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ffmpeg/libavcodec/sonic.c

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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 "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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