Commit e1ba5fc9 authored by Paul B Mahol's avatar Paul B Mahol

dcaenc: update

Long story short: previous code was useless and was port of older
dcaenc, this commit just "sync" with current dcaenc, hopefuly
making this encoder more useful.
Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent 8f0db04b
/*
* DCA encoder
* Copyright (C) 2008 Alexander E. Patrakov
* Copyright (C) 2008-2012 Alexander E. Patrakov
* 2010 Benjamin Larsson
* 2011 Xiang Wang
*
......@@ -21,211 +21,678 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/avassert.h"
#include "avcodec.h"
#include "dca.h"
#include "dcadata.h"
#include "dcaenc.h"
#include "internal.h"
#include "put_bits.h"
#include "dcaenc.h"
#include "dcadata.h"
#include "dca.h"
#undef NDEBUG
#define MAX_CHANNELS 6
#define DCA_SUBBANDS_32 32
#define DCA_MAX_FRAME_SIZE 16383
#define DCA_MAX_FRAME_SIZE 16384
#define DCA_HEADER_SIZE 13
#define DCA_LFE_SAMPLES 8
#define DCA_SUBBANDS 32 ///< Subband activity count
#define QUANTIZER_BITS 16
#define DCA_SUBBANDS 32
#define SUBFRAMES 1
#define SUBSUBFRAMES 4
#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
#define LFE_BITS 8
#define LFE_INTERPOLATION 64
#define LFE_PRESENT 2
#define LFE_MISSING 0
static const int8_t dca_lfe_index[] = {
1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
};
static const int8_t dca_channel_reorder_lfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, 2, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, 3, -1, -1, -1, -1 },
{ 0, 1, -1, 2, 3, -1, -1, -1, -1 },
{ 1, 2, 0, -1, 3, 4, -1, -1, -1 },
{ 2, 3, -1, 0, 1, 4, 5, -1, -1 },
{ 1, 2, 0, -1, 3, 4, 5, -1, -1 },
{ 0, -1, 4, 5, 2, 3, 1, -1, -1 },
{ 3, 4, 1, -1, 0, 2, 5, 6, -1 },
{ 2, 3, -1, 5, 7, 0, 1, 4, 6 },
{ 3, 4, 1, -1, 0, 2, 5, 7, 6 },
};
#define SUBSUBFRAMES 2
#define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
#define AUBANDS 25
static const int8_t dca_channel_reorder_nolfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
{ 0, 1, 2, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, 3, -1, -1, -1, -1, -1 },
{ 0, 1, 2, 3, -1, -1, -1, -1, -1 },
{ 1, 2, 0, 3, 4, -1, -1, -1, -1 },
{ 2, 3, 0, 1, 4, 5, -1, -1, -1 },
{ 1, 2, 0, 3, 4, 5, -1, -1, -1 },
{ 0, 4, 5, 2, 3, 1, -1, -1, -1 },
{ 3, 4, 1, 0, 2, 5, 6, -1, -1 },
{ 2, 3, 5, 7, 0, 1, 4, 6, -1 },
{ 3, 4, 1, 0, 2, 5, 7, 6, -1 },
};
typedef struct {
typedef struct DCAContext {
PutBitContext pb;
int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
int start[MAX_CHANNELS];
int frame_size;
int prim_channels;
int frame_bits;
int fullband_channels;
int channels;
int lfe_channel;
int sample_rate_code;
int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
int samplerate_index;
int bitrate_index;
int channel_config;
const int32_t *band_interpolation;
const int32_t *band_spectrum;
int lfe_scale_factor;
int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
softfloat lfe_quant;
int32_t lfe_peak_cb;
int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */
int32_t subband[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS];
int32_t quantized[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS];
int32_t peak_cb[DCA_SUBBANDS][MAX_CHANNELS];
int32_t downsampled_lfe[DCA_LFE_SAMPLES];
int32_t masking_curve_cb[SUBSUBFRAMES][256];
int abits[DCA_SUBBANDS][MAX_CHANNELS];
int scale_factor[DCA_SUBBANDS][MAX_CHANNELS];
softfloat quant[DCA_SUBBANDS][MAX_CHANNELS];
int32_t eff_masking_curve_cb[256];
int32_t band_masking_cb[32];
int32_t worst_quantization_noise;
int32_t worst_noise_ever;
int consumed_bits;
} DCAContext;
int a_mode; ///< audio channels arrangement
int num_channel;
int lfe_state;
int lfe_offset;
const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
static int32_t cos_table[2048];
static int32_t band_interpolation[2][512];
static int32_t band_spectrum[2][8];
static int32_t auf[9][AUBANDS][256];
static int32_t cb_to_add[256];
static int32_t cb_to_level[2048];
static int32_t lfe_fir_64i[512];
int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
} DCAContext;
/* Transfer function of outer and middle ear, Hz -> dB */
static double hom(double f)
{
double f1 = f / 1000;
return -3.64 * pow(f1, -0.8)
+ 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
- 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
- 0.0006 * (f1 * f1) * (f1 * f1);
}
static double gammafilter(int i, double f)
{
double h = (f - fc[i]) / erb[i];
h = 1 + h * h;
h = 1 / (h * h);
return 20 * log10(h);
}
static int encode_init(AVCodecContext *avctx)
{
DCAContext *c = avctx->priv_data;
uint64_t layout = avctx->channel_layout;
int i, min_frame_bits;
c->fullband_channels = c->channels = avctx->channels;
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
c->band_interpolation = band_interpolation[1];
c->band_spectrum = band_spectrum[1];
c->worst_quantization_noise = -2047;
c->worst_noise_ever = -2047;
if (!layout) {
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
"encoder will guess the layout, but it "
"might be incorrect.\n");
layout = av_get_default_channel_layout(avctx->channels);
}
switch (layout) {
case AV_CH_LAYOUT_MONO: c->channel_config = 0; break;
case AV_CH_LAYOUT_STEREO: c->channel_config = 2; break;
case AV_CH_LAYOUT_2_2: c->channel_config = 8; break;
case AV_CH_LAYOUT_5POINT0: c->channel_config = 9; break;
case AV_CH_LAYOUT_5POINT1: c->channel_config = 9; break;
default:
av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");
return AVERROR_PATCHWELCOME;
}
if (c->lfe_channel)
c->fullband_channels--;
for (i = 0; i < 9; i++) {
if (sample_rates[i] == avctx->sample_rate)
break;
}
if (i == 9)
return AVERROR(EINVAL);
c->samplerate_index = i;
if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {
av_log(avctx, AV_LOG_ERROR, "Bit rate %i not supported.", avctx->bit_rate);
return AVERROR(EINVAL);
}
for (i = 0; dca_bit_rates[i] < avctx->bit_rate; i++)
;
c->bitrate_index = i;
avctx->bit_rate = dca_bit_rates[i];
c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);
min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;
if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))
return AVERROR(EINVAL);
c->frame_size = (c->frame_bits + 7) / 8;
avctx->frame_size = 32 * SUBBAND_SAMPLES;
if (!cos_table[0]) {
int j, k;
for (i = 0; i < 2048; i++) {
cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
cb_to_level[i] = (int32_t)(0x7fffffff * pow(10, -0.005 * i));
}
for (i = 0; i < 512; i++) {
lfe_fir_64i[i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);
band_interpolation[0][i] = (int32_t)(0x1000000000ULL * fir_32bands_perfect[i]);
band_interpolation[1][i] = (int32_t)(0x1000000000ULL * fir_32bands_nonperfect[i]);
}
for (i = 0; i < 9; i++) {
for (j = 0; j < AUBANDS; j++) {
for (k = 0; k < 256; k++) {
double freq = sample_rates[i] * (k + 0.5) / 512;
auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));
}
}
}
for (i = 0; i < 256; i++) {
double add = 1 + pow(10, -0.01 * i);
cb_to_add[i] = (int32_t)(100 * log10(add));
}
for (j = 0; j < 8; j++) {
double accum = 0;
for (i = 0; i < 512; i++) {
double reconst = fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);
accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
}
band_spectrum[0][j] = (int32_t)(200 * log10(accum));
}
for (j = 0; j < 8; j++) {
double accum = 0;
for (i = 0; i < 512; i++) {
double reconst = fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);
accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
}
band_spectrum[1][j] = (int32_t)(200 * log10(accum));
}
}
return 0;
}
static inline int32_t cos_t(int x)
{
return cos_table[x & 2047];
}
static inline int32_t sin_t(int x)
{
return cos_t(x - 512);
}
static int32_t cos_table[128];
static inline int32_t half32(int32_t a)
{
return (a + 1) >> 1;
}
static inline int32_t mul32(int32_t a, int32_t b)
{
int64_t r = (int64_t) a * b;
/* round the result before truncating - improves accuracy */
return (r + 0x80000000) >> 32;
int64_t r = (int64_t)a * b + 0x80000000ULL;
return r >> 32;
}
static void subband_transform(DCAContext *c, const int32_t *input)
{
int ch, subs, i, k, j;
for (ch = 0; ch < c->fullband_channels; ch++) {
/* History is copied because it is also needed for PSY */
int32_t hist[512];
int hist_start = 0;
for (i = 0; i < 512; i++)
hist[i] = c->history[i][ch];
for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
int32_t accum[64];
int32_t resp;
int band;
/* Calculate the convolutions at once */
for (i = 0; i < 64; i++)
accum[i] = 0;
for (k = 0, i = hist_start, j = 0;
i < 512; k = (k + 1) & 63, i++, j++)
accum[k] += mul32(hist[i], c->band_interpolation[j]);
for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)
accum[k] += mul32(hist[i], c->band_interpolation[j]);
for (k = 16; k < 32; k++)
accum[k] = accum[k] - accum[31 - k];
for (k = 32; k < 48; k++)
accum[k] = accum[k] + accum[95 - k];
for (band = 0; band < 32; band++) {
resp = 0;
for (i = 16; i < 48; i++) {
int s = (2 * band + 1) * (2 * (i + 16) + 1);
resp += mul32(accum[i], cos_t(s << 3)) >> 3;
}
c->subband[subs][band][ch] = ((band + 1) & 2) ? -resp : resp;
}
/* Copy in 32 new samples from input */
for (i = 0; i < 32; i++)
hist[i + hist_start] = input[(subs * 32 + i) * c->channels + ch];
hist_start = (hist_start + 32) & 511;
}
}
}
static void lfe_downsample(DCAContext *c, const int32_t *input)
{
/* FIXME: make 128x LFE downsampling possible */
int i, j, lfes;
int32_t hist[512];
int32_t accum;
int hist_start = 0;
for (i = 0; i < 512; i++)
hist[i] = c->history[i][c->channels - 1];
for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
/* Calculate the convolution */
accum = 0;
for (i = hist_start, j = 0; i < 512; i++, j++)
accum += mul32(hist[i], lfe_fir_64i[j]);
for (i = 0; i < hist_start; i++, j++)
accum += mul32(hist[i], lfe_fir_64i[j]);
c->downsampled_lfe[lfes] = accum;
/* Copy in 64 new samples from input */
for (i = 0; i < 64; i++)
hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + c->channels - 1];
hist_start = (hist_start + 64) & 511;
}
}
/* Integer version of the cosine modulated Pseudo QMF */
typedef struct {
int32_t re;
int32_t im;
} cplx32;
static void qmf_init(void)
static void fft(const int32_t in[2 * 256], cplx32 out[256])
{
int i;
int32_t c[17], s[17];
s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */
c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */
for (i = 1; i <= 16; i++) {
s[i] = 2 * (mul32(c[i - 1], 105372028) + mul32(s[i - 1], 2144896908));
c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
cplx32 buf[256], rin[256], rout[256];
int i, j, k, l;
/* do two transforms in parallel */
for (i = 0; i < 256; i++) {
/* Apply the Hann window */
rin[i].re = mul32(in[2 * i], 0x3fffffff - (cos_t(8 * i + 2) >> 1));
rin[i].im = mul32(in[2 * i + 1], 0x3fffffff - (cos_t(8 * i + 6) >> 1));
}
/* pre-rotation */
for (i = 0; i < 256; i++) {
buf[i].re = mul32(cos_t(4 * i + 2), rin[i].re)
- mul32(sin_t(4 * i + 2), rin[i].im);
buf[i].im = mul32(cos_t(4 * i + 2), rin[i].im)
+ mul32(sin_t(4 * i + 2), rin[i].re);
}
for (j = 256, l = 1; j != 1; j >>= 1, l <<= 1) {
for (k = 0; k < 256; k += j) {
for (i = k; i < k + j / 2; i++) {
cplx32 sum, diff;
int t = 8 * l * i;
sum.re = buf[i].re + buf[i + j / 2].re;
sum.im = buf[i].im + buf[i + j / 2].im;
diff.re = buf[i].re - buf[i + j / 2].re;
diff.im = buf[i].im - buf[i + j / 2].im;
buf[i].re = half32(sum.re);
buf[i].im = half32(sum.im);
buf[i + j / 2].re = mul32(diff.re, cos_t(t))
- mul32(diff.im, sin_t(t));
buf[i + j / 2].im = mul32(diff.im, cos_t(t))
+ mul32(diff.re, sin_t(t));
}
}
}
/* post-rotation */
for (i = 0; i < 256; i++) {
int b = ff_reverse[i];
rout[i].re = mul32(buf[b].re, cos_t(4 * i))
- mul32(buf[b].im, sin_t(4 * i));
rout[i].im = mul32(buf[b].im, cos_t(4 * i))
+ mul32(buf[b].re, sin_t(4 * i));
}
for (i = 0; i < 256; i++) {
/* separate the results of the two transforms */
cplx32 o1, o2;
o1.re = rout[i].re - rout[255 - i].re;
o1.im = rout[i].im + rout[255 - i].im;
o2.re = rout[i].im - rout[255 - i].im;
o2.im = -rout[i].re - rout[255 - i].re;
/* combine them into one long transform */
out[i].re = mul32( o1.re + o2.re, cos_t(2 * i + 1))
+ mul32( o1.im - o2.im, sin_t(2 * i + 1));
out[i].im = mul32( o1.im + o2.im, cos_t(2 * i + 1))
+ mul32(-o1.re + o2.re, sin_t(2 * i + 1));
}
}
for (i = 0; i < 16; i++) {
cos_table[i ] = c[i] >> 3; /* avoid output overflow */
cos_table[i + 16] = s[16 - i] >> 3;
cos_table[i + 32] = -s[i] >> 3;
cos_table[i + 48] = -c[16 - i] >> 3;
cos_table[i + 64] = -c[i] >> 3;
cos_table[i + 80] = -s[16 - i] >> 3;
cos_table[i + 96] = s[i] >> 3;
cos_table[i + 112] = c[16 - i] >> 3;
static int32_t get_cb(int32_t in)
{
int i, res;
res = 0;
if (in < 0)
in = -in;
for (i = 1024; i > 0; i >>= 1) {
if (cb_to_level[i + res] >= in)
res += i;
}
return -res;
}
static int32_t band_delta_factor(int band, int sample_num)
static int32_t add_cb(int32_t a, int32_t b)
{
int index = band * (2 * sample_num + 1);
if (band == 0)
return 0x07ffffff;
else
return cos_table[index & 127];
if (a < b)
FFSWAP(int32_t, a, b);
if (a - b >= 256)
return a;
return a + cb_to_add[a - b];
}
static void add_new_samples(DCAContext *c, const int32_t *in,
int count, int channel)
static void adjust_jnd(int samplerate_index,
const int32_t in[512], int32_t out_cb[256])
{
int i;
int32_t power[256];
cplx32 out[256];
int32_t out_cb_unnorm[256];
int32_t denom;
const int32_t ca_cb = -1114;
const int32_t cs_cb = 928;
int i, j;
fft(in, out);
/* Place new samples into the history buffer */
for (i = 0; i < count; i++) {
c->history[channel][c->start[channel] + i] = in[i];
av_assert0(c->start[channel] + i < 512);
for (j = 0; j < 256; j++) {
power[j] = add_cb(get_cb(out[j].re), get_cb(out[j].im));
out_cb_unnorm[j] = -2047; /* and can only grow */
}
c->start[channel] += count;
if (c->start[channel] == 512)
c->start[channel] = 0;
av_assert0(c->start[channel] < 512);
for (i = 0; i < AUBANDS; i++) {
denom = ca_cb; /* and can only grow */
for (j = 0; j < 256; j++)
denom = add_cb(denom, power[j] + auf[samplerate_index][i][j]);
for (j = 0; j < 256; j++)
out_cb_unnorm[j] = add_cb(out_cb_unnorm[j],
-denom + auf[samplerate_index][i][j]);
}
for (j = 0; j < 256; j++)
out_cb[j] = add_cb(out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);
}
static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
int channel)
typedef void (*walk_band_t)(DCAContext *c, int band1, int band2, int f,
int32_t spectrum1, int32_t spectrum2, int channel,
int32_t * arg);
static void walk_band_low(DCAContext *c, int band, int channel,
walk_band_t walk, int32_t *arg)
{
int band, i, j, k;
int32_t resp;
int32_t accum[DCA_SUBBANDS_32] = {0};
int f;
add_new_samples(c, in, DCA_SUBBANDS_32, channel);
if (band == 0) {
for (f = 0; f < 4; f++)
walk(c, 0, 0, f, 0, -2047, channel, arg);
} else {
for (f = 0; f < 8; f++)
walk(c, band, band - 1, 8 * band - 4 + f,
c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);
}
}
/* Calculate the dot product of the signal with the (possibly inverted)
reference decoder's response to this vector:
(0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
so that -1.0 cancels 1.0 from the previous step */
static void walk_band_high(DCAContext *c, int band, int channel,
walk_band_t walk, int32_t *arg)
{
int f;
for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
for (i = 0; i < c->start[channel]; k++, j++, i++)
accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
if (band == 31) {
for (f = 0; f < 4; f++)
walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);
} else {
for (f = 0; f < 8; f++)
walk(c, band, band + 1, 8 * band + 4 + f,
c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);
}
}
resp = 0;
/* TODO: implement FFT instead of this naive calculation */
for (band = 0; band < DCA_SUBBANDS_32; band++) {
for (j = 0; j < 32; j++)
resp += mul32(accum[j], band_delta_factor(band, j));
static void update_band_masking(DCAContext *c, int band1, int band2,
int f, int32_t spectrum1, int32_t spectrum2,
int channel, int32_t * arg)
{
int32_t value = c->eff_masking_curve_cb[f] - spectrum1;
out[band] = (band & 2) ? (-resp) : resp;
if (value < c->band_masking_cb[band1])
c->band_masking_cb[band1] = value;
}
static void calc_masking(DCAContext *c, const int32_t *input)
{
int i, k, band, ch, ssf;
int32_t data[512];
for (i = 0; i < 256; i++)
for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
c->masking_curve_cb[ssf][i] = -2047;
for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
for (ch = 0; ch < c->fullband_channels; ch++) {
for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
data[i] = c->history[k][ch];
for (k -= 512; i < 512; i++, k++)
data[i] = input[k * c->channels + ch];
adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);
}
for (i = 0; i < 256; i++) {
int32_t m = 2048;
for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
if (c->masking_curve_cb[ssf][i] < m)
m = c->masking_curve_cb[ssf][i];
c->eff_masking_curve_cb[i] = m;
}
for (band = 0; band < 32; band++) {
c->band_masking_cb[band] = 2048;
walk_band_low(c, band, 0, update_band_masking, NULL);
walk_band_high(c, band, 0, update_band_masking, NULL);
}
}
static int32_t lfe_fir_64i[512];
static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
static void find_peaks(DCAContext *c)
{
int i, j;
int channel = c->prim_channels;
int32_t accum = 0;
add_new_samples(c, in, LFE_INTERPOLATION, channel);
for (i = c->start[channel], j = 0; i < 512; i++, j++)
accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
for (i = 0; i < c->start[channel]; i++, j++)
accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
return accum;
int band, ch;
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++) {
int sample;
int32_t m = 0;
for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
int32_t s = abs(c->subband[sample][band][ch]);
if (m < s)
m = s;
}
c->peak_cb[band][ch] = get_cb(m);
}
if (c->lfe_channel) {
int sample;
int32_t m = 0;
for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)
if (m < abs(c->downsampled_lfe[sample]))
m = abs(c->downsampled_lfe[sample]);
c->lfe_peak_cb = get_cb(m);
}
}
static void init_lfe_fir(void)
static const int snr_fudge = 128;
#define USED_1ABITS 1
#define USED_NABITS 2
#define USED_26ABITS 4
static int init_quantization_noise(DCAContext *c, int noise)
{
static int initialized = 0;
int i;
if (initialized)
return;
int ch, band, ret = 0;
c->consumed_bits = 132 + 493 * c->fullband_channels;
if (c->lfe_channel)
c->consumed_bits += 72;
/* attempt to guess the bit distribution based on the prevoius frame */
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;
if (snr_cb >= 1312) {
c->abits[band][ch] = 26;
ret |= USED_26ABITS;
} else if (snr_cb >= 222) {
c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);
ret |= USED_NABITS;
} else if (snr_cb >= 0) {
c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);
ret |= USED_NABITS;
} else {
c->abits[band][ch] = 1;
ret |= USED_1ABITS;
}
}
}
for (i = 0; i < 512; i++)
lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
initialized = 1;
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++) {
c->consumed_bits += bit_consumption[c->abits[band][ch]];
}
return ret;
}
static void assign_bits(DCAContext *c)
{
/* Find the bounds where the binary search should work */
int low, high, down;
int used_abits = 0;
init_quantization_noise(c, c->worst_quantization_noise);
low = high = c->worst_quantization_noise;
if (c->consumed_bits > c->frame_bits) {
while (c->consumed_bits > c->frame_bits) {
av_assert0(used_abits != USED_1ABITS);
low = high;
high += snr_fudge;
used_abits = init_quantization_noise(c, high);
}
} else {
while (c->consumed_bits <= c->frame_bits) {
high = low;
if (used_abits == USED_26ABITS)
goto out; /* The requested bitrate is too high, pad with zeros */
low -= snr_fudge;
used_abits = init_quantization_noise(c, low);
}
}
/* Now do a binary search between low and high to see what fits */
for (down = snr_fudge >> 1; down; down >>= 1) {
init_quantization_noise(c, high - down);
if (c->consumed_bits <= c->frame_bits)
high -= down;
}
init_quantization_noise(c, high);
out:
c->worst_quantization_noise = high;
if (high > c->worst_noise_ever)
c->worst_noise_ever = high;
}
static void shift_history(DCAContext *c, const int32_t *input)
{
int k, ch;
for (k = 0; k < 512; k++)
for (ch = 0; ch < c->channels; ch++)
c->history[k][ch] = input[k * c->channels + ch];
}
static int32_t quantize_value(int32_t value, softfloat quant)
{
int32_t offset = 1 << (quant.e - 1);
value = mul32(value, quant.m) + offset;
value = value >> quant.e;
return value;
}
static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
{
int32_t peak;
int our_nscale, try_remove;
softfloat our_quant;
av_assert0(peak_cb <= 0);
av_assert0(peak_cb >= -2047);
our_nscale = 127;
peak = cb_to_level[-peak_cb];
for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
continue;
our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
continue;
our_nscale -= try_remove;
}
if (our_nscale >= 125)
our_nscale = 124;
quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
return our_nscale;
}
static void calc_scales(DCAContext *c)
{
int band, ch;
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++)
c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],
c->abits[band][ch],
&c->quant[band][ch]);
if (c->lfe_channel)
c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);
}
static void quantize_all(DCAContext *c)
{
int sample, band, ch;
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++)
c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);
}
static void put_frame_header(DCAContext *c)
......@@ -244,19 +711,19 @@ static void put_frame_header(DCAContext *c)
put_bits(&c->pb, 1, 0);
/* Number of PCM sample blocks */
put_bits(&c->pb, 7, PCM_SAMPLES-1);
put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
/* Primary frame byte size */
put_bits(&c->pb, 14, c->frame_size-1);
put_bits(&c->pb, 14, c->frame_size - 1);
/* Audio channel arrangement: L + R (stereo) */
put_bits(&c->pb, 6, c->num_channel);
/* Audio channel arrangement */
put_bits(&c->pb, 6, c->channel_config);
/* Core audio sampling frequency */
put_bits(&c->pb, 4, c->sample_rate_code);
put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);
/* Transmission bit rate: 1411.2 kbps */
put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
/* Transmission bit rate */
put_bits(&c->pb, 5, c->bitrate_index);
/* Embedded down mix: disabled */
put_bits(&c->pb, 1, 0);
......@@ -282,8 +749,8 @@ static void put_frame_header(DCAContext *c)
/* Audio sync word insertion flag: after each sub-frame */
put_bits(&c->pb, 1, 0);
/* Low frequency effects flag: not present or interpolation factor=64 */
put_bits(&c->pb, 2, c->lfe_state);
/* Low frequency effects flag: not present or 64x subsampling */
put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);
/* Predictor history switch flag: on */
put_bits(&c->pb, 1, 1);
......@@ -321,82 +788,68 @@ static void put_primary_audio_header(DCAContext *c)
put_bits(&c->pb, 4, SUBFRAMES - 1);
/* Number of primary audio channels */
put_bits(&c->pb, 3, c->prim_channels - 1);
put_bits(&c->pb, 3, c->fullband_channels - 1);
/* Subband activity count */
for (ch = 0; ch < c->prim_channels; ch++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
/* High frequency VQ start subband */
for (ch = 0; ch < c->prim_channels; ch++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
/* Joint intensity coding index: 0, 0 */
for (ch = 0; ch < c->prim_channels; ch++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 3, 0);
/* Transient mode codebook: A4, A4 (arbitrary) */
for (ch = 0; ch < c->prim_channels; ch++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 2, 0);
/* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
for (ch = 0; ch < c->prim_channels; ch++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Bit allocation quantizer select: linear 5-bit */
for (ch = 0; ch < c->prim_channels; ch++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Quantization index codebook select: dummy data
to avoid transmission of scale factor adjustment */
for (i = 1; i < 11; i++)
for (ch = 0; ch < c->prim_channels; ch++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, bitlen[i], thr[i]);
/* Scale factor adjustment index: not transmitted */
/* Audio header CRC check word: not transmitted */
}
/**
* 8-23 bits quantization
* @param sample
* @param bits
*/
static inline uint32_t quantize(int32_t sample, int bits)
{
av_assert0(sample < 1 << (bits - 1));
av_assert0(sample >= -(1 << (bits - 1)));
return sample & ((1 << bits) - 1);
}
static inline int find_scale_factor7(int64_t max_value, int bits)
static void put_subframe_samples(DCAContext *c, int ss, int band, int ch)
{
int i = 0, j = 128, q;
max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
while (i < j) {
q = (i + j) >> 1;
if (max_value < scale_factor_quant7[q])
j = q;
else
i = q + 1;
if (c->abits[band][ch] <= 7) {
int sum, i, j;
for (i = 0; i < 8; i += 4) {
sum = 0;
for (j = 3; j >= 0; j--) {
sum *= quant_levels[c->abits[band][ch]];
sum += c->quantized[ss * 8 + i + j][band][ch];
sum += (quant_levels[c->abits[band][ch]] - 1) / 2;
}
put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);
}
} else {
int i;
for (i = 0; i < 8; i++) {
int bits = bit_consumption[c->abits[band][ch]] / 16;
int32_t mask = (1 << bits) - 1;
put_bits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch] & mask);
}
}
av_assert1(i < 128);
return i;
}
static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
int scale_factor)
{
sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
put_bits(&c->pb, bits, quantize((int) sample, bits));
}
static void put_subframe(DCAContext *c,
int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
int subframe)
static void put_subframe(DCAContext *c, int subframe)
{
int i, sub, ss, ch, max_value;
int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
int i, band, ss, ch;
/* Subsubframes count */
put_bits(&c->pb, 2, SUBSUBFRAMES -1);
......@@ -405,44 +858,27 @@ static void put_subframe(DCAContext *c,
put_bits(&c->pb, 3, 0);
/* Prediction mode: no ADPCM, in each channel and subband */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
put_bits(&c->pb, 1, 0);
/* Prediction VQ addres: not transmitted */
/* Bit allocation index */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 5, QUANTIZER_BITS+3);
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
put_bits(&c->pb, 5, c->abits[band][ch]);
if (SUBSUBFRAMES > 1) {
/* Transition mode: none for each channel and subband */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
put_bits(&c->pb, 1, 0); /* codebook A4 */
}
/* Determine scale_factor */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++) {
max_value = 0;
for (i = 0; i < 8 * SUBSUBFRAMES; i++)
max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
}
if (c->lfe_channel) {
max_value = 0;
for (i = 0; i < 4 * SUBSUBFRAMES; i++)
max_value = FFMAX(max_value, FFABS(lfe_data[i]));
c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
}
/* Scale factors: the same for each channel and subband,
encoded according to Table D.1.2 */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
/* Scale factors */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
put_bits(&c->pb, 7, c->scale_factor[band][ch]);
/* Joint subband scale factor codebook select: not transmitted */
/* Scale factors for joint subband coding: not transmitted */
......@@ -451,152 +887,83 @@ static void put_subframe(DCAContext *c,
/* Stde information CRC check word: not transmitted */
/* VQ encoded high frequency subbands: not transmitted */
/* LFE data */
/* LFE data: 8 samples and scalefactor */
if (c->lfe_channel) {
for (i = 0; i < 4 * SUBSUBFRAMES; i++)
put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
for (i = 0; i < DCA_LFE_SAMPLES; i++)
put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
put_bits(&c->pb, 8, c->lfe_scale_factor);
}
/* Audio data (subsubframes) */
for (ss = 0; ss < SUBSUBFRAMES ; ss++)
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
for (i = 0; i < 8; i++)
put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
put_subframe_samples(c, ss, band, ch);
/* DSYNC */
put_bits(&c->pb, 16, 0xffff);
}
static void put_frame(DCAContext *c,
int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
uint8_t *frame)
{
int i;
init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
put_primary_audio_header(c);
for (i = 0; i < SUBFRAMES; i++)
put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
flush_put_bits(&c->pb);
c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
put_frame_header(c);
flush_put_bits(&c->pb);
}
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int i, k, channel;
DCAContext *c = avctx->priv_data;
const int16_t *samples;
int ret, real_channel = 0;
const int32_t *samples;
int ret, i;
if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)) < 0)
if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size )) < 0)
return ret;
samples = (const int16_t *)frame->data[0];
for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
for (channel = 0; channel < c->prim_channels + 1; channel++) {
real_channel = c->channel_order_tab[channel];
if (real_channel >= 0) {
/* Get 32 PCM samples */
for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
}
/* Put subband samples into the proper place */
qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
}
}
}
samples = (const int32_t *)frame->data[0];
if (c->lfe_channel) {
for (i = 0; i < PCM_SAMPLES / 2; i++) {
for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
c->lfe_data[i] = lfe_downsample(c, c->pcm);
}
}
subband_transform(c, samples);
if (c->lfe_channel)
lfe_downsample(c, samples);
put_frame(c, c->subband, avpkt->data);
calc_masking(c, samples);
find_peaks(c);
assign_bits(c);
calc_scales(c);
quantize_all(c);
shift_history(c, samples);
avpkt->size = c->frame_size;
*got_packet_ptr = 1;
return 0;
}
static int encode_init(AVCodecContext *avctx)
{
DCAContext *c = avctx->priv_data;
int i;
uint64_t layout = avctx->channel_layout;
c->prim_channels = avctx->channels;
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
if (!layout) {
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
"encoder will guess the layout, but it "
"might be incorrect.\n");
layout = av_get_default_channel_layout(avctx->channels);
}
switch (layout) {
case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break;
case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
default:
av_log(avctx, AV_LOG_ERROR,
"Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
return AVERROR_PATCHWELCOME;
}
if (c->lfe_channel) {
init_lfe_fir();
c->prim_channels--;
c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
c->lfe_state = LFE_PRESENT;
c->lfe_offset = dca_lfe_index[c->a_mode];
} else {
c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
c->lfe_state = LFE_MISSING;
}
for (i = 0; i < 16; i++) {
if (avpriv_dca_sample_rates[i] && (avpriv_dca_sample_rates[i] == avctx->sample_rate))
break;
}
if (i == 16) {
av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
for (i = 0; i < 16; i++)
av_log(avctx, AV_LOG_ERROR, "%d, ", avpriv_dca_sample_rates[i]);
av_log(avctx, AV_LOG_ERROR, "supported.\n");
return -1;
}
c->sample_rate_code = i;
init_put_bits(&c->pb, avpkt->data, avpkt->size);
put_frame_header(c);
put_primary_audio_header(c);
for (i = 0; i < SUBFRAMES; i++)
put_subframe(c, i);
avctx->frame_size = 32 * PCM_SAMPLES;
flush_put_bits(&c->pb);
if (!cos_table[127])
qmf_init();
avpkt->pts = frame->pts;
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
avpkt->size = c->frame_size + 1;
*got_packet_ptr = 1;
return 0;
}
static const AVCodecDefault defaults[] = {
{ "b", "1411200" },
{ NULL },
};
AVCodec ff_dca_encoder = {
.name = "dca",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = encode_init,
.encode2 = encode_frame,
.capabilities = CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.name = "dca",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = encode_init,
.encode2 = encode_frame,
.capabilities = CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.supported_samplerates = sample_rates,
.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_2_2,
AV_CH_LAYOUT_5POINT0,
AV_CH_LAYOUT_5POINT1,
0 },
.defaults = defaults,
};
/*
* DCA encoder tables
* Copyright (C) 2008 Alexander E. Patrakov
* Copyright (C) 2008-2012 Alexander E. Patrakov
*
* This file is part of FFmpeg.
*
......@@ -24,523 +24,90 @@
#include <stdint.h>
/* This is a scaled version of the response of the reference decoder to
this vector of subband samples: ( 1.0 0.0 0.0 ... 0.0 )
*/
typedef struct {
int32_t m;
int32_t e;
} softfloat;
static const int32_t UnQMF[512] = {
7,
4,
-961,
-2844,
-8024,
-18978,
-32081,
-15635,
-16582,
-18359,
-17180,
-14868,
-11664,
-8051,
-4477,
-1327,
-1670,
-6019,
-11590,
-18030,
-24762,
-30965,
-35947,
-36145,
-37223,
-86311,
-57024,
-27215,
-11274,
-4684,
42,
108,
188,
250,
-1007,
-596,
-2289,
-12218,
-27191,
-124367,
-184256,
-250538,
-323499,
-397784,
-468855,
-532072,
-583000,
-618041,
-777916,
-783868,
-765968,
-724740,
-662468,
-583058,
-490548,
-401623,
-296090,
-73154,
-36711,
-7766,
-2363,
-4905,
2388,
2681,
5651,
4086,
71110,
139742,
188067,
151237,
101355,
309917,
343690,
358839,
357555,
334606,
289625,
224152,
142063,
48725,
74996,
238425,
411666,
584160,
744276,
880730,
983272,
1041933,
1054396,
789531,
851022,
864032,
675431,
418134,
35762,
66911,
103502,
136403,
-55147,
-245269,
-499595,
-808470,
-1136858,
-2010912,
-2581654,
-3151901,
-3696328,
-4196599,
-4633761,
-4993229,
-5262495,
-5436311,
-477650,
-901314,
-1308090,
-1677468,
-1985525,
-2212848,
-2341196,
-2373915,
-2269552,
-2620489,
-2173858,
-1629954,
-946595,
-193499,
1119459,
1138657,
1335311,
1126544,
2765033,
3139603,
3414913,
3599213,
3676363,
3448981,
3328726,
3111551,
2810887,
2428657,
1973684,
1457278,
893848,
300995,
-292521,
-867621,
-1404936,
-1871278,
-2229831,
-2440932,
-2462684,
-2255006,
-1768898,
-1079574,
82115,
1660302,
3660715,
6123610,
8329598,
11888744,
15722147,
19737089,
25647773,
31039399,
36868007,
43124253,
49737161,
56495958,
63668945,
71039511,
78540240,
86089058,
93600041,
100981151,
108136061,
114970055,
121718321,
127566038,
132774642,
137247294,
140894737,
143635018,
145395599,
146114032,
145742999,
144211606,
141594341,
137808404,
132914122,
126912246,
120243281,
112155281,
103338368,
93904953,
83439152,
72921548,
62192990,
51434918,
40894003,
30786726,
21384955,
12939112,
5718193,
-5790,
-3959261,
-5870978,
-5475538,
-2517061,
3247310,
12042937,
24076729,
39531397,
58562863,
81297002,
107826748,
138209187,
172464115,
210569037,
252468018,
298045453,
347168648,
399634888,
455137189,
513586535,
574537650,
637645129,
702597163,
768856566,
836022040,
903618096,
971159680,
1038137214,
1103987353,
1168195035,
1230223053,
1289539180,
1345620373,
1397957958,
1446063657,
1489474689,
1527740502,
1560502307,
1587383079,
1608071145,
1622301248,
1629859340,
1630584888,
1624373875,
1611178348,
1591018893,
1563948667,
1530105004,
1489673227,
1442904075,
1390107674,
1331590427,
1267779478,
1199115126,
1126053392,
1049146257,
968928307,
885965976,
800851610,
714186243,
626590147,
538672486,
451042824,
364299927,
279026812,
195785029,
115109565,
37503924,
-36564551,
-106668063,
-172421668,
-233487283,
-289575706,
-340448569,
-385919511,
-425854915,
-460174578,
-488840702,
-511893328,
-529405118,
-541489888,
-548312207,
-550036471,
-547005316,
-539436808,
-527630488,
-512084785,
-492941605,
-470665204,
-445668379,
-418328829,
-389072810,
-358293846,
-326396227,
-293769619,
-260792276,
-227825056,
-195208961,
-163262121,
-132280748,
-102533727,
-74230062,
-47600637,
-22817785,
-25786,
20662895,
39167253,
55438413,
69453741,
81242430,
90795329,
98213465,
103540643,
106917392,
108861938,
108539682,
106780704,
103722568,
99043289,
93608686,
87266209,
80212203,
72590022,
64603428,
56362402,
48032218,
39749162,
31638971,
23814664,
16376190,
9409836,
2988017,
-2822356,
-7976595,
-12454837,
-16241147,
-19331944,
-21735011,
-23468284,
-24559822,
-25042936,
-25035583,
-24429587,
-23346408,
-21860411,
-20015718,
-17025330,
-14968728,
-12487138,
-9656319,
-7846681,
-5197816,
-2621904,
-144953,
2144746,
3990570,
5845884,
7454650,
8820394,
9929891,
10784445,
11390921,
11762056,
11916017,
12261189,
12117604,
11815303,
11374622,
10815301,
10157241,
9418799,
8629399,
7780776,
7303680,
6353499,
5392738,
4457895,
3543062,
1305978,
1402521,
1084092,
965652,
-151008,
-666667,
-1032157,
-1231475,
-1319043,
-1006023,
-915720,
-773426,
-612377,
-445864,
-291068,
-161337,
-66484,
-11725,
133453,
388184,
615856,
804033,
942377,
1022911,
1041247,
995854,
891376,
572246,
457992,
316365,
172738,
43037,
-117662,
-98542,
-70279,
-41458,
-535790,
-959038,
-1364456,
-1502265,
-1568530,
-2378681,
-2701111,
-2976407,
-3182552,
-3314415,
-3366600,
-3337701,
-3232252,
-3054999,
1984841,
1925903,
1817377,
1669153,
1490069,
1292040,
1086223,
890983,
699163,
201358,
266971,
296990,
198419,
91119,
4737,
5936,
2553,
2060,
-3828,
-1664,
-4917,
-20796,
-36822,
-131247,
-154923,
-162055,
-161354,
-148762,
-125754,
-94473,
-57821,
-19096,
15172,
43004,
65624,
81354,
89325,
89524,
82766,
71075,
55128,
13686,
6921,
1449,
420,
785,
-215,
-179,
-113,
-49,
6002,
16007,
42978,
100662,
171472,
83975,
93702,
108813,
111893,
110272,
103914,
93973,
81606,
68041,
-54058,
-60695,
-65277,
-67224,
-66213,
-62082,
-55574,
-42988,
-35272,
-63735,
-33501,
-12671,
-4038,
-1232,
5,
7
static const int sample_rates[] = {
8000, 16000, 32000, 11025, 22050, 44100, 12000, 24000, 48000, 0,
};
static const uint8_t bitstream_sfreq[] = { 1, 2, 3, 6, 7, 8, 11, 12, 13 };
/* Auditory filter center frequencies and bandwidths, in Hz.
* The last two are made up, because there is no scientific data.
*/
static uint16_t fc[] = {
50, 150, 250, 350, 450, 570, 700, 840, 1000, 1170, 1370, 1600, 1850, 2150,
2500, 2900, 3400, 4000, 4800, 5800, 7000, 8500, 10500, 13500, 17000
};
static uint16_t erb[] = {
80, 100, 100, 100, 110, 120, 140, 150, 160, 190, 210, 240, 280,
320, 380, 450, 550, 700, 900, 1100, 1300, 1800, 2500, 3500, 4500
};
static const softfloat stepsize_inv[27] = {
{0, 0}, {1342177360, 21}, {2147483647, 21}, {1342177360, 20},
{1819901661, 20}, {2147483647, 20}, {1278263843, 19}, {1579032492, 19},
{1412817763, 18}, {1220162327, 17}, {1118482133, 16}, {1917391412, 16},
{1766017772, 15}, {1525212826, 14}, {1290553940, 13}, {2097179000, 13},
{1677683200, 12}, {1497972244, 11}, {1310893147, 10}, {1165354136, 9},
{1748031204, 9}, {1542092044, 8}, {1636178017, 7}, {1636178017, 6},
{1636178017, 5}, {1636178017, 4}, {1636178017, 3},
};
static const softfloat scalefactor_inv[128] = {
{2147483647, 1}, {2147483647, 1}, {2147483647, 2}, {2147483647, 2},
{2147483647, 2}, {2147483647, 2}, {1431655765, 2}, {1431655765, 2},
{1431655765, 2}, {2147483647, 3}, {2147483647, 3}, {1717986918, 3},
{1431655765, 3}, {1227133513, 3}, {1227133513, 3}, {2147483647, 4},
{1717986918, 4}, {1561806289, 4}, {1431655765, 4}, {1227133513, 4},
{2147483647, 5}, {1908874353, 5}, {1717986918, 5}, {1493901668, 5},
{1321528398, 5}, {1145324612, 5}, {2021161080, 6}, {1808407282, 6},
{1561806289, 6}, {1374389534, 6}, {1227133513, 6}, {2147483647, 7},
{1908874353, 7}, {1676084798, 7}, {1477838209, 7}, {1296593900, 7},
{1145324612, 7}, {2021161080, 8}, {1773405851, 8}, {1561806289, 8},
{1374389534, 8}, {1216273924, 8}, {2139127680, 9}, {1882725390, 9},
{1660893697, 9}, {1462116526, 9}, {1287484341, 9}, {1135859119, 9},
{1999112050, 10}, {1762037865, 10}, {1552982525, 10}, {1367551775, 10},
{1205604855, 10}, {2124660150, 11}, {1871509153, 11}, {1648443220, 11},
{1452459217, 11}, {1279990253, 11}, {1127704233, 11}, {1987368509, 12},
{1750814693, 12}, {1542632939, 12}, {1359099663, 12}, {1197398995, 12},
{2109880792, 13}, {1858853132, 13}, {1638006149, 13}, {1443165385, 13},
{1271479187, 13}, {1120235993, 13}, {1973767086, 14}, {1739045674, 14},
{1532153461, 14}, {1349922194, 14}, {1189384493, 14}, {2095804865, 15},
{1846464029, 15}, {1626872524, 15}, {1433347133, 15}, {1262853884, 15},
{1112619678, 15}, {1960569045, 16}, {1727349015, 16}, {1521881227, 16},
{1340842289, 16}, {1181357555, 16}, {2081669156, 17}, {1834047752, 17},
{1615889229, 17}, {1423675973, 17}, {1254322457, 17}, {1105123583, 17},
{1947330755, 18}, {1715693602, 18}, {1511607799, 18}, {1331801790, 18},
{1173384427, 18}, {2067616532, 19}, {1821667648, 19}, {1604980024, 19},
{1414066955, 19}, {1245861410, 19}, {1097665748, 19}, {1934193616, 20},
{1704119624, 20}, {1501412075, 20}, {1322817107, 20}, {1165466323, 20},
{2053666205, 21}, {1809379407, 21}, {1594151671, 21}, {1404526328, 21},
{1237455941, 21}, {1090259329, 21}, {1921143210, 22}, {1692621231, 22},
{1491281857, 22}, {1313892269, 22}, {1157603482, 22}, {2039810470, 23},
{1797172644, 23}, {1583396912, 23}, {1395050052, 23}, {1229107276, 23},
{1082903494, 23}, {1082903494, 23}, {1082903494, 23}, {1082903494, 23},
};
/* manually derived from
* Table B.5: Selection of quantization levels and codebooks
* FIXME: will become invalid when Huffman codes are introduced.
*/
static const int bit_consumption[27] = {
-8, 28, 40, 48, 52, 60, 68, 76, 80, 96,
112, 128, 144, 160, 176, 192, 208, 224, 240, 256,
272, 288, 304, 320, 336, 352, 368,
};
/* Table B.5: Selection of quantization levels and codebooks */
static const int quant_levels[27] = {
1, 3, 5, 7, 9, 13, 17, 25, 32, 64,
128, 256, 512, 1024, 2048, 4096, 8192, 16384, 32768, 65536,
131072, 262144, 524288, 1048576, 2097152, 4194304, 8388608,
};
#endif /* AVCODEC_DCAENC_H */
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