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ffmpeg/libavcodec/dcaenc.c
Andreas Rheinhardt 211619ad7f avcodec: Remove the FFT_FIXED_32 define 
Since the removal of the 16-bit FFT said define is unnecessary as
FFT_FIXED_32 is always !FFT_FLOAT. But one wouldn't believe it when
looking at the code.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-08-05 19:46:33 +02:00

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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
#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 "encode.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_get_encode_buffer(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);
flush_put_bits(&c->pb);
memset(put_bits_ptr(&c->pb), 0, put_bytes_left(&c->pb, 0));
avpkt->pts = frame->pts;
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,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL,
.priv_data_size = sizeof(DCAEncContext),
.init = encode_init,
.close = encode_close,
.encode2 = encode_frame,
.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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