2 * Simple free lossless/lossy audio codec
3 * Copyright (c) 2004 Alex Beregszaszi
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 #include "rangecoder.h"
30 * Simple free lossless/lossy audio codec
31 * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
32 * Written and designed by Alex Beregszaszi
35 * - CABAC put/get_symbol
36 * - independent quantizer for channels
37 * - >2 channels support
38 * - more decorrelation types
39 * - more tap_quant tests
40 * - selectable intlist writers/readers (bonk-style, golomb, cabac)
43 #define MAX_CHANNELS 2
49 typedef struct SonicContext {
52 int lossless, decorrelation;
54 int num_taps, downsampling;
57 int channels, samplerate, block_align, frame_size;
61 int *coded_samples[MAX_CHANNELS];
71 int *predictor_state[MAX_CHANNELS];
74 #define LATTICE_SHIFT 10
75 #define SAMPLE_SHIFT 4
76 #define LATTICE_FACTOR (1 << LATTICE_SHIFT)
77 #define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)
79 #define BASE_QUANT 0.6
80 #define RATE_VARIATION 3.0
82 static inline int shift(int a,int b)
84 return (a+(1<<(b-1))) >> b;
87 static inline int shift_down(int a,int b)
92 static av_always_inline av_flatten void put_symbol(RangeCoder *c, uint8_t *state, int v, int is_signed, uint64_t rc_stat[256][2], uint64_t rc_stat2[32][2]){
95 #define put_rac(C,S,B) \
99 rc_stat2[(S)-state][B]++;\
105 const int a= FFABS(v);
106 const int e= av_log2(a);
107 put_rac(c, state+0, 0);
110 put_rac(c, state+1+i, 1); //1..10
112 put_rac(c, state+1+i, 0);
114 for(i=e-1; i>=0; i--){
115 put_rac(c, state+22+i, (a>>i)&1); //22..31
119 put_rac(c, state+11 + e, v < 0); //11..21
122 put_rac(c, state+1+FFMIN(i,9), 1); //1..10
124 put_rac(c, state+1+9, 0);
126 for(i=e-1; i>=0; i--){
127 put_rac(c, state+22+FFMIN(i,9), (a>>i)&1); //22..31
131 put_rac(c, state+11 + 10, v < 0); //11..21
134 put_rac(c, state+0, 1);
139 static inline av_flatten int get_symbol(RangeCoder *c, uint8_t *state, int is_signed){
140 if(get_rac(c, state+0))
145 while(get_rac(c, state+1 + FFMIN(e,9))){ //1..10
150 for(i=e-1; i>=0; i--){
151 a += a + get_rac(c, state+22 + FFMIN(i,9)); //22..31
154 e= -(is_signed && get_rac(c, state+11 + FFMIN(e, 10))); //11..21
160 static inline int intlist_write(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part)
164 for (i = 0; i < entries; i++)
165 put_symbol(c, state, buf[i], 1, NULL, NULL);
170 static inline int intlist_read(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part)
174 for (i = 0; i < entries; i++)
175 buf[i] = get_symbol(c, state, 1);
180 static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
184 for (i = 0; i < entries; i++)
185 set_se_golomb(pb, buf[i]);
190 static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
194 for (i = 0; i < entries; i++)
195 buf[i] = get_se_golomb(gb);
202 #define ADAPT_LEVEL 8
204 static int bits_to_store(uint64_t x)
216 static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max)
223 bits = bits_to_store(max);
225 for (i = 0; i < bits-1; i++)
226 put_bits(pb, 1, value & (1 << i));
228 if ( (value | (1 << (bits-1))) <= max)
229 put_bits(pb, 1, value & (1 << (bits-1)));
232 static unsigned int read_uint_max(GetBitContext *gb, int max)
234 int i, bits, value = 0;
239 bits = bits_to_store(max);
241 for (i = 0; i < bits-1; i++)
245 if ( (value | (1<<(bits-1))) <= max)
247 value += 1 << (bits-1);
252 static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
254 int i, j, x = 0, low_bits = 0, max = 0;
255 int step = 256, pos = 0, dominant = 0, any = 0;
258 copy = av_calloc(entries, sizeof(*copy));
260 return AVERROR(ENOMEM);
266 for (i = 0; i < entries; i++)
267 energy += abs(buf[i]);
269 low_bits = bits_to_store(energy / (entries * 2));
273 put_bits(pb, 4, low_bits);
276 for (i = 0; i < entries; i++)
278 put_bits(pb, low_bits, abs(buf[i]));
279 copy[i] = abs(buf[i]) >> low_bits;
284 bits = av_calloc(entries*max, sizeof(*bits));
288 return AVERROR(ENOMEM);
291 for (i = 0; i <= max; i++)
293 for (j = 0; j < entries; j++)
295 bits[x++] = copy[j] > i;
301 int steplet = step >> 8;
303 if (pos + steplet > x)
306 for (i = 0; i < steplet; i++)
307 if (bits[i+pos] != dominant)
310 put_bits(pb, 1, any);
315 step += step / ADAPT_LEVEL;
321 while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
325 write_uint_max(pb, interloper, (step >> 8) - 1);
327 pos += interloper + 1;
328 step -= step / ADAPT_LEVEL;
334 dominant = !dominant;
339 for (i = 0; i < entries; i++)
341 put_bits(pb, 1, buf[i] < 0);
349 static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
351 int i, low_bits = 0, x = 0;
352 int n_zeros = 0, step = 256, dominant = 0;
353 int pos = 0, level = 0;
354 int *bits = av_calloc(entries, sizeof(*bits));
357 return AVERROR(ENOMEM);
361 low_bits = get_bits(gb, 4);
364 for (i = 0; i < entries; i++)
365 buf[i] = get_bits(gb, low_bits);
368 // av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits);
370 while (n_zeros < entries)
372 int steplet = step >> 8;
376 for (i = 0; i < steplet; i++)
377 bits[x++] = dominant;
382 step += step / ADAPT_LEVEL;
386 int actual_run = read_uint_max(gb, steplet-1);
388 // av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run);
390 for (i = 0; i < actual_run; i++)
391 bits[x++] = dominant;
393 bits[x++] = !dominant;
396 n_zeros += actual_run;
400 step -= step / ADAPT_LEVEL;
406 dominant = !dominant;
410 // reconstruct unsigned values
412 for (i = 0; n_zeros < entries; i++)
419 level += 1 << low_bits;
422 if (buf[pos] >= level)
429 buf[pos] += 1 << low_bits;
438 for (i = 0; i < entries; i++)
439 if (buf[i] && get_bits1(gb))
442 // av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos);
448 static void predictor_init_state(int *k, int *state, int order)
452 for (i = order-2; i >= 0; i--)
454 int j, p, x = state[i];
456 for (j = 0, p = i+1; p < order; j++,p++)
458 int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);
459 state[p] += shift_down(k[j]*x, LATTICE_SHIFT);
465 static int predictor_calc_error(int *k, int *state, int order, int error)
467 int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT);
470 int *k_ptr = &(k[order-2]),
471 *state_ptr = &(state[order-2]);
472 for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--)
474 int k_value = *k_ptr, state_value = *state_ptr;
475 x -= shift_down(k_value * state_value, LATTICE_SHIFT);
476 state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
479 for (i = order-2; i >= 0; i--)
481 x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
482 state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT);
486 // don't drift too far, to avoid overflows
487 if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16);
488 if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16);
495 #if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
496 // Heavily modified Levinson-Durbin algorithm which
497 // copes better with quantization, and calculates the
498 // actual whitened result as it goes.
500 static void modified_levinson_durbin(int *window, int window_entries,
501 int *out, int out_entries, int channels, int *tap_quant)
504 int *state = av_calloc(window_entries, sizeof(*state));
506 memcpy(state, window, 4* window_entries);
508 for (i = 0; i < out_entries; i++)
510 int step = (i+1)*channels, k, j;
511 double xx = 0.0, xy = 0.0;
513 int *x_ptr = &(window[step]);
514 int *state_ptr = &(state[0]);
515 j = window_entries - step;
516 for (;j>0;j--,x_ptr++,state_ptr++)
518 double x_value = *x_ptr;
519 double state_value = *state_ptr;
520 xx += state_value*state_value;
521 xy += x_value*state_value;
524 for (j = 0; j <= (window_entries - step); j++);
526 double stepval = window[step+j];
527 double stateval = window[j];
528 // xx += (double)window[j]*(double)window[j];
529 // xy += (double)window[step+j]*(double)window[j];
530 xx += stateval*stateval;
531 xy += stepval*stateval;
537 k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));
539 if (k > (LATTICE_FACTOR/tap_quant[i]))
540 k = LATTICE_FACTOR/tap_quant[i];
541 if (-k > (LATTICE_FACTOR/tap_quant[i]))
542 k = -(LATTICE_FACTOR/tap_quant[i]);
548 x_ptr = &(window[step]);
549 state_ptr = &(state[0]);
550 j = window_entries - step;
551 for (;j>0;j--,x_ptr++,state_ptr++)
553 int x_value = *x_ptr;
554 int state_value = *state_ptr;
555 *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
556 *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
559 for (j=0; j <= (window_entries - step); j++)
561 int stepval = window[step+j];
562 int stateval=state[j];
563 window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
564 state[j] += shift_down(k * stepval, LATTICE_SHIFT);
572 static inline int code_samplerate(int samplerate)
576 case 44100: return 0;
577 case 22050: return 1;
578 case 11025: return 2;
579 case 96000: return 3;
580 case 48000: return 4;
581 case 32000: return 5;
582 case 24000: return 6;
583 case 16000: return 7;
586 return AVERROR(EINVAL);
589 static av_cold int sonic_encode_init(AVCodecContext *avctx)
591 SonicContext *s = avctx->priv_data;
597 if (avctx->channels > MAX_CHANNELS)
599 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
600 return AVERROR(EINVAL); /* only stereo or mono for now */
603 if (avctx->channels == 2)
604 s->decorrelation = MID_SIDE;
606 s->decorrelation = 3;
608 if (avctx->codec->id == AV_CODEC_ID_SONIC_LS)
613 s->quantization = 0.0;
619 s->quantization = 1.0;
623 if (s->num_taps < 32 || s->num_taps > 1024 || s->num_taps % 32) {
624 av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
625 return AVERROR_INVALIDDATA;
629 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
630 for (i = 0; i < s->num_taps; i++)
631 s->tap_quant[i] = ff_sqrt(i+1);
633 s->channels = avctx->channels;
634 s->samplerate = avctx->sample_rate;
636 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
637 s->frame_size = s->channels*s->block_align*s->downsampling;
639 s->tail_size = s->num_taps*s->channels;
640 s->tail = av_calloc(s->tail_size, sizeof(*s->tail));
642 return AVERROR(ENOMEM);
644 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) );
646 return AVERROR(ENOMEM);
648 for (i = 0; i < s->channels; i++)
650 s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
651 if (!s->coded_samples[i])
652 return AVERROR(ENOMEM);
655 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
657 s->window_size = ((2*s->tail_size)+s->frame_size);
658 s->window = av_calloc(s->window_size, sizeof(*s->window));
660 return AVERROR(ENOMEM);
662 avctx->extradata = av_mallocz(16);
663 if (!avctx->extradata)
664 return AVERROR(ENOMEM);
665 init_put_bits(&pb, avctx->extradata, 16*8);
667 put_bits(&pb, 2, s->version); // version
670 if (s->version >= 2) {
671 put_bits(&pb, 8, s->version);
672 put_bits(&pb, 8, s->minor_version);
674 put_bits(&pb, 2, s->channels);
675 put_bits(&pb, 4, code_samplerate(s->samplerate));
677 put_bits(&pb, 1, s->lossless);
679 put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
680 put_bits(&pb, 2, s->decorrelation);
681 put_bits(&pb, 2, s->downsampling);
682 put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024
683 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table
686 avctx->extradata_size = put_bits_count(&pb)/8;
688 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
689 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
691 avctx->frame_size = s->block_align*s->downsampling;
696 static av_cold int sonic_encode_close(AVCodecContext *avctx)
698 SonicContext *s = avctx->priv_data;
701 for (i = 0; i < s->channels; i++)
702 av_freep(&s->coded_samples[i]);
704 av_freep(&s->predictor_k);
706 av_freep(&s->tap_quant);
707 av_freep(&s->window);
708 av_freep(&s->int_samples);
713 static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
714 const AVFrame *frame, int *got_packet_ptr)
716 SonicContext *s = avctx->priv_data;
718 int i, j, ch, quant = 0, x = 0;
720 const short *samples = (const int16_t*)frame->data[0];
723 if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0)
726 ff_init_range_encoder(&c, avpkt->data, avpkt->size);
727 ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8);
728 memset(state, 128, sizeof(state));
731 for (i = 0; i < s->frame_size; i++)
732 s->int_samples[i] = samples[i];
735 for (i = 0; i < s->frame_size; i++)
736 s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
738 switch(s->decorrelation)
741 for (i = 0; i < s->frame_size; i += s->channels)
743 s->int_samples[i] += s->int_samples[i+1];
744 s->int_samples[i+1] -= shift(s->int_samples[i], 1);
748 for (i = 0; i < s->frame_size; i += s->channels)
749 s->int_samples[i+1] -= s->int_samples[i];
752 for (i = 0; i < s->frame_size; i += s->channels)
753 s->int_samples[i] -= s->int_samples[i+1];
757 memset(s->window, 0, 4* s->window_size);
759 for (i = 0; i < s->tail_size; i++)
760 s->window[x++] = s->tail[i];
762 for (i = 0; i < s->frame_size; i++)
763 s->window[x++] = s->int_samples[i];
765 for (i = 0; i < s->tail_size; i++)
768 for (i = 0; i < s->tail_size; i++)
769 s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];
772 modified_levinson_durbin(s->window, s->window_size,
773 s->predictor_k, s->num_taps, s->channels, s->tap_quant);
774 if ((ret = intlist_write(&c, state, s->predictor_k, s->num_taps, 0)) < 0)
777 for (ch = 0; ch < s->channels; ch++)
780 for (i = 0; i < s->block_align; i++)
783 for (j = 0; j < s->downsampling; j++, x += s->channels)
785 s->coded_samples[ch][i] = sum;
789 // simple rate control code
792 double energy1 = 0.0, energy2 = 0.0;
793 for (ch = 0; ch < s->channels; ch++)
795 for (i = 0; i < s->block_align; i++)
797 double sample = s->coded_samples[ch][i];
798 energy2 += sample*sample;
799 energy1 += fabs(sample);
803 energy2 = sqrt(energy2/(s->channels*s->block_align));
804 energy1 = M_SQRT2*energy1/(s->channels*s->block_align);
806 // increase bitrate when samples are like a gaussian distribution
807 // reduce bitrate when samples are like a two-tailed exponential distribution
809 if (energy2 > energy1)
810 energy2 += (energy2-energy1)*RATE_VARIATION;
812 quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
813 // av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);
815 quant = av_clip(quant, 1, 65534);
817 put_symbol(&c, state, quant, 0, NULL, NULL);
819 quant *= SAMPLE_FACTOR;
822 // write out coded samples
823 for (ch = 0; ch < s->channels; ch++)
826 for (i = 0; i < s->block_align; i++)
827 s->coded_samples[ch][i] = ROUNDED_DIV(s->coded_samples[ch][i], quant);
829 if ((ret = intlist_write(&c, state, s->coded_samples[ch], s->block_align, 1)) < 0)
833 // av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
835 avpkt->size = ff_rac_terminate(&c);
840 #endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */
842 #if CONFIG_SONIC_DECODER
843 static const int samplerate_table[] =
844 { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
846 static av_cold int sonic_decode_init(AVCodecContext *avctx)
848 SonicContext *s = avctx->priv_data;
852 s->channels = avctx->channels;
853 s->samplerate = avctx->sample_rate;
855 if (!avctx->extradata)
857 av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
858 return AVERROR_INVALIDDATA;
861 init_get_bits8(&gb, avctx->extradata, avctx->extradata_size);
863 s->version = get_bits(&gb, 2);
864 if (s->version >= 2) {
865 s->version = get_bits(&gb, 8);
866 s->minor_version = get_bits(&gb, 8);
870 av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
871 return AVERROR_INVALIDDATA;
876 s->channels = get_bits(&gb, 2);
877 s->samplerate = samplerate_table[get_bits(&gb, 4)];
878 av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
879 s->channels, s->samplerate);
882 if (s->channels > MAX_CHANNELS)
884 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
885 return AVERROR_INVALIDDATA;
888 s->lossless = get_bits1(&gb);
890 skip_bits(&gb, 3); // XXX FIXME
891 s->decorrelation = get_bits(&gb, 2);
892 if (s->decorrelation != 3 && s->channels != 2) {
893 av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation);
894 return AVERROR_INVALIDDATA;
897 s->downsampling = get_bits(&gb, 2);
898 if (!s->downsampling) {
899 av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n");
900 return AVERROR_INVALIDDATA;
903 s->num_taps = (get_bits(&gb, 5)+1)<<5;
904 if (get_bits1(&gb)) // XXX FIXME
905 av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
907 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
908 s->frame_size = s->channels*s->block_align*s->downsampling;
909 // avctx->frame_size = s->block_align;
911 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
912 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
915 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
916 for (i = 0; i < s->num_taps; i++)
917 s->tap_quant[i] = ff_sqrt(i+1);
919 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k));
921 for (i = 0; i < s->channels; i++)
923 s->predictor_state[i] = av_calloc(s->num_taps, sizeof(**s->predictor_state));
924 if (!s->predictor_state[i])
925 return AVERROR(ENOMEM);
928 for (i = 0; i < s->channels; i++)
930 s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
931 if (!s->coded_samples[i])
932 return AVERROR(ENOMEM);
934 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
936 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
940 static av_cold int sonic_decode_close(AVCodecContext *avctx)
942 SonicContext *s = avctx->priv_data;
945 av_freep(&s->int_samples);
946 av_freep(&s->tap_quant);
947 av_freep(&s->predictor_k);
949 for (i = 0; i < s->channels; i++)
951 av_freep(&s->predictor_state[i]);
952 av_freep(&s->coded_samples[i]);
958 static int sonic_decode_frame(AVCodecContext *avctx,
959 void *data, int *got_frame_ptr,
962 const uint8_t *buf = avpkt->data;
963 int buf_size = avpkt->size;
964 SonicContext *s = avctx->priv_data;
967 int i, quant, ch, j, ret;
969 AVFrame *frame = data;
971 if (buf_size == 0) return 0;
973 frame->nb_samples = s->frame_size / avctx->channels;
974 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
976 samples = (int16_t *)frame->data[0];
978 // av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
980 memset(state, 128, sizeof(state));
981 ff_init_range_decoder(&c, buf, buf_size);
982 ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8);
984 intlist_read(&c, state, s->predictor_k, s->num_taps, 0);
987 for (i = 0; i < s->num_taps; i++)
988 s->predictor_k[i] *= s->tap_quant[i];
993 quant = get_symbol(&c, state, 0) * SAMPLE_FACTOR;
995 // av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
997 for (ch = 0; ch < s->channels; ch++)
1001 predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);
1003 intlist_read(&c, state, s->coded_samples[ch], s->block_align, 1);
1005 for (i = 0; i < s->block_align; i++)
1007 for (j = 0; j < s->downsampling - 1; j++)
1009 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
1013 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
1017 for (i = 0; i < s->num_taps; i++)
1018 s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
1021 switch(s->decorrelation)
1024 for (i = 0; i < s->frame_size; i += s->channels)
1026 s->int_samples[i+1] += shift(s->int_samples[i], 1);
1027 s->int_samples[i] -= s->int_samples[i+1];
1031 for (i = 0; i < s->frame_size; i += s->channels)
1032 s->int_samples[i+1] += s->int_samples[i];
1035 for (i = 0; i < s->frame_size; i += s->channels)
1036 s->int_samples[i] += s->int_samples[i+1];
1041 for (i = 0; i < s->frame_size; i++)
1042 s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);
1044 // internal -> short
1045 for (i = 0; i < s->frame_size; i++)
1046 samples[i] = av_clip_int16(s->int_samples[i]);
1053 AVCodec ff_sonic_decoder = {
1055 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
1056 .type = AVMEDIA_TYPE_AUDIO,
1057 .id = AV_CODEC_ID_SONIC,
1058 .priv_data_size = sizeof(SonicContext),
1059 .init = sonic_decode_init,
1060 .close = sonic_decode_close,
1061 .decode = sonic_decode_frame,
1062 .capabilities = CODEC_CAP_DR1 | CODEC_CAP_EXPERIMENTAL,
1064 #endif /* CONFIG_SONIC_DECODER */
1066 #if CONFIG_SONIC_ENCODER
1067 AVCodec ff_sonic_encoder = {
1069 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
1070 .type = AVMEDIA_TYPE_AUDIO,
1071 .id = AV_CODEC_ID_SONIC,
1072 .priv_data_size = sizeof(SonicContext),
1073 .init = sonic_encode_init,
1074 .encode2 = sonic_encode_frame,
1075 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
1076 .capabilities = CODEC_CAP_EXPERIMENTAL,
1077 .close = sonic_encode_close,
1081 #if CONFIG_SONIC_LS_ENCODER
1082 AVCodec ff_sonic_ls_encoder = {
1084 .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
1085 .type = AVMEDIA_TYPE_AUDIO,
1086 .id = AV_CODEC_ID_SONIC_LS,
1087 .priv_data_size = sizeof(SonicContext),
1088 .init = sonic_encode_init,
1089 .encode2 = sonic_encode_frame,
1090 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
1091 .capabilities = CODEC_CAP_EXPERIMENTAL,
1092 .close = sonic_encode_close,