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.
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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
28 * Simple free lossless/lossy audio codec
29 * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
30 * Written and designed by Alex Beregszaszi
33 * - CABAC put/get_symbol
34 * - independent quantizer for channels
35 * - >2 channels support
36 * - more decorrelation types
37 * - more tap_quant tests
38 * - selectable intlist writers/readers (bonk-style, golomb, cabac)
41 #define MAX_CHANNELS 2
47 typedef struct SonicContext {
48 int lossless, decorrelation;
50 int num_taps, downsampling;
53 int channels, samplerate, block_align, frame_size;
57 int *coded_samples[MAX_CHANNELS];
67 int *predictor_state[MAX_CHANNELS];
70 #define LATTICE_SHIFT 10
71 #define SAMPLE_SHIFT 4
72 #define LATTICE_FACTOR (1 << LATTICE_SHIFT)
73 #define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)
75 #define BASE_QUANT 0.6
76 #define RATE_VARIATION 3.0
78 static inline int divide(int a, int b)
81 return -( (-a + b/2)/b );
86 static inline int shift(int a,int b)
88 return (a+(1<<(b-1))) >> b;
91 static inline int shift_down(int a,int b)
93 return (a>>b)+((a<0)?1:0);
97 static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
101 for (i = 0; i < entries; i++)
102 set_se_golomb(pb, buf[i]);
107 static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
111 for (i = 0; i < entries; i++)
112 buf[i] = get_se_golomb(gb);
119 #define ADAPT_LEVEL 8
121 static int bits_to_store(uint64_t x)
133 static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max)
140 bits = bits_to_store(max);
142 for (i = 0; i < bits-1; i++)
143 put_bits(pb, 1, value & (1 << i));
145 if ( (value | (1 << (bits-1))) <= max)
146 put_bits(pb, 1, value & (1 << (bits-1)));
149 static unsigned int read_uint_max(GetBitContext *gb, int max)
151 int i, bits, value = 0;
156 bits = bits_to_store(max);
158 for (i = 0; i < bits-1; i++)
162 if ( (value | (1<<(bits-1))) <= max)
164 value += 1 << (bits-1);
169 static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
171 int i, j, x = 0, low_bits = 0, max = 0;
172 int step = 256, pos = 0, dominant = 0, any = 0;
175 copy = av_mallocz(4* entries);
183 for (i = 0; i < entries; i++)
184 energy += abs(buf[i]);
186 low_bits = bits_to_store(energy / (entries * 2));
190 put_bits(pb, 4, low_bits);
193 for (i = 0; i < entries; i++)
195 put_bits(pb, low_bits, abs(buf[i]));
196 copy[i] = abs(buf[i]) >> low_bits;
201 bits = av_mallocz(4* entries*max);
208 for (i = 0; i <= max; i++)
210 for (j = 0; j < entries; j++)
212 bits[x++] = copy[j] > i;
218 int steplet = step >> 8;
220 if (pos + steplet > x)
223 for (i = 0; i < steplet; i++)
224 if (bits[i+pos] != dominant)
227 put_bits(pb, 1, any);
232 step += step / ADAPT_LEVEL;
238 while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
242 write_uint_max(pb, interloper, (step >> 8) - 1);
244 pos += interloper + 1;
245 step -= step / ADAPT_LEVEL;
251 dominant = !dominant;
256 for (i = 0; i < entries; i++)
258 put_bits(pb, 1, buf[i] < 0);
266 static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
268 int i, low_bits = 0, x = 0;
269 int n_zeros = 0, step = 256, dominant = 0;
270 int pos = 0, level = 0;
271 int *bits = av_mallocz(4* entries);
278 low_bits = get_bits(gb, 4);
281 for (i = 0; i < entries; i++)
282 buf[i] = get_bits(gb, low_bits);
285 // av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits);
287 while (n_zeros < entries)
289 int steplet = step >> 8;
293 for (i = 0; i < steplet; i++)
294 bits[x++] = dominant;
299 step += step / ADAPT_LEVEL;
303 int actual_run = read_uint_max(gb, steplet-1);
305 // av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run);
307 for (i = 0; i < actual_run; i++)
308 bits[x++] = dominant;
310 bits[x++] = !dominant;
313 n_zeros += actual_run;
317 step -= step / ADAPT_LEVEL;
323 dominant = !dominant;
327 // reconstruct unsigned values
329 for (i = 0; n_zeros < entries; i++)
336 level += 1 << low_bits;
339 if (buf[pos] >= level)
346 buf[pos] += 1 << low_bits;
355 for (i = 0; i < entries; i++)
356 if (buf[i] && get_bits1(gb))
359 // av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos);
365 static void predictor_init_state(int *k, int *state, int order)
369 for (i = order-2; i >= 0; i--)
371 int j, p, x = state[i];
373 for (j = 0, p = i+1; p < order; j++,p++)
375 int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);
376 state[p] += shift_down(k[j]*x, LATTICE_SHIFT);
382 static int predictor_calc_error(int *k, int *state, int order, int error)
384 int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT);
387 int *k_ptr = &(k[order-2]),
388 *state_ptr = &(state[order-2]);
389 for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--)
391 int k_value = *k_ptr, state_value = *state_ptr;
392 x -= shift_down(k_value * state_value, LATTICE_SHIFT);
393 state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
396 for (i = order-2; i >= 0; i--)
398 x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
399 state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT);
403 // don't drift too far, to avoid overflows
404 if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16);
405 if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16);
412 #if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
413 // Heavily modified Levinson-Durbin algorithm which
414 // copes better with quantization, and calculates the
415 // actual whitened result as it goes.
417 static void modified_levinson_durbin(int *window, int window_entries,
418 int *out, int out_entries, int channels, int *tap_quant)
421 int *state = av_mallocz(4* window_entries);
423 memcpy(state, window, 4* window_entries);
425 for (i = 0; i < out_entries; i++)
427 int step = (i+1)*channels, k, j;
428 double xx = 0.0, xy = 0.0;
430 int *x_ptr = &(window[step]), *state_ptr = &(state[0]);
431 j = window_entries - step;
432 for (;j>=0;j--,x_ptr++,state_ptr++)
434 double x_value = *x_ptr, state_value = *state_ptr;
435 xx += state_value*state_value;
436 xy += x_value*state_value;
439 for (j = 0; j <= (window_entries - step); j++);
441 double stepval = window[step+j], stateval = window[j];
442 // xx += (double)window[j]*(double)window[j];
443 // xy += (double)window[step+j]*(double)window[j];
444 xx += stateval*stateval;
445 xy += stepval*stateval;
451 k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));
453 if (k > (LATTICE_FACTOR/tap_quant[i]))
454 k = LATTICE_FACTOR/tap_quant[i];
455 if (-k > (LATTICE_FACTOR/tap_quant[i]))
456 k = -(LATTICE_FACTOR/tap_quant[i]);
462 x_ptr = &(window[step]);
463 state_ptr = &(state[0]);
464 j = window_entries - step;
465 for (;j>=0;j--,x_ptr++,state_ptr++)
467 int x_value = *x_ptr, state_value = *state_ptr;
468 *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
469 *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
472 for (j=0; j <= (window_entries - step); j++)
474 int stepval = window[step+j], stateval=state[j];
475 window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
476 state[j] += shift_down(k * stepval, LATTICE_SHIFT);
484 static inline int code_samplerate(int samplerate)
488 case 44100: return 0;
489 case 22050: return 1;
490 case 11025: return 2;
491 case 96000: return 3;
492 case 48000: return 4;
493 case 32000: return 5;
494 case 24000: return 6;
495 case 16000: return 7;
501 static av_cold int sonic_encode_init(AVCodecContext *avctx)
503 SonicContext *s = avctx->priv_data;
507 if (avctx->channels > MAX_CHANNELS)
509 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
510 return -1; /* only stereo or mono for now */
513 if (avctx->channels == 2)
514 s->decorrelation = MID_SIDE;
516 s->decorrelation = 3;
518 if (avctx->codec->id == AV_CODEC_ID_SONIC_LS)
523 s->quantization = 0.0;
529 s->quantization = 1.0;
533 if ((s->num_taps < 32) || (s->num_taps > 1024) ||
534 ((s->num_taps>>5)<<5 != s->num_taps))
536 av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
541 s->tap_quant = av_mallocz(4* s->num_taps);
542 for (i = 0; i < s->num_taps; i++)
543 s->tap_quant[i] = (int)(sqrt(i+1));
545 s->channels = avctx->channels;
546 s->samplerate = avctx->sample_rate;
548 s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling;
549 s->frame_size = s->channels*s->block_align*s->downsampling;
551 s->tail_size = s->num_taps*s->channels;
552 s->tail = av_mallocz(4 * s->tail_size);
556 s->predictor_k = av_mallocz(4 * s->num_taps);
560 for (i = 0; i < s->channels; i++)
562 s->coded_samples[i] = av_mallocz(4* s->block_align);
563 if (!s->coded_samples[i])
567 s->int_samples = av_mallocz(4* s->frame_size);
569 s->window_size = ((2*s->tail_size)+s->frame_size);
570 s->window = av_mallocz(4* s->window_size);
574 avctx->extradata = av_mallocz(16);
575 if (!avctx->extradata)
577 init_put_bits(&pb, avctx->extradata, 16*8);
579 put_bits(&pb, 2, version); // version
582 put_bits(&pb, 2, s->channels);
583 put_bits(&pb, 4, code_samplerate(s->samplerate));
585 put_bits(&pb, 1, s->lossless);
587 put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
588 put_bits(&pb, 2, s->decorrelation);
589 put_bits(&pb, 2, s->downsampling);
590 put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024
591 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table
594 avctx->extradata_size = put_bits_count(&pb)/8;
596 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
597 version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
599 avctx->coded_frame = avcodec_alloc_frame();
600 if (!avctx->coded_frame)
601 return AVERROR(ENOMEM);
602 avctx->coded_frame->key_frame = 1;
603 avctx->frame_size = s->block_align*s->downsampling;
608 static av_cold int sonic_encode_close(AVCodecContext *avctx)
610 SonicContext *s = avctx->priv_data;
613 av_freep(&avctx->coded_frame);
615 for (i = 0; i < s->channels; i++)
616 av_free(s->coded_samples[i]);
618 av_free(s->predictor_k);
620 av_free(s->tap_quant);
622 av_free(s->int_samples);
627 static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
628 const AVFrame *frame, int *got_packet_ptr)
630 SonicContext *s = avctx->priv_data;
632 int i, j, ch, quant = 0, x = 0;
634 const short *samples = (const int16_t*)frame->data[0];
636 if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0)
639 init_put_bits(&pb, avpkt->data, avpkt->size);
642 for (i = 0; i < s->frame_size; i++)
643 s->int_samples[i] = samples[i];
646 for (i = 0; i < s->frame_size; i++)
647 s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
649 switch(s->decorrelation)
652 for (i = 0; i < s->frame_size; i += s->channels)
654 s->int_samples[i] += s->int_samples[i+1];
655 s->int_samples[i+1] -= shift(s->int_samples[i], 1);
659 for (i = 0; i < s->frame_size; i += s->channels)
660 s->int_samples[i+1] -= s->int_samples[i];
663 for (i = 0; i < s->frame_size; i += s->channels)
664 s->int_samples[i] -= s->int_samples[i+1];
668 memset(s->window, 0, 4* s->window_size);
670 for (i = 0; i < s->tail_size; i++)
671 s->window[x++] = s->tail[i];
673 for (i = 0; i < s->frame_size; i++)
674 s->window[x++] = s->int_samples[i];
676 for (i = 0; i < s->tail_size; i++)
679 for (i = 0; i < s->tail_size; i++)
680 s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];
683 modified_levinson_durbin(s->window, s->window_size,
684 s->predictor_k, s->num_taps, s->channels, s->tap_quant);
685 if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0)
688 for (ch = 0; ch < s->channels; ch++)
691 for (i = 0; i < s->block_align; i++)
694 for (j = 0; j < s->downsampling; j++, x += s->channels)
696 s->coded_samples[ch][i] = sum;
700 // simple rate control code
703 double energy1 = 0.0, energy2 = 0.0;
704 for (ch = 0; ch < s->channels; ch++)
706 for (i = 0; i < s->block_align; i++)
708 double sample = s->coded_samples[ch][i];
709 energy2 += sample*sample;
710 energy1 += fabs(sample);
714 energy2 = sqrt(energy2/(s->channels*s->block_align));
715 energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align);
717 // increase bitrate when samples are like a gaussian distribution
718 // reduce bitrate when samples are like a two-tailed exponential distribution
720 if (energy2 > energy1)
721 energy2 += (energy2-energy1)*RATE_VARIATION;
723 quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
724 // av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);
731 set_ue_golomb(&pb, quant);
733 quant *= SAMPLE_FACTOR;
736 // write out coded samples
737 for (ch = 0; ch < s->channels; ch++)
740 for (i = 0; i < s->block_align; i++)
741 s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant);
743 if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0)
747 // av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
750 avpkt->size = (put_bits_count(&pb)+7)/8;
754 #endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */
756 #if CONFIG_SONIC_DECODER
757 static const int samplerate_table[] =
758 { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
760 static av_cold int sonic_decode_init(AVCodecContext *avctx)
762 SonicContext *s = avctx->priv_data;
766 s->channels = avctx->channels;
767 s->samplerate = avctx->sample_rate;
769 if (!avctx->extradata)
771 av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
775 init_get_bits(&gb, avctx->extradata, avctx->extradata_size);
777 version = get_bits(&gb, 2);
780 av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
786 s->channels = get_bits(&gb, 2);
787 s->samplerate = samplerate_table[get_bits(&gb, 4)];
788 av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
789 s->channels, s->samplerate);
792 if (s->channels > MAX_CHANNELS)
794 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
798 s->lossless = get_bits1(&gb);
800 skip_bits(&gb, 3); // XXX FIXME
801 s->decorrelation = get_bits(&gb, 2);
802 if (s->decorrelation != 3 && s->channels != 2) {
803 av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation);
804 return AVERROR_INVALIDDATA;
807 s->downsampling = get_bits(&gb, 2);
808 if (!s->downsampling) {
809 av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n");
810 return AVERROR_INVALIDDATA;
813 s->num_taps = (get_bits(&gb, 5)+1)<<5;
814 if (get_bits1(&gb)) // XXX FIXME
815 av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
817 s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling;
818 s->frame_size = s->channels*s->block_align*s->downsampling;
819 // avctx->frame_size = s->block_align;
821 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
822 version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
825 s->tap_quant = av_mallocz(4* s->num_taps);
826 for (i = 0; i < s->num_taps; i++)
827 s->tap_quant[i] = (int)(sqrt(i+1));
829 s->predictor_k = av_mallocz(4* s->num_taps);
831 for (i = 0; i < s->channels; i++)
833 s->predictor_state[i] = av_mallocz(4* s->num_taps);
834 if (!s->predictor_state[i])
838 for (i = 0; i < s->channels; i++)
840 s->coded_samples[i] = av_mallocz(4* s->block_align);
841 if (!s->coded_samples[i])
844 s->int_samples = av_mallocz(4* s->frame_size);
846 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
850 static av_cold int sonic_decode_close(AVCodecContext *avctx)
852 SonicContext *s = avctx->priv_data;
855 av_free(s->int_samples);
856 av_free(s->tap_quant);
857 av_free(s->predictor_k);
859 for (i = 0; i < s->channels; i++)
861 av_free(s->predictor_state[i]);
862 av_free(s->coded_samples[i]);
868 static int sonic_decode_frame(AVCodecContext *avctx,
869 void *data, int *got_frame_ptr,
872 const uint8_t *buf = avpkt->data;
873 int buf_size = avpkt->size;
874 SonicContext *s = avctx->priv_data;
876 int i, quant, ch, j, ret;
878 AVFrame *frame = data;
880 if (buf_size == 0) return 0;
882 frame->nb_samples = s->frame_size / avctx->channels;
883 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
885 samples = (int16_t *)frame->data[0];
887 // av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
889 init_get_bits(&gb, buf, buf_size*8);
891 intlist_read(&gb, s->predictor_k, s->num_taps, 0);
894 for (i = 0; i < s->num_taps; i++)
895 s->predictor_k[i] *= s->tap_quant[i];
900 quant = get_ue_golomb(&gb) * SAMPLE_FACTOR;
902 // av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
904 for (ch = 0; ch < s->channels; ch++)
908 predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);
910 intlist_read(&gb, s->coded_samples[ch], s->block_align, 1);
912 for (i = 0; i < s->block_align; i++)
914 for (j = 0; j < s->downsampling - 1; j++)
916 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
920 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
924 for (i = 0; i < s->num_taps; i++)
925 s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
928 switch(s->decorrelation)
931 for (i = 0; i < s->frame_size; i += s->channels)
933 s->int_samples[i+1] += shift(s->int_samples[i], 1);
934 s->int_samples[i] -= s->int_samples[i+1];
938 for (i = 0; i < s->frame_size; i += s->channels)
939 s->int_samples[i+1] += s->int_samples[i];
942 for (i = 0; i < s->frame_size; i += s->channels)
943 s->int_samples[i] += s->int_samples[i+1];
948 for (i = 0; i < s->frame_size; i++)
949 s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);
952 for (i = 0; i < s->frame_size; i++)
953 samples[i] = av_clip_int16(s->int_samples[i]);
959 return (get_bits_count(&gb)+7)/8;
962 AVCodec ff_sonic_decoder = {
964 .type = AVMEDIA_TYPE_AUDIO,
965 .id = AV_CODEC_ID_SONIC,
966 .priv_data_size = sizeof(SonicContext),
967 .init = sonic_decode_init,
968 .close = sonic_decode_close,
969 .decode = sonic_decode_frame,
970 .capabilities = CODEC_CAP_DR1 | CODEC_CAP_EXPERIMENTAL,
971 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
973 #endif /* CONFIG_SONIC_DECODER */
975 #if CONFIG_SONIC_ENCODER
976 AVCodec ff_sonic_encoder = {
978 .type = AVMEDIA_TYPE_AUDIO,
979 .id = AV_CODEC_ID_SONIC,
980 .priv_data_size = sizeof(SonicContext),
981 .init = sonic_encode_init,
982 .encode2 = sonic_encode_frame,
983 .capabilities = CODEC_CAP_EXPERIMENTAL,
984 .close = sonic_encode_close,
985 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
989 #if CONFIG_SONIC_LS_ENCODER
990 AVCodec ff_sonic_ls_encoder = {
992 .type = AVMEDIA_TYPE_AUDIO,
993 .id = AV_CODEC_ID_SONIC_LS,
994 .priv_data_size = sizeof(SonicContext),
995 .init = sonic_encode_init,
996 .encode2 = sonic_encode_frame,
997 .capabilities = CODEC_CAP_EXPERIMENTAL,
998 .close = sonic_encode_close,
999 .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),