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commit ca1d253621d4643b7b9ba13ded6dfc68c329fda5 Author: Andreas Rheinhardt <[email protected]> AuthorDate: Sun Mar 8 20:30:22 2026 +0100 Commit: Andreas Rheinhardt <[email protected]> CommitDate: Tue Jun 23 17:15:02 2026 +0200 avcodec/sonic: Remove sonic codecs The original plan from commit d89fbfd4df6fd64f604a8373224aa396149a3784 (merged on 2024-11-25) was to remove the encoders on the then next major version bump (i.e. the bump to lavc 62, the current version) and to disable the decoders by default on said bump. Both did not happen; one deadline has already passed and the other is right upon us, so just remove all sonic codecs now. Note: Codec IDs and descriptors are kept. Signed-off-by: Andreas Rheinhardt <[email protected]> --- MAINTAINERS | 1 - configure | 6 - libavcodec/Makefile | 3 - libavcodec/allcodecs.c | 3 - libavcodec/sonic.c | 831 --------------------------------------------- libavcodec/version_major.h | 4 - 6 files changed, 848 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index cdaf9b251a..88a6478a64 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -253,7 +253,6 @@ Codecs: sanm.c Manuel Lauss smc.c Mike Melanson snow* Michael Niedermayer, Loren Merritt - sonic.c Alex Beregszaszi speedhq.c Steinar H. Gunderson srt* Aurelien Jacobs sunrast.c Ivo van Poorten diff --git a/configure b/configure index 53eff1f154..d3ea7ca6b5 100755 --- a/configure +++ b/configure @@ -3329,9 +3329,6 @@ sipr_decoder_select="lsp celp_math" smvjpeg_decoder_select="mjpeg_decoder" snow_decoder_select="dwt h264qpel rangecoder videodsp" snow_encoder_select="dwt h264qpel hpeldsp me_cmp mpegvideoencdsp rangecoder videodsp" -sonic_decoder_select="golomb rangecoder" -sonic_encoder_select="golomb rangecoder" -sonic_ls_encoder_select="golomb rangecoder" sp5x_decoder_select="mjpeg_decoder" speedhq_decoder_select="blockdsp idctdsp" speedhq_encoder_select="mpegvideoenc" @@ -4693,9 +4690,6 @@ do_random(){ $action $(rand_list "$@" | awk "BEGIN { srand($random_seed) } \$1 == \"prob\" { prob = \$2; next } rand() < prob { print }") } -# deprecated components (disabled by default) -disable sonic_encoder sonic_ls_encoder - for opt do optval="${opt#*=}" case "$opt" in diff --git a/libavcodec/Makefile b/libavcodec/Makefile index ffbacc2ed3..7abf2d2d89 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -724,9 +724,6 @@ OBJS-$(CONFIG_SNOW_ENCODER) += snowenc.o snow.o snow_dwt.o h263data.o ituh263enc.o \ motion_est.o ratecontrol.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-$(CONFIG_SPEEDHQ_DECODER) += speedhqdec.o speedhq.o mpeg12.o \ mpeg12data.o OBJS-$(CONFIG_SPEEDHQ_ENCODER) += speedhq.o mpeg12data.o mpeg12enc.o speedhqenc.o diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index 314cb230a4..592b026c9f 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -538,9 +538,6 @@ extern const FFCodec ff_shorten_decoder; extern const FFCodec ff_sipr_decoder; extern const FFCodec ff_siren_decoder; extern const FFCodec ff_smackaud_decoder; -extern const FFCodec ff_sonic_encoder; -extern const FFCodec ff_sonic_decoder; -extern const FFCodec ff_sonic_ls_encoder; extern const FFCodec ff_tak_decoder; extern const FFCodec ff_truehd_encoder; extern const FFCodec ff_truehd_decoder; diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c deleted file mode 100644 index 08549aacfe..0000000000 --- a/libavcodec/sonic.c +++ /dev/null @@ -1,831 +0,0 @@ -/* - * 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 "config_components.h" - -#include "libavutil/mem.h" -#include "avcodec.h" -#include "codec_internal.h" -#include "decode.h" -#include "encode.h" -#include "get_bits.h" -#include "golomb.h" -#include "put_golomb.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); -} - - -#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 = window + window_entries; - - memcpy(state, window, window_entries * sizeof(*state)); - - for (i = 0; i < out_entries; i++) - { - int step = (i+1)*channels, k, j; - double xx = 0.0, xy = 0.0; - 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; - } - 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]; - - 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); - } - } -} - -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; - int *coded_samples; - PutBitContext pb; - int i; - - s->version = 2; - - if (avctx->ch_layout.nb_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->ch_layout.nb_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->ch_layout.nb_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); - - coded_samples = av_calloc(s->block_align, s->channels * sizeof(**s->coded_samples)); - if (!coded_samples) - return AVERROR(ENOMEM); - for (i = 0; i < s->channels; i++, coded_samples += s->block_align) - s->coded_samples[i] = coded_samples; - - 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, 2 * 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_bytes_output(&pb); - - 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; - - av_freep(&s->coded_samples[0]); - 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 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 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 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_packet(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, s->window_size * sizeof(*s->window)); - - 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; - } - - avpkt->size = ff_rac_terminate(&c, 0); - *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; - int *tmp; - GetBitContext gb; - int i; - int ret; - - s->channels = avctx->ch_layout.nb_channels; - s->samplerate = avctx->sample_rate; - - if (!avctx->extradata) - { - av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n"); - return AVERROR_INVALIDDATA; - } - - ret = init_get_bits8(&gb, avctx->extradata, avctx->extradata_size); - if (ret < 0) - return ret; - - 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; - } - av_channel_layout_uninit(&avctx->ch_layout); - avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC; - avctx->ch_layout.nb_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"); - - if (s->num_taps > 128) - return AVERROR_INVALIDDATA; - - 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; - - if (s->num_taps * s->channels > s->frame_size) { - av_log(avctx, AV_LOG_ERROR, - "number of taps times channels (%d * %d) larger than frame size %d\n", - s->num_taps, s->channels, s->frame_size); - return AVERROR_INVALIDDATA; - } - - 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)); - - tmp = av_calloc(s->num_taps, s->channels * sizeof(**s->predictor_state)); - if (!tmp) - return AVERROR(ENOMEM); - for (i = 0; i < s->channels; i++, tmp += s->num_taps) - s->predictor_state[i] = tmp; - - tmp = av_calloc(s->block_align, s->channels * sizeof(**s->coded_samples)); - if (!tmp) - return AVERROR(ENOMEM); - for (i = 0; i < s->channels; i++, tmp += s->block_align) - s->coded_samples[i] = tmp; - - 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; - - av_freep(&s->int_samples); - av_freep(&s->tap_quant); - av_freep(&s->predictor_k); - av_freep(&s->predictor_state[0]); - av_freep(&s->coded_samples[0]); - - return 0; -} - -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; - unsigned a; - e= 0; - while(get_rac(c, state+1 + FFMIN(e,9))){ //1..10 - e++; - if (e > 31) - return AVERROR_INVALIDDATA; - } - - 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; - } -} - -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; -} - -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] * (unsigned)state[p], LATTICE_SHIFT); - state[p] += shift_down(k[j]* (unsigned)x, LATTICE_SHIFT); - x = tmp; - } - } -} - -static int predictor_calc_error(int *k, int *state, int order, int error) -{ - int i, x = error - (unsigned)shift_down(k[order-1] * (unsigned)state[order-1], LATTICE_SHIFT); - - 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 -= (unsigned)shift_down(k_value * (unsigned)state_value, LATTICE_SHIFT); - state_ptr[1] = state_value + shift_down(k_value * (unsigned)x, LATTICE_SHIFT); - } - - // 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; -} - -static int sonic_decode_frame(AVCodecContext *avctx, AVFrame *frame, - 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; - - if (buf_size == 0) return 0; - - frame->nb_samples = s->frame_size / avctx->ch_layout.nb_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] *= (unsigned) s->tap_quant[i]; - - if (s->lossless) - quant = 1; - else - quant = get_symbol(&c, state, 0) * (unsigned)SAMPLE_FACTOR; - -// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant); - - for (ch = 0; ch < s->channels; ch++) - { - int x = ch; - - if (c.overread > MAX_OVERREAD) - return AVERROR_INVALIDDATA; - - 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] * (unsigned)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; -} - -const FFCodec ff_sonic_decoder = { - .p.name = "sonic", - CODEC_LONG_NAME("Sonic"), - .p.type = AVMEDIA_TYPE_AUDIO, - .p.id = AV_CODEC_ID_SONIC, - .priv_data_size = sizeof(SonicContext), - .init = sonic_decode_init, - .close = sonic_decode_close, - FF_CODEC_DECODE_CB(sonic_decode_frame), - .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL | AV_CODEC_CAP_CHANNEL_CONF, - .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, -}; -#endif /* CONFIG_SONIC_DECODER */ - -#if CONFIG_SONIC_ENCODER -const FFCodec ff_sonic_encoder = { - .p.name = "sonic", - CODEC_LONG_NAME("Sonic"), - .p.type = AVMEDIA_TYPE_AUDIO, - .p.id = AV_CODEC_ID_SONIC, - .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL | - AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE, - .priv_data_size = sizeof(SonicContext), - .init = sonic_encode_init, - FF_CODEC_ENCODE_CB(sonic_encode_frame), - CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_S16), - .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, - .close = sonic_encode_close, -}; -#endif - -#if CONFIG_SONIC_LS_ENCODER -const FFCodec ff_sonic_ls_encoder = { - .p.name = "sonicls", - CODEC_LONG_NAME("Sonic lossless"), - .p.type = AVMEDIA_TYPE_AUDIO, - .p.id = AV_CODEC_ID_SONIC_LS, - .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL | - AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE, - .priv_data_size = sizeof(SonicContext), - .init = sonic_encode_init, - FF_CODEC_ENCODE_CB(sonic_encode_frame), - CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_S16), - .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, - .close = sonic_encode_close, -}; -#endif diff --git a/libavcodec/version_major.h b/libavcodec/version_major.h index 52f6d629dd..2723676245 100644 --- a/libavcodec/version_major.h +++ b/libavcodec/version_major.h @@ -53,10 +53,6 @@ // reminder to remove the OMX encoder on next major bump #define FF_CODEC_OMX (LIBAVCODEC_VERSION_MAJOR < 63) -// reminder to remove Sonic Lossy/Lossless encoders on next major bump -#define FF_CODEC_SONIC_ENC (LIBAVCODEC_VERSION_MAJOR < 63) -// reminder to remove Sonic decoder on next-next major bump -#define FF_CODEC_SONIC_DEC (LIBAVCODEC_VERSION_MAJOR < 63) #define FF_API_NVENC_H264_MAIN (LIBAVCODEC_VERSION_MAJOR < 63) _______________________________________________ ffmpeg-cvslog mailing list -- [email protected] To unsubscribe send an email to [email protected]
