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提交 42cfb383 编写于 作者: M Mans Rullgard
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Remove Sonic experimental audio codec 

Since initially committed in 2004, this codec has only been touched
for maintenanance.  Functionally, it contains no novel ideas and
its intended audience is better served by existing mature codecs.
Signed-off-by: NMans Rullgard <mans@mansr.com>
上级 5a15602a
隐藏空白更改
内联 并排
Showing with 0 addition and 989 deletion
configure
浏览文件 @ 42cfb383
......@@ -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"
......
doc/general.texi
浏览文件 @ 42cfb383
......@@ -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
......
libavcodec/Makefile
浏览文件 @ 42cfb383
......@@ -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
......
libavcodec/allcodecs.c
浏览文件 @ 42cfb383
......@@ -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);
......
libavcodec/sonic.c 已删除 100644 → 0
浏览文件 @ 5a15602a
/*
* 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
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