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ffmpeg/libavcodec/sonic.c
Andreas Rheinhardt 56e9e0273a avcodec/encode: Always use intermediate buffer in ff_alloc_packet2() 
Up until now, ff_alloc_packet2() has a min_size parameter:
It is supposed to be a lower bound on the final size of the packet
to allocate. If it is not too far from the upper bound (namely,
if it is at least half the upper bound), then ff_alloc_packet2()
already allocates the final, already refcounted packet; if it is
not, then the packet is not refcounted and its data only points to
a buffer owned by the AVCodecContext (in this case, the packet will
be made refcounted in encode_simple_internal() in libavcodec/encode.c).
The goal of this was to avoid data copies and intermediate buffers
if one has a precise lower bound.

Yet those encoders for which precise lower bounds exist have recently
been switched to ff_get_encode_buffer() (which automatically allocates
final buffers), leaving only two encoders to actually set the min_size
to something else than zero (namely aliaspixenc and hapenc). Both of
these encoders use a very low lower bound that is not helpful in any
nontrivial case.

This commit therefore removes the min_size parameter as well as the
codepath in ff_alloc_packet2() for the allocation of final buffers.
Furthermore, the function has been renamed to ff_alloc_packet() and
moved to encode.h alongside ff_get_encode_buffer().

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-06-08 12:52:50 +02:00

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/*
* 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 "encode.h"
#include "get_bits.h"
#include "golomb.h"
#include "internal.h"
#include "rangecoder.h"
/**
* @file
* Simple free lossless/lossy audio codec
* Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
* Written and designed by Alex Beregszaszi
*
* TODO:
* - CABAC put/get_symbol
* - independent quantizer for channels
* - >2 channels support
* - more decorrelation types
* - more tap_quant tests
* - selectable intlist writers/readers (bonk-style, golomb, cabac)
*/
#define MAX_CHANNELS 2
#define MID_SIDE 0
#define LEFT_SIDE 1
#define RIGHT_SIDE 2
typedef struct SonicContext {
int version;
int minor_version;
int lossless, decorrelation;
int num_taps, downsampling;
double quantization;
int channels, samplerate, block_align, frame_size;
int *tap_quant;
int *int_samples;
int *coded_samples[MAX_CHANNELS];
// for encoding
int *tail;
int tail_size;
int *window;
int window_size;
// for decoding
int *predictor_k;
int *predictor_state[MAX_CHANNELS];
} SonicContext;
#define LATTICE_SHIFT 10
#define SAMPLE_SHIFT 4
#define LATTICE_FACTOR (1 << LATTICE_SHIFT)
#define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)
#define BASE_QUANT 0.6
#define RATE_VARIATION 3.0
static inline int shift(int a,int b)
{
return (a+(1<<(b-1))) >> b;
}
static inline int shift_down(int a,int b)
{
return (a>>b)+(a<0);
}
static av_always_inline av_flatten void put_symbol(RangeCoder *c, uint8_t *state, int v, int is_signed, uint64_t rc_stat[256][2], uint64_t rc_stat2[32][2]){
int i;
#define put_rac(C,S,B) \
do{\
if(rc_stat){\
rc_stat[*(S)][B]++;\
rc_stat2[(S)-state][B]++;\
}\
put_rac(C,S,B);\
}while(0)
if(v){
const int a= FFABS(v);
const int e= av_log2(a);
put_rac(c, state+0, 0);
if(e<=9){
for(i=0; i<e; i++){
put_rac(c, state+1+i, 1); //1..10
}
put_rac(c, state+1+i, 0);
for(i=e-1; i>=0; i--){
put_rac(c, state+22+i, (a>>i)&1); //22..31
}
if(is_signed)
put_rac(c, state+11 + e, v < 0); //11..21
}else{
for(i=0; i<e; i++){
put_rac(c, state+1+FFMIN(i,9), 1); //1..10
}
put_rac(c, state+1+9, 0);
for(i=e-1; i>=0; i--){
put_rac(c, state+22+FFMIN(i,9), (a>>i)&1); //22..31
}
if(is_signed)
put_rac(c, state+11 + 10, v < 0); //11..21
}
}else{
put_rac(c, state+0, 1);
}
#undef put_rac
}
static inline av_flatten int get_symbol(RangeCoder *c, uint8_t *state, int is_signed){
if(get_rac(c, state+0))
return 0;
else{
int i, e;
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 - 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->channels > MAX_CHANNELS)
{
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
return AVERROR(EINVAL); /* only stereo or mono for now */
}
if (avctx->channels == 2)
s->decorrelation = MID_SIDE;
else
s->decorrelation = 3;
if (avctx->codec->id == AV_CODEC_ID_SONIC_LS)
{
s->lossless = 1;
s->num_taps = 32;
s->downsampling = 1;
s->quantization = 0.0;
}
else
{
s->num_taps = 128;
s->downsampling = 2;
s->quantization = 1.0;
}
// max tap 2048
if (s->num_taps < 32 || s->num_taps > 1024 || s->num_taps % 32) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
return AVERROR_INVALIDDATA;
}
// generate taps
s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
if (!s->tap_quant)
return AVERROR(ENOMEM);
for (i = 0; i < s->num_taps; i++)
s->tap_quant[i] = ff_sqrt(i+1);
s->channels = avctx->channels;
s->samplerate = avctx->sample_rate;
s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
s->frame_size = s->channels*s->block_align*s->downsampling;
s->tail_size = s->num_taps*s->channels;
s->tail = av_calloc(s->tail_size, sizeof(*s->tail));
if (!s->tail)
return AVERROR(ENOMEM);
s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) );
if (!s->predictor_k)
return AVERROR(ENOMEM);
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->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;
}
avctx->channels = s->channels;
s->lossless = get_bits1(&gb);
if (!s->lossless)
skip_bits(&gb, 3); // XXX FIXME
s->decorrelation = get_bits(&gb, 2);
if (s->decorrelation != 3 && s->channels != 2) {
av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation);
return AVERROR_INVALIDDATA;
}
s->downsampling = get_bits(&gb, 2);
if (!s->downsampling) {
av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n");
return AVERROR_INVALIDDATA;
}
s->num_taps = (get_bits(&gb, 5)+1)<<5;
if (get_bits1(&gb)) // XXX FIXME
av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
s->frame_size = s->channels*s->block_align*s->downsampling;
// avctx->frame_size = s->block_align;
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,
void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
SonicContext *s = avctx->priv_data;
RangeCoder c;
uint8_t state[32];
int i, quant, ch, j, ret;
int16_t *samples;
AVFrame *frame = data;
if (buf_size == 0) return 0;
frame->nb_samples = s->frame_size / avctx->channels;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
samples = (int16_t *)frame->data[0];
// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
memset(state, 128, sizeof(state));
ff_init_range_decoder(&c, buf, buf_size);
ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8);
intlist_read(&c, state, s->predictor_k, s->num_taps, 0);
// dequantize
for (i = 0; i < s->num_taps; i++)
s->predictor_k[i] *= s->tap_quant[i];
if (s->lossless)
quant = 1;
else
quant = get_symbol(&c, state, 0) * SAMPLE_FACTOR;
// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
for (ch = 0; ch < s->channels; ch++)
{
int x = ch;
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 AVCodec ff_sonic_decoder = {
.name = "sonic",
.long_name = NULL_IF_CONFIG_SMALL("Sonic"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_SONIC,
.priv_data_size = sizeof(SonicContext),
.init = sonic_decode_init,
.close = sonic_decode_close,
.decode = sonic_decode_frame,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};
#endif /* CONFIG_SONIC_DECODER */
#if CONFIG_SONIC_ENCODER
const AVCodec ff_sonic_encoder = {
.name = "sonic",
.long_name = NULL_IF_CONFIG_SMALL("Sonic"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_SONIC,
.priv_data_size = sizeof(SonicContext),
.init = sonic_encode_init,
.encode2 = sonic_encode_frame,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
.capabilities = AV_CODEC_CAP_EXPERIMENTAL,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.close = sonic_encode_close,
};
#endif
#if CONFIG_SONIC_LS_ENCODER
const AVCodec ff_sonic_ls_encoder = {
.name = "sonicls",
.long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_SONIC_LS,
.priv_data_size = sizeof(SonicContext),
.init = sonic_encode_init,
.encode2 = sonic_encode_frame,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
.capabilities = AV_CODEC_CAP_EXPERIMENTAL,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.close = sonic_encode_close,
};
#endif
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