mirror of
https://github.com/FFmpeg/FFmpeg.git
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avcodec/dcaenc: Initial implementation of ADPCM encoding for DCA encoder
This commit is contained in:
Daniil Cherednik
Rostislav Pehlivanov
parent
5f928c5201
commit
b8c2b9c392
@ -244,7 +244,8 @@ OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
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OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadata.o dcahuff.o \
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dca_core.o dca_exss.o dca_xll.o dca_lbr.o \
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dcadsp.o dcadct.o synth_filter.o
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OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o dcahuff.o
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OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o dcahuff.o \
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dcaadpcm.o
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OBJS-$(CONFIG_DDS_DECODER) += dds.o
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OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o diractab.o \
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dirac_arith.o dirac_dwt.o dirac_vlc.o
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@ -18,6 +18,7 @@
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "dcaadpcm.h"
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#include "dcadec.h"
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#include "dcadata.h"
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#include "dcahuff.h"
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@ -670,46 +671,21 @@ static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, in
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return 0;
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}
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static inline void dequantize(int32_t *output, const int32_t *input,
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int32_t step_size, int32_t scale, int residual)
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{
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// Account for quantizer step size
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int64_t step_scale = (int64_t)step_size * scale;
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int n, shift = 0;
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// Limit scale factor resolution to 22 bits
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if (step_scale > (1 << 23)) {
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shift = av_log2(step_scale >> 23) + 1;
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step_scale >>= shift;
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}
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// Scale the samples
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if (residual) {
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for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
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output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
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} else {
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for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
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output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
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}
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}
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static inline void inverse_adpcm(int32_t **subband_samples,
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const int16_t *vq_index,
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const int8_t *prediction_mode,
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int sb_start, int sb_end,
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int ofs, int len)
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{
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int i, j, k;
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int i, j;
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for (i = sb_start; i < sb_end; i++) {
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if (prediction_mode[i]) {
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const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]];
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const int pred_id = vq_index[i];
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int32_t *ptr = subband_samples[i] + ofs;
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for (j = 0; j < len; j++) {
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int64_t err = 0;
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for (k = 0; k < DCA_ADPCM_COEFFS; k++)
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err += (int64_t)ptr[j - k - 1] * coeff[k];
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ptr[j] = clip23(ptr[j] + clip23(norm13(err)));
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int32_t x = ff_dcaadpcm_predict(pred_id, ptr + j - DCA_ADPCM_COEFFS);
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ptr[j] = clip23(ptr[j] + x);
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}
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}
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}
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@ -817,8 +793,8 @@ static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType heade
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scale = clip23(adj * scale >> 22);
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}
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dequantize(s->subband_samples[ch][band] + ofs,
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audio, step_size, scale, 0);
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ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs,
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audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES);
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}
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}
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@ -1146,8 +1122,8 @@ static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchann
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else
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scale = xbr_scale_factors[ch][band][1];
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dequantize(s->subband_samples[ch][band] + ofs,
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audio, step_size, scale, 1);
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ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs,
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audio, step_size, scale, 1, DCA_SUBBAND_SAMPLES);
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}
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}
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@ -1326,8 +1302,8 @@ static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int
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// Get the scale factor
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scale = s->scale_factors[ch][band >> 1][band & 1];
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dequantize(s->x96_subband_samples[ch][band] + ofs,
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audio, step_size, scale, 0);
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ff_dca_core_dequantize(s->x96_subband_samples[ch][band] + ofs,
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audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES);
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}
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}
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@ -33,6 +33,7 @@
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#include "dca_exss.h"
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#include "dcadsp.h"
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#include "dcadct.h"
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#include "dcamath.h"
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#include "dcahuff.h"
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#include "fft.h"
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#include "synth_filter.h"
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@ -43,7 +44,6 @@
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#define DCA_SUBFRAMES 16
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#define DCA_SUBBAND_SAMPLES 8
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#define DCA_PCMBLOCK_SAMPLES 32
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#define DCA_ADPCM_COEFFS 4
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#define DCA_LFE_HISTORY 8
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#define DCA_ABITS_MAX 26
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@ -195,6 +195,29 @@ static inline int ff_dca_core_map_spkr(DCACoreDecoder *core, int spkr)
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return -1;
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}
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static inline void ff_dca_core_dequantize(int32_t *output, const int32_t *input,
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int32_t step_size, int32_t scale, int residual, int len)
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{
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// Account for quantizer step size
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int64_t step_scale = (int64_t)step_size * scale;
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int n, shift = 0;
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// Limit scale factor resolution to 22 bits
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if (step_scale > (1 << 23)) {
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shift = av_log2(step_scale >> 23) + 1;
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step_scale >>= shift;
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}
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// Scale the samples
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if (residual) {
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for (n = 0; n < len; n++)
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output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
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} else {
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for (n = 0; n < len; n++)
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output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
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}
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}
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int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size);
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int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset);
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int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth);
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228
libavcodec/dcaadpcm.c
Normal file
228
libavcodec/dcaadpcm.c
Normal file
@ -0,0 +1,228 @@
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/*
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* DCA ADPCM engine
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* Copyright (C) 2017 Daniil Cherednik
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "dcaadpcm.h"
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#include "dcaenc.h"
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#include "dca_core.h"
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#include "mathops.h"
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typedef int32_t premultiplied_coeffs[10];
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//assume we have DCA_ADPCM_COEFFS values before x
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static inline int64_t calc_corr(const int32_t *x, int len, int j, int k)
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{
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int n;
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int64_t s = 0;
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for (n = 0; n < len; n++)
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s += MUL64(x[n-j], x[n-k]);
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return s;
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}
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static inline int64_t apply_filter(const int16_t a[DCA_ADPCM_COEFFS], const int64_t corr[15], const int32_t aa[10])
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{
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int64_t err = 0;
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int64_t tmp = 0;
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err = corr[0];
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tmp += MUL64(a[0], corr[1]);
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tmp += MUL64(a[1], corr[2]);
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tmp += MUL64(a[2], corr[3]);
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tmp += MUL64(a[3], corr[4]);
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tmp = norm__(tmp, 13);
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tmp += tmp;
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err -= tmp;
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tmp = 0;
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tmp += MUL64(corr[5], aa[0]);
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tmp += MUL64(corr[6], aa[1]);
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tmp += MUL64(corr[7], aa[2]);
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tmp += MUL64(corr[8], aa[3]);
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tmp += MUL64(corr[9], aa[4]);
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tmp += MUL64(corr[10], aa[5]);
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tmp += MUL64(corr[11], aa[6]);
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tmp += MUL64(corr[12], aa[7]);
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tmp += MUL64(corr[13], aa[8]);
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tmp += MUL64(corr[14], aa[9]);
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tmp = norm__(tmp, 26);
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err += tmp;
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return llabs(err);
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}
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static int64_t find_best_filter(const DCAADPCMEncContext *s, const int32_t *in, int len)
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{
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const premultiplied_coeffs *precalc_data = s->private_data;
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int i, j, k = 0;
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int vq;
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int64_t err;
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int64_t min_err = 1ll << 62;
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int64_t corr[15];
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for (i = 0; i <= DCA_ADPCM_COEFFS; i++)
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for (j = i; j <= DCA_ADPCM_COEFFS; j++)
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corr[k++] = calc_corr(in+4, len, i, j);
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for (i = 0; i < DCA_ADPCM_VQCODEBOOK_SZ; i++) {
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err = apply_filter(ff_dca_adpcm_vb[i], corr, *precalc_data);
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if (err < min_err) {
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min_err = err;
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vq = i;
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}
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precalc_data++;
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}
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return vq;
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}
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static inline int64_t calc_prediction_gain(int pred_vq, const int32_t *in, int32_t *out, int len)
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{
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int i;
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int32_t error;
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int64_t signal_energy = 0;
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int64_t error_energy = 0;
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for (i = 0; i < len; i++) {
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error = in[DCA_ADPCM_COEFFS + i] - ff_dcaadpcm_predict(pred_vq, in + i);
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out[i] = error;
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signal_energy += MUL64(in[DCA_ADPCM_COEFFS + i], in[DCA_ADPCM_COEFFS + i]);
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error_energy += MUL64(error, error);
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}
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if (!error_energy)
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return -1;
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return signal_energy / error_energy;
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}
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int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *in, int len, int *diff)
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{
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int pred_vq, i;
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int32_t input_buffer[16 + DCA_ADPCM_COEFFS];
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int32_t input_buffer2[16 + DCA_ADPCM_COEFFS];
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int32_t max = 0;
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int shift_bits;
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uint64_t pg = 0;
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for (i = 0; i < len + DCA_ADPCM_COEFFS; i++)
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max |= FFABS(in[i]);
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// normalize input to simplify apply_filter
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shift_bits = av_log2(max) - 11;
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for (i = 0; i < len + DCA_ADPCM_COEFFS; i++) {
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input_buffer[i] = norm__(in[i], 7);
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input_buffer2[i] = norm__(in[i], shift_bits);
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}
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pred_vq = find_best_filter(s, input_buffer2, len);
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if (pred_vq < 0)
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return -1;
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pg = calc_prediction_gain(pred_vq, input_buffer, diff, len);
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// Greater than 10db (10*log(10)) prediction gain to use ADPCM.
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// TODO: Tune it.
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if (pg < 10)
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return -1;
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for (i = 0; i < len; i++)
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diff[i] <<= 7;
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return pred_vq;
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}
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static void precalc(premultiplied_coeffs *data)
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{
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int i, j, k;
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for (i = 0; i < DCA_ADPCM_VQCODEBOOK_SZ; i++) {
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int id = 0;
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int32_t t = 0;
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for (j = 0; j < DCA_ADPCM_COEFFS; j++) {
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for (k = j; k < DCA_ADPCM_COEFFS; k++) {
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t = (int32_t)ff_dca_adpcm_vb[i][j] * (int32_t)ff_dca_adpcm_vb[i][k];
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if (j != k)
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t *= 2;
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(*data)[id++] = t;
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}
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}
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data++;
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}
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}
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int ff_dcaadpcm_do_real(int pred_vq_index,
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softfloat quant, int32_t scale_factor, int32_t step_size,
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const int32_t *prev_hist, const int32_t *in, int32_t *next_hist, int32_t *out,
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int len, int32_t peak)
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{
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int i;
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int64_t delta;
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int32_t dequant_delta;
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int32_t work_bufer[16 + DCA_ADPCM_COEFFS];
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memcpy(work_bufer, prev_hist, sizeof(int32_t) * DCA_ADPCM_COEFFS);
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for (i = 0; i < len; i++) {
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work_bufer[DCA_ADPCM_COEFFS + i] = ff_dcaadpcm_predict(pred_vq_index, &work_bufer[i]);
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delta = (int64_t)in[i] - ((int64_t)work_bufer[DCA_ADPCM_COEFFS + i] << 7);
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out[i] = quantize_value(av_clip64(delta, -peak, peak), quant);
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ff_dca_core_dequantize(&dequant_delta, &out[i], step_size, scale_factor, 0, 1);
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work_bufer[DCA_ADPCM_COEFFS+i] += dequant_delta;
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}
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memcpy(next_hist, &work_bufer[len], sizeof(int32_t) * DCA_ADPCM_COEFFS);
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return 0;
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}
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av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s)
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{
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if (!s)
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return -1;
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s->private_data = av_malloc(sizeof(premultiplied_coeffs) * DCA_ADPCM_VQCODEBOOK_SZ);
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precalc(s->private_data);
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return 0;
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}
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av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s)
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{
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if (!s)
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return;
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av_freep(&s->private_data);
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}
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54
libavcodec/dcaadpcm.h
Normal file
54
libavcodec/dcaadpcm.h
Normal file
@ -0,0 +1,54 @@
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/*
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* DCA ADPCM engine
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* Copyright (C) 2017 Daniil Cherednik
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*
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* This file is part of FFmpeg.
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*
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* 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
|
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* 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
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*/
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#ifndef AVCODEC_DCAADPCM_H
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#define AVCODEC_DCAADPCM_H
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#include "dcamath.h"
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#include "dcadata.h"
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#include "dcaenc.h"
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typedef struct DCAADPCMEncContext {
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void *private_data;
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} DCAADPCMEncContext;
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static inline int64_t ff_dcaadpcm_predict(int pred_vq_index, const int32_t *input)
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{
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int i;
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const int16_t *coeff = ff_dca_adpcm_vb[pred_vq_index];
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int64_t pred = 0;
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for (i = 0; i < DCA_ADPCM_COEFFS; i++)
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pred += (int64_t)input[DCA_ADPCM_COEFFS - 1 - i] * coeff[i];
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return clip23(norm13(pred));
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}
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int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *input, int len, int *diff);
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int ff_dcaadpcm_do_real(int pred_vq_index,
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softfloat quant, int32_t scale_factor, int32_t step_size,
|
||||
const int32_t *prev_hist, const int32_t *in, int32_t *next_hist, int32_t *out,
|
||||
int len, int32_t peak);
|
||||
|
||||
av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s);
|
||||
av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s);
|
||||
|
||||
#endif /* AVCODEC_DCAADPCM_H */
|
||||
@ -61,7 +61,7 @@ const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS] = {
|
||||
/* ADPCM data */
|
||||
|
||||
/* 16 bits signed fractional Q13 binary codes */
|
||||
const int16_t ff_dca_adpcm_vb[4096][4] = {
|
||||
const int16_t ff_dca_adpcm_vb[DCA_ADPCM_VQCODEBOOK_SZ][DCA_ADPCM_COEFFS] = {
|
||||
{ 9928, -2618, -1093, -1263 },
|
||||
{ 11077, -2876, -1747, -308 },
|
||||
{ 10503, -1082, -1426, -1167 },
|
||||
|
||||
@ -25,6 +25,9 @@
|
||||
|
||||
#include "dcahuff.h"
|
||||
|
||||
#define DCA_ADPCM_COEFFS 4
|
||||
#define DCA_ADPCM_VQCODEBOOK_SZ 4096
|
||||
|
||||
extern const uint32_t ff_dca_bit_rates[32];
|
||||
|
||||
extern const uint8_t ff_dca_channels[16];
|
||||
@ -36,7 +39,7 @@ extern const uint8_t ff_dca_dmix_primary_nch[8];
|
||||
extern const uint8_t ff_dca_quant_index_sel_nbits[DCA_CODE_BOOKS];
|
||||
extern const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS];
|
||||
|
||||
extern const int16_t ff_dca_adpcm_vb[4096][4];
|
||||
extern const int16_t ff_dca_adpcm_vb[DCA_ADPCM_VQCODEBOOK_SZ][DCA_ADPCM_COEFFS];
|
||||
|
||||
extern const uint32_t ff_dca_scale_factor_quant6[64];
|
||||
extern const uint32_t ff_dca_scale_factor_quant7[128];
|
||||
|
||||
@ -25,8 +25,12 @@
|
||||
#include "libavutil/channel_layout.h"
|
||||
#include "libavutil/common.h"
|
||||
#include "libavutil/ffmath.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 "internal.h"
|
||||
@ -44,8 +48,15 @@
|
||||
#define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
|
||||
#define AUBANDS 25
|
||||
|
||||
typedef struct CompressionOptions {
|
||||
int adpcm_mode;
|
||||
} CompressionOptions;
|
||||
|
||||
typedef struct DCAEncContext {
|
||||
AVClass *class;
|
||||
PutBitContext pb;
|
||||
DCAADPCMEncContext adpcm_ctx;
|
||||
CompressionOptions options;
|
||||
int frame_size;
|
||||
int frame_bits;
|
||||
int fullband_channels;
|
||||
@ -61,10 +72,13 @@ typedef struct DCAEncContext {
|
||||
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][SUBBAND_SAMPLES];
|
||||
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];
|
||||
@ -77,6 +91,7 @@ typedef struct DCAEncContext {
|
||||
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
|
||||
} DCAEncContext;
|
||||
|
||||
static int32_t cos_table[2048];
|
||||
@ -107,18 +122,52 @@ static double gammafilter(int i, double f)
|
||||
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 -1;
|
||||
|
||||
/* 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)
|
||||
{
|
||||
int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS;
|
||||
av_freep(&bufer);
|
||||
}
|
||||
|
||||
static int encode_init(AVCodecContext *avctx)
|
||||
{
|
||||
DCAEncContext *c = avctx->priv_data;
|
||||
uint64_t layout = avctx->channel_layout;
|
||||
int i, j, min_frame_bits;
|
||||
|
||||
if (subband_bufer_alloc(c))
|
||||
return AVERROR(ENOMEM);
|
||||
|
||||
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;
|
||||
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 "
|
||||
@ -150,6 +199,12 @@ static int encode_init(AVCodecContext *avctx)
|
||||
}
|
||||
/* 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++) {
|
||||
@ -238,6 +293,16 @@ static int encode_init(AVCodecContext *avctx)
|
||||
return 0;
|
||||
}
|
||||
|
||||
static av_cold int encode_close(AVCodecContext *avctx)
|
||||
{
|
||||
if (avctx->priv_data) {
|
||||
DCAEncContext *c = avctx->priv_data;
|
||||
subband_bufer_free(c);
|
||||
ff_dcaadpcm_free(&c->adpcm_ctx);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
static inline int32_t cos_t(int x)
|
||||
{
|
||||
return cos_table[x & 2047];
|
||||
@ -253,12 +318,6 @@ static inline int32_t half32(int32_t a)
|
||||
return (a + 1) >> 1;
|
||||
}
|
||||
|
||||
static inline int32_t mul32(int32_t a, int32_t b)
|
||||
{
|
||||
int64_t r = (int64_t)a * b + 0x80000000ULL;
|
||||
return r >> 32;
|
||||
}
|
||||
|
||||
static void subband_transform(DCAEncContext *c, const int32_t *input)
|
||||
{
|
||||
int ch, subs, i, k, j;
|
||||
@ -545,31 +604,53 @@ static void calc_masking(DCAEncContext *c, const int32_t *input)
|
||||
}
|
||||
}
|
||||
|
||||
static inline int32_t find_peak(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(m);
|
||||
}
|
||||
|
||||
static void find_peaks(DCAEncContext *c)
|
||||
{
|
||||
int band, ch;
|
||||
|
||||
for (ch = 0; ch < c->fullband_channels; ch++)
|
||||
for (ch = 0; ch < c->fullband_channels; ch++) {
|
||||
for (band = 0; band < 32; band++) {
|
||||
int sample;
|
||||
int32_t m = 0;
|
||||
|
||||
for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
|
||||
int32_t s = abs(c->subband[ch][band][sample]);
|
||||
if (m < s)
|
||||
m = s;
|
||||
}
|
||||
c->peak_cb[ch][band] = get_cb(m);
|
||||
c->peak_cb[ch][band] = find_peak(c->subband[ch][band], SUBBAND_SAMPLES);
|
||||
}
|
||||
}
|
||||
|
||||
if (c->lfe_channel) {
|
||||
int sample;
|
||||
int32_t m = 0;
|
||||
c->lfe_peak_cb = find_peak(c->downsampled_lfe, DCA_LFE_SAMPLES);
|
||||
}
|
||||
}
|
||||
|
||||
for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)
|
||||
if (m < abs(c->downsampled_lfe[sample]))
|
||||
m = abs(c->downsampled_lfe[sample]);
|
||||
c->lfe_peak_cb = get_cb(m);
|
||||
static void 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(estimated_diff, 16);
|
||||
} else {
|
||||
c->prediction_mode[ch][band] = -1;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
@ -578,13 +659,16 @@ static const int snr_fudge = 128;
|
||||
#define USED_NABITS 2
|
||||
#define USED_26ABITS 4
|
||||
|
||||
static int32_t quantize_value(int32_t value, softfloat quant)
|
||||
static inline int32_t get_step_size(const DCAEncContext *c, int ch, int band)
|
||||
{
|
||||
int32_t offset = 1 << (quant.e - 1);
|
||||
int32_t step_size;
|
||||
|
||||
value = mul32(value, quant.m) + offset;
|
||||
value = value >> quant.e;
|
||||
return value;
|
||||
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(int32_t peak_cb, int abits, softfloat *quant)
|
||||
@ -619,14 +703,40 @@ static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
|
||||
return our_nscale;
|
||||
}
|
||||
|
||||
static void quantize_all(DCAEncContext *c)
|
||||
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(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, 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++)
|
||||
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
|
||||
c->quantized[ch][band][sample] = quantize_value(c->subband[ch][band][sample], c->quant[ch][band]);
|
||||
if (c->prediction_mode[ch][band] == -1)
|
||||
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
|
||||
c->quantized[ch][band][sample] = quantize_value(c->subband[ch][band][sample], c->quant[ch][band]);
|
||||
}
|
||||
|
||||
static void accumulate_huff_bit_consumption(int abits, int32_t *quantized, uint32_t *result)
|
||||
@ -710,6 +820,7 @@ static int init_quantization_noise(DCAEncContext *c, int noise)
|
||||
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;
|
||||
|
||||
@ -740,12 +851,15 @@ static int init_quantization_noise(DCAEncContext *c, int noise)
|
||||
/* TODO: May be cache scaled values */
|
||||
for (ch = 0; ch < c->fullband_channels; ch++) {
|
||||
for (band = 0; band < 32; band++) {
|
||||
c->scale_factor[ch][band] = calc_one_scale(c->peak_cb[ch][band],
|
||||
c->abits[ch][band],
|
||||
&c->quant[ch][band]);
|
||||
if (c->prediction_mode[ch][band] == -1) {
|
||||
c->scale_factor[ch][band] = calc_one_scale(c->peak_cb[ch][band],
|
||||
c->abits[ch][band],
|
||||
&c->quant[ch][band]);
|
||||
}
|
||||
}
|
||||
}
|
||||
quantize_all(c);
|
||||
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));
|
||||
@ -819,6 +933,41 @@ static void shift_history(DCAEncContext *c, const int32_t *input)
|
||||
}
|
||||
}
|
||||
|
||||
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] << 7;
|
||||
samples[1] = c->adpcm_history[ch][band][1] << 7;
|
||||
samples[2] = c->adpcm_history[ch][band][2] << 7;
|
||||
samples[3] = c->adpcm_history[ch][band][3] << 7;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static void calc_lfe_scales(DCAEncContext *c)
|
||||
{
|
||||
if (c->lfe_channel)
|
||||
@ -1001,9 +1150,14 @@ static void put_subframe(DCAEncContext *c, int subframe)
|
||||
/* Prediction mode: no ADPCM, in each channel and subband */
|
||||
for (ch = 0; ch < c->fullband_channels; ch++)
|
||||
for (band = 0; band < DCAENC_SUBBANDS; band++)
|
||||
put_bits(&c->pb, 1, 0);
|
||||
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]);
|
||||
|
||||
/* Prediction VQ address: not transmitted */
|
||||
/* Bit allocation index */
|
||||
for (ch = 0; ch < c->fullband_channels; ch++) {
|
||||
if (c->bit_allocation_sel[ch] == 6) {
|
||||
@ -1068,12 +1222,15 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
||||
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++)
|
||||
@ -1092,6 +1249,20 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
||||
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 },
|
||||
@ -1104,6 +1275,7 @@ AVCodec ff_dca_encoder = {
|
||||
.id = AV_CODEC_ID_DTS,
|
||||
.priv_data_size = sizeof(DCAEncContext),
|
||||
.init = encode_init,
|
||||
.close = encode_close,
|
||||
.encode2 = encode_frame,
|
||||
.capabilities = AV_CODEC_CAP_EXPERIMENTAL,
|
||||
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
|
||||
@ -1116,4 +1288,5 @@ AVCodec ff_dca_encoder = {
|
||||
AV_CH_LAYOUT_5POINT1,
|
||||
0 },
|
||||
.defaults = defaults,
|
||||
.priv_class = &dcaenc_class,
|
||||
};
|
||||
|
||||
@ -24,6 +24,8 @@
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
#include "dcamath.h"
|
||||
|
||||
typedef struct {
|
||||
int32_t m;
|
||||
int32_t e;
|
||||
@ -144,4 +146,13 @@ static const int8_t channel_reorder_nolfe[16][9] = {
|
||||
{ 3, 2, 4, 0, 1, 5, 7, 6, -1 },
|
||||
};
|
||||
|
||||
static inline 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;
|
||||
}
|
||||
|
||||
#endif /* AVCODEC_DCAENC_H */
|
||||
|
||||
@ -49,6 +49,7 @@ static inline int32_t mul17(int32_t a, int32_t b) { return mul__(a, b, 17); }
|
||||
static inline int32_t mul22(int32_t a, int32_t b) { return mul__(a, b, 22); }
|
||||
static inline int32_t mul23(int32_t a, int32_t b) { return mul__(a, b, 23); }
|
||||
static inline int32_t mul31(int32_t a, int32_t b) { return mul__(a, b, 31); }
|
||||
static inline int32_t mul32(int32_t a, int32_t b) { return mul__(a, b, 32); }
|
||||
|
||||
static inline int32_t clip23(int32_t a) { return av_clip_intp2(a, 23); }
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user