Mercurial > libavcodec.hg
view sonic.c @ 10554:deabe9f7571b libavcodec
Add support for two pass encoding in libtheora
author | conrad |
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date | Sun, 22 Nov 2009 21:08:40 +0000 |
parents | 4ebcb6c121e4 |
children | 8a4984c5cacc |
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/* * Simple free lossless/lossy audio codec * Copyright (c) 2004 Alex Beregszaszi * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avcodec.h" #include "get_bits.h" #include "golomb.h" /** * @file libavcodec/sonic.c * Simple free lossless/lossy audio codec * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk) * Written and designed by Alex Beregszaszi * * TODO: * - CABAC put/get_symbol * - independent quantizer for channels * - >2 channels support * - more decorrelation types * - more tap_quant tests * - selectable intlist writers/readers (bonk-style, golomb, cabac) */ #define MAX_CHANNELS 2 #define MID_SIDE 0 #define LEFT_SIDE 1 #define RIGHT_SIDE 2 typedef struct SonicContext { int lossless, decorrelation; int num_taps, downsampling; double quantization; int channels, samplerate, block_align, frame_size; int *tap_quant; int *int_samples; int *coded_samples[MAX_CHANNELS]; // for encoding int *tail; int tail_size; int *window; int window_size; // for decoding int *predictor_k; int *predictor_state[MAX_CHANNELS]; } SonicContext; #define LATTICE_SHIFT 10 #define SAMPLE_SHIFT 4 #define LATTICE_FACTOR (1 << LATTICE_SHIFT) #define SAMPLE_FACTOR (1 << SAMPLE_SHIFT) #define BASE_QUANT 0.6 #define RATE_VARIATION 3.0 static inline int divide(int a, int b) { if (a < 0) return -( (-a + b/2)/b ); else return (a + b/2)/b; } static inline int shift(int a,int b) { return (a+(1<<(b-1))) >> b; } static inline int shift_down(int a,int b) { return (a>>b)+((a<0)?1:0); } #if 1 static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part) { int i; for (i = 0; i < entries; i++) set_se_golomb(pb, buf[i]); return 1; } static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part) { int i; for (i = 0; i < entries; i++) buf[i] = get_se_golomb(gb); return 1; } #else #define ADAPT_LEVEL 8 static int bits_to_store(uint64_t x) { int res = 0; while(x) { res++; x >>= 1; } return res; } static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max) { int i, bits; if (!max) return; bits = bits_to_store(max); for (i = 0; i < bits-1; i++) put_bits(pb, 1, value & (1 << i)); if ( (value | (1 << (bits-1))) <= max) put_bits(pb, 1, value & (1 << (bits-1))); } static unsigned int read_uint_max(GetBitContext *gb, int max) { int i, bits, value = 0; if (!max) return 0; bits = bits_to_store(max); for (i = 0; i < bits-1; i++) if (get_bits1(gb)) value += 1 << i; if ( (value | (1<<(bits-1))) <= max) if (get_bits1(gb)) value += 1 << (bits-1); return value; } static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part) { int i, j, x = 0, low_bits = 0, max = 0; int step = 256, pos = 0, dominant = 0, any = 0; int *copy, *bits; copy = av_mallocz(4* entries); if (!copy) return -1; if (base_2_part) { int energy = 0; for (i = 0; i < entries; i++) energy += abs(buf[i]); low_bits = bits_to_store(energy / (entries * 2)); if (low_bits > 15) low_bits = 15; put_bits(pb, 4, low_bits); } for (i = 0; i < entries; i++) { put_bits(pb, low_bits, abs(buf[i])); copy[i] = abs(buf[i]) >> low_bits; if (copy[i] > max) max = abs(copy[i]); } bits = av_mallocz(4* entries*max); if (!bits) { // av_free(copy); return -1; } for (i = 0; i <= max; i++) { for (j = 0; j < entries; j++) if (copy[j] >= i) bits[x++] = copy[j] > i; } // store bitstream while (pos < x) { int steplet = step >> 8; if (pos + steplet > x) steplet = x - pos; for (i = 0; i < steplet; i++) if (bits[i+pos] != dominant) any = 1; put_bits(pb, 1, any); if (!any) { pos += steplet; step += step / ADAPT_LEVEL; } else { int interloper = 0; while (((pos + interloper) < x) && (bits[pos + interloper] == dominant)) interloper++; // note change write_uint_max(pb, interloper, (step >> 8) - 1); pos += interloper + 1; step -= step / ADAPT_LEVEL; } if (step < 256) { step = 65536 / step; dominant = !dominant; } } // store signs for (i = 0; i < entries; i++) if (buf[i]) put_bits(pb, 1, buf[i] < 0); // av_free(bits); // av_free(copy); return 0; } static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part) { int i, low_bits = 0, x = 0; int n_zeros = 0, step = 256, dominant = 0; int pos = 0, level = 0; int *bits = av_mallocz(4* entries); if (!bits) return -1; if (base_2_part) { low_bits = get_bits(gb, 4); if (low_bits) for (i = 0; i < entries; i++) buf[i] = get_bits(gb, low_bits); } // av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits); while (n_zeros < entries) { int steplet = step >> 8; if (!get_bits1(gb)) { for (i = 0; i < steplet; i++) bits[x++] = dominant; if (!dominant) n_zeros += steplet; step += step / ADAPT_LEVEL; } else { int actual_run = read_uint_max(gb, steplet-1); // av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run); for (i = 0; i < actual_run; i++) bits[x++] = dominant; bits[x++] = !dominant; if (!dominant) n_zeros += actual_run; else n_zeros++; step -= step / ADAPT_LEVEL; } if (step < 256) { step = 65536 / step; dominant = !dominant; } } // reconstruct unsigned values n_zeros = 0; for (i = 0; n_zeros < entries; i++) { while(1) { if (pos >= entries) { pos = 0; level += 1 << low_bits; } if (buf[pos] >= level) break; pos++; } if (bits[i]) buf[pos] += 1 << low_bits; else n_zeros++; pos++; } // av_free(bits); // read signs for (i = 0; i < entries; i++) if (buf[i] && get_bits1(gb)) buf[i] = -buf[i]; // av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos); return 0; } #endif static void predictor_init_state(int *k, int *state, int order) { int i; for (i = order-2; i >= 0; i--) { int j, p, x = state[i]; for (j = 0, p = i+1; p < order; j++,p++) { int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT); state[p] += shift_down(k[j]*x, LATTICE_SHIFT); x = tmp; } } } static int predictor_calc_error(int *k, int *state, int order, int error) { int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT); #if 1 int *k_ptr = &(k[order-2]), *state_ptr = &(state[order-2]); for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--) { int k_value = *k_ptr, state_value = *state_ptr; x -= shift_down(k_value * state_value, LATTICE_SHIFT); state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT); } #else for (i = order-2; i >= 0; i--) { x -= shift_down(k[i] * state[i], LATTICE_SHIFT); state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT); } #endif // don't drift too far, to avoid overflows if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16); if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16); state[0] = x; return x; } #if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER // Heavily modified Levinson-Durbin algorithm which // copes better with quantization, and calculates the // actual whitened result as it goes. static void modified_levinson_durbin(int *window, int window_entries, int *out, int out_entries, int channels, int *tap_quant) { int i; int *state = av_mallocz(4* window_entries); memcpy(state, window, 4* window_entries); for (i = 0; i < out_entries; i++) { int step = (i+1)*channels, k, j; double xx = 0.0, xy = 0.0; #if 1 int *x_ptr = &(window[step]), *state_ptr = &(state[0]); j = window_entries - step; for (;j>=0;j--,x_ptr++,state_ptr++) { double x_value = *x_ptr, state_value = *state_ptr; xx += state_value*state_value; xy += x_value*state_value; } #else for (j = 0; j <= (window_entries - step); j++); { double stepval = window[step+j], stateval = window[j]; // xx += (double)window[j]*(double)window[j]; // xy += (double)window[step+j]*(double)window[j]; xx += stateval*stateval; xy += stepval*stateval; } #endif if (xx == 0.0) k = 0; else k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5)); if (k > (LATTICE_FACTOR/tap_quant[i])) k = LATTICE_FACTOR/tap_quant[i]; if (-k > (LATTICE_FACTOR/tap_quant[i])) k = -(LATTICE_FACTOR/tap_quant[i]); out[i] = k; k *= tap_quant[i]; #if 1 x_ptr = &(window[step]); state_ptr = &(state[0]); j = window_entries - step; for (;j>=0;j--,x_ptr++,state_ptr++) { int x_value = *x_ptr, state_value = *state_ptr; *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT); *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT); } #else for (j=0; j <= (window_entries - step); j++) { int stepval = window[step+j], stateval=state[j]; window[step+j] += shift_down(k * stateval, LATTICE_SHIFT); state[j] += shift_down(k * stepval, LATTICE_SHIFT); } #endif } av_free(state); } static inline int code_samplerate(int samplerate) { switch (samplerate) { case 44100: return 0; case 22050: return 1; case 11025: return 2; case 96000: return 3; case 48000: return 4; case 32000: return 5; case 24000: return 6; case 16000: return 7; case 8000: return 8; } return -1; } static av_cold int sonic_encode_init(AVCodecContext *avctx) { SonicContext *s = avctx->priv_data; PutBitContext pb; int i, version = 0; if (avctx->channels > MAX_CHANNELS) { av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); return -1; /* only stereo or mono for now */ } if (avctx->channels == 2) s->decorrelation = MID_SIDE; if (avctx->codec->id == CODEC_ID_SONIC_LS) { s->lossless = 1; s->num_taps = 32; s->downsampling = 1; s->quantization = 0.0; } else { s->num_taps = 128; s->downsampling = 2; s->quantization = 1.0; } // max tap 2048 if ((s->num_taps < 32) || (s->num_taps > 1024) || ((s->num_taps>>5)<<5 != s->num_taps)) { av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n"); return -1; } // generate taps s->tap_quant = av_mallocz(4* s->num_taps); for (i = 0; i < s->num_taps; i++) s->tap_quant[i] = (int)(sqrt(i+1)); s->channels = avctx->channels; s->samplerate = avctx->sample_rate; s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling; s->frame_size = s->channels*s->block_align*s->downsampling; s->tail = av_mallocz(4* s->num_taps*s->channels); if (!s->tail) return -1; s->tail_size = s->num_taps*s->channels; s->predictor_k = av_mallocz(4 * s->num_taps); if (!s->predictor_k) return -1; for (i = 0; i < s->channels; i++) { s->coded_samples[i] = av_mallocz(4* s->block_align); if (!s->coded_samples[i]) return -1; } s->int_samples = av_mallocz(4* s->frame_size); s->window_size = ((2*s->tail_size)+s->frame_size); s->window = av_mallocz(4* s->window_size); if (!s->window) return -1; avctx->extradata = av_mallocz(16); if (!avctx->extradata) return -1; init_put_bits(&pb, avctx->extradata, 16*8); put_bits(&pb, 2, version); // version if (version == 1) { put_bits(&pb, 2, s->channels); put_bits(&pb, 4, code_samplerate(s->samplerate)); } put_bits(&pb, 1, s->lossless); if (!s->lossless) put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision put_bits(&pb, 2, s->decorrelation); put_bits(&pb, 2, s->downsampling); put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table flush_put_bits(&pb); avctx->extradata_size = put_bits_count(&pb)/8; av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); avctx->coded_frame = avcodec_alloc_frame(); if (!avctx->coded_frame) return AVERROR(ENOMEM); avctx->coded_frame->key_frame = 1; avctx->frame_size = s->block_align*s->downsampling; return 0; } static av_cold int sonic_encode_close(AVCodecContext *avctx) { SonicContext *s = avctx->priv_data; int i; av_freep(&avctx->coded_frame); for (i = 0; i < s->channels; i++) av_free(s->coded_samples[i]); av_free(s->predictor_k); av_free(s->tail); av_free(s->tap_quant); av_free(s->window); av_free(s->int_samples); return 0; } static int sonic_encode_frame(AVCodecContext *avctx, uint8_t *buf, int buf_size, void *data) { SonicContext *s = avctx->priv_data; PutBitContext pb; int i, j, ch, quant = 0, x = 0; short *samples = data; init_put_bits(&pb, buf, buf_size*8); // short -> internal for (i = 0; i < s->frame_size; i++) s->int_samples[i] = samples[i]; if (!s->lossless) for (i = 0; i < s->frame_size; i++) s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT; switch(s->decorrelation) { case MID_SIDE: for (i = 0; i < s->frame_size; i += s->channels) { s->int_samples[i] += s->int_samples[i+1]; s->int_samples[i+1] -= shift(s->int_samples[i], 1); } break; case LEFT_SIDE: for (i = 0; i < s->frame_size; i += s->channels) s->int_samples[i+1] -= s->int_samples[i]; break; case RIGHT_SIDE: for (i = 0; i < s->frame_size; i += s->channels) s->int_samples[i] -= s->int_samples[i+1]; break; } memset(s->window, 0, 4* s->window_size); for (i = 0; i < s->tail_size; i++) s->window[x++] = s->tail[i]; for (i = 0; i < s->frame_size; i++) s->window[x++] = s->int_samples[i]; for (i = 0; i < s->tail_size; i++) s->window[x++] = 0; for (i = 0; i < s->tail_size; i++) s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i]; // generate taps modified_levinson_durbin(s->window, s->window_size, s->predictor_k, s->num_taps, s->channels, s->tap_quant); if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0) return -1; for (ch = 0; ch < s->channels; ch++) { x = s->tail_size+ch; for (i = 0; i < s->block_align; i++) { int sum = 0; for (j = 0; j < s->downsampling; j++, x += s->channels) sum += s->window[x]; s->coded_samples[ch][i] = sum; } } // simple rate control code if (!s->lossless) { double energy1 = 0.0, energy2 = 0.0; for (ch = 0; ch < s->channels; ch++) { for (i = 0; i < s->block_align; i++) { double sample = s->coded_samples[ch][i]; energy2 += sample*sample; energy1 += fabs(sample); } } energy2 = sqrt(energy2/(s->channels*s->block_align)); energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align); // increase bitrate when samples are like a gaussian distribution // reduce bitrate when samples are like a two-tailed exponential distribution if (energy2 > energy1) energy2 += (energy2-energy1)*RATE_VARIATION; quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR); // av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2); if (quant < 1) quant = 1; if (quant > 65535) quant = 65535; set_ue_golomb(&pb, quant); quant *= SAMPLE_FACTOR; } // write out coded samples for (ch = 0; ch < s->channels; ch++) { if (!s->lossless) for (i = 0; i < s->block_align; i++) s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant); if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0) return -1; } // av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8); flush_put_bits(&pb); return (put_bits_count(&pb)+7)/8; } #endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */ #if CONFIG_SONIC_DECODER static const int samplerate_table[] = { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 }; static av_cold int sonic_decode_init(AVCodecContext *avctx) { SonicContext *s = avctx->priv_data; GetBitContext gb; int i, version; s->channels = avctx->channels; s->samplerate = avctx->sample_rate; if (!avctx->extradata) { av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n"); return -1; } init_get_bits(&gb, avctx->extradata, avctx->extradata_size); version = get_bits(&gb, 2); if (version > 1) { av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n"); return -1; } if (version == 1) { s->channels = get_bits(&gb, 2); s->samplerate = samplerate_table[get_bits(&gb, 4)]; av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n", s->channels, s->samplerate); } if (s->channels > MAX_CHANNELS) { av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); return -1; } s->lossless = get_bits1(&gb); if (!s->lossless) skip_bits(&gb, 3); // XXX FIXME s->decorrelation = get_bits(&gb, 2); s->downsampling = get_bits(&gb, 2); s->num_taps = (get_bits(&gb, 5)+1)<<5; if (get_bits1(&gb)) // XXX FIXME av_log(avctx, AV_LOG_INFO, "Custom quant table\n"); s->block_align = (int)(2048.0*(s->samplerate/44100))/s->downsampling; s->frame_size = s->channels*s->block_align*s->downsampling; // avctx->frame_size = s->block_align; av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); // generate taps s->tap_quant = av_mallocz(4* s->num_taps); for (i = 0; i < s->num_taps; i++) s->tap_quant[i] = (int)(sqrt(i+1)); s->predictor_k = av_mallocz(4* s->num_taps); for (i = 0; i < s->channels; i++) { s->predictor_state[i] = av_mallocz(4* s->num_taps); if (!s->predictor_state[i]) return -1; } for (i = 0; i < s->channels; i++) { s->coded_samples[i] = av_mallocz(4* s->block_align); if (!s->coded_samples[i]) return -1; } s->int_samples = av_mallocz(4* s->frame_size); avctx->sample_fmt = SAMPLE_FMT_S16; return 0; } static av_cold int sonic_decode_close(AVCodecContext *avctx) { SonicContext *s = avctx->priv_data; int i; av_free(s->int_samples); av_free(s->tap_quant); av_free(s->predictor_k); for (i = 0; i < s->channels; i++) { av_free(s->predictor_state[i]); av_free(s->coded_samples[i]); } return 0; } static int sonic_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; SonicContext *s = avctx->priv_data; GetBitContext gb; int i, quant, ch, j; short *samples = data; if (buf_size == 0) return 0; // av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size); init_get_bits(&gb, buf, buf_size*8); intlist_read(&gb, s->predictor_k, s->num_taps, 0); // dequantize for (i = 0; i < s->num_taps; i++) s->predictor_k[i] *= s->tap_quant[i]; if (s->lossless) quant = 1; else quant = get_ue_golomb(&gb) * SAMPLE_FACTOR; // av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant); for (ch = 0; ch < s->channels; ch++) { int x = ch; predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps); intlist_read(&gb, s->coded_samples[ch], s->block_align, 1); for (i = 0; i < s->block_align; i++) { for (j = 0; j < s->downsampling - 1; j++) { s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0); x += s->channels; } s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant); x += s->channels; } for (i = 0; i < s->num_taps; i++) s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels]; } switch(s->decorrelation) { case MID_SIDE: for (i = 0; i < s->frame_size; i += s->channels) { s->int_samples[i+1] += shift(s->int_samples[i], 1); s->int_samples[i] -= s->int_samples[i+1]; } break; case LEFT_SIDE: for (i = 0; i < s->frame_size; i += s->channels) s->int_samples[i+1] += s->int_samples[i]; break; case RIGHT_SIDE: for (i = 0; i < s->frame_size; i += s->channels) s->int_samples[i] += s->int_samples[i+1]; break; } if (!s->lossless) for (i = 0; i < s->frame_size; i++) s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT); // internal -> short for (i = 0; i < s->frame_size; i++) samples[i] = av_clip_int16(s->int_samples[i]); align_get_bits(&gb); *data_size = s->frame_size * 2; return (get_bits_count(&gb)+7)/8; } AVCodec sonic_decoder = { "sonic", CODEC_TYPE_AUDIO, CODEC_ID_SONIC, sizeof(SonicContext), sonic_decode_init, NULL, sonic_decode_close, sonic_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("Sonic"), }; #endif /* CONFIG_SONIC_DECODER */ #if CONFIG_SONIC_ENCODER AVCodec sonic_encoder = { "sonic", CODEC_TYPE_AUDIO, CODEC_ID_SONIC, sizeof(SonicContext), sonic_encode_init, sonic_encode_frame, sonic_encode_close, NULL, .long_name = NULL_IF_CONFIG_SMALL("Sonic"), }; #endif #if CONFIG_SONIC_LS_ENCODER AVCodec sonic_ls_encoder = { "sonicls", CODEC_TYPE_AUDIO, CODEC_ID_SONIC_LS, sizeof(SonicContext), sonic_encode_init, sonic_encode_frame, sonic_encode_close, NULL, .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"), }; #endif