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ffmpeg/libavcodec/dcaenc.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs 
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:15 -03:00

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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
*/
#define FFT_FLOAT 0
#define FFT_FIXED_32 1
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/ffmath.h"
#include "libavutil/mem_internal.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "dca.h"
#include "dcaadpcm.h"
#include "dcamath.h"
#include "dca_core.h"
#include "dcadata.h"
#include "dcaenc.h"
#include "fft.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
#define COS_T(x) (c->cos_table[(x) & 2047])
typedef struct CompressionOptions {
int adpcm_mode;
} CompressionOptions;
typedef struct DCAEncContext {
AVClass *class;
PutBitContext pb;
DCAADPCMEncContext adpcm_ctx;
FFTContext mdct;
CompressionOptions options;
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 prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS * 2];
int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
int32_t *subband[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES];
int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; ///< expected peak of residual signal
int32_t downsampled_lfe[DCA_LFE_SAMPLES];
int32_t masking_curve_cb[SUBSUBFRAMES][256];
int32_t bit_allocation_sel[MAX_CHANNELS];
int abits[MAX_CHANNELS][DCAENC_SUBBANDS];
int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS];
softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS];
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;
int consumed_adpcm_bits; ///< Number of bits to transmit ADPCM related info
int32_t cos_table[2048];
int32_t band_interpolation_tab[2][512];
int32_t band_spectrum_tab[2][8];
int32_t auf[9][AUBANDS][256];
int32_t cb_to_add[256];
int32_t cb_to_level[2048];
int32_t lfe_fir_64i[512];
} DCAEncContext;
/* 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 subband_bufer_alloc(DCAEncContext *c)
{
int ch, band;
int32_t *bufer = av_calloc(MAX_CHANNELS * DCAENC_SUBBANDS *
(SUBBAND_SAMPLES + DCA_ADPCM_COEFFS),
sizeof(int32_t));
if (!bufer)
return AVERROR(ENOMEM);
/* we need a place for DCA_ADPCM_COEFF samples from previous frame
* to calc prediction coefficients for each subband */
for (ch = 0; ch < MAX_CHANNELS; ch++) {
for (band = 0; band < DCAENC_SUBBANDS; band++) {
c->subband[ch][band] = bufer +
ch * DCAENC_SUBBANDS * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) +
band * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + DCA_ADPCM_COEFFS;
}
}
return 0;
}
static void subband_bufer_free(DCAEncContext *c)
{
if (c->subband[0][0]) {
int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS;
av_free(bufer);
c->subband[0][0] = NULL;
}
}
static int encode_init(AVCodecContext *avctx)
{
DCAEncContext *c = avctx->priv_data;
uint64_t layout = avctx->channel_layout;
int i, j, k, min_frame_bits;
int ret;
if ((ret = subband_bufer_alloc(c)) < 0)
return ret;
c->fullband_channels = c->channels = avctx->channels;
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
c->band_interpolation = c->band_interpolation_tab[1];
c->band_spectrum = c->band_spectrum_tab[1];
c->worst_quantization_noise = -2047;
c->worst_noise_ever = -2047;
c->consumed_adpcm_bits = 0;
if (ff_dcaadpcm_init(&c->adpcm_ctx))
return AVERROR(ENOMEM);
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 = channel_reorder_lfe[c->channel_config];
} else {
c->channel_order_tab = channel_reorder_nolfe[c->channel_config];
}
for (i = 0; i < MAX_CHANNELS; i++) {
for (j = 0; j < DCA_CODE_BOOKS; j++) {
c->quant_index_sel[i][j] = ff_dca_quant_index_group_size[j];
}
/* 6 - no Huffman */
c->bit_allocation_sel[i] = 6;
for (j = 0; j < DCAENC_SUBBANDS; j++) {
/* -1 - no ADPCM */
c->prediction_mode[i][j] = -1;
memset(c->adpcm_history[i][j], 0, sizeof(int32_t)*DCA_ADPCM_COEFFS);
}
}
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.", 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 ((ret = ff_mdct_init(&c->mdct, 9, 0, 1.0)) < 0)
return ret;
/* Init all tables */
c->cos_table[0] = 0x7fffffff;
c->cos_table[512] = 0;
c->cos_table[1024] = -c->cos_table[0];
for (i = 1; i < 512; i++) {
c->cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
c->cos_table[1024-i] = -c->cos_table[i];
c->cos_table[1024+i] = -c->cos_table[i];
c->cos_table[2048-i] = +c->cos_table[i];
}
for (i = 0; i < 2048; i++)
c->cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i));
for (k = 0; k < 32; k++) {
for (j = 0; j < 8; j++) {
c->lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
c->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++) {
c->band_interpolation_tab[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
c->band_interpolation_tab[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;
c->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);
c->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);
}
c->band_spectrum_tab[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);
}
c->band_spectrum_tab[1][j] = (int32_t)(200 * log10(accum));
}
return 0;
}
static av_cold int encode_close(AVCodecContext *avctx)
{
DCAEncContext *c = avctx->priv_data;
ff_mdct_end(&c->mdct);
subband_bufer_free(c);
ff_dcaadpcm_free(&c->adpcm_ctx);
return 0;
}
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];
memcpy(hist, &c->history[ch][0], 512 * sizeof(int32_t));
for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
int32_t accum[64];
int32_t resp;
int band;
/* Calculate the convolutions at once */
memset(accum, 0, 64 * sizeof(int32_t));
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[ch][band][subs] = ((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 = lfe_index[c->channel_config];
int i, j, lfes;
int32_t hist[512];
int32_t accum;
int hist_start = 0;
memcpy(hist, &c->history[c->channels - 1][0], 512 * sizeof(int32_t));
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], c->lfe_fir_64i[j]);
for (i = 0; i < hist_start; i++, j++)
accum += mul32(hist[i], c->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;
}
}
static int32_t get_cb(DCAEncContext *c, int32_t in)
{
int i, res = 0;
in = FFABS(in);
for (i = 1024; i > 0; i >>= 1) {
if (c->cb_to_level[i + res] >= in)
res += i;
}
return -res;
}
static int32_t add_cb(DCAEncContext *c, int32_t a, int32_t b)
{
if (a < b)
FFSWAP(int32_t, a, b);
if (a - b >= 256)
return a;
return a + c->cb_to_add[a - b];
}
static void calc_power(DCAEncContext *c,
const int32_t in[2 * 256], int32_t power[256])
{
int i;
LOCAL_ALIGNED_32(int32_t, data, [512]);
LOCAL_ALIGNED_32(int32_t, coeff, [256]);
for (i = 0; i < 512; i++)
data[i] = norm__(mul32(in[i], 0x3fffffff - (COS_T(4 * i + 2) >> 1)), 4);
c->mdct.mdct_calc(&c->mdct, coeff, data);
for (i = 0; i < 256; i++) {
const int32_t cb = get_cb(c, coeff[i]);
power[i] = add_cb(c, cb, cb);
}
}
static void adjust_jnd(DCAEncContext *c,
const int32_t in[512], int32_t out_cb[256])
{
int32_t power[256];
int32_t out_cb_unnorm[256];
int32_t denom;
const int32_t ca_cb = -1114;
const int32_t cs_cb = 928;
const int samplerate_index = c->samplerate_index;
int i, j;
calc_power(c, in, power);
for (j = 0; j < 256; j++)
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(c, denom, power[j] + c->auf[samplerate_index][i][j]);
for (j = 0; j < 256; j++)
out_cb_unnorm[j] = add_cb(c, out_cb_unnorm[j],
-denom + c->auf[samplerate_index][i][j]);
}
for (j = 0; j < 256; j++)
out_cb[j] = add_cb(c, 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[ch][k];
for (k -= 512; i < 512; i++, k++)
data[i] = input[k * c->channels + chi];
adjust_jnd(c, 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 inline int32_t find_peak(DCAEncContext *c, const int32_t *in, int len)
{
int sample;
int32_t m = 0;
for (sample = 0; sample < len; sample++) {
int32_t s = abs(in[sample]);
if (m < s)
m = s;
}
return get_cb(c, m);
}
static void find_peaks(DCAEncContext *c)
{
int band, ch;
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++)
c->peak_cb[ch][band] = find_peak(c, c->subband[ch][band],
SUBBAND_SAMPLES);
}
if (c->lfe_channel)
c->lfe_peak_cb = find_peak(c, c->downsampled_lfe, DCA_LFE_SAMPLES);
}
static void adpcm_analysis(DCAEncContext *c)
{
int ch, band;
int pred_vq_id;
int32_t *samples;
int32_t estimated_diff[SUBBAND_SAMPLES];
c->consumed_adpcm_bits = 0;
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
pred_vq_id = ff_dcaadpcm_subband_analysis(&c->adpcm_ctx, samples,
SUBBAND_SAMPLES, estimated_diff);
if (pred_vq_id >= 0) {
c->prediction_mode[ch][band] = pred_vq_id;
c->consumed_adpcm_bits += 12; //12 bits to transmit prediction vq index
c->diff_peak_cb[ch][band] = find_peak(c, estimated_diff, 16);
} else {
c->prediction_mode[ch][band] = -1;
}
}
}
}
static const int snr_fudge = 128;
#define USED_1ABITS 1
#define USED_26ABITS 4
static inline int32_t get_step_size(DCAEncContext *c, int ch, int band)
{
int32_t step_size;
if (c->bitrate_index == 3)
step_size = ff_dca_lossless_quant[c->abits[ch][band]];
else
step_size = ff_dca_lossy_quant[c->abits[ch][band]];
return step_size;
}
static int calc_one_scale(DCAEncContext *c, 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 = c->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 ((ff_dca_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((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
return our_nscale;
}
static inline void quantize_adpcm_subband(DCAEncContext *c, int ch, int band)
{
int32_t step_size;
int32_t diff_peak_cb = c->diff_peak_cb[ch][band];
c->scale_factor[ch][band] = calc_one_scale(c, diff_peak_cb,
c->abits[ch][band],
&c->quant[ch][band]);
step_size = get_step_size(c, ch, band);
ff_dcaadpcm_do_real(c->prediction_mode[ch][band],
c->quant[ch][band],
ff_dca_scale_factor_quant7[c->scale_factor[ch][band]],
step_size, c->adpcm_history[ch][band], c->subband[ch][band],
c->adpcm_history[ch][band] + 4, c->quantized[ch][band],
SUBBAND_SAMPLES, c->cb_to_level[-diff_peak_cb]);
}
static void quantize_adpcm(DCAEncContext *c)
{
int band, ch;
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < 32; band++)
if (c->prediction_mode[ch][band] >= 0)
quantize_adpcm_subband(c, ch, band);
}
static void quantize_pcm(DCAEncContext *c)
{
int sample, band, ch;
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
if (c->prediction_mode[ch][band] == -1) {
for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
int32_t val = quantize_value(c->subband[ch][band][sample],
c->quant[ch][band]);
c->quantized[ch][band][sample] = val;
}
}
}
}
}
static void accumulate_huff_bit_consumption(int abits, int32_t *quantized,
uint32_t *result)
{
uint8_t sel, id = abits - 1;
for (sel = 0; sel < ff_dca_quant_index_group_size[id]; sel++)
result[sel] += ff_dca_vlc_calc_quant_bits(quantized, SUBBAND_SAMPLES,
sel, id);
}
static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7],
uint32_t clc_bits[DCA_CODE_BOOKS],
int32_t res[DCA_CODE_BOOKS])
{
uint8_t i, sel;
uint32_t best_sel_bits[DCA_CODE_BOOKS];
int32_t best_sel_id[DCA_CODE_BOOKS];
uint32_t t, bits = 0;
for (i = 0; i < DCA_CODE_BOOKS; i++) {
av_assert0(!((!!vlc_bits[i][0]) ^ (!!clc_bits[i])));
if (vlc_bits[i][0] == 0) {
/* do not transmit adjustment index for empty codebooks */
res[i] = ff_dca_quant_index_group_size[i];
/* and skip it */
continue;
}
best_sel_bits[i] = vlc_bits[i][0];
best_sel_id[i] = 0;
for (sel = 0; sel < ff_dca_quant_index_group_size[i]; sel++) {
if (best_sel_bits[i] > vlc_bits[i][sel] && vlc_bits[i][sel]) {
best_sel_bits[i] = vlc_bits[i][sel];
best_sel_id[i] = sel;
}
}
/* 2 bits to transmit scale factor adjustment index */
t = best_sel_bits[i] + 2;
if (t < clc_bits[i]) {
res[i] = best_sel_id[i];
bits += t;
} else {
res[i] = ff_dca_quant_index_group_size[i];
bits += clc_bits[i];
}
}
return bits;
}
static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands,
int32_t *res)
{
uint8_t i;
uint32_t t;
int32_t best_sel = 6;
int32_t best_bits = bands * 5;
/* Check do we have subband which cannot be encoded by Huffman tables */
for (i = 0; i < bands; i++) {
if (abits[i] > 12 || abits[i] == 0) {
*res = best_sel;
return best_bits;
}
}
for (i = 0; i < DCA_BITALLOC_12_COUNT; i++) {
t = ff_dca_vlc_calc_alloc_bits(abits, bands, i);
if (t < best_bits) {
best_bits = t;
best_sel = i;
}
}
*res = best_sel;
return best_bits;
}
static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero)
{
int ch, band, ret = USED_26ABITS | USED_1ABITS;
uint32_t huff_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS][7];
uint32_t clc_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS];
uint32_t bits_counter = 0;
c->consumed_bits = 132 + 333 * c->fullband_channels;
c->consumed_bits += c->consumed_adpcm_bits;
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[ch][band] - c->band_masking_cb[band] - noise;
if (snr_cb >= 1312) {
c->abits[ch][band] = 26;
ret &= ~USED_1ABITS;
} else if (snr_cb >= 222) {
c->abits[ch][band] = 8 + mul32(snr_cb - 222, 69000000);
ret &= ~(USED_26ABITS | USED_1ABITS);
} else if (snr_cb >= 0) {
c->abits[ch][band] = 2 + mul32(snr_cb, 106000000);
ret &= ~(USED_26ABITS | USED_1ABITS);
} else if (forbid_zero || snr_cb >= -140) {
c->abits[ch][band] = 1;
ret &= ~USED_26ABITS;
} else {
c->abits[ch][band] = 0;
ret &= ~(USED_26ABITS | USED_1ABITS);
}
}
c->consumed_bits += set_best_abits_code(c->abits[ch], 32,
&c->bit_allocation_sel[ch]);
}
/* Recalc scale_factor each time to get bits consumption in case of Huffman coding.
It is suboptimal solution */
/* TODO: May be cache scaled values */
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
if (c->prediction_mode[ch][band] == -1) {
c->scale_factor[ch][band] = calc_one_scale(c, c->peak_cb[ch][band],
c->abits[ch][band],
&c->quant[ch][band]);
}
}
}
quantize_adpcm(c);
quantize_pcm(c);
memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t));
memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t));
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
if (c->abits[ch][band] && c->abits[ch][band] <= DCA_CODE_BOOKS) {
accumulate_huff_bit_consumption(c->abits[ch][band],
c->quantized[ch][band],
huff_bit_count_accum[ch][c->abits[ch][band] - 1]);
clc_bit_count_accum[ch][c->abits[ch][band] - 1] += bit_consumption[c->abits[ch][band]];
} else {
bits_counter += bit_consumption[c->abits[ch][band]];
}
}
}
for (ch = 0; ch < c->fullband_channels; ch++) {
bits_counter += set_best_code(huff_bit_count_accum[ch],
clc_bit_count_accum[ch],
c->quant_index_sel[ch]);
}
c->consumed_bits += bits_counter;
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;
int forbid_zero = 1;
restart:
init_quantization_noise(c, c->worst_quantization_noise, forbid_zero);
low = high = c->worst_quantization_noise;
if (c->consumed_bits > c->frame_bits) {
while (c->consumed_bits > c->frame_bits) {
if (used_abits == USED_1ABITS && forbid_zero) {
forbid_zero = 0;
goto restart;
}
low = high;
high += snr_fudge;
used_abits = init_quantization_noise(c, high, forbid_zero);
}
} 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, forbid_zero);
}
}
/* 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, forbid_zero);
if (c->consumed_bits <= c->frame_bits)
high -= down;
}
init_quantization_noise(c, high, forbid_zero);
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[ch][k] = input[k * c->channels + chi];
}
}
static void fill_in_adpcm_bufer(DCAEncContext *c)
{
int ch, band;
int32_t step_size;
/* We fill in ADPCM work buffer for subbands which hasn't been ADPCM coded
* in current frame - we need this data if subband of next frame is
* ADPCM
*/
for (ch = 0; ch < c->channels; ch++) {
for (band = 0; band < 32; band++) {
int32_t *samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
if (c->prediction_mode[ch][band] == -1) {
step_size = get_step_size(c, ch, band);
ff_dca_core_dequantize(c->adpcm_history[ch][band],
c->quantized[ch][band]+12, step_size,
ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], 0, 4);
} else {
AV_COPY128U(c->adpcm_history[ch][band], c->adpcm_history[ch][band]+4);
}
/* Copy dequantized values for LPC analysis.
* It reduces artifacts in case of extreme quantization,
* example: in current frame abits is 1 and has no prediction flag,
* but end of this frame is sine like signal. In this case, if LPC analysis uses
* original values, likely LPC analysis returns good prediction gain, and sets prediction flag.
* But there are no proper value in decoder history, so likely result will be no good.
* Bitstream has "Predictor history flag switch", but this flag disables history for all subbands
*/
samples[0] = c->adpcm_history[ch][band][0] * (1 << 7);
samples[1] = c->adpcm_history[ch][band][1] * (1 << 7);
samples[2] = c->adpcm_history[ch][band][2] * (1 << 7);
samples[3] = c->adpcm_history[ch][band][3] * (1 << 7);
}
}
}
static void calc_lfe_scales(DCAEncContext *c)
{
if (c->lfe_channel)
c->lfe_scale_factor = calc_one_scale(c, c->lfe_peak_cb, 11, &c->lfe_quant);
}
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)
{
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, c->bit_allocation_sel[ch]);
/* Quantization index codebook select */
for (i = 0; i < DCA_CODE_BOOKS; i++)
for (ch = 0; ch < c->fullband_channels; ch++)
put_bits(&c->pb, ff_dca_quant_index_sel_nbits[i], c->quant_index_sel[ch][i]);
/* Scale factor adjustment index: transmitted in case of Huffman coding */
for (i = 0; i < DCA_CODE_BOOKS; i++)
for (ch = 0; ch < c->fullband_channels; ch++)
if (c->quant_index_sel[ch][i] < ff_dca_quant_index_group_size[i])
put_bits(&c->pb, 2, 0);
/* Audio header CRC check word: not transmitted */
}
static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
{
int i, j, sum, bits, sel;
if (c->abits[ch][band] <= DCA_CODE_BOOKS) {
av_assert0(c->abits[ch][band] > 0);
sel = c->quant_index_sel[ch][c->abits[ch][band] - 1];
// Huffman codes
if (sel < ff_dca_quant_index_group_size[c->abits[ch][band] - 1]) {
ff_dca_vlc_enc_quant(&c->pb, &c->quantized[ch][band][ss * 8], 8,
sel, c->abits[ch][band] - 1);
return;
}
// Block codes
if (c->abits[ch][band] <= 7) {
for (i = 0; i < 8; i += 4) {
sum = 0;
for (j = 3; j >= 0; j--) {
sum *= ff_dca_quant_levels[c->abits[ch][band]];
sum += c->quantized[ch][band][ss * 8 + i + j];
sum += (ff_dca_quant_levels[c->abits[ch][band]] - 1) / 2;
}
put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum);
}
return;
}
}
for (i = 0; i < 8; i++) {
bits = bit_consumption[c->abits[ch][band]] / 16;
put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]);
}
}
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, !(c->prediction_mode[ch][band] == -1));
/* Prediction VQ address */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCAENC_SUBBANDS; band++)
if (c->prediction_mode[ch][band] >= 0)
put_bits(&c->pb, 12, c->prediction_mode[ch][band]);
/* Bit allocation index */
for (ch = 0; ch < c->fullband_channels; ch++) {
if (c->bit_allocation_sel[ch] == 6) {
for (band = 0; band < DCAENC_SUBBANDS; band++) {
put_bits(&c->pb, 5, c->abits[ch][band]);
}
} else {
ff_dca_vlc_enc_alloc(&c->pb, c->abits[ch], DCAENC_SUBBANDS,
c->bit_allocation_sel[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++)
if (c->abits[ch][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++)
if (c->abits[ch][band])
put_bits(&c->pb, 7, c->scale_factor[ch][band]);
/* 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++)
if (c->abits[ch][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);
if (c->options.adpcm_mode)
adpcm_analysis(c);
find_peaks(c);
assign_bits(c);
calc_lfe_scales(c);
shift_history(c, samples);
init_put_bits(&c->pb, avpkt->data, avpkt->size);
fill_in_adpcm_bufer(c);
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->size = put_bytes_output(&c->pb);
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
*got_packet_ptr = 1;
return 0;
}
#define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption options[] = {
{ "dca_adpcm", "Use ADPCM encoding", offsetof(DCAEncContext, options.adpcm_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DCAENC_FLAGS },
{ NULL },
};
static const AVClass dcaenc_class = {
.class_name = "DCA (DTS Coherent Acoustics)",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault defaults[] = {
{ "b", "1411200" },
{ NULL },
};
const 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,
.close = encode_close,
.encode2 = encode_frame,
.capabilities = AV_CODEC_CAP_EXPERIMENTAL,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.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,
.priv_class = &dcaenc_class,
};
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