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/*
* Simple free lossless/lossy audio codec
* Copyright (c) 2004 Alex Beregszaszi
* |
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* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* 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 |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/
#include "avcodec.h" |
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#include "get_bits.h" |
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#include "golomb.h" |
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#include "internal.h" |
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/** |
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* @file |
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* 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
|
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#define MID_SIDE 0
#define LEFT_SIDE 1
#define RIGHT_SIDE 2
|
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typedef struct SonicContext { |
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AVFrame frame; |
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int lossless, decorrelation; |
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|
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int num_taps, downsampling;
double quantization; |
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|
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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;
|
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#define LATTICE_SHIFT 10
#define SAMPLE_SHIFT 4
#define LATTICE_FACTOR (1 << LATTICE_SHIFT)
#define SAMPLE_FACTOR (1 << SAMPLE_SHIFT) |
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|
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#define BASE_QUANT 0.6
#define RATE_VARIATION 3.0 |
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static inline int divide(int a, int b)
{
if (a < 0) |
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return -( (-a + b/2)/b ); |
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else |
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return (a + b/2)/b; |
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}
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++) |
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set_se_golomb(pb, buf[i]); |
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return 1;
}
static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
{
int i; |
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|
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for (i = 0; i < entries; i++) |
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buf[i] = get_se_golomb(gb); |
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return 1;
}
#else
#define ADAPT_LEVEL 8
static int bits_to_store(uint64_t x)
{
int res = 0; |
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|
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while(x)
{ |
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res++;
x >>= 1; |
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}
return res;
}
static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max)
{
int i, bits;
if (!max) |
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return; |
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bits = bits_to_store(max);
for (i = 0; i < bits-1; i++) |
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put_bits(pb, 1, value & (1 << i)); |
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if ( (value | (1 << (bits-1))) <= max) |
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put_bits(pb, 1, value & (1 << (bits-1))); |
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}
static unsigned int read_uint_max(GetBitContext *gb, int max)
{
int i, bits, value = 0; |
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|
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if (!max) |
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return 0; |
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bits = bits_to_store(max);
for (i = 0; i < bits-1; i++) |
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if (get_bits1(gb))
value += 1 << i; |
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if ( (value | (1<<(bits-1))) <= max) |
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if (get_bits1(gb))
value += 1 << (bits-1); |
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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) |
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return -1; |
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|
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if (base_2_part)
{ |
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int energy = 0; |
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|
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for (i = 0; i < entries; i++)
energy += abs(buf[i]); |
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|
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low_bits = bits_to_store(energy / (entries * 2));
if (low_bits > 15)
low_bits = 15; |
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|
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put_bits(pb, 4, low_bits); |
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} |
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|
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for (i = 0; i < entries; i++)
{ |
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put_bits(pb, low_bits, abs(buf[i]));
copy[i] = abs(buf[i]) >> low_bits;
if (copy[i] > max)
max = abs(copy[i]); |
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}
bits = av_mallocz(4* entries*max);
if (!bits)
{ |
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// av_free(copy);
return -1; |
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} |
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|
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for (i = 0; i <= max; i++)
{ |
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for (j = 0; j < entries; j++)
if (copy[j] >= i)
bits[x++] = copy[j] > i; |
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}
// store bitstream
while (pos < x)
{ |
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int steplet = step >> 8; |
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|
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if (pos + steplet > x)
steplet = x - pos; |
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|
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for (i = 0; i < steplet; i++)
if (bits[i+pos] != dominant)
any = 1; |
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|
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put_bits(pb, 1, any); |
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|
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if (!any)
{
pos += steplet;
step += step / ADAPT_LEVEL;
}
else
{
int interloper = 0; |
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|
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while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
interloper++; |
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|
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// note change
write_uint_max(pb, interloper, (step >> 8) - 1); |
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|
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pos += interloper + 1;
step -= step / ADAPT_LEVEL;
} |
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|
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if (step < 256)
{
step = 65536 / step;
dominant = !dominant;
} |
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} |
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|
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// store signs
for (i = 0; i < entries; i++) |
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if (buf[i])
put_bits(pb, 1, buf[i] < 0); |
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// 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) |
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return -1; |
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|
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if (base_2_part)
{ |
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low_bits = get_bits(gb, 4); |
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|
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if (low_bits)
for (i = 0; i < entries; i++)
buf[i] = get_bits(gb, low_bits); |
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}
// av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits);
while (n_zeros < entries)
{ |
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int steplet = step >> 8; |
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|
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if (!get_bits1(gb))
{
for (i = 0; i < steplet; i++)
bits[x++] = dominant; |
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|
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if (!dominant)
n_zeros += steplet; |
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|
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step += step / ADAPT_LEVEL;
}
else
{
int actual_run = read_uint_max(gb, steplet-1); |
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|
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// av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run); |
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|
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for (i = 0; i < actual_run; i++)
bits[x++] = dominant; |
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|
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bits[x++] = !dominant; |
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|
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if (!dominant)
n_zeros += actual_run;
else
n_zeros++; |
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|
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step -= step / ADAPT_LEVEL;
} |
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|
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if (step < 256)
{
step = 65536 / step;
dominant = !dominant;
} |
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} |
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|
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// reconstruct unsigned values
n_zeros = 0;
for (i = 0; n_zeros < entries; i++)
{ |
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while(1)
{
if (pos >= entries)
{
pos = 0;
level += 1 << low_bits;
} |
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|
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if (buf[pos] >= level)
break; |
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|
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pos++;
} |
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|
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if (bits[i])
buf[pos] += 1 << low_bits;
else
n_zeros++; |
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|
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pos++; |
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}
// av_free(bits); |
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|
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// read signs
for (i = 0; i < entries; i++) |
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if (buf[i] && get_bits1(gb))
buf[i] = -buf[i]; |
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// 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--)
{ |
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int j, p, x = state[i]; |
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|
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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;
} |
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}
}
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]), |
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*state_ptr = &(state[order-2]); |
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for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--)
{ |
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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); |
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}
#else
for (i = order-2; i >= 0; i--)
{ |
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x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT); |
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}
#endif
|
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// don't drift too far, to avoid overflows |
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if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16);
if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16);
state[0] = x;
return x;
}
|
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#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER |
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// 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, |
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int *out, int out_entries, int channels, int *tap_quant) |
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{
int i;
int *state = av_mallocz(4* window_entries); |
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|
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memcpy(state, window, 4* window_entries); |
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|
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for (i = 0; i < out_entries; i++)
{ |
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int step = (i+1)*channels, k, j;
double xx = 0.0, xy = 0.0; |
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#if 1 |
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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;
} |
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#else |
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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;
} |
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#endif |
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if (xx == 0.0)
k = 0;
else
k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5)); |
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|
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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]); |
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|
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out[i] = k;
k *= tap_quant[i]; |
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#if 1 |
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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);
} |
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#else |
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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);
} |
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#endif
} |
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|
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av_free(state);
}
static inline int code_samplerate(int samplerate)
{
switch (samplerate)
{ |
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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; |
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}
return -1;
}
|
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static av_cold int sonic_encode_init(AVCodecContext *avctx) |
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{
SonicContext *s = avctx->priv_data;
PutBitContext pb;
int i, version = 0;
if (avctx->channels > MAX_CHANNELS) |
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{ |
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av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); |
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return -1; /* only stereo or mono for now */ |
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} |
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if (avctx->channels == 2) |
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s->decorrelation = MID_SIDE; |
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|
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if (avctx->codec->id == AV_CODEC_ID_SONIC_LS) |
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{ |
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s->lossless = 1;
s->num_taps = 32;
s->downsampling = 1;
s->quantization = 0.0; |
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}
else
{ |
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s->num_taps = 128;
s->downsampling = 2;
s->quantization = 1.0; |
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}
// max tap 2048
if ((s->num_taps < 32) || (s->num_taps > 1024) || |
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((s->num_taps>>5)<<5 != s->num_taps)) |
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{ |
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av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
return -1; |
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}
// generate taps
s->tap_quant = av_mallocz(4* s->num_taps);
for (i = 0; i < s->num_taps; i++) |
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s->tap_quant[i] = (int)(sqrt(i+1)); |
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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;
|
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s->tail_size = s->num_taps*s->channels;
s->tail = av_mallocz(4 * s->tail_size); |
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if (!s->tail) |
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return -1; |
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s->predictor_k = av_mallocz(4 * s->num_taps);
if (!s->predictor_k) |
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return -1; |
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for (i = 0; i < s->channels; i++)
{ |
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s->coded_samples[i] = av_mallocz(4* s->block_align);
if (!s->coded_samples[i])
return -1; |
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} |
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|
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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) |
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return -1; |
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avctx->extradata = av_mallocz(16);
if (!avctx->extradata) |
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return -1; |
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init_put_bits(&pb, avctx->extradata, 16*8);
put_bits(&pb, 2, version); // version
if (version == 1)
{ |
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put_bits(&pb, 2, s->channels);
put_bits(&pb, 4, code_samplerate(s->samplerate)); |
54f5fd22 |
}
put_bits(&pb, 1, s->lossless);
if (!s->lossless) |
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put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision |
cc078b9e |
put_bits(&pb, 2, s->decorrelation); |
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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;
|
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av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
bb270c08 |
version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); |
54f5fd22 |
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) |
8fa36ae0 |
return AVERROR(ENOMEM); |
54f5fd22 |
avctx->coded_frame->key_frame = 1;
avctx->frame_size = s->block_align*s->downsampling;
return 0;
}
|
98a6fff9 |
static av_cold int sonic_encode_close(AVCodecContext *avctx) |
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{
SonicContext *s = avctx->priv_data;
int i;
av_freep(&avctx->coded_frame);
for (i = 0; i < s->channels; i++) |
bb270c08 |
av_free(s->coded_samples[i]); |
54f5fd22 |
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;
}
|
a44cbc1c |
static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr) |
54f5fd22 |
{
SonicContext *s = avctx->priv_data;
PutBitContext pb;
int i, j, ch, quant = 0, x = 0; |
a44cbc1c |
int ret;
const short *samples = (const int16_t*)frame->data[0]; |
54f5fd22 |
|
bcaf64b6 |
if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0) |
a44cbc1c |
return ret;
init_put_bits(&pb, avpkt->data, avpkt->size); |
54f5fd22 |
// short -> internal
for (i = 0; i < s->frame_size; i++) |
bb270c08 |
s->int_samples[i] = samples[i]; |
54f5fd22 |
if (!s->lossless) |
bb270c08 |
for (i = 0; i < s->frame_size; i++)
s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT; |
54f5fd22 |
|
cc078b9e |
switch(s->decorrelation)
{ |
bb270c08 |
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; |
cc078b9e |
} |
54f5fd22 |
memset(s->window, 0, 4* s->window_size); |
115329f1 |
|
54f5fd22 |
for (i = 0; i < s->tail_size; i++) |
bb270c08 |
s->window[x++] = s->tail[i]; |
54f5fd22 |
for (i = 0; i < s->frame_size; i++) |
bb270c08 |
s->window[x++] = s->int_samples[i]; |
115329f1 |
|
54f5fd22 |
for (i = 0; i < s->tail_size; i++) |
bb270c08 |
s->window[x++] = 0; |
54f5fd22 |
for (i = 0; i < s->tail_size; i++) |
bb270c08 |
s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i]; |
54f5fd22 |
// generate taps
modified_levinson_durbin(s->window, s->window_size, |
bb270c08 |
s->predictor_k, s->num_taps, s->channels, s->tap_quant); |
54f5fd22 |
if (intlist_write(&pb, s->predictor_k, s->num_taps, 0) < 0) |
bb270c08 |
return -1; |
54f5fd22 |
for (ch = 0; ch < s->channels; ch++)
{ |
bb270c08 |
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;
} |
54f5fd22 |
} |
115329f1 |
// simple rate control code |
54f5fd22 |
if (!s->lossless)
{ |
bb270c08 |
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);
}
} |
115329f1 |
|
bb270c08 |
energy2 = sqrt(energy2/(s->channels*s->block_align));
energy1 = sqrt(2.0)*energy1/(s->channels*s->block_align); |
115329f1 |
|
bb270c08 |
// increase bitrate when samples are like a gaussian distribution
// reduce bitrate when samples are like a two-tailed exponential distribution |
115329f1 |
|
bb270c08 |
if (energy2 > energy1)
energy2 += (energy2-energy1)*RATE_VARIATION; |
115329f1 |
|
bb270c08 |
quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
// av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2); |
54f5fd22 |
|
bb270c08 |
if (quant < 1)
quant = 1; |
512beea5 |
if (quant > 65534)
quant = 65534; |
115329f1 |
|
bb270c08 |
set_ue_golomb(&pb, quant); |
115329f1 |
|
bb270c08 |
quant *= SAMPLE_FACTOR; |
54f5fd22 |
}
// write out coded samples
for (ch = 0; ch < s->channels; ch++)
{ |
bb270c08 |
if (!s->lossless)
for (i = 0; i < s->block_align; i++)
s->coded_samples[ch][i] = divide(s->coded_samples[ch][i], quant); |
54f5fd22 |
|
bb270c08 |
if (intlist_write(&pb, s->coded_samples[ch], s->block_align, 1) < 0)
return -1; |
54f5fd22 |
}
// av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
flush_put_bits(&pb); |
a44cbc1c |
avpkt->size = (put_bits_count(&pb)+7)/8;
*got_packet_ptr = 1;
return 0; |
54f5fd22 |
} |
b250f9c6 |
#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */ |
54f5fd22 |
|
b250f9c6 |
#if CONFIG_SONIC_DECODER |
359a9979 |
static const int samplerate_table[] =
{ 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
|
98a6fff9 |
static av_cold int sonic_decode_init(AVCodecContext *avctx) |
54f5fd22 |
{
SonicContext *s = avctx->priv_data;
GetBitContext gb;
int i, version; |
115329f1 |
|
54f5fd22 |
s->channels = avctx->channels;
s->samplerate = avctx->sample_rate; |
115329f1 |
|
6f9803e5 |
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
|
54f5fd22 |
if (!avctx->extradata)
{ |
bb270c08 |
av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
return -1; |
54f5fd22 |
} |
115329f1 |
|
54f5fd22 |
init_get_bits(&gb, avctx->extradata, avctx->extradata_size); |
115329f1 |
|
54f5fd22 |
version = get_bits(&gb, 2);
if (version > 1)
{ |
bb270c08 |
av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
return -1; |
54f5fd22 |
}
if (version == 1)
{ |
bb270c08 |
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); |
54f5fd22 |
}
if (s->channels > MAX_CHANNELS)
{ |
bb270c08 |
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
return -1; |
54f5fd22 |
}
s->lossless = get_bits1(&gb);
if (!s->lossless) |
bb270c08 |
skip_bits(&gb, 3); // XXX FIXME |
cc078b9e |
s->decorrelation = get_bits(&gb, 2); |
54f5fd22 |
s->downsampling = get_bits(&gb, 2); |
8a0cd587 |
if (!s->downsampling) {
av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n");
return AVERROR_INVALIDDATA;
}
|
54f5fd22 |
s->num_taps = (get_bits(&gb, 5)+1)<<5;
if (get_bits1(&gb)) // XXX FIXME |
bb270c08 |
av_log(avctx, AV_LOG_INFO, "Custom quant table\n"); |
115329f1 |
|
d7c91c4c |
s->block_align = (int)(2048.0*s->samplerate/44100)/s->downsampling; |
54f5fd22 |
s->frame_size = s->channels*s->block_align*s->downsampling;
// avctx->frame_size = s->block_align;
|
cc078b9e |
av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", |
bb270c08 |
version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); |
54f5fd22 |
// generate taps
s->tap_quant = av_mallocz(4* s->num_taps);
for (i = 0; i < s->num_taps; i++) |
bb270c08 |
s->tap_quant[i] = (int)(sqrt(i+1)); |
115329f1 |
|
54f5fd22 |
s->predictor_k = av_mallocz(4* s->num_taps); |
115329f1 |
|
54f5fd22 |
for (i = 0; i < s->channels; i++)
{ |
bb270c08 |
s->predictor_state[i] = av_mallocz(4* s->num_taps);
if (!s->predictor_state[i])
return -1; |
54f5fd22 |
}
for (i = 0; i < s->channels; i++)
{ |
bb270c08 |
s->coded_samples[i] = av_mallocz(4* s->block_align);
if (!s->coded_samples[i])
return -1; |
54f5fd22 |
}
s->int_samples = av_mallocz(4* s->frame_size);
|
5d6e4c16 |
avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
54f5fd22 |
return 0;
}
|
98a6fff9 |
static av_cold int sonic_decode_close(AVCodecContext *avctx) |
54f5fd22 |
{
SonicContext *s = avctx->priv_data;
int i; |
115329f1 |
|
54f5fd22 |
av_free(s->int_samples);
av_free(s->tap_quant);
av_free(s->predictor_k); |
115329f1 |
|
54f5fd22 |
for (i = 0; i < s->channels; i++)
{ |
bb270c08 |
av_free(s->predictor_state[i]);
av_free(s->coded_samples[i]); |
54f5fd22 |
} |
115329f1 |
|
54f5fd22 |
return 0;
}
static int sonic_decode_frame(AVCodecContext *avctx, |
6f9803e5 |
void *data, int *got_frame_ptr, |
7a00bbad |
AVPacket *avpkt) |
54f5fd22 |
{ |
7a00bbad |
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size; |
54f5fd22 |
SonicContext *s = avctx->priv_data;
GetBitContext gb; |
6f9803e5 |
int i, quant, ch, j, ret; |
89cd95b1 |
int16_t *samples; |
54f5fd22 |
if (buf_size == 0) return 0;
|
6f9803e5 |
s->frame.nb_samples = s->frame_size; |
874c5b02 |
if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) { |
6f9803e5 |
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
} |
89cd95b1 |
samples = (int16_t *)s->frame.data[0]; |
6f9803e5 |
|
54f5fd22 |
// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size); |
115329f1 |
|
54f5fd22 |
init_get_bits(&gb, buf, buf_size*8); |
115329f1 |
|
54f5fd22 |
intlist_read(&gb, s->predictor_k, s->num_taps, 0);
// dequantize
for (i = 0; i < s->num_taps; i++) |
bb270c08 |
s->predictor_k[i] *= s->tap_quant[i]; |
54f5fd22 |
if (s->lossless) |
bb270c08 |
quant = 1; |
54f5fd22 |
else |
bb270c08 |
quant = get_ue_golomb(&gb) * SAMPLE_FACTOR; |
54f5fd22 |
// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
for (ch = 0; ch < s->channels; ch++)
{ |
bb270c08 |
int x = ch; |
54f5fd22 |
|
bb270c08 |
predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps); |
115329f1 |
|
bb270c08 |
intlist_read(&gb, s->coded_samples[ch], s->block_align, 1); |
54f5fd22 |
|
bb270c08 |
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;
} |
115329f1 |
|
bb270c08 |
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;
} |
54f5fd22 |
|
bb270c08 |
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]; |
54f5fd22 |
} |
115329f1 |
|
cc078b9e |
switch(s->decorrelation)
{ |
bb270c08 |
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; |
cc078b9e |
} |
54f5fd22 |
if (!s->lossless) |
bb270c08 |
for (i = 0; i < s->frame_size; i++)
s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT); |
54f5fd22 |
// internal -> short
for (i = 0; i < s->frame_size; i++) |
aee481ce |
samples[i] = av_clip_int16(s->int_samples[i]); |
54f5fd22 |
align_get_bits(&gb);
|
6f9803e5 |
*got_frame_ptr = 1;
*(AVFrame*)data = s->frame; |
54f5fd22 |
return (get_bits_count(&gb)+7)/8;
} |
359a9979 |
|
e7e2df27 |
AVCodec ff_sonic_decoder = { |
ba10207b |
.name = "sonic",
.type = AVMEDIA_TYPE_AUDIO, |
7a72695c |
.id = AV_CODEC_ID_SONIC, |
ba10207b |
.priv_data_size = sizeof(SonicContext),
.init = sonic_decode_init,
.close = sonic_decode_close,
.decode = sonic_decode_frame, |
7ed9abf7 |
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_EXPERIMENTAL, |
359a9979 |
.long_name = NULL_IF_CONFIG_SMALL("Sonic"),
}; |
2a43a093 |
#endif /* CONFIG_SONIC_DECODER */ |
54f5fd22 |
|
b250f9c6 |
#if CONFIG_SONIC_ENCODER |
e7e2df27 |
AVCodec ff_sonic_encoder = { |
ba10207b |
.name = "sonic",
.type = AVMEDIA_TYPE_AUDIO, |
7a72695c |
.id = AV_CODEC_ID_SONIC, |
ba10207b |
.priv_data_size = sizeof(SonicContext),
.init = sonic_encode_init, |
a44cbc1c |
.encode2 = sonic_encode_frame, |
7ed9abf7 |
.capabilities = CODEC_CAP_EXPERIMENTAL, |
ba10207b |
.close = sonic_encode_close, |
fe4bf374 |
.long_name = NULL_IF_CONFIG_SMALL("Sonic"), |
54f5fd22 |
}; |
f544a5fc |
#endif |
54f5fd22 |
|
b250f9c6 |
#if CONFIG_SONIC_LS_ENCODER |
e7e2df27 |
AVCodec ff_sonic_ls_encoder = { |
ba10207b |
.name = "sonicls",
.type = AVMEDIA_TYPE_AUDIO, |
7a72695c |
.id = AV_CODEC_ID_SONIC_LS, |
ba10207b |
.priv_data_size = sizeof(SonicContext),
.init = sonic_encode_init, |
a44cbc1c |
.encode2 = sonic_encode_frame, |
7ed9abf7 |
.capabilities = CODEC_CAP_EXPERIMENTAL, |
ba10207b |
.close = sonic_encode_close, |
fe4bf374 |
.long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"), |
54f5fd22 |
};
#endif |