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.
21028 Commits
37 Branches
400 Tags
464 MiB
 
 
 
 
Tag: Branch: Tree: 95ce961d88
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from '95ce961d88'
${ noResults }
FFmpeg/libavcodec/atrac3.c

1025 lines
32 KiB
Raw Blame History

/*
* Atrac 3 compatible decoder
* Copyright (c) 2006-2008 Maxim Poliakovski
* Copyright (c) 2006-2008 Benjamin Larsson
*
* 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 libavcodec/atrac3.c
* Atrac 3 compatible decoder.
* This decoder handles Sony's ATRAC3 data.
*
* Container formats used to store atrac 3 data:
* RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
*
* To use this decoder, a calling application must supply the extradata
* bytes provided in the containers above.
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "bytestream.h"
#include "atrac.h"
#include "atrac3data.h"
#define JOINT_STEREO 0x12
#define STEREO 0x2
/* 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[1024];
int gcBlkSwitch;
gain_block gainBlock[2];
DECLARE_ALIGNED_16(float, spectrum[1024]);
DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);
float delayBuf1[46]; ///<qmf delay buffers
float delayBuf2[46];
float delayBuf3[46];
} channel_unit;
typedef struct {
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;
//@}
//@{
/** 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];
//@}
//@{
/** data buffers */
float outSamples[2048];
uint8_t* decoded_bytes_buffer;
float tempBuf[1070];
//@}
//@{
/** extradata */
int atrac3version;
int delay;
int scrambled_stream;
int frame_factor;
//@}
} ATRAC3Context;
static DECLARE_ALIGNED_16(float,mdct_window[512]);
static VLC spectral_coeff_tab[7];
static float gain_tab1[16];
static float gain_tab2[31];
static FFTContext mdct_ctx;
static DSPContext dsp;
/**
* 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(float *pInput, float *pOutput, int odd_band)
{
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 ff_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]);
}
ff_imdct_calc(&mdct_ctx,pOutput,pInput);
/* Perform windowing on the output. */
dsp.vector_fmul(pOutput,mdct_window,512);
}
/**
* Atrac 3 indata descrambling, only used for data coming from the rm container
*
* @param in pointer to 8 bit array of indata
* @param bits amount of bits
* @param out pointer to 8 bit array of outdata
*/
static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
int i, off;
uint32_t c;
const uint32_t* buf;
uint32_t* obuf = (uint32_t*) out;
off = (intptr_t)inbuffer & 3;
buf = (const uint32_t*) (inbuffer - off);
c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
bytes += 3 + off;
for (i = 0; i < bytes/4; i++)
obuf[i] = c ^ buf[i];
if (off)
av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
return off;
}
static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
float enc_window[256];
int i;
/* Generate the mdct window, for details see
* http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
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];
}
/* Initialize the MDCT transform. */
ff_mdct_init(&mdct_ctx, 9, 1, 1.0);
}
/**
* 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->decoded_bytes_buffer);
return 0;
}
/**
/ * 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
*/
static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
{
int numBits, cnt, code, huffSymb;
if (selector == 1)
numCodes /= 2;
if (codingFlag != 0) {
/* constant length coding (CLC) */
numBits = CLCLengthTab[selector];
if (selector > 1) {
for (cnt = 0; cnt < numCodes; cnt++) {
if (numBits)
code = get_sbits(gb, numBits);
else
code = 0;
mantissas[cnt] = code;
}
} else {
for (cnt = 0; cnt < numCodes; cnt++) {
if (numBits)
code = get_bits(gb, numBits); //numBits is always 4 in this case
else
code = 0;
mantissas[cnt*2] = seTab_0[code >> 2];
mantissas[cnt*2+1] = seTab_0[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)
code = -code;
mantissas[cnt] = 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];
}
}
}
}
/**
* 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
*/
static int decodeSpectrum (GetBitContext *gb, float *pOut)
{
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);
}
for (cnt = 0; cnt <= numSubbands; cnt++) {
first = subbandTab[cnt];
last = subbandTab[cnt+1];
subbWidth = last - first;
if (subband_vlc_index[cnt] != 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);
/* Decode the scale factor for this subband. */
SF = sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
/* Inverse quantize the coefficients. */
for (pIn=mantissas ; first<last; first++, pIn++)
pOut[first] = *pIn * SF;
} else {
/* This subband was not coded, so zero the entire subband. */
memset(pOut+first, 0, subbWidth*sizeof(float));
}
}
/* Clear the subbands that were not coded. */
first = subbandTab[cnt];
memset(pOut+first, 0, (1024 - first) * sizeof(float));
return numSubbands;
}
/**
* Restore the quantized tonal components
*
* @param gb the GetBit context
* @param pComponent tone component
* @param numBands amount of coded bands
*/
static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
{
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;
components = get_bits(gb,5);
/* no tonal components */
if (components == 0)
return 0;
coding_mode_selector = get_bits(gb,2);
if (coding_mode_selector == 2)
return -1;
coding_mode = coding_mode_selector & 1;
for (i = 0; i < components; i++) {
for (cnt = 0; cnt <= numBands; cnt++)
band_flags[cnt] = get_bits1(gb);
coded_values_per_component = get_bits(gb,3);
quant_step_index = get_bits(gb,3);
if (quant_step_index <= 1)
return -1;
if (coding_mode_selector == 3)
coding_mode = get_bits1(gb);
for (j = 0; j < (numBands + 1) * 4; j++) {
if (band_flags[j >> 2] == 0)
continue;
coded_components = get_bits(gb,3);
for (k=0; k<coded_components; k++) {
sfIndx = get_bits(gb,6);
pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
max_coded_values = 1024 - pComponent[component_count].pos;
coded_values = coded_values_per_component + 1;
coded_values = FFMIN(max_coded_values,coded_values);
scalefactor = sf_table[sfIndx] * iMaxQuant[quant_step_index];
readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
pComponent[component_count].numCoefs = coded_values;
/* inverse quant */
pCoef = pComponent[component_count].coef;
for (cnt = 0; cnt < coded_values; cnt++)
pCoef[cnt] = mantissa[cnt] * scalefactor;
component_count++;
}
}
}
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
*/
static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
{
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])
return -1;
}
}
/* Clear the unused blocks. */
for (; i<4 ; i++)
pGain[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
*/
static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
{
/* gain compensation function */
float gain1, gain2, gain_inc;
int cnt, numdata, nsample, startLoc, endLoc;
if (pGain2->num_gain_data == 0)
gain1 = 1.0;
else
gain1 = gain_tab1[pGain2->levcode[0]];
if (pGain1->num_gain_data == 0) {
for (cnt = 0; cnt < 256; cnt++)
pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
} else {
numdata = pGain1->num_gain_data;
pGain1->loccode[numdata] = 32;
pGain1->levcode[numdata] = 4;
nsample = 0; // current sample = 0
for (cnt = 0; cnt < numdata; cnt++) {
startLoc = pGain1->loccode[cnt] * 8;
endLoc = startLoc + 8;
gain2 = gain_tab1[pGain1->levcode[cnt]];
gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
/* interpolate */
for (; nsample < startLoc; nsample++)
pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
/* interpolation is done over eight samples */
for (; nsample < endLoc; nsample++) {
pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
gain2 *= gain_inc;
}
}
for (; nsample < 256; nsample++)
pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
}
/* Delay for the overlapping part. */
memcpy(pPrev, &pIn[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
*/
static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
{
int cnt, i, lastPos = -1;
float *pIn, *pOut;
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<pComponent[cnt].numCoefs ; i++)
pOut[i] += pIn[i];
}
return lastPos;
}
#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
{
int i, band, nsample, s1, s2;
float c1, c2;
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];
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];
/* 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;
}
}
/* 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);
}
}
}
static void getChannelWeights (int indx, int flag, float ch[2]){
if (indx == 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)
FFSWAP(float, ch[0], ch[1]);
}
}
static void channelWeighting (float *su1, float *su2, int *p3)
{
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]);
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(; nsample < 256; nsample++) {
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)
*/
static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
{
int band, result=0, numSubbands, lastTonal, numBands;
if (codingMode == JOINT_STEREO && channelNum == 1) {
if (get_bits(gb,2) != 3) {
av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
return -1;
}
} else {
if (get_bits(gb,6) != 0x28) {
av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
return -1;
}
}
/* number of coded QMF bands */
pSnd->bandsCoded = get_bits(gb,2);
result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
if (result) return result;
pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
if (pSnd->numComponents == -1) return -1;
numSubbands = decodeSpectrum (gb, pSnd->spectrum);
/* Merge the decoded spectrum and tonal components. */
lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->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);
/* Reconstruct time domain samples. */
for (band=0; band<4; band++) {
/* Perform the IMDCT step without overlapping. */
if (band <= numBands) {
IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
} else
memset(pSnd->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]));
}
/* Swap the gain control buffers for the next frame. */
pSnd->gcBlkSwitch ^= 1;
return 0;
}
/**
* Frame handling
*
* @param q Atrac3 private context
* @param databuf the input data
*/
static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
{
int result, i;
float *p1, *p2, *p3, *p4;
uint8_t *ptr1;
if (q->codingMode == 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, q->outSamples, 0, JOINT_STEREO);
if (result != 0)
return (result);
/* 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);
}
} else {
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--;
}
/* Skip the sync codes (0xF8). */
ptr1 = q->decoded_bytes_buffer;
for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
if (i >= q->bytes_per_frame)
return -1;
}
/* set the bitstream reader at the start of the second Sound Unit*/
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));
q->weighting_delay[4] = get_bits1(&q->gb);
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);
}
/* Decode Sound Unit 2. */
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
if (result != 0)
return (result);
/* Reconstruct the channel coefficients. */
reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
} else {
/* normal stereo mode or mono */
/* Decode the channel sound units. */
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], &q->outSamples[i*1024], i, q->codingMode);
if (result != 0)
return (result);
}
}
/* Apply the iQMF synthesis filter. */
p1= q->outSamples;
for (i=0 ; i<q->channels ; i++) {
p2= p1+256;
p3= p2+256;
p4= p3+256;
atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
p1 +=1024;
}
return 0;
}
/**
* Atrac frame decoding
*
* @param avctx pointer to the AVCodecContext
*/
static int atrac3_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt) {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ATRAC3Context *q = avctx->priv_data;
int result = 0, i;
const uint8_t* databuf;
int16_t* samples = data;
if (buf_size < avctx->block_align)
return buf_size;
/* Check if we need to descramble and what buffer to pass on. */
if (q->scrambled_stream) {
decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
databuf = q->decoded_bytes_buffer;
} else {
databuf = buf;
}
result = decodeFrame(q, databuf);
if (result != 0) {
av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
return -1;
}
if (q->channels == 1) {
/* mono */
for (i = 0; i<1024; i++)
samples[i] = av_clip_int16(round(q->outSamples[i]));
*data_size = 1024 * sizeof(int16_t);
} else {
/* stereo */
for (i = 0; i < 1024; i++) {
samples[i*2] = av_clip_int16(round(q->outSamples[i]));
samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
}
*data_size = 2048 * sizeof(int16_t);
}
return avctx->block_align;
}
/**
* Atrac3 initialization
*
* @param avctx pointer to the AVCodecContext
*/
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
int i;
const uint8_t *edata_ptr = avctx->extradata;
ATRAC3Context *q = avctx->priv_data;
static VLC_TYPE atrac3_vlc_table[4096][2];
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->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
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
/* setup */
q->samples_per_frame = 1024 * 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);
return -1;
}
} 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->samples_per_channel = q->samples_per_frame / q->channels;
q->scrambled_stream = 1;
} else {
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);
return -1;
}
if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
return -1;
}
if (q->delay != 0x88E) {
av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
return -1;
}
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);
return -1;
}
if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
return -1;
}
if(avctx->block_align >= UINT_MAX/2)
return -1;
/* 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)
return AVERROR(ENOMEM);
/* Initialize the VLC tables. */
if (!vlcs_initialized) {
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);
}
vlcs_initialized = 1;
}
init_atrac3_transforms(q);
atrac_generate_tables();
/* 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);
/* init the joint-stereo decoding data */
q->weighting_delay[0] = 0;
q->weighting_delay[1] = 7;
q->weighting_delay[2] = 0;
q->weighting_delay[3] = 7;
q->weighting_delay[4] = 0;
q->weighting_delay[5] = 7;
for (i=0; i<4; i++) {
q->matrix_coeff_index_prev[i] = 3;
q->matrix_coeff_index_now[i] = 3;
q->matrix_coeff_index_next[i] = 3;
}
dsputil_init(&dsp, avctx);
q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
if (!q->pUnits) {
av_free(q->decoded_bytes_buffer);
return AVERROR(ENOMEM);
}
avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
AVCodec atrac3_decoder =
{
.name = "atrac3",
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_ATRAC3,
.priv_data_size = sizeof(ATRAC3Context),
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
};