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ffmpeg/libavcodec/dcaenc.c

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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 "avcodec.h"
#include "dca.h"
#include "dcadata.h"
#include "dcaenc.h"
#include "internal.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 DCA_SUBBANDS 32
#define SUBFRAMES 1
#define SUBSUBFRAMES 2
#define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
#define AUBANDS 25
typedef struct DCAContext {
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;
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;
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)
{
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));
}
/* FIXME: probably incorrect */
for (i = 0; i < 256; i++) {
lfe_fir_64i[i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);
lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);
}
for (i = 0; i < 512; 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 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(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;
}
}
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)(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 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(DCAContext *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(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;
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 void find_peaks(DCAContext *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(DCAContext *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(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)
{
/* 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(DCAContext *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, DCA_SUBBANDS - 2);
/* High frequency VQ start subband */
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->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(DCAContext *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;
int32_t mask = (1 << bits) - 1;
put_bits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch] & mask);
}
}
}
static void put_subframe(DCAContext *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 < DCA_SUBBANDS; band++)
put_bits(&c->pb, 1, 0);
/* Prediction VQ addres: not transmitted */
/* Bit allocation index */
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->fullband_channels; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
put_bits(&c->pb, 1, 0); /* codebook A4 */
}
/* 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 */
/* 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 < DCA_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)
{
DCAContext *c = avctx->priv_data;
const int32_t *samples;
int ret, i;
if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size )) < 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);
flush_put_bits(&c->pb);
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_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,
};
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