;
924 for (
i = min_order-1;
i < max_order;
i++) {
925 if (
s->bps_code * 4 +
s->options.lpc_coeff_precision +
av_log2(
i) <= 32) {
926 s->flac_dsp.lpc16_encode(res, smp,
n,
i+1, coefs[
i],
shift[
i]);
928 s->flac_dsp.lpc32_encode(res, smp,
n,
i+1, coefs[
i],
shift[
i]);
939 opt_order = min_order - 1 + (max_order-min_order)/3;
943 int last = opt_order;
945 if (i < min_order-1 || i >= max_order ||
bits[
i] < UINT32_MAX)
947 if (
s->bps_code * 4 +
s->options.lpc_coeff_precision +
av_log2(
i) <= 32) {
948 s->flac_dsp.lpc32_encode(res, smp,
n,
i+1, coefs[
i],
shift[
i]);
950 s->flac_dsp.lpc16_encode(res, smp,
n,
i+1, coefs[
i],
shift[
i]);
960 if (
s->options.multi_dim_quant) {
962 int i,
step, improved;
963 int64_t best_score = INT64_MAX;
966 qmax = (1 << (
s->options.lpc_coeff_precision - 1)) - 1;
968 for (
i=0;
i<opt_order;
i++)
979 for (
i=0;
i<opt_order;
i++) {
980 int diff = ((
tmp + 1) % 3) - 1;
981 lpc_try[
i] = av_clip(coefs[opt_order - 1][
i] +
diff, -qmax, qmax);
988 if (
s->bps_code * 4 +
s->options.lpc_coeff_precision +
av_log2(opt_order - 1) <= 32) {
989 s->flac_dsp.lpc16_encode(res, smp,
n, opt_order, lpc_try,
shift[opt_order-1]);
991 s->flac_dsp.lpc32_encode(res, smp,
n, opt_order, lpc_try,
shift[opt_order-1]);
994 if (score < best_score) {
996 memcpy(coefs[opt_order-1], lpc_try,
sizeof(*coefs));
1003 sub->
order = opt_order;
1009 if (
s->bps_code * 4 +
s->options.lpc_coeff_precision +
av_log2(opt_order) <= 32) {
1042 if (
s->frame.bs_code[0] == 6)
1044 else if (
s->frame.bs_code[0] == 7)
1048 count += ((
s->sr_code[0] == 12) + (
s->sr_code[0] > 12) * 2) * 8;
1064 for (
ch = 0;
ch <
s->channels;
ch++)
1071 if (
count > INT_MAX)
1081 for (
ch = 0;
ch <
s->channels;
ch++) {
1085 for (
i = 0;
i <
s->frame.blocksize;
i++) {
1091 if (v && !(v & 1)) {
1094 for (
i = 0;
i <
s->frame.blocksize;
i++)
1119 sum[0] = sum[1] = sum[2] = sum[3] = 0;
1120 for (
i = 2;
i <
n;
i++) {
1121 lt = left_ch[
i] - 2*left_ch[
i-1] + left_ch[
i-2];
1122 rt = right_ch[
i] - 2*right_ch[
i-1] + right_ch[
i-2];
1123 sum[2] +=
FFABS((lt + rt) >> 1);
1124 sum[3] +=
FFABS(lt - rt);
1125 sum[0] +=
FFABS(lt);
1126 sum[1] +=
FFABS(rt);
1129 for (
i = 0;
i < 4;
i++) {
1135 score[0] = sum[0] + sum[1];
1136 score[1] = sum[0] + sum[3];
1137 score[2] = sum[1] + sum[3];
1138 score[3] = sum[2] + sum[3];
1142 for (
i = 1;
i < 4;
i++)
1143 if (score[
i] < score[best])
1162 right =
frame->subframes[1].samples;
1164 if (
s->channels != 2) {
1169 if (
s->options.ch_mode < 0) {
1170 int max_rice_param = (1 <<
frame->subframes[0].rc.coding_mode) - 2;
1173 frame->ch_mode =
s->options.ch_mode;
1180 for (
i = 0;
i <
n;
i++) {
1183 right[
i] =
tmp - right[
i];
1185 frame->subframes[1].obits++;
1187 for (
i = 0;
i <
n;
i++)
1188 right[
i] =
left[
i] - right[
i];
1189 frame->subframes[1].obits++;
1191 for (
i = 0;
i <
n;
i++)
1193 frame->subframes[0].obits++;
1225 if (
frame->bs_code[0] == 6)
1227 else if (
frame->bs_code[0] == 7)
1230 if (
s->sr_code[0] == 12)
1232 else if (
s->sr_code[0] > 12)
1246 for (
ch = 0;
ch <
s->channels;
ch++) {
1248 int i, p, porder, psize;
1273 int cbits =
s->options.lpc_coeff_precision;
1285 psize =
s->frame.blocksize >> porder;
1290 for (p = 0; p < 1 << porder; p++) {
1293 while (res < part_end)
1326 int buf_size =
s->frame.blocksize *
s->channels *
1327 ((
s->avctx->bits_per_raw_sample + 7) / 8);
1329 if (
s->avctx->bits_per_raw_sample > 16 || HAVE_BIGENDIAN) {
1335 if (
s->avctx->bits_per_raw_sample <= 16) {
1338 s->bdsp.bswap16_buf((uint16_t *)
s->md5_buffer,
1339 (
const uint16_t *)
samples, buf_size / 2);
1340 buf =
s->md5_buffer;
1347 for (
i = 0;
i <
s->frame.blocksize *
s->channels;
i++) {
1351 buf =
s->md5_buffer;
1363 int frame_bytes, out_bytes,
ret;
1369 s->max_framesize =
s->max_encoded_framesize;
1373 #if FF_API_SIDEDATA_ONLY_PKT
1386 avpkt->
pts =
s->next_pts;
1388 *got_packet_ptr = 1;
1396 if (
frame->nb_samples <
s->frame.blocksize) {
1414 if (frame_bytes < 0 || frame_bytes >
s->max_framesize) {
1415 s->frame.verbatim_only = 1;
1417 if (frame_bytes < 0) {
1429 s->sample_count +=
frame->nb_samples;
1434 if (out_bytes >
s->max_encoded_framesize)
1435 s->max_encoded_framesize = out_bytes;
1436 if (out_bytes < s->min_framesize)
1437 s->min_framesize = out_bytes;
1441 avpkt->
size = out_bytes;
1445 *got_packet_ptr = 1;
1463 #define FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1466 {
"lpc_type",
"LPC algorithm", offsetof(
FlacEncodeContext,
options.lpc_type),
AV_OPT_TYPE_INT, {.i64 =
FF_LPC_TYPE_DEFAULT },
FF_LPC_TYPE_DEFAULT,
FF_LPC_TYPE_NB-1,
FLAGS,
"lpc_type" },
1474 {
"prediction_order_method",
"Search method for selecting prediction order", offsetof(
FlacEncodeContext,
options.prediction_order_method),
AV_OPT_TYPE_INT, {.i64 = -1 }, -1,
ORDER_METHOD_LOG,
FLAGS,
"predm" },
1481 {
"ch_mode",
"Stereo decorrelation mode", offsetof(
FlacEncodeContext,
options.ch_mode),
AV_OPT_TYPE_INT, { .i64 = -1 }, -1,
FLAC_CHMODE_MID_SIDE,
FLAGS,
"ch_mode" },
int frame_size
Number of samples per channel in an audio frame.
#define FF_ENABLE_DEPRECATION_WARNINGS
#define AV_LOG_WARNING
Something somehow does not look correct.
#define PUT_UTF8(val, tmp, PUT_BYTE)
FFLPCType
LPC analysis type.
#define AV_CH_LAYOUT_5POINT0_BACK
int32_t samples[FLAC_MAX_BLOCKSIZE]
static av_cold int init(AVCodecContext *avctx)
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
static av_cold int flac_encode_init(AVCodecContext *avctx)
uint64_t channel_layout
Audio channel layout.
int sample_rate
samples per second
static void set_sr_golomb_flac(PutBitContext *pb, int i, int k, int limit, int esc_len)
write signed golomb rice code (flac).
int exact_rice_parameters
#define MAX_PARTITION_ORDER
static enum AVSampleFormat sample_fmts[]
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
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.
static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
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
@ FF_LPC_TYPE_CHOLESKY
Cholesky factorization.
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
int prediction_order_method
static int select_blocksize(int samplerate, int block_time_ms)
Set blocksize based on samplerate.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
static uint64_t find_subframe_rice_params(FlacEncodeContext *s, FlacSubframe *sub, int pred_order)
#define ORDER_METHOD_4LEVEL
#define AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_QUAD
@ FF_LPC_TYPE_DEFAULT
use the codec default LPC type
const int32_t ff_flac_blocksize_table[16]
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void write_subframes(FlacEncodeContext *s)
int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
Calculate an estimate for the maximum frame size based on verbatim mode.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static void remove_wasted_bits(FlacEncodeContext *s)
#define FLAC_SUBFRAME_LPC
#define COPY_SAMPLES(bits)
static uint64_t calc_optimal_rice_params(RiceContext *rc, int porder, uint64_t sums[32][MAX_PARTITIONS], int n, int pred_order, int max_param, int exact)
#define FLAC_SUBFRAME_VERBATIM
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
#define FLAC_SUBFRAME_CONSTANT
const int ff_flac_sample_rate_table[16]
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
FlacSubframe subframes[FLAC_MAX_CHANNELS]
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
attribute_deprecated int max_prediction_order
#define FLAC_SUBFRAME_FIXED
const char * av_default_item_name(void *ptr)
Return the context name.
#define AV_CH_LAYOUT_5POINT1
#define FLAC_STREAMINFO_SIZE
#define ORDER_METHOD_SEARCH
int ff_lpc_calc_coefs(LPCContext *s, const int32_t *samples, int blocksize, int min_order, int max_order, int precision, int32_t coefs[][MAX_LPC_ORDER], int *shift, enum FFLPCType lpc_type, int lpc_passes, int omethod, int min_shift, int max_shift, int zero_shift)
Calculate LPC coefficients for multiple orders.
#define AV_CH_FRONT_CENTER
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 uint64_t rice_count_exact(const int32_t *res, int n, int k)
int max_encoded_framesize
static int encode_residual_ch(FlacEncodeContext *s, int ch)
static int get_max_p_order(int max_porder, int n, int order)
#define ORDER_METHOD_8LEVEL
static int find_optimal_param_exact(uint64_t sums[32][MAX_PARTITIONS], int i, int max_param)
unsigned int md5_buffer_size
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static void channel_decorrelation(FlacEncodeContext *s)
Perform stereo channel decorrelation.
@ FF_LPC_TYPE_NB
Not part of ABI.
enum AVSampleFormat sample_fmt
audio sample format
static int encode_frame(FlacEncodeContext *s)
static void calc_sum_top(int pmax, int kmax, const uint32_t *data, int n, int pred_order, uint64_t sums[32][MAX_PARTITIONS])
int32_t residual[FLAC_MAX_BLOCKSIZE+11]
CompressionOptions options
const char const char void * val
static const uint8_t header[24]
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
#define AV_CH_LAYOUT_5POINT1_BACK
static int write_frame(FlacEncodeContext *s, AVPacket *avpkt)
static const AVOption options[]
int32_t coefs[MAX_LPC_ORDER]
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
int channels
number of audio channels
#define AV_CH_LAYOUT_5POINT0
static void calc_sum_next(int level, uint64_t sums[32][MAX_PARTITIONS], int kmax)
void av_md5_init(AVMD5 *ctx)
Initialize MD5 hashing.
#define i(width, name, range_min, range_max)
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
static int put_bits_count(PutBitContext *s)
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static void write_utf8(PutBitContext *pb, uint32_t val)
static int count_frame_header(FlacEncodeContext *s)
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
AVSampleFormat
Audio sample formats.
static void write_frame_header(FlacEncodeContext *s)
static void write_frame_footer(FlacEncodeContext *s)
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
static void copy_samples(FlacEncodeContext *s, const void *samples)
Copy channel-interleaved input samples into separate subframes.
void av_md5_final(AVMD5 *ctx, uint8_t *dst)
Finish hashing and output digest value.
static int estimate_stereo_mode(const int32_t *left_ch, const int32_t *right_ch, int n, int max_rice_param)
static uint64_t calc_rice_params(RiceContext *rc, uint32_t udata[FLAC_MAX_BLOCKSIZE], uint64_t sums[32][MAX_PARTITIONS], int pmin, int pmax, const int32_t *data, int n, int pred_order, int exact)
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
void av_md5_update(AVMD5 *ctx, const uint8_t *src, int len)
Update hash value.
static void encode_residual_fixed(int32_t *res, const int32_t *smp, int n, int order)
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
#define FLAC_MAX_CHANNELS
#define MAX_LPC_PRECISION
main external API structure.
uint64_t rc_sums[32][MAX_PARTITIONS]
uint32_t rc_udata[FLAC_MAX_BLOCKSIZE]
uint32_t av_crc(const AVCRC *ctx, uint32_t crc, const uint8_t *buffer, size_t length)
Calculate the CRC of a block.
struct AVMD5 * av_md5_alloc(void)
Allocate an AVMD5 context.
@ AV_PKT_DATA_NEW_EXTRADATA
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
static uint64_t subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub, int pred_order)
static void frame_end(MpegEncContext *s)
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Filter the word “frame” indicates either a video frame or a group of audio samples
#define AV_CODEC_CAP_LOSSLESS
Codec is lossless.
static void init_frame(FlacEncodeContext *s, int nb_samples)
int params[MAX_PARTITIONS]
static int shift(int a, int b)
#define FF_DISABLE_DEPRECATION_WARNINGS
#define FLAC_MAX_BLOCKSIZE
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
static av_always_inline int diff(const uint32_t a, const uint32_t b)
This structure stores compressed data.
enum CodingMode coding_mode
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
static int find_optimal_param(uint64_t sum, int n, int max_param)
Solve for d/dk(rice_encode_count) = n-((sum-(n>>1))>>(k+1)) = 0.
static av_cold void dprint_compression_options(FlacEncodeContext *s)
static int update_md5_sum(FlacEncodeContext *s, const void *samples)
static av_cold int flac_encode_close(AVCodecContext *avctx)
attribute_deprecated int min_prediction_order
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
static const AVClass flac_encoder_class
static void write_streaminfo(FlacEncodeContext *s, uint8_t *header)
Write streaminfo metadata block to byte array.
#define ORDER_METHOD_2LEVEL
#define FLAC_MIN_BLOCKSIZE
@ FF_LPC_TYPE_NONE
do not use LPC prediction or use all zero coefficients
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
@ AV_SAMPLE_FMT_S32
signed 32 bits
@ FF_LPC_TYPE_LEVINSON
Levinson-Durbin recursion.
#define rice_encode_count(sum, n, k)
@ FLAC_CHMODE_INDEPENDENT
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
@ FF_LPC_TYPE_FIXED
fixed LPC coefficients
attribute_deprecated int side_data_only_packets
Encoding only and set by default.