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ffmpeg/libavcodec/dcaenc.c
Ganesh Ajjanagadde db1a642cd2 all: move ff_exp10, ff_exp10f, ff_fast_powf to lavu/ffmath.h 
The idea is to use ffmath.h for internal implementations of math functions.
Currently, it is used for variants of libm functions, but is by no means
limited to such things.

Note that this is not exported; use lavu/mathematics for such purposes.

Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanag@gmail.com>
2016-03-22 10:15:31 -07:00

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/*
* DCA encoder
* Copyright (C) 2008-2012 Alexander E. Patrakov
* 2010 Benjamin Larsson
* 2011 Xiang Wang
*
* 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 "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/ffmath.h"
#include "avcodec.h"
#include "dca.h"
#include "dcadata.h"
#include "dcaenc.h"
#include "internal.h"
#include "mathops.h"
#include "put_bits.h"
#define MAX_CHANNELS 6
#define DCA_MAX_FRAME_SIZE 16384
#define DCA_HEADER_SIZE 13
#define DCA_LFE_SAMPLES 8
#define DCAENC_SUBBANDS 32
#define SUBFRAMES 1
#define SUBSUBFRAMES 2
#define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
#define AUBANDS 25
typedef struct DCAEncContext {
PutBitContext pb;
int frame_size;
int frame_bits;
int fullband_channels;
int channels;
int lfe_channel;
int samplerate_index;
int bitrate_index;
int channel_config;
const int32_t *band_interpolation;
const int32_t *band_spectrum;
int lfe_scale_factor;
softfloat lfe_quant;
int32_t lfe_peak_cb;
const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */
int32_t subband[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
int32_t quantized[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
int32_t peak_cb[DCAENC_SUBBANDS][MAX_CHANNELS];
int32_t downsampled_lfe[DCA_LFE_SAMPLES];
int32_t masking_curve_cb[SUBSUBFRAMES][256];
int abits[DCAENC_SUBBANDS][MAX_CHANNELS];
int scale_factor[DCAENC_SUBBANDS][MAX_CHANNELS];
softfloat quant[DCAENC_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;
} DCAEncContext;
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];
/* 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)
{
DCAEncContext *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--;
c->channel_order_tab = ff_dca_channel_reorder_lfe[c->channel_config];
} else {
c->channel_order_tab = ff_dca_channel_reorder_nolfe[c->channel_config];
}
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 %"PRId64" not supported.", (int64_t)avctx->bit_rate);
return AVERROR(EINVAL);
}
for (i = 0; ff_dca_bit_rates[i] < avctx->bit_rate; i++)
;
c->bitrate_index = 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;
cos_table[0] = 0x7fffffff;
cos_table[512] = 0;
cos_table[1024] = -cos_table[0];
for (i = 1; i < 512; i++) {
cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
cos_table[1024-i] = -cos_table[i];
cos_table[1024+i] = -cos_table[i];
cos_table[2048-i] = cos_table[i];
}
for (i = 0; i < 2048; i++) {
cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i));
}
for (k = 0; k < 32; k++) {
for (j = 0; j < 8; j++) {
lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
lfe_fir_64i[64 * (7-j) + (63 - k)] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
}
}
for (i = 0; i < 512; i++) {
band_interpolation[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
band_interpolation[1][i] = (int32_t)(0x1000000000ULL * ff_dca_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 + ff_exp10(-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 = ff_dca_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 = ff_dca_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 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 + 0x80000000ULL;
return r >> 32;
}
static void subband_transform(DCAEncContext *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;
const int chi = c->channel_order_tab[ch];
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 + chi];
hist_start = (hist_start + 32) & 511;
}
}
}
static void lfe_downsample(DCAEncContext *c, const int32_t *input)
{
/* FIXME: make 128x LFE downsampling possible */
const int lfech = ff_dca_lfe_index[c->channel_config];
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 + lfech];
hist_start = (hist_start + 64) & 511;
}
}
typedef struct {
int32_t re;
int32_t im;
} cplx32;
static void fft(const int32_t in[2 * 256], cplx32 out[256])
{
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));
}
}
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 add_cb(int32_t a, int32_t b)
{
if (a < b)
FFSWAP(int32_t, a, b);
if (a - b >= 256)
return a;
return a + cb_to_add[a - b];
}
static void adjust_jnd(int samplerate_index,
const int32_t in[512], int32_t out_cb[256])
{
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);
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 */
}
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);
}
typedef void (*walk_band_t)(DCAEncContext *c, int band1, int band2, int f,
int32_t spectrum1, int32_t spectrum2, int channel,
int32_t * arg);
static void walk_band_low(DCAEncContext *c, int band, int channel,
walk_band_t walk, int32_t *arg)
{
int f;
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);
}
}
static void walk_band_high(DCAEncContext *c, int band, int channel,
walk_band_t walk, int32_t *arg)
{
int f;
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);
}
}
static void update_band_masking(DCAEncContext *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;
if (value < c->band_masking_cb[band1])
c->band_masking_cb[band1] = value;
}
static void calc_masking(DCAEncContext *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++) {
const int chi = c->channel_order_tab[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 + chi];
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 void find_peaks(DCAEncContext *c)
{
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 const int snr_fudge = 128;
#define USED_1ABITS 1
#define USED_NABITS 2
#define USED_26ABITS 4
static int init_quantization_noise(DCAEncContext *c, int noise)
{
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 (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(DCAEncContext *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(DCAEncContext *c, const int32_t *input)
{
int k, ch;
for (k = 0; k < 512; k++)
for (ch = 0; ch < c->channels; ch++) {
const int chi = c->channel_order_tab[ch];
c->history[k][ch] = input[k * c->channels + chi];
}
}
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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *c)
{
/* SYNC */
put_bits(&c->pb, 16, 0x7ffe);
put_bits(&c->pb, 16, 0x8001);
/* Frame type: normal */
put_bits(&c->pb, 1, 1);
/* Deficit sample count: none */
put_bits(&c->pb, 5, 31);
/* CRC is not present */
put_bits(&c->pb, 1, 0);
/* Number of PCM sample blocks */
put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
/* Primary frame byte size */
put_bits(&c->pb, 14, c->frame_size - 1);
/* Audio channel arrangement */
put_bits(&c->pb, 6, c->channel_config);
/* Core audio sampling frequency */
put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);
/* Transmission bit rate */
put_bits(&c->pb, 5, c->bitrate_index);
/* Embedded down mix: disabled */
put_bits(&c->pb, 1, 0);
/* Embedded dynamic range flag: not present */
put_bits(&c->pb, 1, 0);
/* Embedded time stamp flag: not present */
put_bits(&c->pb, 1, 0);
/* Auxiliary data flag: not present */
put_bits(&c->pb, 1, 0);
/* HDCD source: no */
put_bits(&c->pb, 1, 0);
/* Extension audio ID: N/A */
put_bits(&c->pb, 3, 0);
/* Extended audio data: not present */
put_bits(&c->pb, 1, 0);
/* Audio sync word insertion flag: after each sub-frame */
put_bits(&c->pb, 1, 0);
/* 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);
/* No CRC */
/* Multirate interpolator switch: non-perfect reconstruction */
put_bits(&c->pb, 1, 0);
/* Encoder software revision: 7 */
put_bits(&c->pb, 4, 7);
/* Copy history: 0 */
put_bits(&c->pb, 2, 0);
/* Source PCM resolution: 16 bits, not DTS ES */
put_bits(&c->pb, 3, 0);
/* Front sum/difference coding: no */
put_bits(&c->pb, 1, 0);
/* Surrounds sum/difference coding: no */
put_bits(&c->pb, 1, 0);
/* Dialog normalization: 0 dB */
put_bits(&c->pb, 4, 0);
}
static void put_primary_audio_header(DCAEncContext *c)
{
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
int ch, i;
/* Number of subframes */
put_bits(&c->pb, 4, SUBFRAMES - 1);
/* Number of primary audio channels */
put_bits(&c->pb, 3, c->fullband_channels - 1);
/* Subband activity count */
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 5, DCAENC_SUBBANDS - 2);
/* High frequency VQ start subband */
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, 5, DCAENC_SUBBANDS - 1);
/* Joint intensity coding index: 0, 0 */
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->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->fullband_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Bit allocation quantizer select: linear 5-bit */
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->fullband_channels; ch++)
put_bits(&c->pb, bitlen[i], thr[i]);
/* Scale factor adjustment index: not transmitted */
/* Audio header CRC check word: not transmitted */
}
static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
{
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;
put_sbits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch]);
}
}
}
static void put_subframe(DCAEncContext *c, int subframe)
{
int i, band, ss, ch;
/* Subsubframes count */
put_bits(&c->pb, 2, SUBSUBFRAMES -1);
/* Partial subsubframe sample count: dummy */
put_bits(&c->pb, 3, 0);
/* Prediction mode: no ADPCM, in each channel and subband */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCAENC_SUBBANDS; band++)
put_bits(&c->pb, 1, 0);
/* Prediction VQ address: not transmitted */
/* Bit allocation index */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCAENC_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->fullband_channels; ch++)
for (band = 0; band < DCAENC_SUBBANDS; band++)
put_bits(&c->pb, 1, 0); /* codebook A4 */
}
/* Scale factors */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCAENC_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 */
/* Stereo down-mix coefficients: not transmitted */
/* Dynamic range coefficient: not transmitted */
/* Stde information CRC check word: not transmitted */
/* VQ encoded high frequency subbands: not transmitted */
/* LFE data: 8 samples and scalefactor */
if (c->lfe_channel) {
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->fullband_channels; ch++)
for (band = 0; band < DCAENC_SUBBANDS; band++)
put_subframe_samples(c, ss, band, ch);
/* DSYNC */
put_bits(&c->pb, 16, 0xffff);
}
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
DCAEncContext *c = avctx->priv_data;
const int32_t *samples;
int ret, i;
if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size, 0)) < 0)
return ret;
samples = (const int32_t *)frame->data[0];
subband_transform(c, samples);
if (c->lfe_channel)
lfe_downsample(c, samples);
calc_masking(c, samples);
find_peaks(c);
assign_bits(c);
calc_scales(c);
quantize_all(c);
shift_history(c, samples);
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);
for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++)
put_bits(&c->pb, 1, 0);
flush_put_bits(&c->pb);
avpkt->pts = frame->pts;
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
avpkt->size = put_bits_count(&c->pb) >> 3;
*got_packet_ptr = 1;
return 0;
}
static const AVCodecDefault defaults[] = {
{ "b", "1411200" },
{ NULL },
};
AVCodec ff_dca_encoder = {
.name = "dca",
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DTS,
.priv_data_size = sizeof(DCAEncContext),
.init = encode_init,
.encode2 = encode_frame,
.capabilities = AV_CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE },
.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,
};
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