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Diffstat (limited to 'libavcodec/sonic.c')
| -rw-r--r-- | libavcodec/sonic.c | 1001 |
1 files changed, 1001 insertions, 0 deletions
diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c new file mode 100644 index 0000000..b67434f --- /dev/null +++ b/libavcodec/sonic.c @@ -0,0 +1,1001 @@ +/* + * 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" + +/** + * @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 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; + 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>>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_size = s->num_taps*s->channels; + s->tail = av_mallocz(4 * s->tail_size); + if (!s->tail) + return -1; + + 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, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) +{ + SonicContext *s = avctx->priv_data; + PutBitContext pb; + int i, j, ch, quant = 0, x = 0; + int ret; + const short *samples = (const int16_t*)frame->data[0]; + + if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0) + return ret; + + init_put_bits(&pb, avpkt->data, avpkt->size); + + // 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 > 65534) + quant = 65534; + + 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); + avpkt->size = (put_bits_count(&pb)+7)/8; + *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, 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); + 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 = (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 = AV_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 *got_frame_ptr, + 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, 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); + + 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); + + *got_frame_ptr = 1; + + return (get_bits_count(&gb)+7)/8; +} + +AVCodec ff_sonic_decoder = { + .name = "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, + .long_name = NULL_IF_CONFIG_SMALL("Sonic"), +}; +#endif /* CONFIG_SONIC_DECODER */ + +#if CONFIG_SONIC_ENCODER +AVCodec ff_sonic_encoder = { + .name = "sonic", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_SONIC, + .priv_data_size = sizeof(SonicContext), + .init = sonic_encode_init, + .encode2 = sonic_encode_frame, + .capabilities = CODEC_CAP_EXPERIMENTAL, + .close = sonic_encode_close, + .long_name = NULL_IF_CONFIG_SMALL("Sonic"), +}; +#endif + +#if CONFIG_SONIC_LS_ENCODER +AVCodec ff_sonic_ls_encoder = { + .name = "sonicls", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_SONIC_LS, + .priv_data_size = sizeof(SonicContext), + .init = sonic_encode_init, + .encode2 = sonic_encode_frame, + .capabilities = CODEC_CAP_EXPERIMENTAL, + .close = sonic_encode_close, + .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"), +}; +#endif |
