This was an experimental/research codec of which ffmpeg is the only
encoder and decoder, development has stalled and these files don't exist
in the wild.
Signed-off-by: J. Dekker <j...@itanimul.li>
---
Changelog | 1 +
configure | 3 -
libavcodec/Makefile | 3 -
libavcodec/allcodecs.c | 3 -
libavcodec/codec_desc.c | 14 -
libavcodec/sonic.c | 1125 ---------------------------------------
6 files changed, 1 insertion(+), 1148 deletions(-)
delete mode 100644 libavcodec/sonic.c
diff --git a/Changelog b/Changelog
index 610ee61dd6..e2096f249a 100644
--- a/Changelog
+++ b/Changelog
@@ -27,6 +27,7 @@ version <next>:
- a C11-compliant compiler is now required; note that this requirement
will be bumped to C17 in the near future, so consider updating your
build environment if it lacks C17 support
+- sonic lossy/lossless audio codec removed
version 6.1:
- libaribcaption decoder
diff --git a/configure b/configure
index bb5e630bad..e639a5e2b7 100755
--- a/configure
+++ b/configure
@@ -2991,9 +2991,6 @@ sipr_decoder_select="lsp"
smvjpeg_decoder_select="mjpeg_decoder"
snow_decoder_select="dwt h264qpel rangecoder videodsp"
snow_encoder_select="dwt h264qpel hpeldsp me_cmp mpegvideoenc 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"
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 09ae5270b3..3fc716ee68 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -687,9 +687,6 @@ OBJS-$(CONFIG_SNOW_DECODER) += snowdec.o snow.o
snow_dwt.o
OBJS-$(CONFIG_SNOW_ENCODER) += snowenc.o snow.o snow_dwt.o
\
h263.o h263data.o ituh263enc.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 ef8c3a6d7d..e0a4a5421d 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -535,9 +535,6 @@ 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/codec_desc.c b/libavcodec/codec_desc.c
index 033344304c..9b456616be 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -3175,20 +3175,6 @@ static const AVCodecDescriptor codec_descriptors[] = {
.long_name = NULL_IF_CONFIG_SMALL("Wave synthesis pseudo-codec"),
.props = AV_CODEC_PROP_INTRA_ONLY,
},
- {
- .id = AV_CODEC_ID_SONIC,
- .type = AVMEDIA_TYPE_AUDIO,
- .name = "sonic",
- .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
- .props = AV_CODEC_PROP_INTRA_ONLY,
- },
- {
- .id = AV_CODEC_ID_SONIC_LS,
- .type = AVMEDIA_TYPE_AUDIO,
- .name = "sonicls",
- .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
- .props = AV_CODEC_PROP_INTRA_ONLY,
- },
{
.id = AV_CODEC_ID_EVRC,
.type = AVMEDIA_TYPE_AUDIO,
diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c
deleted file mode 100644
index 0544fecf46..0000000000
--- a/libavcodec/sonic.c
+++ /dev/null
@@ -1,1125 +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 "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);
-}
-
-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;
- 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;
- }
-}
-
-#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] * (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);
-
-#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 -= (unsigned)shift_down(k_value * (unsigned)state_value,
LATTICE_SHIFT);
- state_ptr[1] = state_value + shift_down(k_value * (unsigned)x,
LATTICE_SHIFT);
- }
-#else
- for (i = order-2; i >= 0; i--)
- {
- x -= (unsigned)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 = 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;
-#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
- }
-}
-
-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 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");
-
- 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 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),
- .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
- .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),
- .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
- .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
- .close = sonic_encode_close,
-};
-#endif
--
2.43.2
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