diff options
| -rwxr-xr-x | configure | 3 | ||||
| -rw-r--r-- | doc/general.texi | 4 | ||||
| -rw-r--r-- | libavcodec/Makefile | 3 | ||||
| -rw-r--r-- | libavcodec/allcodecs.c | 2 | ||||
| -rw-r--r-- | libavcodec/sonic.c | 977 |
5 files changed, 0 insertions, 989 deletions
@@ -1324,9 +1324,6 @@ shorten_decoder_select="golomb" sipr_decoder_select="lsp" snow_decoder_select="dwt" snow_encoder_select="aandct dwt" -sonic_decoder_select="golomb" -sonic_encoder_select="golomb" -sonic_ls_encoder_select="golomb" svq1_encoder_select="aandct" svq3_decoder_select="golomb h264dsp h264pred" svq3_decoder_suggest="zlib" diff --git a/doc/general.texi b/doc/general.texi index 3ef4d678eb..f6c61a2342 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -665,10 +665,6 @@ following image formats are supported: @item Sierra VMD audio @tab @tab X @tab Used in Sierra VMD files. @item Smacker audio @tab @tab X -@item Sonic @tab X @tab X - @tab experimental codec -@item Sonic lossless @tab X @tab X - @tab experimental codec @item Speex @tab @tab E @tab supported through external library libspeex @item True Audio (TTA) @tab @tab X diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 21bdbf42d1..d72a34062e 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -337,9 +337,6 @@ OBJS-$(CONFIG_SNOW_ENCODER) += snow.o rangecoder.o motion_est.o \ ituh263enc.o mpegvideo_enc.o \ mpeg12data.o OBJS-$(CONFIG_SOL_DPCM_DECODER) += dpcm.o -OBJS-$(CONFIG_SONIC_DECODER) += sonic.o -OBJS-$(CONFIG_SONIC_ENCODER) += sonic.o -OBJS-$(CONFIG_SONIC_LS_ENCODER) += sonic.o OBJS-$(CONFIG_SP5X_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o OBJS-$(CONFIG_SRT_DECODER) += srtdec.o ass.o OBJS-$(CONFIG_SUNRAST_DECODER) += sunrast.o diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index 767a50283c..8de6ad8429 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -266,8 +266,6 @@ void avcodec_register_all(void) REGISTER_DECODER (SHORTEN, shorten); REGISTER_DECODER (SIPR, sipr); REGISTER_DECODER (SMACKAUD, smackaud); - REGISTER_ENCDEC (SONIC, sonic); - REGISTER_ENCODER (SONIC_LS, sonic_ls); REGISTER_DECODER (TRUEHD, truehd); REGISTER_DECODER (TRUESPEECH, truespeech); REGISTER_DECODER (TTA, tta); diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c deleted file mode 100644 index bd6691f61b..0000000000 --- a/libavcodec/sonic.c +++ /dev/null @@ -1,977 +0,0 @@ -/* - * Simple free lossless/lossy audio codec - * Copyright (c) 2004 Alex Beregszaszi - * - * This file is part of Libav. - * - * Libav 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. - * - * Libav 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 Libav; 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 - * 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 = 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 *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 ff_sonic_decoder = { - "sonic", - AVMEDIA_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 ff_sonic_encoder = { - "sonic", - AVMEDIA_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 ff_sonic_ls_encoder = { - "sonicls", - AVMEDIA_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 |