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Commit e55d5390 authored by Justin Ruggles's avatar Justin Ruggles
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atrac3: cosmetics: pretty-printing and renaming 

also does some minor refactoring.
parent abdee952
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Showing with 572 additions and 585 deletions
libavcodec/atrac3.c
View file @ e55d5390
......@@ -38,10 +38,10 @@
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "fft.h"
#include "fmtconvert.h"
#include "get_bits.h"
#include "atrac.h"
#include "atrac3data.h"
......@@ -52,142 +52,130 @@
#define SAMPLES_PER_FRAME 1024
#define MDCT_SIZE 512
/* These structures are needed to store the parsed gain control data. */
typedef struct {
int num_gain_data;
int levcode[8];
int loccode[8];
} gain_info;
typedef struct {
gain_info gBlock[4];
} gain_block;
typedef struct {
int pos;
int numCoefs;
float coef[8];
} tonal_component;
typedef struct {
int bandsCoded;
int numComponents;
tonal_component components[64];
float prevFrame[SAMPLES_PER_FRAME];
int gcBlkSwitch;
gain_block gainBlock[2];
typedef struct GainInfo {
int num_gain_data;
int lev_code[8];
int loc_code[8];
} GainInfo;
typedef struct GainBlock {
GainInfo g_block[4];
} GainBlock;
typedef struct TonalComponent {
int pos;
int num_coefs;
float coef[8];
} TonalComponent;
typedef struct ChannelUnit {
int bands_coded;
int num_components;
float prev_frame[SAMPLES_PER_FRAME];
int gc_blk_switch;
TonalComponent components[64];
GainBlock gain_block[2];
DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
DECLARE_ALIGNED(32, float, IMDCT_buf)[SAMPLES_PER_FRAME];
DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
float delayBuf1[46]; ///<qmf delay buffers
float delayBuf2[46];
float delayBuf3[46];
} channel_unit;
float delay_buf1[46]; ///<qmf delay buffers
float delay_buf2[46];
float delay_buf3[46];
} ChannelUnit;
typedef struct {
AVFrame frame;
GetBitContext gb;
typedef struct ATRAC3Context {
AVFrame frame;
GetBitContext gb;
//@{
/** stream data */
int channels;
int codingMode;
int bit_rate;
int sample_rate;
int samples_per_channel;
int samples_per_frame;
int bits_per_frame;
int bytes_per_frame;
int pBs;
channel_unit* pUnits;
int channels;
int coding_mode;
int bit_rate;
int sample_rate;
int samples_per_channel;
int samples_per_frame;
int bits_per_frame;
int bytes_per_frame;
ChannelUnit *units;
//@}
//@{
/** joint-stereo related variables */
int matrix_coeff_index_prev[4];
int matrix_coeff_index_now[4];
int matrix_coeff_index_next[4];
int weighting_delay[6];
int matrix_coeff_index_prev[4];
int matrix_coeff_index_now[4];
int matrix_coeff_index_next[4];
int weighting_delay[6];
//@}
//@{
/** data buffers */
uint8_t* decoded_bytes_buffer;
float tempBuf[1070];
uint8_t *decoded_bytes_buffer;
float temp_buf[1070];
//@}
//@{
/** extradata */
int atrac3version;
int delay;
int scrambled_stream;
int frame_factor;
int version;
int delay;
int scrambled_stream;
int frame_factor;
//@}
FFTContext mdct_ctx;
FmtConvertContext fmt_conv;
AVFloatDSPContext fdsp;
FFTContext mdct_ctx;
FmtConvertContext fmt_conv;
AVFloatDSPContext fdsp;
} ATRAC3Context;
static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
static VLC spectral_coeff_tab[7];
static float gain_tab1[16];
static float gain_tab2[31];
static VLC spectral_coeff_tab[7];
static float gain_tab1[16];
static float gain_tab2[31];
/**
* Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
* caused by the reverse spectra of the QMF.
/*
* Regular 512 points IMDCT without overlapping, with the exception of the
* swapping of odd bands caused by the reverse spectra of the QMF.
*
* @param pInput float input
* @param pOutput float output
* @param odd_band 1 if the band is an odd band
*/
static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
{
int i;
int i;
if (odd_band) {
/**
* Reverse the odd bands before IMDCT, this is an effect of the QMF transform
* or it gives better compression to do it this way.
* FIXME: It should be possible to handle this in imdct_calc
* for that to happen a modification of the prerotation step of
* all SIMD code and C code is needed.
* Or fix the functions before so they generate a pre reversed spectrum.
*/
for (i=0; i<128; i++)
FFSWAP(float, pInput[i], pInput[255-i]);
* Reverse the odd bands before IMDCT, this is an effect of the QMF
* transform or it gives better compression to do it this way.
* FIXME: It should be possible to handle this in imdct_calc
* for that to happen a modification of the prerotation step of
* all SIMD code and C code is needed.
* Or fix the functions before so they generate a pre reversed spectrum.
*/
for (i = 0; i < 128; i++)
FFSWAP(float, input[i], input[255 - i]);
}
q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
/* Perform windowing on the output. */
q->fdsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);
}
/**
* Atrac 3 indata descrambling, only used for data coming from the rm container
*
* @param inbuffer pointer to 8 bit array of indata
* @param out pointer to 8 bit array of outdata
* @param bytes amount of bytes
/*
* indata descrambling, only used for data coming from the rm container
*/
static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
{
int i, off;
uint32_t c;
const uint32_t* buf;
uint32_t* obuf = (uint32_t*) out;
const uint32_t *buf;
uint32_t *output = (uint32_t *)out;
off = (intptr_t)inbuffer & 3;
buf = (const uint32_t*) (inbuffer - off);
c = av_be2ne32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
off = (intptr_t)input & 3;
buf = (const uint32_t *)(input - off);
c = av_be2ne32((0x537F6103 >> (off * 8)) | (0x537F6103 << (32 - (off * 8))));
bytes += 3 + off;
for (i = 0; i < bytes/4; i++)
obuf[i] = c ^ buf[i];
for (i = 0; i < bytes / 4; i++)
output[i] = c ^ buf[i];
if (off)
av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);
......@@ -195,35 +183,34 @@ static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
return off;
}
static av_cold int init_atrac3_transforms(ATRAC3Context *q) {
static av_cold int init_atrac3_transforms(ATRAC3Context *q)
{
float enc_window[256];
int i;
/* Generate the mdct window, for details see
/* generate the mdct window, for details see
* http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
for (i=0 ; i<256; i++)
for (i = 0; i < 256; i++)
enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
if (!mdct_window[0])
for (i=0 ; i<256; i++) {
mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
mdct_window[511-i] = mdct_window[i];
if (!mdct_window[0]) {
for (i = 0; i < 256; i++) {
mdct_window[i] = enc_window[i] /
(enc_window[ i] * enc_window[ i] +
enc_window[255 - i] * enc_window[255 - i]);
mdct_window[511 - i] = mdct_window[i];
}
}
/* Initialize the MDCT transform. */
/* initialize the MDCT transform */
return ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768);
}
/**
* Atrac3 uninit, free all allocated memory
*/
static av_cold int atrac3_decode_close(AVCodecContext *avctx)
{
ATRAC3Context *q = avctx->priv_data;
av_free(q->pUnits);
av_free(q->units);
av_free(q->decoded_bytes_buffer);
ff_mdct_end(&q->mdct_ctx);
......@@ -231,192 +218,200 @@ static av_cold int atrac3_decode_close(AVCodecContext *avctx)
return 0;
}
/**
/ * Mantissa decoding
/*
* Mantissa decoding
*
* @param gb the GetBit context
* @param selector what table is the output values coded with
* @param codingFlag constant length coding or variable length coding
* @param mantissas mantissa output table
* @param numCodes amount of values to get
* @param selector which table the output values are coded with
* @param coding_flag constant length coding or variable length coding
* @param mantissas mantissa output table
* @param num_codes number of values to get
*/
static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
int coding_flag, int *mantissas,
int num_codes)
{
int numBits, cnt, code, huffSymb;
int i, code, huff_symb;
if (selector == 1)
numCodes /= 2;
num_codes /= 2;
if (codingFlag != 0) {
if (coding_flag != 0) {
/* constant length coding (CLC) */
numBits = CLCLengthTab[selector];
int num_bits = clc_length_tab[selector];
if (selector > 1) {
for (cnt = 0; cnt < numCodes; cnt++) {
if (numBits)
code = get_sbits(gb, numBits);
for (i = 0; i < num_codes; i++) {
if (num_bits)
code = get_sbits(gb, num_bits);
else
code = 0;
mantissas[cnt] = code;
mantissas[i] = code;
}
} else {
for (cnt = 0; cnt < numCodes; cnt++) {
if (numBits)
code = get_bits(gb, numBits); //numBits is always 4 in this case
for (i = 0; i < num_codes; i++) {
if (num_bits)
code = get_bits(gb, num_bits); // num_bits is always 4 in this case
else
code = 0;
mantissas[cnt*2] = seTab_0[code >> 2];
mantissas[cnt*2+1] = seTab_0[code & 3];
mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
}
}
} else {
/* variable length coding (VLC) */
if (selector != 1) {
for (cnt = 0; cnt < numCodes; cnt++) {
huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
huffSymb += 1;
code = huffSymb >> 1;
if (huffSymb & 1)
for (i = 0; i < num_codes; i++) {
huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
spectral_coeff_tab[selector-1].bits, 3);
huff_symb += 1;
code = huff_symb >> 1;
if (huff_symb & 1)
code = -code;
mantissas[cnt] = code;
mantissas[i] = code;
}
} else {
for (cnt = 0; cnt < numCodes; cnt++) {
huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
mantissas[cnt*2] = decTable1[huffSymb*2];
mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
for (i = 0; i < num_codes; i++) {
huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
spectral_coeff_tab[selector - 1].bits, 3);
mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
}
}
}
}
/**
/*
* Restore the quantized band spectrum coefficients
*
* @param gb the GetBit context
* @param pOut decoded band spectrum
* @return outSubbands subband counter, fix for broken specification/files
* @return subband count, fix for broken specification/files
*/
static int decodeSpectrum (GetBitContext *gb, float *pOut)
static int decode_spectrum(GetBitContext *gb, float *output)
{
int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
int subband_vlc_index[32], SF_idxs[32];
int mantissas[128];
float SF;
numSubbands = get_bits(gb, 5); // number of coded subbands
codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
/* Get the VLC selector table for the subbands, 0 means not coded. */
for (cnt = 0; cnt <= numSubbands; cnt++)
subband_vlc_index[cnt] = get_bits(gb, 3);
/* Read the scale factor indexes from the stream. */
for (cnt = 0; cnt <= numSubbands; cnt++) {
if (subband_vlc_index[cnt] != 0)
SF_idxs[cnt] = get_bits(gb, 6);
int num_subbands, coding_mode, i, j, first, last, subband_size;
int subband_vlc_index[32], sf_index[32];
int mantissas[128];
float scale_factor;
num_subbands = get_bits(gb, 5); // number of coded subbands
coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
/* get the VLC selector table for the subbands, 0 means not coded */
for (i = 0; i <= num_subbands; i++)
subband_vlc_index[i] = get_bits(gb, 3);
/* read the scale factor indexes from the stream */
for (i = 0; i <= num_subbands; i++) {
if (subband_vlc_index[i] != 0)
sf_index[i] = get_bits(gb, 6);
}
for (cnt = 0; cnt <= numSubbands; cnt++) {
first = subbandTab[cnt];
last = subbandTab[cnt+1];
for (i = 0; i <= num_subbands; i++) {
first = subband_tab[i ];
last = subband_tab[i + 1];
subbWidth = last - first;
subband_size = last - first;
if (subband_vlc_index[cnt] != 0) {
/* Decode spectral coefficients for this subband. */
if (subband_vlc_index[i] != 0) {
/* decode spectral coefficients for this subband */
/* TODO: This can be done faster is several blocks share the
* same VLC selector (subband_vlc_index) */
readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
mantissas, subband_size);
/* Decode the scale factor for this subband. */
SF = ff_atrac_sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
/* decode the scale factor for this subband */
scale_factor = ff_atrac_sf_table[sf_index[i]] *
inv_max_quant[subband_vlc_index[i]];
/* Inverse quantize the coefficients. */
for (pIn=mantissas ; first<last; first++, pIn++)
pOut[first] = *pIn * SF;
/* inverse quantize the coefficients */
for (j = 0; first < last; first++, j++)
output[first] = mantissas[j] * scale_factor;
} else {
/* This subband was not coded, so zero the entire subband. */
memset(pOut+first, 0, subbWidth*sizeof(float));
/* this subband was not coded, so zero the entire subband */
memset(output + first, 0, subband_size * sizeof(float));
}
}
/* Clear the subbands that were not coded. */
first = subbandTab[cnt];
memset(pOut+first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float));
return numSubbands;
/* clear the subbands that were not coded */
first = subband_tab[i];
memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float));
return num_subbands;
}
/**
/*
* Restore the quantized tonal components
*
* @param gb the GetBit context
* @param pComponent tone component
* @param numBands amount of coded bands
* @param components tonal components
* @param num_bands number of coded bands
*/
static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
static int decode_tonal_components(GetBitContext *gb,
TonalComponent *components, int num_bands)
{
int i,j,k,cnt;
int components, coding_mode_selector, coding_mode, coded_values_per_component;
int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
int band_flags[4], mantissa[8];
float *pCoef;
float scalefactor;
int component_count = 0;
int i, b, c, m;
int nb_components, coding_mode_selector, coding_mode;
int band_flags[4], mantissa[8];
int component_count = 0;
components = get_bits(gb,5);
nb_components = get_bits(gb, 5);
/* no tonal components */
if (components == 0)
if (nb_components == 0)
return 0;
coding_mode_selector = get_bits(gb,2);
coding_mode_selector = get_bits(gb, 2);
if (coding_mode_selector == 2)
return AVERROR_INVALIDDATA;
coding_mode = coding_mode_selector & 1;
for (i = 0; i < components; i++) {
for (cnt = 0; cnt <= numBands; cnt++)
band_flags[cnt] = get_bits1(gb);
for (i = 0; i < nb_components; i++) {
int coded_values_per_component, quant_step_index;
for (b = 0; b <= num_bands; b++)
band_flags[b] = get_bits1(gb);
coded_values_per_component = get_bits(gb,3);
coded_values_per_component = get_bits(gb, 3);
quant_step_index = get_bits(gb,3);
quant_step_index = get_bits(gb, 3);
if (quant_step_index <= 1)
return AVERROR_INVALIDDATA;
if (coding_mode_selector == 3)
coding_mode = get_bits1(gb);
for (j = 0; j < (numBands + 1) * 4; j++) {
if (band_flags[j >> 2] == 0)
for (b = 0; b < (num_bands + 1) * 4; b++) {
int coded_components;
if (band_flags[b >> 2] == 0)
continue;
coded_components = get_bits(gb,3);
coded_components = get_bits(gb, 3);
for (c = 0; c < coded_components; c++) {
TonalComponent *cmp = &components[component_count];
int sf_index, coded_values, max_coded_values;
float scale_factor;
for (k=0; k<coded_components; k++) {
sfIndx = get_bits(gb,6);
sf_index = get_bits(gb, 6);
if (component_count >= 64)
return AVERROR_INVALIDDATA;
pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
coded_values = coded_values_per_component + 1;
coded_values = FFMIN(max_coded_values,coded_values);
scalefactor = ff_atrac_sf_table[sfIndx] * iMaxQuant[quant_step_index];
cmp->pos = b * 64 + get_bits(gb, 6);
max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
coded_values = coded_values_per_component + 1;
coded_values = FFMIN(max_coded_values, coded_values);
readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
scale_factor = ff_atrac_sf_table[sf_index] *
inv_max_quant[quant_step_index];
pComponent[component_count].numCoefs = coded_values;
read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
mantissa, coded_values);
cmp->num_coefs = coded_values;
/* inverse quant */
pCoef = pComponent[component_count].coef;
for (cnt = 0; cnt < coded_values; cnt++)
pCoef[cnt] = mantissa[cnt] * scalefactor;
for (m = 0; m < coded_values; m++)
cmp->coef[m] = mantissa[m] * scale_factor;
component_count++;
}
......@@ -426,334 +421,326 @@ static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent
return component_count;
}
/**
/*
* Decode gain parameters for the coded bands
*
* @param gb the GetBit context
* @param pGb the gainblock for the current band
* @param numBands amount of coded bands
* @param block the gainblock for the current band
* @param num_bands amount of coded bands
*/
static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
static int decode_gain_control(GetBitContext *gb, GainBlock *block,
int num_bands)
{
int i, cf, numData;
int *pLevel, *pLoc;
gain_info *pGain = pGb->gBlock;
for (i=0 ; i<=numBands; i++)
{
numData = get_bits(gb,3);
pGain[i].num_gain_data = numData;
pLevel = pGain[i].levcode;
pLoc = pGain[i].loccode;
for (cf = 0; cf < numData; cf++){
pLevel[cf]= get_bits(gb,4);
pLoc [cf]= get_bits(gb,5);
if(cf && pLoc[cf] <= pLoc[cf-1])
int i, cf, num_data;
int *level, *loc;
GainInfo *gain = block->g_block;
for (i = 0; i <= num_bands; i++) {
num_data = get_bits(gb, 3);
gain[i].num_gain_data = num_data;
level = gain[i].lev_code;
loc = gain[i].loc_code;
for (cf = 0; cf < gain[i].num_gain_data; cf++) {
level[cf] = get_bits(gb, 4);
loc [cf] = get_bits(gb, 5);
if (cf && loc[cf] <= loc[cf - 1])
return AVERROR_INVALIDDATA;
}
}
/* Clear the unused blocks. */
for (; i<4 ; i++)
pGain[i].num_gain_data = 0;
for (; i < 4 ; i++)
gain[i].num_gain_data = 0;
return 0;
}
/**
/*
* Apply gain parameters and perform the MDCT overlapping part
*
* @param pIn input float buffer
* @param pPrev previous float buffer to perform overlap against
* @param pOut output float buffer
* @param pGain1 current band gain info
* @param pGain2 next band gain info
* @param input input buffer
* @param prev previous buffer to perform overlap against
* @param output output buffer
* @param gain1 current band gain info
* @param gain2 next band gain info
*/
static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
static void gain_compensate_and_overlap(float *input, float *prev,
float *output, GainInfo *gain1,
GainInfo *gain2)
{
/* gain compensation function */
float gain1, gain2, gain_inc;
int cnt, numdata, nsample, startLoc, endLoc;
float g1, g2, gain_inc;
int i, j, num_data, start_loc, end_loc;
if (pGain2->num_gain_data == 0)
gain1 = 1.0;
if (gain2->num_gain_data == 0)
g1 = 1.0;
else
gain1 = gain_tab1[pGain2->levcode[0]];
g1 = gain_tab1[gain2->lev_code[0]];
if (pGain1->num_gain_data == 0) {
for (cnt = 0; cnt < 256; cnt++)
pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
if (gain1->num_gain_data == 0) {
for (i = 0; i < 256; i++)
output[i] = input[i] * g1 + prev[i];
} else {
numdata = pGain1->num_gain_data;
pGain1->loccode[numdata] = 32;
pGain1->levcode[numdata] = 4;
nsample = 0; // current sample = 0
num_data = gain1->num_gain_data;
gain1->loc_code[num_data] = 32;
gain1->lev_code[num_data] = 4;
for (cnt = 0; cnt < numdata; cnt++) {
startLoc = pGain1->loccode[cnt] * 8;
endLoc = startLoc + 8;
for (i = 0, j = 0; i < num_data; i++) {
start_loc = gain1->loc_code[i] * 8;
end_loc = start_loc + 8;
gain2 = gain_tab1[pGain1->levcode[cnt]];
gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
g2 = gain_tab1[gain1->lev_code[i]];
gain_inc = gain_tab2[gain1->lev_code[i + 1] -
gain1->lev_code[i ] + 15];
/* interpolate */
for (; nsample < startLoc; nsample++)
pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
for (; j < start_loc; j++)
output[j] = (input[j] * g1 + prev[j]) * g2;
/* interpolation is done over eight samples */
for (; nsample < endLoc; nsample++) {
pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
gain2 *= gain_inc;
for (; j < end_loc; j++) {
output[j] = (input[j] * g1 + prev[j]) * g2;
g2 *= gain_inc;
}
}
for (; nsample < 256; nsample++)
pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
for (; j < 256; j++)
output[j] = input[j] * g1 + prev[j];
}
/* Delay for the overlapping part. */
memcpy(pPrev, &pIn[256], 256*sizeof(float));
memcpy(prev, &input[256], 256 * sizeof(float));
}
/**
/*
* Combine the tonal band spectrum and regular band spectrum
* Return position of the last tonal coefficient
*
* @param pSpectrum output spectrum buffer
* @param numComponents amount of tonal components
* @param pComponent tonal components for this band
* @param spectrum output spectrum buffer
* @param num_components number of tonal components
* @param components tonal components for this band
* @return position of the last tonal coefficient
*/
static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
static int add_tonal_components(float *spectrum, int num_components,
TonalComponent *components)
{
int cnt, i, lastPos = -1;
float *pIn, *pOut;
int i, j, last_pos = -1;
float *input, *output;
for (cnt = 0; cnt < numComponents; cnt++){
lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
pIn = pComponent[cnt].coef;
pOut = &(pSpectrum[pComponent[cnt].pos]);
for (i = 0; i < num_components; i++) {
last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
input = components[i].coef;
output = &spectrum[components[i].pos];
for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
pOut[i] += pIn[i];
for (j = 0; j < components[i].num_coefs; j++)
output[i] += input[i];
}
return lastPos;
return last_pos;
}
#define INTERPOLATE(old, new, nsample) \
((old) + (nsample) * 0.125 * ((new) - (old)))
#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
static void reverse_matrixing(float *su1, float *su2, int *prev_code,
int *curr_code)
{
int i, band, nsample, s1, s2;
float c1, c2;
float mc1_l, mc1_r, mc2_l, mc2_r;
int i, nsample, band;
float mc1_l, mc1_r, mc2_l, mc2_r;
for (i=0,band = 0; band < 4*256; band+=256,i++) {
s1 = pPrevCode[i];
s2 = pCurrCode[i];
for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
int s1 = prev_code[i];
int s2 = curr_code[i];
nsample = 0;
if (s1 != s2) {
/* Selector value changed, interpolation needed. */
mc1_l = matrixCoeffs[s1*2];
mc1_r = matrixCoeffs[s1*2+1];
mc2_l = matrixCoeffs[s2*2];
mc2_r = matrixCoeffs[s2*2+1];
mc1_l = matrix_coeffs[s1 * 2 ];
mc1_r = matrix_coeffs[s1 * 2 + 1];
mc2_l = matrix_coeffs[s2 * 2 ];
mc2_r = matrix_coeffs[s2 * 2 + 1];
/* Interpolation is done over the first eight samples. */
for(; nsample < 8; nsample++) {
c1 = su1[band+nsample];
c2 = su2[band+nsample];
c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
su1[band+nsample] = c2;
su2[band+nsample] = c1 * 2.0 - c2;
for (; nsample < 8; nsample++) {
float c1 = su1[band + nsample];
float c2 = su2[band + nsample];
c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample) +
c2 * INTERPOLATE(mc1_r, mc2_r, nsample);
su1[band + nsample] = c2;
su2[band + nsample] = c1 * 2.0 - c2;
}
}
/* Apply the matrix without interpolation. */
switch (s2) {
case 0: /* M/S decoding */
for (; nsample < 256; nsample++) {
c1 = su1[band+nsample];
c2 = su2[band+nsample];
su1[band+nsample] = c2 * 2.0;
su2[band+nsample] = (c1 - c2) * 2.0;
}
break;
case 1:
for (; nsample < 256; nsample++) {
c1 = su1[band+nsample];
c2 = su2[band+nsample];
su1[band+nsample] = (c1 + c2) * 2.0;
su2[band+nsample] = c2 * -2.0;
}
break;
case 2:
case 3:
for (; nsample < 256; nsample++) {
c1 = su1[band+nsample];
c2 = su2[band+nsample];
su1[band+nsample] = c1 + c2;
su2[band+nsample] = c1 - c2;
}
break;
default:
assert(0);
case 0: /* M/S decoding */
for (; nsample < 256; nsample++) {
float c1 = su1[band + nsample];
float c2 = su2[band + nsample];
su1[band + nsample] = c2 * 2.0;
su2[band + nsample] = (c1 - c2) * 2.0;
}
break;
case 1:
for (; nsample < 256; nsample++) {
float c1 = su1[band + nsample];
float c2 = su2[band + nsample];
su1[band + nsample] = (c1 + c2) * 2.0;
su2[band + nsample] = c2 * -2.0;
}
break;
case 2:
case 3:
for (; nsample < 256; nsample++) {
float c1 = su1[band + nsample];
float c2 = su2[band + nsample];
su1[band + nsample] = c1 + c2;
su2[band + nsample] = c1 - c2;
}
break;
default:
assert(0);
}
}
}
static void getChannelWeights (int indx, int flag, float ch[2]){
if (indx == 7) {
static void get_channel_weights(int index, int flag, float ch[2])
{
if (index == 7) {
ch[0] = 1.0;
ch[1] = 1.0;
} else {
ch[0] = (float)(indx & 7) / 7.0;
ch[1] = sqrt(2 - ch[0]*ch[0]);
if(flag)
ch[0] = (index & 7) / 7.0;
ch[1] = sqrt(2 - ch[0] * ch[0]);
if (flag)
FFSWAP(float, ch[0], ch[1]);
}
}
static void channelWeighting (float *su1, float *su2, int *p3)
static void channel_weighting(float *su1, float *su2, int *p3)
{
int band, nsample;
int band, nsample;
/* w[x][y] y=0 is left y=1 is right */
float w[2][2];
if (p3[1] != 7 || p3[3] != 7){
getChannelWeights(p3[1], p3[0], w[0]);
getChannelWeights(p3[3], p3[2], w[1]);
if (p3[1] != 7 || p3[3] != 7) {
get_channel_weights(p3[1], p3[0], w[0]);
get_channel_weights(p3[3], p3[2], w[1]);
for(band = 1; band < 4; band++) {
/* scale the channels by the weights */
for(nsample = 0; nsample < 8; nsample++) {
su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
for (band = 1; band < 4; band++) {
for (nsample = 0; nsample < 8; nsample++) {
su1[band * 256 + nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
su2[band * 256 + nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
}
for(; nsample < 256; nsample++) {
su1[band*256+nsample] *= w[1][0];
su2[band*256+nsample] *= w[1][1];
su1[band * 256 + nsample] *= w[1][0];
su2[band * 256 + nsample] *= w[1][1];
}
}
}
}
/**
/*
* Decode a Sound Unit
*
* @param gb the GetBit context
* @param pSnd the channel unit to be used
* @param pOut the decoded samples before IQMF in float representation
* @param channelNum channel number
* @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono)
* @param snd the channel unit to be used
* @param output the decoded samples before IQMF in float representation
* @param channel_num channel number
* @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
*/
static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
ChannelUnit *snd, float *output,
int channel_num, int coding_mode)
{
int band, result=0, numSubbands, lastTonal, numBands;
int band, ret, num_subbands, last_tonal, num_bands;
GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
if (codingMode == JOINT_STEREO && channelNum == 1) {
if (get_bits(gb,2) != 3) {
if (coding_mode == JOINT_STEREO && channel_num == 1) {
if (get_bits(gb, 2) != 3) {
av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
return AVERROR_INVALIDDATA;
}
} else {
if (get_bits(gb,6) != 0x28) {
if (get_bits(gb, 6) != 0x28) {
av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
return AVERROR_INVALIDDATA;
}
}
/* number of coded QMF bands */
pSnd->bandsCoded = get_bits(gb,2);
snd->bands_coded = get_bits(gb, 2);
result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
if (result) return result;
ret = decode_gain_control(gb, gain2, snd->bands_coded);
if (ret)
return ret;
pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
if (pSnd->numComponents == -1) return -1;
snd->num_components = decode_tonal_components(gb, snd->components,
snd->bands_coded);
if (snd->num_components == -1)
return -1;
numSubbands = decodeSpectrum (gb, pSnd->spectrum);
num_subbands = decode_spectrum(gb, snd->spectrum);
/* Merge the decoded spectrum and tonal components. */
lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
snd->components);
/* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
numBands = (subbandTab[numSubbands] - 1) >> 8;
if (lastTonal >= 0)
numBands = FFMAX((lastTonal + 256) >> 8, numBands);
/* calculate number of used MLT/QMF bands according to the amount of coded
spectral lines */
num_bands = (subband_tab[num_subbands] - 1) >> 8;
if (last_tonal >= 0)
num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
/* Reconstruct time domain samples. */
for (band=0; band<4; band++) {
for (band = 0; band < 4; band++) {
/* Perform the IMDCT step without overlapping. */
if (band <= numBands) {
IMLT(q, &(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
} else
memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
if (band <= num_bands)
imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
else
memset(snd->imdct_buf, 0, 512 * sizeof(float));
/* gain compensation and overlapping */
gainCompensateAndOverlap(pSnd->IMDCT_buf, &pSnd->prevFrame[band * 256],
&pOut[band * 256],
&pSnd->gainBlock[1 - pSnd->gcBlkSwitch].gBlock[band],
&pSnd->gainBlock[ pSnd->gcBlkSwitch].gBlock[band]);
gain_compensate_and_overlap(snd->imdct_buf,
&snd->prev_frame[band * 256],
&output[band * 256],
&gain1->g_block[band],
&gain2->g_block[band]);
}
/* Swap the gain control buffers for the next frame. */
pSnd->gcBlkSwitch ^= 1;
snd->gc_blk_switch ^= 1;
return 0;
}
/**
* Frame handling
*
* @param q Atrac3 private context
* @param databuf the input data
*/
static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
float **out_samples)
static int decode_frame(ATRAC3Context *q, const uint8_t *databuf,
float **out_samples)
{
int result, i;
float *p1, *p2, *p3, *p4;
int ret, i;
uint8_t *ptr1;
if (q->codingMode == JOINT_STEREO) {
if (q->coding_mode == JOINT_STEREO) {
/* channel coupling mode */
/* decode Sound Unit 1 */
init_get_bits(&q->gb,databuf,q->bits_per_frame);
result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
if (result != 0)
return result;
ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
JOINT_STEREO);
if (ret != 0)
return ret;
/* Framedata of the su2 in the joint-stereo mode is encoded in
* reverse byte order so we need to swap it first. */
if (databuf == q->decoded_bytes_buffer) {
uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
ptr1 = q->decoded_bytes_buffer;
for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
FFSWAP(uint8_t,*ptr1,*ptr2);
}
uint8_t *ptr2 = q->decoded_bytes_buffer + q->bytes_per_frame - 1;
ptr1 = q->decoded_bytes_buffer;
for (i = 0; i < q->bytes_per_frame / 2; i++, ptr1++, ptr2--)
FFSWAP(uint8_t, *ptr1, *ptr2);
} else {
const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
const uint8_t *ptr2 = databuf + q->bytes_per_frame - 1;
for (i = 0; i < q->bytes_per_frame; i++)
q->decoded_bytes_buffer[i] = *ptr2--;
}
......@@ -767,74 +754,69 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
/* set the bitstream reader at the start of the second Sound Unit*/
init_get_bits(&q->gb,ptr1,q->bits_per_frame);
init_get_bits(&q->gb, ptr1, q->bits_per_frame);
/* Fill the Weighting coeffs delay buffer */
memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
memmove(q->weighting_delay, &q->weighting_delay[2], 4 * sizeof(int));
q->weighting_delay[4] = get_bits1(&q->gb);
q->weighting_delay[5] = get_bits(&q->gb,3);
q->weighting_delay[5] = get_bits(&q->gb, 3);
for (i = 0; i < 4; i++) {
q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
}
/* Decode Sound Unit 2. */
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
if (result != 0)
return result;
ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
out_samples[1], 1, JOINT_STEREO);
if (ret != 0)
return ret;
/* Reconstruct the channel coefficients. */
reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
channelWeighting(out_samples[0], out_samples[1], q->weighting_delay);
reverse_matrixing(out_samples[0], out_samples[1],
q->matrix_coeff_index_prev,
q->matrix_coeff_index_now);
channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
} else {
/* normal stereo mode or mono */
/* Decode the channel sound units. */
for (i=0 ; i<q->channels ; i++) {
for (i = 0; i < q->channels; i++) {
/* Set the bitstream reader at the start of a channel sound unit. */
init_get_bits(&q->gb,
databuf + i * q->bytes_per_frame / q->channels,
q->bits_per_frame / q->channels);
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
if (result != 0)
return result;
ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
out_samples[i], i, q->coding_mode);
if (ret != 0)
return ret;
}
}
/* Apply the iQMF synthesis filter. */
for (i=0 ; i<q->channels ; i++) {
p1 = out_samples[i];
p2= p1+256;
p3= p2+256;
p4= p3+256;
ff_atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
ff_atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
ff_atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
for (i = 0; i < q->channels; i++) {
float *p1 = out_samples[i];
float *p2 = p1 + 256;
float *p3 = p2 + 256;
float *p4 = p3 + 256;
ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
}
return 0;
}
/**
* Atrac frame decoding
*
* @param avctx pointer to the AVCodecContext
*/
static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ATRAC3Context *q = avctx->priv_data;
int result;
const uint8_t* databuf;
int ret;
const uint8_t *databuf;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
......@@ -844,9 +826,9 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
/* get output buffer */
q->frame.nb_samples = SAMPLES_PER_FRAME;
if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) {
if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return result;
return ret;
}
/* Check if we need to descramble and what buffer to pass on. */
......@@ -857,11 +839,10 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
databuf = buf;
}
result = decodeFrame(q, databuf, (float **)q->frame.extended_data);
if (result != 0) {
av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
return result;
ret = decode_frame(q, databuf, (float **)q->frame.extended_data);
if (ret) {
av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
return ret;
}
*got_frame_ptr = 1;
......@@ -870,13 +851,6 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
return avctx->block_align;
}
/**
* Atrac3 initialization
*
* @param avctx pointer to the AVCodecContext
*/
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
int i, ret;
......@@ -886,101 +860,108 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
static int vlcs_initialized = 0;
/* Take data from the AVCodecContext (RM container). */
q->sample_rate = avctx->sample_rate;
q->channels = avctx->channels;
q->bit_rate = avctx->bit_rate;
q->bits_per_frame = avctx->block_align * 8;
q->sample_rate = avctx->sample_rate;
q->channels = avctx->channels;
q->bit_rate = avctx->bit_rate;
q->bits_per_frame = avctx->block_align * 8;
q->bytes_per_frame = avctx->block_align;
/* Take care of the codec-specific extradata. */
if (avctx->extradata_size == 14) {
/* Parse the extradata, WAV format */
av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1
av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
bytestream_get_le16(&edata_ptr)); // Unknown value always 1
q->samples_per_channel = bytestream_get_le32(&edata_ptr);
q->codingMode = bytestream_get_le16(&edata_ptr);
av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1
av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0
q->coding_mode = bytestream_get_le16(&edata_ptr);
av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
q->frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
bytestream_get_le16(&edata_ptr)); // Unknown always 0
/* setup */
q->samples_per_frame = SAMPLES_PER_FRAME * q->channels;
q->atrac3version = 4;
q->delay = 0x88E;
if (q->codingMode)
q->codingMode = JOINT_STEREO;
else
q->codingMode = STEREO;
q->scrambled_stream = 0;
if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) {
} else {
av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor);
q->version = 4;
q->delay = 0x88E;
q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO;
q->scrambled_stream = 0;
if (q->bytes_per_frame != 96 * q->channels * q->frame_factor &&
q->bytes_per_frame != 152 * q->channels * q->frame_factor &&
q->bytes_per_frame != 192 * q->channels * q->frame_factor) {
av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
"configuration %d/%d/%d\n", q->bytes_per_frame, q->channels,
q->frame_factor);
return AVERROR_INVALIDDATA;
}
} else if (avctx->extradata_size == 10) {
/* Parse the extradata, RM format. */
q->atrac3version = bytestream_get_be32(&edata_ptr);
q->samples_per_frame = bytestream_get_be16(&edata_ptr);
q->delay = bytestream_get_be16(&edata_ptr);
q->codingMode = bytestream_get_be16(&edata_ptr);
q->version = bytestream_get_be32(&edata_ptr);
q->samples_per_frame = bytestream_get_be16(&edata_ptr);
q->delay = bytestream_get_be16(&edata_ptr);
q->coding_mode = bytestream_get_be16(&edata_ptr);
q->samples_per_channel = q->samples_per_frame / q->channels;
q->scrambled_stream = 1;
q->scrambled_stream = 1;
} else {
av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
avctx->extradata_size);
}
/* Check the extradata. */
if (q->atrac3version != 4) {
av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
/* Check the extradata */
if (q->version != 4) {
av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", q->version);
return AVERROR_INVALIDDATA;
}
if (q->samples_per_frame != SAMPLES_PER_FRAME && q->samples_per_frame != SAMPLES_PER_FRAME*2) {
av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
if (q->samples_per_frame != SAMPLES_PER_FRAME &&
q->samples_per_frame != SAMPLES_PER_FRAME * 2) {
av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
q->samples_per_frame);
return AVERROR_INVALIDDATA;
}
if (q->delay != 0x88E) {
av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
q->delay);
return AVERROR_INVALIDDATA;
}
if (q->codingMode == STEREO) {
av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
} else if (q->codingMode == JOINT_STEREO) {
av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
} else {
av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
if (q->coding_mode == STEREO)
av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
else if (q->coding_mode == JOINT_STEREO)
av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
else {
av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
q->coding_mode);
return AVERROR_INVALIDDATA;
}
if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
if (avctx->channels <= 0 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
return AVERROR(EINVAL);
}
if(avctx->block_align >= UINT_MAX/2)
if (avctx->block_align >= UINT_MAX / 2)
return AVERROR(EINVAL);
/* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
* this is for the bitstream reader. */
if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL)
q->decoded_bytes_buffer = av_mallocz(avctx->block_align +
(4 - avctx->block_align % 4) +
FF_INPUT_BUFFER_PADDING_SIZE);
if (q->decoded_bytes_buffer == NULL)
return AVERROR(ENOMEM);
/* Initialize the VLC tables. */
if (!vlcs_initialized) {
for (i=0 ; i<7 ; i++) {
for (i = 0; i < 7; i++) {
spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i];
init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
huff_bits[i], 1, 1,
huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
atrac3_vlc_offs[i ];
init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
huff_bits[i], 1, 1,
huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
}
vlcs_initialized = 1;
}
......@@ -995,12 +976,12 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
ff_atrac_generate_tables();
/* Generate gain tables. */
for (i=0 ; i<16 ; i++)
gain_tab1[i] = powf (2.0, (4 - i));
/* Generate gain tables */
for (i = 0; i < 16; i++)
gain_tab1[i] = powf(2.0, (4 - i));
for (i=-15 ; i<16 ; i++)
gain_tab2[i+15] = powf (2.0, i * -0.125);
for (i = -15; i < 16; i++)
gain_tab2[i + 15] = powf(2.0, i * -0.125);
/* init the joint-stereo decoding data */
q->weighting_delay[0] = 0;
......@@ -1010,17 +991,17 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
q->weighting_delay[4] = 0;
q->weighting_delay[5] = 7;
for (i=0; i<4; i++) {
for (i = 0; i < 4; i++) {
q->matrix_coeff_index_prev[i] = 3;
q->matrix_coeff_index_now[i] = 3;
q->matrix_coeff_index_now[i] = 3;
q->matrix_coeff_index_next[i] = 3;
}
avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
if (!q->pUnits) {
q->units = av_mallocz(sizeof(ChannelUnit) * q->channels);
if (!q->units) {
atrac3_decode_close(avctx);
return AVERROR(ENOMEM);
}
......@@ -1031,18 +1012,16 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
return 0;
}
AVCodec ff_atrac3_decoder =
{
.name = "atrac3",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ATRAC3,
.priv_data_size = sizeof(ATRAC3Context),
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
AVCodec ff_atrac3_decoder = {
.name = "atrac3",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ATRAC3,
.priv_data_size = sizeof(ATRAC3Context),
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
libavcodec/atrac3data.h
View file @ e55d5390
......@@ -33,101 +33,109 @@
/* VLC tables */
static const uint8_t huffcode1[9] = {
0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
0x0, 0x4, 0x5, 0xC, 0xD, 0x1C, 0x1D, 0x1E, 0x1F
};
static const uint8_t huffbits1[9] = {
1,3,3,4,4,5,5,5,5,
};
static const uint8_t huffbits1[9] = { 1, 3, 3, 4, 4, 5, 5, 5, 5 };
static const uint8_t huffcode2[5] = {
0x0,0x4,0x5,0x6,0x7,
};
static const uint8_t huffcode2[5] = { 0x0, 0x4, 0x5, 0x6, 0x7 };
static const uint8_t huffbits2[5] = {
1,3,3,3,3,
};
static const uint8_t huffbits2[5] = { 1, 3, 3, 3, 3 };
static const uint8_t huffcode3[7] = {
0x0,0x4,0x5,0xC,0xD,0xE,0xF,
};
static const uint8_t huffcode3[7] = { 0x0, 0x4, 0x5, 0xC, 0xD, 0xE, 0xF };
static const uint8_t huffbits3[7] = {
1,3,3,4,4,4,4,
};
static const uint8_t huffbits3[7] = { 1, 3, 3, 4, 4, 4, 4 };
static const uint8_t huffcode4[9] = {
0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
0x0, 0x4, 0x5, 0xC, 0xD, 0x1C, 0x1D, 0x1E, 0x1F
};
static const uint8_t huffbits4[9] = {
1,3,3,4,4,5,5,5,5,
};
static const uint8_t huffbits4[9] = { 1, 3, 3, 4, 4, 5, 5, 5, 5 };
static const uint8_t huffcode5[15] = {
0x0,0x2,0x3,0x8,0x9,0xA,0xB,0x1C,0x1D,0x3C,0x3D,0x3E,0x3F,0xC,0xD,
0x00, 0x02, 0x03, 0x08, 0x09, 0x0A, 0x0B, 0x1C,
0x1D, 0x3C, 0x3D, 0x3E, 0x3F, 0x0C, 0x0D
};
static const uint8_t huffbits5[15] = {
2,3,3,4,4,4,4,5,5,6,6,6,6,4,4
2, 3, 3, 4, 4, 4, 4, 5, 5, 6, 6, 6, 6, 4, 4
};
static const uint8_t huffcode6[31] = {
0x0,0x2,0x3,0x4,0x5,0x6,0x7,0x14,0x15,0x16,0x17,0x18,0x19,0x34,0x35,
0x36,0x37,0x38,0x39,0x3A,0x3B,0x78,0x79,0x7A,0x7B,0x7C,0x7D,0x7E,0x7F,0x8,0x9,
0x00, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x14,
0x15, 0x16, 0x17, 0x18, 0x19, 0x34, 0x35, 0x36,
0x37, 0x38, 0x39, 0x3A, 0x3B, 0x78, 0x79, 0x7A,
0x7B, 0x7C, 0x7D, 0x7E, 0x7F, 0x08, 0x09
};
static const uint8_t huffbits6[31] = {
3,4,4,4,4,4,4,5,5,5,5,5,5,6,6,6,6,6,6,6,6,7,7,7,7,7,7,7,7,4,4
3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 6, 6, 6,
6, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 7, 4, 4
};
static const uint8_t huffcode7[63] = {
0x0,0x8,0x9,0xA,0xB,0xC,0xD,0xE,0xF,0x10,0x11,0x24,0x25,0x26,0x27,0x28,
0x29,0x2A,0x2B,0x2C,0x2D,0x2E,0x2F,0x30,0x31,0x32,0x33,0x68,0x69,0x6A,0x6B,0x6C,
0x6D,0x6E,0x6F,0x70,0x71,0x72,0x73,0x74,0x75,0xEC,0xED,0xEE,0xEF,0xF0,0xF1,0xF2,
0xF3,0xF4,0xF5,0xF6,0xF7,0xF8,0xF9,0xFA,0xFB,0xFC,0xFD,0xFE,0xFF,0x2,0x3,
0x00, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
0x0F, 0x10, 0x11, 0x24, 0x25, 0x26, 0x27, 0x28,
0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F, 0x30,
0x31, 0x32, 0x33, 0x68, 0x69, 0x6A, 0x6B, 0x6C,
0x6D, 0x6E, 0x6F, 0x70, 0x71, 0x72, 0x73, 0x74,
0x75, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2,
0xF3, 0xF4, 0xF5, 0xF6, 0xF7, 0xF8, 0xF9, 0xFA,
0xFB, 0xFC, 0xFD, 0xFE, 0xFF, 0x02, 0x03
};
static const uint8_t huffbits7[63] = {
3,5,5,5,5,5,5,5,5,5,5,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,4,4
3, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 8,
8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 4, 4
};
static const uint8_t huff_tab_sizes[7] = {
9, 5, 7, 9, 15, 31, 63,
9, 5, 7, 9, 15, 31, 63,
};
static const uint8_t* const huff_codes[7] = {
huffcode1,huffcode2,huffcode3,huffcode4,huffcode5,huffcode6,huffcode7,
huffcode1, huffcode2, huffcode3, huffcode4, huffcode5, huffcode6, huffcode7
};
static const uint8_t* const huff_bits[7] = {
huffbits1,huffbits2,huffbits3,huffbits4,huffbits5,huffbits6,huffbits7,
huffbits1, huffbits2, huffbits3, huffbits4, huffbits5, huffbits6, huffbits7,
};
static const uint16_t atrac3_vlc_offs[] = {
0,512,1024,1536,2048,2560,3072,3584,4096
static const uint16_t atrac3_vlc_offs[9] = {
0, 512, 1024, 1536, 2048, 2560, 3072, 3584, 4096
};
/* selector tables */
static const uint8_t CLCLengthTab[8] = {0, 4, 3, 3, 4, 4, 5, 6};
static const int8_t seTab_0[4] = {0, 1, -2, -1};
static const int8_t decTable1[18] = {0,0, 0,1, 0,-1, 1,0, -1,0, 1,1, 1,-1, -1,1, -1,-1};
static const uint8_t clc_length_tab[8] = { 0, 4, 3, 3, 4, 4, 5, 6 };
static const int8_t mantissa_clc_tab[4] = { 0, 1, -2, -1 };
static const int8_t mantissa_vlc_tab[18] = {
0, 0, 0, 1, 0, -1, 1, 0, -1, 0, 1, 1, 1, -1, -1, 1, -1, -1
};
/* tables for the scalefactor decoding */
static const float iMaxQuant[8] = {
0.0, 1.0/1.5, 1.0/2.5, 1.0/3.5, 1.0/4.5, 1.0/7.5, 1.0/15.5, 1.0/31.5
static const float inv_max_quant[8] = {
0.0, 1.0 / 1.5, 1.0 / 2.5, 1.0 / 3.5,
1.0 / 4.5, 1.0 / 7.5, 1.0 / 15.5, 1.0 / 31.5
};
static const uint16_t subbandTab[33] = {
0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224,
256, 288, 320, 352, 384, 416, 448, 480, 512, 576, 640, 704, 768, 896, 1024
static const uint16_t subband_tab[33] = {
0, 8, 16, 24, 32, 40, 48, 56,
64, 80, 96, 112, 128, 144, 160, 176,
192, 224, 256, 288, 320, 352, 384, 416,
448, 480, 512, 576, 640, 704, 768, 896,
1024
};
/* joint stereo related tables */
static const float matrixCoeffs[8] = {0.0, 2.0, 2.0, 2.0, 0.0, 0.0, 1.0, 1.0};
static const float matrix_coeffs[8] = {
0.0, 2.0, 2.0, 2.0, 0.0, 0.0, 1.0, 1.0
};
#endif /* AVCODEC_ATRAC3DATA_H */
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