Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
12905 Commits
37 Branches
400 Tags
464 MiB
 
 
 
 
Tag: Branch: Tree: 88855b51cd
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from '88855b51cd'
${ noResults }
FFmpeg/libavcodec/dca.c

1260 lines
44 KiB
Raw Blame History

/*
* DCA compatible decoder
* Copyright (C) 2004 Gildas Bazin
* Copyright (C) 2004 Benjamin Zores
* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
*
* 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
*/
/**
* @file dca.c
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "avcodec.h"
#include "dsputil.h"
#include "bitstream.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca.h"
//#define TRACE
#define DCA_PRIM_CHANNELS_MAX (5)
#define DCA_SUBBANDS (32)
#define DCA_ABITS_MAX (32) /* Should be 28 */
#define DCA_SUBSUBFAMES_MAX (4)
#define DCA_LFE_MAX (3)
enum DCAMode {
DCA_MONO = 0,
DCA_CHANNEL,
DCA_STEREO,
DCA_STEREO_SUMDIFF,
DCA_STEREO_TOTAL,
DCA_3F,
DCA_2F1R,
DCA_3F1R,
DCA_2F2R,
DCA_3F2R,
DCA_4F2R
};
#define DCA_DOLBY 101 /* FIXME */
#define DCA_CHANNEL_BITS 6
#define DCA_CHANNEL_MASK 0x3F
#define DCA_LFE 0x80
#define HEADER_SIZE 14
#define CONVERT_BIAS 384
#define DCA_MAX_FRAME_SIZE 16383
/** Bit allocation */
typedef struct {
int offset; ///< code values offset
int maxbits[8]; ///< max bits in VLC
int wrap; ///< wrap for get_vlc2()
VLC vlc[8]; ///< actual codes
} BitAlloc;
static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
static BitAlloc dca_tmode; ///< transition mode VLCs
static BitAlloc dca_scalefactor; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
/** Pre-calculated cosine modulation coefs for the QMF */
static float cos_mod[544];
static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
{
return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
}
typedef struct {
AVCodecContext *avctx;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
int crc_present; ///< crc is present in the bitstream
int sample_blocks; ///< number of PCM sample blocks
int frame_size; ///< primary frame byte size
int amode; ///< audio channels arrangement
int sample_rate; ///< audio sampling rate
int bit_rate; ///< transmission bit rate
int downmix; ///< embedded downmix enabled
int dynrange; ///< embedded dynamic range flag
int timestamp; ///< embedded time stamp flag
int aux_data; ///< auxiliary data flag
int hdcd; ///< source material is mastered in HDCD
int ext_descr; ///< extension audio descriptor flag
int ext_coding; ///< extended coding flag
int aspf; ///< audio sync word insertion flag
int lfe; ///< low frequency effects flag
int predictor_history; ///< predictor history flag
int header_crc; ///< header crc check bytes
int multirate_inter; ///< multirate interpolator switch
int version; ///< encoder software revision
int copy_history; ///< copy history
int source_pcm_res; ///< source pcm resolution
int front_sum; ///< front sum/difference flag
int surround_sum; ///< surround sum/difference flag
int dialog_norm; ///< dialog normalisation parameter
/* Primary audio coding header */
int subframes; ///< number of subframes
int total_channels; ///< number of channels including extensions
int prim_channels; ///< number of primary audio channels
int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
/* Primary audio coding side information */
int subsubframes; ///< number of subsubframes
int partial_samples; ///< partial subsubframe samples count
int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
int dynrange_coef; ///< dynamic range coefficient
int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
2 /*history */ ]; ///< Low frequency effect data
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
int output; ///< type of output
int bias; ///< output bias
DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
GetBitContext gb;
/* Current position in DCA frame */
int current_subframe;
int current_subsubframe;
int debug_flag; ///< used for suppressing repeated error messages output
DSPContext dsp;
} DCAContext;
static void dca_init_vlcs(void)
{
static int vlcs_initialized = 0;
int i, j;
if (vlcs_initialized)
return;
dca_bitalloc_index.offset = 1;
dca_bitalloc_index.wrap = 2;
for (i = 0; i < 5; i++)
init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
bitalloc_12_bits[i], 1, 1,
bitalloc_12_codes[i], 2, 2, 1);
dca_scalefactor.offset = -64;
dca_scalefactor.wrap = 2;
for (i = 0; i < 5; i++)
init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
scales_bits[i], 1, 1,
scales_codes[i], 2, 2, 1);
dca_tmode.offset = 0;
dca_tmode.wrap = 1;
for (i = 0; i < 4; i++)
init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
tmode_bits[i], 1, 1,
tmode_codes[i], 2, 2, 1);
for(i = 0; i < 10; i++)
for(j = 0; j < 7; j++){
if(!bitalloc_codes[i][j]) break;
dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i],
bitalloc_bits[i][j], 1, 1,
bitalloc_codes[i][j], 2, 2, 1);
}
vlcs_initialized = 1;
}
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
{
while(len--)
*dst++ = get_bits(gb, bits);
}
static int dca_parse_frame_header(DCAContext * s)
{
int i, j;
static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
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 };
s->bias = CONVERT_BIAS;
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
get_bits(&s->gb, 32);
/* Frame header */
s->frame_type = get_bits(&s->gb, 1);
s->samples_deficit = get_bits(&s->gb, 5) + 1;
s->crc_present = get_bits(&s->gb, 1);
s->sample_blocks = get_bits(&s->gb, 7) + 1;
s->frame_size = get_bits(&s->gb, 14) + 1;
if (s->frame_size < 95)
return -1;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
return -1;
s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)];
if (!s->bit_rate)
return -1;
s->downmix = get_bits(&s->gb, 1);
s->dynrange = get_bits(&s->gb, 1);
s->timestamp = get_bits(&s->gb, 1);
s->aux_data = get_bits(&s->gb, 1);
s->hdcd = get_bits(&s->gb, 1);
s->ext_descr = get_bits(&s->gb, 3);
s->ext_coding = get_bits(&s->gb, 1);
s->aspf = get_bits(&s->gb, 1);
s->lfe = get_bits(&s->gb, 2);
s->predictor_history = get_bits(&s->gb, 1);
/* TODO: check CRC */
if (s->crc_present)
s->header_crc = get_bits(&s->gb, 16);
s->multirate_inter = get_bits(&s->gb, 1);
s->version = get_bits(&s->gb, 4);
s->copy_history = get_bits(&s->gb, 2);
s->source_pcm_res = get_bits(&s->gb, 3);
s->front_sum = get_bits(&s->gb, 1);
s->surround_sum = get_bits(&s->gb, 1);
s->dialog_norm = get_bits(&s->gb, 4);
/* FIXME: channels mixing levels */
s->output = s->amode;
if(s->lfe) s->output |= DCA_LFE;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
s->sample_blocks, s->sample_blocks * 32);
av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
s->amode, dca_channels[s->amode]);
av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n",
s->sample_rate, dca_sample_rates[s->sample_rate]);
av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
s->bit_rate, dca_bit_rates[s->bit_rate]);
av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
s->predictor_history);
av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
s->multirate_inter);
av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
av_log(s->avctx, AV_LOG_DEBUG,
"source pcm resolution: %i (%i bits/sample)\n",
s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
#endif
/* Primary audio coding header */
s->subframes = get_bits(&s->gb, 4) + 1;
s->total_channels = get_bits(&s->gb, 3) + 1;
s->prim_channels = s->total_channels;
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */
for (i = 0; i < s->prim_channels; i++) {
s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->subband_activity[i] > DCA_SUBBANDS)
s->subband_activity[i] = DCA_SUBBANDS;
}
for (i = 0; i < s->prim_channels; i++) {
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->vq_start_subband[i] > DCA_SUBBANDS)
s->vq_start_subband[i] = DCA_SUBBANDS;
}
get_array(&s->gb, s->joint_intensity, s->prim_channels, 3);
get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
/* Get codebooks quantization indexes */
memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
for (j = 1; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
s->scalefactor_adj[i][j] = 1;
for (j = 1; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
if (s->quant_index_huffman[i][j] < thr[j])
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (s->crc_present) {
/* Audio header CRC check */
get_bits(&s->gb, 16);
}
s->current_subframe = 0;
s->current_subsubframe = 0;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
for(i = 0; i < s->prim_channels; i++){
av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %i",
s->quant_index_huffman[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
#endif
return 0;
}
static inline int get_scale(GetBitContext *gb, int level, int value)
{
if (level < 5) {
/* huffman encoded */
value += get_bitalloc(gb, &dca_scalefactor, level);
} else if(level < 8)
value = get_bits(gb, level + 1);
return value;
}
static int dca_subframe_header(DCAContext * s)
{
/* Primary audio coding side information */
int j, k;
s->subsubframes = get_bits(&s->gb, 2) + 1;
s->partial_samples = get_bits(&s->gb, 3);
for (j = 0; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
s->prediction_mode[j][k] = get_bits(&s->gb, 1);
}
/* Get prediction codebook */
for (j = 0; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
if (s->prediction_mode[j][k] > 0) {
/* (Prediction coefficient VQ address) */
s->prediction_vq[j][k] = get_bits(&s->gb, 12);
}
}
}
/* Bit allocation index */
for (j = 0; j < s->prim_channels; j++) {
for (k = 0; k < s->vq_start_subband[j]; k++) {
if (s->bitalloc_huffman[j] == 6)
s->bitalloc[j][k] = get_bits(&s->gb, 5);
else if (s->bitalloc_huffman[j] == 5)
s->bitalloc[j][k] = get_bits(&s->gb, 4);
else if (s->bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
return -1;
} else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
}
if (s->bitalloc[j][k] > 26) {
// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
// j, k, s->bitalloc[j][k]);
return -1;
}
}
}
/* Transition mode */
for (j = 0; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
s->transition_mode[j][k] = 0;
if (s->subsubframes > 1 &&
k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
s->transition_mode[j][k] =
get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
}
}
}
for (j = 0; j < s->prim_channels; j++) {
const uint32_t *scale_table;
int scale_sum;
memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
if (s->scalefactor_huffman[j] == 6)
scale_table = scale_factor_quant7;
else
scale_table = scale_factor_quant6;
/* When huffman coded, only the difference is encoded */
scale_sum = 0;
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
s->scale_factor[j][k][0] = scale_table[scale_sum];
}
if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
/* Get second scale factor */
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
s->scale_factor[j][k][1] = scale_table[scale_sum];
}
}
}
/* Joint subband scale factor codebook select */
for (j = 0; j < s->prim_channels; j++) {
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0)
s->joint_huff[j] = get_bits(&s->gb, 3);
}
/* Scale factors for joint subband coding */
for (j = 0; j < s->prim_channels; j++) {
int source_channel;
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0) {
int scale = 0;
source_channel = s->joint_intensity[j] - 1;
/* When huffman coded, only the difference is encoded
* (is this valid as well for joint scales ???) */
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
scale = get_scale(&s->gb, s->joint_huff[j], 0);
scale += 64; /* bias */
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
}
if (!s->debug_flag & 0x02) {
av_log(s->avctx, AV_LOG_DEBUG,
"Joint stereo coding not supported\n");
s->debug_flag |= 0x02;
}
}
}
/* Stereo downmix coefficients */
if (s->prim_channels > 2) {
if(s->downmix) {
for (j = 0; j < s->prim_channels; j++) {
s->downmix_coef[j][0] = get_bits(&s->gb, 7);
s->downmix_coef[j][1] = get_bits(&s->gb, 7);
}
} else {
int am = s->amode & DCA_CHANNEL_MASK;
for (j = 0; j < s->prim_channels; j++) {
s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
}
}
}
/* Dynamic range coefficient */
if (s->dynrange)
s->dynrange_coef = get_bits(&s->gb, 8);
/* Side information CRC check word */
if (s->crc_present) {
get_bits(&s->gb, 16);
}
/*
* Primary audio data arrays
*/
/* VQ encoded high frequency subbands */
for (j = 0; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
/* 1 vector -> 32 samples */
s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
/* Low frequency effect data */
if (s->lfe) {
/* LFE samples */
int lfe_samples = 2 * s->lfe * s->subsubframes;
float lfe_scale;
for (j = lfe_samples; j < lfe_samples * 2; j++) {
/* Signed 8 bits int */
s->lfe_data[j] = get_sbits(&s->gb, 8);
}
/* Scale factor index */
s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s->lfe_scale_factor;
for (j = lfe_samples; j < lfe_samples * 2; j++)
s->lfe_data[j] *= lfe_scale;
}
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
s->partial_samples);
for (j = 0; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = 0; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG,
"prediction coefs: %f, %f, %f, %f\n",
(float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
(float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
(float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
(float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
}
for (j = 0; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
for (k = 0; k < s->vq_start_subband[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = 0; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
for (k = 0; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = 0; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = 0; j < s->prim_channels; j++) {
if (s->joint_intensity[j] > 0) {
int source_channel = s->joint_intensity[j] - 1;
av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
}
if (s->prim_channels > 2 && s->downmix) {
av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
for (j = 0; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
for (j = 0; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
if(s->lfe){
int lfe_samples = 2 * s->lfe * s->subsubframes;
av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
for (j = lfe_samples; j < lfe_samples * 2; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
#endif
return 0;
}
static void qmf_32_subbands(DCAContext * s, int chans,
float samples_in[32][8], float *samples_out,
float scale, float bias)
{
const float *prCoeff;
int i, j, k;
float praXin[33], *raXin = &praXin[1];
float *subband_fir_hist = s->subband_fir_hist[chans];
float *subband_fir_hist2 = s->subband_fir_noidea[chans];
int chindex = 0, subindex;
praXin[0] = 0.0;
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
prCoeff = fir_32bands_nonperfect;
else /* Perfect reconstruction */
prCoeff = fir_32bands_perfect;
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
float t1, t2, sum[16], diff[16];
/* Load in one sample from each subband and clear inactive subbands */
for (i = 0; i < s->subband_activity[chans]; i++)
raXin[i] = samples_in[i][subindex];
for (; i < 32; i++)
raXin[i] = 0.0;
/* Multiply by cosine modulation coefficients and
* create temporary arrays SUM and DIFF */
for (j = 0, k = 0; k < 16; k++) {
t1 = 0.0;
t2 = 0.0;
for (i = 0; i < 16; i++, j++){
t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
}
sum[k] = t1 + t2;
diff[k] = t1 - t2;
}
j = 512;
/* Store history */
for (k = 0; k < 16; k++)
subband_fir_hist[k] = cos_mod[j++] * sum[k];
for (k = 0; k < 16; k++)
subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
/* Multiply by filter coefficients */
for (k = 31, i = 0; i < 32; i++, k--)
for (j = 0; j < 512; j += 64){
subband_fir_hist2[i] += prCoeff[i+j] * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
}
/* Create 32 PCM output samples */
for (i = 0; i < 32; i++)
samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
/* Update working arrays */
memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
}
}
static void lfe_interpolation_fir(int decimation_select,
int num_deci_sample, float *samples_in,
float *samples_out, float scale,
float bias)
{
/* samples_in: An array holding decimated samples.
* Samples in current subframe starts from samples_in[0],
* while samples_in[-1], samples_in[-2], ..., stores samples
* from last subframe as history.
*
* samples_out: An array holding interpolated samples
*/
int decifactor, k, j;
const float *prCoeff;
int interp_index = 0; /* Index to the interpolated samples */
int deciindex;
/* Select decimation filter */
if (decimation_select == 1) {
decifactor = 128;
prCoeff = lfe_fir_128;
} else {
decifactor = 64;
prCoeff = lfe_fir_64;
}
/* Interpolation */
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
/* One decimated sample generates decifactor interpolated ones */
for (k = 0; k < decifactor; k++) {
float rTmp = 0.0;
//FIXME the coeffs are symetric, fix that
for (j = 0; j < 512 / decifactor; j++)
rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
samples_out[interp_index++] = rTmp / scale + bias;
}
}
}
/* downmixing routines */
#define MIX_REAR1(samples, si1, rs, coef) \
samples[i] += samples[si1] * coef[rs][0]; \
samples[i+256] += samples[si1] * coef[rs][1];
#define MIX_REAR2(samples, si1, si2, rs, coef) \
samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
#define MIX_FRONT3(samples, coef) \
t = samples[i]; \
samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
for(i = 0; i < 256; i++){ \
op1 \
op2 \
}
static void dca_downmix(float *samples, int srcfmt,
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2])
{
int i;
float t;
float coef[DCA_PRIM_CHANNELS_MAX][2];
for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
}
switch (srcfmt) {
case DCA_MONO:
case DCA_CHANNEL:
case DCA_STEREO_TOTAL:
case DCA_STEREO_SUMDIFF:
case DCA_4F2R:
av_log(NULL, 0, "Not implemented!\n");
break;
case DCA_STEREO:
break;
case DCA_3F:
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
break;
case DCA_2F1R:
DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
break;
case DCA_3F1R:
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR1(samples, i + 768, 3, coef));
break;
case DCA_2F2R:
DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
break;
case DCA_3F2R:
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
break;
}
}
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int *values)
{
int i;
int offset = (levels - 1) >> 1;
for (i = 0; i < 4; i++) {
values[i] = (code % levels) - offset;
code /= levels;
}
if (code == 0)
return 0;
else {
av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
return -1;
}
}
static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
static int dca_subsubframe(DCAContext * s)
{
int k, l;
int subsubframe = s->current_subsubframe;
const float *quant_step_table;
/* FIXME */
float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
/*
* Audio data
*/
/* Select quantization step size table */
if (s->bit_rate == 0x1f)
quant_step_table = lossless_quant_d;
else
quant_step_table = lossy_quant_d;
for (k = 0; k < s->prim_channels; k++) {
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
/* Select the mid-tread linear quantizer */
int abits = s->bitalloc[k][l];
float quant_step_size = quant_step_table[abits];
float rscale;
/*
* Determine quantization index code book and its type
*/
/* Select quantization index code book */
int sel = s->quant_index_huffman[k][abits];
/*
* Extract bits from the bit stream
*/
if(!abits){
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
}else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
if(abits <= 7){
/* Block code */
int block_code1, block_code2, size, levels;
int block[8];
size = abits_sizes[abits-1];
levels = abits_levels[abits-1];
block_code1 = get_bits(&s->gb, size);
/* FIXME Should test return value */
decode_blockcode(block_code1, levels, block);
block_code2 = get_bits(&s->gb, size);
decode_blockcode(block_code2, levels, &block[4]);
for (m = 0; m < 8; m++)
subband_samples[k][l][m] = block[m];
}else{
/* no coding */
for (m = 0; m < 8; m++)
subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
}
}else{
/* Huffman coded */
for (m = 0; m < 8; m++)
subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
}
/* Deal with transients */
if (s->transition_mode[k][l] &&
subsubframe >= s->transition_mode[k][l])
rscale = quant_step_size * s->scale_factor[k][l][1];
else
rscale = quant_step_size * s->scale_factor[k][l][0];
rscale *= s->scalefactor_adj[k][sel];
for (m = 0; m < 8; m++)
subband_samples[k][l][m] *= rscale;
/*
* Inverse ADPCM if in prediction mode
*/
if (s->prediction_mode[k][l]) {
int n;
for (m = 0; m < 8; m++) {
for (n = 1; n <= 4; n++)
if (m >= n)
subband_samples[k][l][m] +=
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
subband_samples[k][l][m - n] / 8192);
else if (s->predictor_history)
subband_samples[k][l][m] +=
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
s->subband_samples_hist[k][l][m - n +
4] / 8192);
}
}
}
/*
* Decode VQ encoded high frequencies
*/
for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
/* 1 vector -> 32 samples but we only need the 8 samples
* for this subsubframe. */
int m;
if (!s->debug_flag & 0x01) {
av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
for (m = 0; m < 8; m++) {
subband_samples[k][l][m] =
high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
m]
* (float) s->scale_factor[k][l][0] / 16.0;
}
}
}
/* Check for DSYNC after subsubframe */
if (s->aspf || subsubframe == s->subsubframes - 1) {
if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
#endif
} else {
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
}
}
/* Backup predictor history for adpcm */
for (k = 0; k < s->prim_channels; k++)
for (l = 0; l < s->vq_start_subband[k]; l++)
memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
4 * sizeof(subband_samples[0][0][0]));
/* 32 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
/* static float pcm_to_double[8] =
{32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k],
2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
0 /*s->bias */ );
}
/* Down mixing */
if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
dca_downmix(s->samples, s->amode, s->downmix_coef);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->output & DCA_LFE) {
int lfe_samples = 2 * s->lfe * s->subsubframes;
int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
lfe_interpolation_fir(s->lfe, 2 * s->lfe,
s->lfe_data + lfe_samples +
2 * s->lfe * subsubframe,
&s->samples[256 * i_channels],
256.0, 0 /* s->bias */);
/* Outputs 20bits pcm samples */
}
return 0;
}
static int dca_subframe_footer(DCAContext * s)
{
int aux_data_count = 0, i;
int lfe_samples;
/*
* Unpack optional information
*/
if (s->timestamp)
get_bits(&s->gb, 32);
if (s->aux_data)
aux_data_count = get_bits(&s->gb, 6);
for (i = 0; i < aux_data_count; i++)
get_bits(&s->gb, 8);
if (s->crc_present && (s->downmix || s->dynrange))
get_bits(&s->gb, 16);
lfe_samples = 2 * s->lfe * s->subsubframes;
for (i = 0; i < lfe_samples; i++) {
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
}
return 0;
}
/**
* Decode a dca frame block
*
* @param s pointer to the DCAContext
*/
static int dca_decode_block(DCAContext * s)
{
/* Sanity check */
if (s->current_subframe >= s->subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->subframes);
return -1;
}
if (!s->current_subsubframe) {
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
if (dca_subframe_header(s))
return -1;
}
/* Read subsubframe */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
if (dca_subsubframe(s))
return -1;
/* Update state */
s->current_subsubframe++;
if (s->current_subsubframe >= s->subsubframes) {
s->current_subsubframe = 0;
s->current_subframe++;
}
if (s->current_subframe >= s->subframes) {
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
/* Read subframe footer */
if (dca_subframe_footer(s))
return -1;
}
return 0;
}
/**
* Convert bitstream to one representation based on sync marker
*/
static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
int max_size)
{
uint32_t mrk;
int i, tmp;
const uint16_t *ssrc = (const uint16_t *) src;
uint16_t *sdst = (uint16_t *) dst;
PutBitContext pb;
if((unsigned)src_size > (unsigned)max_size) {
av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
return -1;
}
mrk = AV_RB32(src);
switch (mrk) {
case DCA_MARKER_RAW_BE:
memcpy(dst, src, FFMIN(src_size, max_size));
return FFMIN(src_size, max_size);
case DCA_MARKER_RAW_LE:
for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
*sdst++ = bswap_16(*ssrc++);
return FFMIN(src_size, max_size);
case DCA_MARKER_14B_BE:
case DCA_MARKER_14B_LE:
init_put_bits(&pb, dst, max_size);
for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
put_bits(&pb, 14, tmp);
}
flush_put_bits(&pb);
return (put_bits_count(&pb) + 7) >> 3;
default:
return -1;
}
}
/**
* Main frame decoding function
* FIXME add arguments
*/
static int dca_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
const uint8_t * buf, int buf_size)
{
int i, j, k;
int16_t *samples = data;
DCAContext *s = avctx->priv_data;
int channels;
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
if (s->dca_buffer_size == -1) {
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
return -1;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
if (dca_parse_frame_header(s) < 0) {
//seems like the frame is corrupt, try with the next one
*data_size=0;
return buf_size;
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
channels = s->prim_channels + !!s->lfe;
if(avctx->request_channels == 2 && s->prim_channels > 2) {
channels = 2;
s->output = DCA_STEREO;
}
avctx->channels = channels;
if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
return -1;
*data_size = 0;
for (i = 0; i < (s->sample_blocks / 8); i++) {
dca_decode_block(s);
s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
/* interleave samples */
for (j = 0; j < 256; j++) {
for (k = 0; k < channels; k++)
samples[k] = s->tsamples[j + k * 256];
samples += channels;
}
*data_size += 256 * sizeof(int16_t) * channels;
}
return buf_size;
}
/**
* Build the cosine modulation tables for the QMF
*
* @param s pointer to the DCAContext
*/
static void pre_calc_cosmod(DCAContext * s)
{
int i, j, k;
static int cosmod_initialized = 0;
if(cosmod_initialized) return;
for (j = 0, k = 0; k < 16; k++)
for (i = 0; i < 16; i++)
cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64);
for (k = 0; k < 16; k++)
for (i = 0; i < 16; i++)
cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
for (k = 0; k < 16; k++)
cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
for (k = 0; k < 16; k++)
cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
cosmod_initialized = 1;
}
/**
* DCA initialization
*
* @param avctx pointer to the AVCodecContext
*/
static int dca_decode_init(AVCodecContext * avctx)
{
DCAContext *s = avctx->priv_data;
s->avctx = avctx;
dca_init_vlcs();
pre_calc_cosmod(s);
dsputil_init(&s->dsp, avctx);
/* allow downmixing to stereo */
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
avctx->request_channels == 2) {
avctx->channels = avctx->request_channels;
}
return 0;
}
AVCodec dca_decoder = {
.name = "dca",
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = dca_decode_init,
.decode = dca_decode_frame,
};