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FFmpeg/libavcodec/dca.c
Stefano Sabatini fe4bf37455 Make AVCodec long_names definition conditional depending on CONFIG_SMALL. 
Originally committed as revision 13759 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-12 21:50:13 +00:00

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/*
* 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 av_cold 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;
}
/* There is nothing that prevents a dts frame to change channel configuration
but FFmpeg doesn't support that so only set the channels if it is previously
unset. Ideally during the first probe for channels the crc should be checked
and only set avctx->channels when the crc is ok. Right now the decoder could
set the channels based on a broken first frame.*/
if (!avctx->channels)
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 av_cold 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 av_cold 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,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
};
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