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ffmpeg.wasm-core
Commit d439ba15 authored by Alexander E. Patrakov's avatar Alexander E. Patrakov Committed by Michael Niedermayer
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DCA/DTA encoder

parent d84f191d
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Showing with 1134 additions and 2 deletions
doc/general.texi
View file @ d439ba15
......@@ -604,7 +604,7 @@ following image formats are supported:
@tab Codec used in Delphine Software International games.
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
@item DCA (DTS Coherent Acoustics) @tab @tab X
@item DCA (DTS Coherent Acoustics) @tab X @tab X
@item DPCM id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item DPCM Interplay @tab @tab X
......
libavcodec/Makefile
View file @ d439ba15
......@@ -105,6 +105,7 @@ OBJS-$(CONFIG_COOK_DECODER) += cook.o
OBJS-$(CONFIG_CSCD_DECODER) += cscd.o
OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o dcadsp.o
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o
OBJS-$(CONFIG_DFA_DECODER) += dfa.o
OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o \
......
libavcodec/allcodecs.c
View file @ d439ba15
......@@ -246,7 +246,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER (COOK, cook);
REGISTER_DECODER (DCA, dca);
REGISTER_ENCDEC (DCA, dca);
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
REGISTER_DECODER (EAC3, eac3);
REGISTER_ENCDEC (FLAC, flac);
......
libavcodec/dcaenc.c 0 → 100644
View file @ d439ba15
/*
* DCA encoder
* Copyright (C) 2008 Alexander E. Patrakov
* 2010 Benjamin Larsson
* 2011 Xiang Wang
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/common.h"
#include "libavutil/avassert.h"
#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#include "dcaenc.h"
#include "dcadata.h"
#undef NDEBUG
#define MAX_CHANNELS 6
#define DCA_SUBBANDS_32 32
#define DCA_MAX_FRAME_SIZE 16383
#define DCA_HEADER_SIZE 13
#define DCA_SUBBANDS 32 ///< Subband activity count
#define QUANTIZER_BITS 16
#define SUBFRAMES 1
#define SUBSUBFRAMES 4
#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
#define LFE_BITS 8
#define LFE_INTERPOLATION 64
#define LFE_PRESENT 2
#define LFE_MISSING 0
static const int8_t dca_lfe_index[] = {
1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
};
static const int8_t dca_channel_reorder_lfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, 2, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, 3, -1, -1, -1, -1 },
{ 0, 1, -1, 2, 3, -1, -1, -1, -1 },
{ 1, 2, 0, -1, 3, 4, -1, -1, -1 },
{ 2, 3, -1, 0, 1, 4, 5, -1, -1 },
{ 1, 2, 0, -1, 3, 4, 5, -1, -1 },
{ 0, -1, 4, 5, 2, 3, 1, -1, -1 },
{ 3, 4, 1, -1, 0, 2, 5, 6, -1 },
{ 2, 3, -1, 5, 7, 0, 1, 4, 6 },
{ 3, 4, 1, -1, 0, 2, 5, 7, 6 },
};
static const int8_t dca_channel_reorder_nolfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
{ 0, 1, 2, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, 3, -1, -1, -1, -1, -1 },
{ 0, 1, 2, 3, -1, -1, -1, -1, -1 },
{ 1, 2, 0, 3, 4, -1, -1, -1, -1 },
{ 2, 3, 0, 1, 4, 5, -1, -1, -1 },
{ 1, 2, 0, 3, 4, 5, -1, -1, -1 },
{ 0, 4, 5, 2, 3, 1, -1, -1, -1 },
{ 3, 4, 1, 0, 2, 5, 6, -1, -1 },
{ 2, 3, 5, 7, 0, 1, 4, 6, -1 },
{ 3, 4, 1, 0, 2, 5, 7, 6, -1 },
};
typedef struct {
PutBitContext pb;
int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
int start[MAX_CHANNELS];
int frame_size;
int prim_channels;
int lfe_channel;
int sample_rate_code;
int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
int lfe_scale_factor;
int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
int a_mode; ///< audio channels arrangement
int num_channel;
int lfe_state;
int lfe_offset;
const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
} DCAContext;
static int32_t cos_table[128];
static inline int32_t mul32(int32_t a, int32_t b)
{
int64_t r = (int64_t) a * b;
/* round the result before truncating - improves accuracy */
return (r + 0x80000000) >> 32;
}
/* Integer version of the cosine modulated Pseudo QMF */
static void qmf_init(void)
{
int i;
int32_t c[17], s[17];
s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */
c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */
for (i = 1; i <= 16; i++) {
s[i] = 2 * (mul32(c[i - 1], 105372028) + mul32(s[i - 1], 2144896908));
c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
}
for (i = 0; i < 16; i++) {
cos_table[i ] = c[i] >> 3; /* avoid output overflow */
cos_table[i + 16] = s[16 - i] >> 3;
cos_table[i + 32] = -s[i] >> 3;
cos_table[i + 48] = -c[16 - i] >> 3;
cos_table[i + 64] = -c[i] >> 3;
cos_table[i + 80] = -s[16 - i] >> 3;
cos_table[i + 96] = s[i] >> 3;
cos_table[i + 112] = c[16 - i] >> 3;
}
}
static int32_t band_delta_factor(int band, int sample_num)
{
int index = band * (2 * sample_num + 1);
if (band == 0)
return 0x07ffffff;
else
return cos_table[index & 127];
}
static void add_new_samples(DCAContext *c, const int32_t *in,
int count, int channel)
{
int i;
/* Place new samples into the history buffer */
for (i = 0; i < count; i++) {
c->history[channel][c->start[channel] + i] = in[i];
av_assert0(c->start[channel] + i < 512);
}
c->start[channel] += count;
if (c->start[channel] == 512)
c->start[channel] = 0;
av_assert0(c->start[channel] < 512);
}
static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
int channel)
{
int band, i, j, k;
int32_t resp;
int32_t accum[DCA_SUBBANDS_32] = {0};
add_new_samples(c, in, DCA_SUBBANDS_32, channel);
/* Calculate the dot product of the signal with the (possibly inverted)
reference decoder's response to this vector:
(0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
so that -1.0 cancels 1.0 from the previous step */
for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
for (i = 0; i < c->start[channel]; k++, j++, i++)
accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
resp = 0;
/* TODO: implement FFT instead of this naive calculation */
for (band = 0; band < DCA_SUBBANDS_32; band++) {
for (j = 0; j < 32; j++)
resp += mul32(accum[j], band_delta_factor(band, j));
out[band] = (band & 2) ? (-resp) : resp;
}
}
static int32_t lfe_fir_64i[512];
static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
{
int i, j;
int channel = c->prim_channels;
int32_t accum = 0;
add_new_samples(c, in, LFE_INTERPOLATION, channel);
for (i = c->start[channel], j = 0; i < 512; i++, j++)
accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
for (i = 0; i < c->start[channel]; i++, j++)
accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
return accum;
}
static void init_lfe_fir(void)
{
static int initialized = 0;
int i;
if (initialized)
return;
for (i = 0; i < 512; i++)
lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
initialized = 1;
}
static void put_frame_header(DCAContext *c)
{
/* SYNC */
put_bits(&c->pb, 16, 0x7ffe);
put_bits(&c->pb, 16, 0x8001);
/* Frame type: normal */
put_bits(&c->pb, 1, 1);
/* Deficit sample count: none */
put_bits(&c->pb, 5, 31);
/* CRC is not present */
put_bits(&c->pb, 1, 0);
/* Number of PCM sample blocks */
put_bits(&c->pb, 7, PCM_SAMPLES-1);
/* Primary frame byte size */
put_bits(&c->pb, 14, c->frame_size-1);
/* Audio channel arrangement: L + R (stereo) */
put_bits(&c->pb, 6, c->num_channel);
/* Core audio sampling frequency */
put_bits(&c->pb, 4, c->sample_rate_code);
/* Transmission bit rate: 1411.2 kbps */
put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
/* 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 interpolation factor=64 */
put_bits(&c->pb, 2, c->lfe_state);
/* Predictor history switch flag: on */
put_bits(&c->pb, 1, 1);
/* No CRC */
/* Multirate interpolator switch: non-perfect reconstruction */
put_bits(&c->pb, 1, 0);
/* Encoder software revision: 7 */
put_bits(&c->pb, 4, 7);
/* Copy history: 0 */
put_bits(&c->pb, 2, 0);
/* Source PCM resolution: 16 bits, not DTS ES */
put_bits(&c->pb, 3, 0);
/* Front sum/difference coding: no */
put_bits(&c->pb, 1, 0);
/* Surrounds sum/difference coding: no */
put_bits(&c->pb, 1, 0);
/* Dialog normalization: 0 dB */
put_bits(&c->pb, 4, 0);
}
static void put_primary_audio_header(DCAContext *c)
{
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
int ch, i;
/* Number of subframes */
put_bits(&c->pb, 4, SUBFRAMES - 1);
/* Number of primary audio channels */
put_bits(&c->pb, 3, c->prim_channels - 1);
/* Subband activity count */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
/* High frequency VQ start subband */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
/* Joint intensity coding index: 0, 0 */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 3, 0);
/* Transient mode codebook: A4, A4 (arbitrary) */
for (ch = 0; ch < c->prim_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->prim_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Bit allocation quantizer select: linear 5-bit */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Quantization index codebook select: dummy data
to avoid transmission of scale factor adjustment */
for (i = 1; i < 11; i++)
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, bitlen[i], thr[i]);
/* Scale factor adjustment index: not transmitted */
}
/**
* 8-23 bits quantization
* @param sample
* @param bits
*/
static inline uint32_t quantize(int32_t sample, int bits)
{
av_assert0(sample < 1 << (bits - 1));
av_assert0(sample >= -(1 << (bits - 1)));
sample &= sample & ((1 << bits) - 1);
return sample;
}
static inline int find_scale_factor7(int64_t max_value, int bits)
{
int i = 0, j = 128, q;
max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
while (i < j) {
q = (i + j) >> 1;
if (max_value < scale_factor_quant7[q])
j = q;
else
i = q + 1;
}
av_assert1(i < 128);
return i;
}
static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
int scale_factor)
{
sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
put_bits(&c->pb, bits, quantize((int) sample, bits));
}
static void put_subframe(DCAContext *c,
int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
int subframe)
{
int i, sub, ss, ch, max_value;
int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
/* 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->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 1, 0);
/* Prediction VQ addres: not transmitted */
/* Bit allocation index */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 5, QUANTIZER_BITS+3);
if (SUBSUBFRAMES > 1) {
/* Transition mode: none for each channel and subband */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 1, 0); /* codebook A4 */
}
/* Determine scale_factor */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++) {
max_value = 0;
for (i = 0; i < 8 * SUBSUBFRAMES; i++)
max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
}
if (c->lfe_channel) {
max_value = 0;
for (i = 0; i < 4 * SUBSUBFRAMES; i++)
max_value = FFMAX(max_value, FFABS(lfe_data[i]));
c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
}
/* Scale factors: the same for each channel and subband,
encoded according to Table D.1.2 */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
/* 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 */
if (c->lfe_channel) {
for (i = 0; i < 4 * SUBSUBFRAMES; i++)
put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
put_bits(&c->pb, 8, c->lfe_scale_factor);
}
/* Audio data (subsubframes) */
for (ss = 0; ss < SUBSUBFRAMES ; ss++)
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
for (i = 0; i < 8; i++)
put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
/* DSYNC */
put_bits(&c->pb, 16, 0xffff);
}
static void put_frame(DCAContext *c,
int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
uint8_t *frame)
{
int i;
init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
put_primary_audio_header(c);
for (i = 0; i < SUBFRAMES; i++)
put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
flush_put_bits(&c->pb);
c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
put_frame_header(c);
flush_put_bits(&c->pb);
}
static int encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
{
int i, k, channel;
DCAContext *c = avctx->priv_data;
int16_t *samples = data;
int real_channel = 0;
for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
for (channel = 0; channel < c->prim_channels + 1; channel++) {
/* Get 32 PCM samples */
for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
}
/* Put subband samples into the proper place */
real_channel = c->channel_order_tab[channel];
if (real_channel >= 0) {
qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
}
}
}
if (c->lfe_channel) {
for (i = 0; i < PCM_SAMPLES / 2; i++) {
for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
c->lfe_data[i] = lfe_downsample(c, c->pcm);
}
}
put_frame(c, c->subband, frame);
return c->frame_size;
}
static int encode_init(AVCodecContext *avctx)
{
DCAContext *c = avctx->priv_data;
int i;
c->prim_channels = avctx->channels;
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
switch (avctx->channel_layout) {
case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break;
case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
default:
av_log(avctx, AV_LOG_ERROR,
"Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
return AVERROR_PATCHWELCOME;
}
if (c->lfe_channel) {
init_lfe_fir();
c->prim_channels--;
c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
c->lfe_state = LFE_PRESENT;
c->lfe_offset = dca_lfe_index[c->a_mode];
} else {
c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
c->lfe_state = LFE_MISSING;
}
for (i = 0; i < 16; i++) {
if (dca_sample_rates[i] && (dca_sample_rates[i] == avctx->sample_rate))
break;
}
if (i == 16) {
av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
for (i = 0; i < 16; i++)
av_log(avctx, AV_LOG_ERROR, "%d, ", dca_sample_rates[i]);
av_log(avctx, AV_LOG_ERROR, "supported.\n");
return -1;
}
c->sample_rate_code = i;
avctx->frame_size = 32 * PCM_SAMPLES;
if (!cos_table[127])
qmf_init();
return 0;
}
AVCodec ff_dca_encoder = {
.name = "dca",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = encode_init,
.encode = encode_frame,
.capabilities = CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
};
libavcodec/dcaenc.h 0 → 100644
View file @ d439ba15
/*
* DCA encoder tables
* Copyright (C) 2008 Alexander E. Patrakov
*
* 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
*/
#ifndef AVCODEC_DCAENC_H
#define AVCODEC_DCAENC_H
/* This is a scaled version of the response of the reference decoder to
this vector of subband samples: ( 1.0 0.0 0.0 ... 0.0 )
*/
static const int32_t UnQMF[512] = {
7,
4,
-961,
-2844,
-8024,
-18978,
-32081,
-15635,
-16582,
-18359,
-17180,
-14868,
-11664,
-8051,
-4477,
-1327,
-1670,
-6019,
-11590,
-18030,
-24762,
-30965,
-35947,
-36145,
-37223,
-86311,
-57024,
-27215,
-11274,
-4684,
42,
108,
188,
250,
-1007,
-596,
-2289,
-12218,
-27191,
-124367,
-184256,
-250538,
-323499,
-397784,
-468855,
-532072,
-583000,
-618041,
-777916,
-783868,
-765968,
-724740,
-662468,
-583058,
-490548,
-401623,
-296090,
-73154,
-36711,
-7766,
-2363,
-4905,
2388,
2681,
5651,
4086,
71110,
139742,
188067,
151237,
101355,
309917,
343690,
358839,
357555,
334606,
289625,
224152,
142063,
48725,
74996,
238425,
411666,
584160,
744276,
880730,
983272,
1041933,
1054396,
789531,
851022,
864032,
675431,
418134,
35762,
66911,
103502,
136403,
-55147,
-245269,
-499595,
-808470,
-1136858,
-2010912,
-2581654,
-3151901,
-3696328,
-4196599,
-4633761,
-4993229,
-5262495,
-5436311,
-477650,
-901314,
-1308090,
-1677468,
-1985525,
-2212848,
-2341196,
-2373915,
-2269552,
-2620489,
-2173858,
-1629954,
-946595,
-193499,
1119459,
1138657,
1335311,
1126544,
2765033,
3139603,
3414913,
3599213,
3676363,
3448981,
3328726,
3111551,
2810887,
2428657,
1973684,
1457278,
893848,
300995,
-292521,
-867621,
-1404936,
-1871278,
-2229831,
-2440932,
-2462684,
-2255006,
-1768898,
-1079574,
82115,
1660302,
3660715,
6123610,
8329598,
11888744,
15722147,
19737089,
25647773,
31039399,
36868007,
43124253,
49737161,
56495958,
63668945,
71039511,
78540240,
86089058,
93600041,
100981151,
108136061,
114970055,
121718321,
127566038,
132774642,
137247294,
140894737,
143635018,
145395599,
146114032,
145742999,
144211606,
141594341,
137808404,
132914122,
126912246,
120243281,
112155281,
103338368,
93904953,
83439152,
72921548,
62192990,
51434918,
40894003,
30786726,
21384955,
12939112,
5718193,
-5790,
-3959261,
-5870978,
-5475538,
-2517061,
3247310,
12042937,
24076729,
39531397,
58562863,
81297002,
107826748,
138209187,
172464115,
210569037,
252468018,
298045453,
347168648,
399634888,
455137189,
513586535,
574537650,
637645129,
702597163,
768856566,
836022040,
903618096,
971159680,
1038137214,
1103987353,
1168195035,
1230223053,
1289539180,
1345620373,
1397957958,
1446063657,
1489474689,
1527740502,
1560502307,
1587383079,
1608071145,
1622301248,
1629859340,
1630584888,
1624373875,
1611178348,
1591018893,
1563948667,
1530105004,
1489673227,
1442904075,
1390107674,
1331590427,
1267779478,
1199115126,
1126053392,
1049146257,
968928307,
885965976,
800851610,
714186243,
626590147,
538672486,
451042824,
364299927,
279026812,
195785029,
115109565,
37503924,
-36564551,
-106668063,
-172421668,
-233487283,
-289575706,
-340448569,
-385919511,
-425854915,
-460174578,
-488840702,
-511893328,
-529405118,
-541489888,
-548312207,
-550036471,
-547005316,
-539436808,
-527630488,
-512084785,
-492941605,
-470665204,
-445668379,
-418328829,
-389072810,
-358293846,
-326396227,
-293769619,
-260792276,
-227825056,
-195208961,
-163262121,
-132280748,
-102533727,
-74230062,
-47600637,
-22817785,
-25786,
20662895,
39167253,
55438413,
69453741,
81242430,
90795329,
98213465,
103540643,
106917392,
108861938,
108539682,
106780704,
103722568,
99043289,
93608686,
87266209,
80212203,
72590022,
64603428,
56362402,
48032218,
39749162,
31638971,
23814664,
16376190,
9409836,
2988017,
-2822356,
-7976595,
-12454837,
-16241147,
-19331944,
-21735011,
-23468284,
-24559822,
-25042936,
-25035583,
-24429587,
-23346408,
-21860411,
-20015718,
-17025330,
-14968728,
-12487138,
-9656319,
-7846681,
-5197816,
-2621904,
-144953,
2144746,
3990570,
5845884,
7454650,
8820394,
9929891,
10784445,
11390921,
11762056,
11916017,
12261189,
12117604,
11815303,
11374622,
10815301,
10157241,
9418799,
8629399,
7780776,
7303680,
6353499,
5392738,
4457895,
3543062,
1305978,
1402521,
1084092,
965652,
-151008,
-666667,
-1032157,
-1231475,
-1319043,
-1006023,
-915720,
-773426,
-612377,
-445864,
-291068,
-161337,
-66484,
-11725,
133453,
388184,
615856,
804033,
942377,
1022911,
1041247,
995854,
891376,
572246,
457992,
316365,
172738,
43037,
-117662,
-98542,
-70279,
-41458,
-535790,
-959038,
-1364456,
-1502265,
-1568530,
-2378681,
-2701111,
-2976407,
-3182552,
-3314415,
-3366600,
-3337701,
-3232252,
-3054999,
1984841,
1925903,
1817377,
1669153,
1490069,
1292040,
1086223,
890983,
699163,
201358,
266971,
296990,
198419,
91119,
4737,
5936,
2553,
2060,
-3828,
-1664,
-4917,
-20796,
-36822,
-131247,
-154923,
-162055,
-161354,
-148762,
-125754,
-94473,
-57821,
-19096,
15172,
43004,
65624,
81354,
89325,
89524,
82766,
71075,
55128,
13686,
6921,
1449,
420,
785,
-215,
-179,
-113,
-49,
6002,
16007,
42978,
100662,
171472,
83975,
93702,
108813,
111893,
110272,
103914,
93973,
81606,
68041,
-54058,
-60695,
-65277,
-67224,
-66213,
-62082,
-55574,
-42988,
-35272,
-63735,
-33501,
-12671,
-4038,
-1232,
5,
7
};
#endif /* AVCODEC_DCAENC_H */
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