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FFmpeg/libavcodec/atrac3.c
Stefano Sabatini 72415b2adb Define AVMediaType enum, and use it instead of enum CodecType, which 
is deprecated and will be dropped at the next major bump.

Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-30 23:30:55 +00:00

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/*
* 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 "fft.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 = AVMEDIA_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)"),
};
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