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Diffstat (limited to 'libavcodec/sonic.c')
| -rw-r--r-- | libavcodec/sonic.c | 1116 |
1 files changed, 1116 insertions, 0 deletions
diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c new file mode 100644 index 0000000..c5076f9 --- /dev/null +++ b/libavcodec/sonic.c @@ -0,0 +1,1116 @@ +/* + * 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 int 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)); + + if (!state) + return AVERROR(ENOMEM); + + 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); + return 0; +} + +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)); + if (!s->tap_quant) + return AVERROR(ENOMEM); + + 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 || !s->int_samples) + 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 + ret = modified_levinson_durbin(s->window, s->window_size, + s->predictor_k, s->num_taps, s->channels, s->tap_quant); + if (ret < 0) + return ret; + + 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) + { + int sample_rate_index; + s->channels = get_bits(&gb, 2); + sample_rate_index = get_bits(&gb, 4); + if (sample_rate_index >= FF_ARRAY_ELEMS(samplerate_table)) { + av_log(avctx, AV_LOG_ERROR, "Invalid sample_rate_index %d\n", sample_rate_index); + return AVERROR_INVALIDDATA; + } + s->samplerate = samplerate_table[sample_rate_index]; + av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n", + s->channels, s->samplerate); + } + + if (s->channels > MAX_CHANNELS || s->channels < 1) + { + av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); + return AVERROR_INVALIDDATA; + } + avctx->channels = s->channels; + + 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)); + if (!s->tap_quant) + return AVERROR(ENOMEM); + + 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)); + if (!s->int_samples) + return AVERROR(ENOMEM); + + 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 |
