/* * 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" #include "internal.h" #include "rangecoder.h" /** * @file * 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 version; int minor_version; 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 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); } 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]){ int i; #define put_rac(C,S,B) \ do{\ if(rc_stat){\ rc_stat[*(S)][B]++;\ rc_stat2[(S)-state][B]++;\ }\ put_rac(C,S,B);\ }while(0) if(v){ const int a= FFABS(v); const int e= av_log2(a); put_rac(c, state+0, 0); if(e<=9){ for(i=0; i<e; i++){ put_rac(c, state+1+i, 1); //1..10 } put_rac(c, state+1+i, 0); for(i=e-1; i>=0; i--){ put_rac(c, state+22+i, (a>>i)&1); //22..31 } if(is_signed) put_rac(c, state+11 + e, v < 0); //11..21 }else{ for(i=0; i<e; i++){ put_rac(c, state+1+FFMIN(i,9), 1); //1..10 } put_rac(c, state+1+9, 0); for(i=e-1; i>=0; i--){ put_rac(c, state+22+FFMIN(i,9), (a>>i)&1); //22..31 } if(is_signed) put_rac(c, state+11 + 10, v < 0); //11..21 } }else{ put_rac(c, state+0, 1); } #undef put_rac } static inline av_flatten int get_symbol(RangeCoder *c, uint8_t *state, int is_signed){ if(get_rac(c, state+0)) return 0; else{ int i, e, a; e= 0; while(get_rac(c, state+1 + FFMIN(e,9))){ //1..10 e++; } a= 1; for(i=e-1; i>=0; i--){ a += a + get_rac(c, state+22 + FFMIN(i,9)); //22..31 } e= -(is_signed && get_rac(c, state+11 + FFMIN(e, 10))); //11..21 return (a^e)-e; } } #if 1 static inline int intlist_write(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part) { int i; for (i = 0; i < entries; i++) put_symbol(c, state, buf[i], 1, NULL, NULL); return 1; } static inline int intlist_read(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part) { int i; for (i = 0; i < entries; i++) buf[i] = get_symbol(c, state, 1); return 1; } #elif 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_calloc(entries, sizeof(*copy)); if (!copy) return AVERROR(ENOMEM); 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_calloc(entries*max, sizeof(*bits)); if (!bits) { av_free(copy); return AVERROR(ENOMEM); } 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_calloc(entries, sizeof(*bits)); if (!bits) return AVERROR(ENOMEM); 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_calloc(window_entries, sizeof(*state)); 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]); int *state_ptr = &(state[0]); j = window_entries - step; for (;j>0;j--,x_ptr++,state_ptr++) { double x_value = *x_ptr; double 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]; double 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; int 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]; int 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 AVERROR(EINVAL); } static av_cold int sonic_encode_init(AVCodecContext *avctx) { SonicContext *s = avctx->priv_data; PutBitContext pb; int i; s->version = 2; if (avctx->channels > MAX_CHANNELS) { av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); return AVERROR(EINVAL); /* only stereo or mono for now */ } if (avctx->channels == 2) s->decorrelation = MID_SIDE; else s->decorrelation = 3; if (avctx->codec->id == AV_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 % 32) { av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n"); return AVERROR_INVALIDDATA; } // generate taps s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant)); for (i = 0; i < s->num_taps; i++) s->tap_quant[i] = ff_sqrt(i+1); s->channels = avctx->channels; s->samplerate = avctx->sample_rate; s->block_align = 2048LL*s->samplerate/(44100*s->downsampling); s->frame_size = s->channels*s->block_align*s->downsampling; s->tail_size = s->num_taps*s->channels; s->tail = av_calloc(s->tail_size, sizeof(*s->tail)); if (!s->tail) return AVERROR(ENOMEM); s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) ); if (!s->predictor_k) return AVERROR(ENOMEM); for (i = 0; i < s->channels; i++) { s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples)); if (!s->coded_samples[i]) return AVERROR(ENOMEM); } s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples)); s->window_size = ((2*s->tail_size)+s->frame_size); s->window = av_calloc(s->window_size, sizeof(*s->window)); if (!s->window) return AVERROR(ENOMEM); avctx->extradata = av_mallocz(16); if (!avctx->extradata) return AVERROR(ENOMEM); init_put_bits(&pb, avctx->extradata, 16*8); put_bits(&pb, 2, s->version); // version if (s->version >= 1) { if (s->version >= 2) { put_bits(&pb, 8, s->version); put_bits(&pb, 8, s->minor_version); } 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.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); 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; for (i = 0; i < s->channels; i++) av_freep(&s->coded_samples[i]); av_freep(&s->predictor_k); av_freep(&s->tail); av_freep(&s->tap_quant); av_freep(&s->window); av_freep(&s->int_samples); return 0; } static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { SonicContext *s = avctx->priv_data; RangeCoder c; int i, j, ch, quant = 0, x = 0; int ret; const short *samples = (const int16_t*)frame->data[0]; uint8_t state[32]; if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0) return ret; ff_init_range_encoder(&c, avpkt->data, avpkt->size); ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8); memset(state, 128, sizeof(state)); // 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 ((ret = intlist_write(&c, state, s->predictor_k, s->num_taps, 0)) < 0) return ret; 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 = M_SQRT2*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); quant = av_clip(quant, 1, 65534); put_symbol(&c, state, quant, 0, NULL, NULL); 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] = ROUNDED_DIV(s->coded_samples[ch][i], quant); if ((ret = intlist_write(&c, state, s->coded_samples[ch], s->block_align, 1)) < 0) return ret; } // av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8); avpkt->size = ff_rac_terminate(&c); *got_packet_ptr = 1; return 0; } #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; s->channels = avctx->channels; s->samplerate = avctx->sample_rate; if (!avctx->extradata) { av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n"); return AVERROR_INVALIDDATA; } init_get_bits8(&gb, avctx->extradata, avctx->extradata_size); s->version = get_bits(&gb, 2); if (s->version >= 2) { s->version = get_bits(&gb, 8); s->minor_version = get_bits(&gb, 8); } if (s->version != 2) { av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n"); return AVERROR_INVALIDDATA; } if (s->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 AVERROR_INVALIDDATA; } s->lossless = get_bits1(&gb); if (!s->lossless) skip_bits(&gb, 3); // XXX FIXME s->decorrelation = get_bits(&gb, 2); if (s->decorrelation != 3 && s->channels != 2) { av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation); return AVERROR_INVALIDDATA; } s->downsampling = get_bits(&gb, 2); if (!s->downsampling) { av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n"); return AVERROR_INVALIDDATA; } 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 = 2048LL*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.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); // generate taps s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant)); for (i = 0; i < s->num_taps; i++) s->tap_quant[i] = ff_sqrt(i+1); s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k)); for (i = 0; i < s->channels; i++) { s->predictor_state[i] = av_calloc(s->num_taps, sizeof(**s->predictor_state)); if (!s->predictor_state[i]) return AVERROR(ENOMEM); } for (i = 0; i < s->channels; i++) { s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples)); if (!s->coded_samples[i]) return AVERROR(ENOMEM); } s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples)); avctx->sample_fmt = AV_SAMPLE_FMT_S16; return 0; } static av_cold int sonic_decode_close(AVCodecContext *avctx) { SonicContext *s = avctx->priv_data; int i; av_freep(&s->int_samples); av_freep(&s->tap_quant); av_freep(&s->predictor_k); for (i = 0; i < s->channels; i++) { av_freep(&s->predictor_state[i]); av_freep(&s->coded_samples[i]); } return 0; } static int sonic_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; SonicContext *s = avctx->priv_data; RangeCoder c; uint8_t state[32]; int i, quant, ch, j, ret; int16_t *samples; AVFrame *frame = data; if (buf_size == 0) return 0; frame->nb_samples = s->frame_size / avctx->channels; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; samples = (int16_t *)frame->data[0]; // av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size); memset(state, 128, sizeof(state)); ff_init_range_decoder(&c, buf, buf_size); ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8); intlist_read(&c, state, 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_symbol(&c, state, 0) * 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(&c, state, 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]); *got_frame_ptr = 1; return buf_size; } AVCodec ff_sonic_decoder = { .name = "sonic", .long_name = NULL_IF_CONFIG_SMALL("Sonic"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_SONIC, .priv_data_size = sizeof(SonicContext), .init = sonic_decode_init, .close = sonic_decode_close, .decode = sonic_decode_frame, .capabilities = CODEC_CAP_DR1 | CODEC_CAP_EXPERIMENTAL, }; #endif /* CONFIG_SONIC_DECODER */ #if CONFIG_SONIC_ENCODER AVCodec ff_sonic_encoder = { .name = "sonic", .long_name = NULL_IF_CONFIG_SMALL("Sonic"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_SONIC, .priv_data_size = sizeof(SonicContext), .init = sonic_encode_init, .encode2 = sonic_encode_frame, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .capabilities = CODEC_CAP_EXPERIMENTAL, .close = sonic_encode_close, }; #endif #if CONFIG_SONIC_LS_ENCODER AVCodec ff_sonic_ls_encoder = { .name = "sonicls", .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_SONIC_LS, .priv_data_size = sizeof(SonicContext), .init = sonic_encode_init, .encode2 = sonic_encode_frame, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .capabilities = CODEC_CAP_EXPERIMENTAL, .close = sonic_encode_close, }; #endif