2
* Atrac 3 compatible decoder
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* Copyright (c) 2006-2007 Maxim Poliakovski
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* Copyright (c) 2006-2007 Benjamin Larsson
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* This file is part of FFmpeg.
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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* Atrac 3 compatible decoder.
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* This decoder handles RealNetworks, RealAudio atrc data.
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* Atrac 3 is identified by the codec name atrc in RealMedia files.
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* To use this decoder, a calling application must supply the extradata
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* bytes provided from the RealMedia container: 10 bytes or 14 bytes
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* from the WAV container.
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#include "bitstream.h"
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#include "bytestream.h"
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#include "atrac3data.h"
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#define JOINT_STEREO 0x12
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/* These structures are needed to store the parsed gain control data. */
69
tonal_component components[64];
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float prevFrame[1024];
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gain_block gainBlock[2];
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DECLARE_ALIGNED_16(float, spectrum[1024]);
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DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);
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float delayBuf1[46]; ///<qmf delay buffers
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int samples_per_channel;
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int samples_per_frame;
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/** joint-stereo related variables */
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int matrix_coeff_index_prev[4];
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int matrix_coeff_index_now[4];
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int matrix_coeff_index_next[4];
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int weighting_delay[6];
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float outSamples[2048];
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uint8_t* decoded_bytes_buffer;
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DECLARE_ALIGNED_16(float,mdct_tmp[512]);
116
int scrambled_stream;
121
static DECLARE_ALIGNED_16(float,mdct_window[512]);
122
static float qmf_window[48];
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static VLC spectral_coeff_tab[7];
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static float SFTable[64];
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static float gain_tab1[16];
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static float gain_tab2[31];
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static MDCTContext mdct_ctx;
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static DSPContext dsp;
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/* quadrature mirror synthesis filter */
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* Quadrature mirror synthesis filter.
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* @param inlo lower part of spectrum
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* @param inhi higher part of spectrum
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* @param nIn size of spectrum buffer
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* @param pOut out buffer
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* @param delayBuf delayBuf buffer
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* @param temp temp buffer
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static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
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memcpy(temp, delayBuf, 46*sizeof(float));
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for(i=0; i<nIn; i+=2){
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p3[2*i+0] = inlo[i ] + inhi[i ];
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p3[2*i+1] = inlo[i ] - inhi[i ];
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p3[2*i+2] = inlo[i+1] + inhi[i+1];
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p3[2*i+3] = inlo[i+1] - inhi[i+1];
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for (j = nIn; j != 0; j--) {
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for (i = 0; i < 48; i += 2) {
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s1 += p1[i] * qmf_window[i];
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s2 += p1[i+1] * qmf_window[i+1];
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/* Update the delay buffer. */
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memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
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* Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
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* caused by the reverse spectra of the QMF.
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* @param pInput float input
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* @param pOutput float output
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* @param odd_band 1 if the band is an odd band
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* @param mdct_tmp aligned temporary buffer for the mdct
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static void IMLT(float *pInput, float *pOutput, int odd_band, float* mdct_tmp)
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* Reverse the odd bands before IMDCT, this is an effect of the QMF transform
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* or it gives better compression to do it this way.
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* FIXME: It should be possible to handle this in ff_imdct_calc
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* for that to happen a modification of the prerotation step of
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* all SIMD code and C code is needed.
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* Or fix the functions before so they generate a pre reversed spectrum.
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for (i=0; i<128; i++)
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FFSWAP(float, pInput[i], pInput[255-i]);
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mdct_ctx.fft.imdct_calc(&mdct_ctx,pOutput,pInput,mdct_tmp);
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/* Perform windowing on the output. */
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dsp.vector_fmul(pOutput,mdct_window,512);
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* Atrac 3 indata descrambling, only used for data coming from the rm container
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* @param in pointer to 8 bit array of indata
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* @param bits amount of bits
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* @param out pointer to 8 bit array of outdata
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static int decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){
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uint32_t* obuf = (uint32_t*) out;
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off = (int)((long)inbuffer & 3);
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buf = (uint32_t*) (inbuffer - off);
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c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
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for (i = 0; i < bytes/4; i++)
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obuf[i] = c ^ buf[i];
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av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
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static void init_atrac3_transforms(ATRAC3Context *q) {
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float enc_window[256];
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/* Generate the mdct window, for details see
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* http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
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for (i=0 ; i<256; i++)
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enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
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for (i=0 ; i<256; i++) {
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mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
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mdct_window[511-i] = mdct_window[i];
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/* Generate the QMF window. */
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for (i=0 ; i<24; i++) {
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s = qmf_48tap_half[i] * 2.0;
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qmf_window[47 - i] = s;
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/* Initialize the MDCT transform. */
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ff_mdct_init(&mdct_ctx, 9, 1);
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* Atrac3 uninit, free all allocated memory
279
static int atrac3_decode_close(AVCodecContext *avctx)
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ATRAC3Context *q = avctx->priv_data;
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av_free(q->decoded_bytes_buffer);
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/ * Mantissa decoding
292
* @param gb the GetBit context
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* @param selector what table is the output values coded with
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* @param codingFlag constant length coding or variable length coding
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* @param mantissas mantissa output table
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* @param numCodes amount of values to get
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static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
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int numBits, cnt, code, huffSymb;
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if (codingFlag != 0) {
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/* constant length coding (CLC) */
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//FIXME we don't have any samples coded in CLC mode
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numBits = CLCLengthTab[selector];
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for (cnt = 0; cnt < numCodes; cnt++) {
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code = get_sbits(gb, numBits);
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mantissas[cnt] = code;
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for (cnt = 0; cnt < numCodes; cnt++) {
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code = get_bits(gb, numBits); //numBits is always 4 in this case
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mantissas[cnt*2] = seTab_0[code >> 2];
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mantissas[cnt*2+1] = seTab_0[code & 3];
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/* variable length coding (VLC) */
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for (cnt = 0; cnt < numCodes; cnt++) {
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huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
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code = huffSymb >> 1;
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mantissas[cnt] = code;
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for (cnt = 0; cnt < numCodes; cnt++) {
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huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
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mantissas[cnt*2] = decTable1[huffSymb*2];
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mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
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* Restore the quantized band spectrum coefficients
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* @param gb the GetBit context
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* @param pOut decoded band spectrum
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* @return outSubbands subband counter, fix for broken specification/files
358
static int decodeSpectrum (GetBitContext *gb, float *pOut)
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int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
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int subband_vlc_index[32], SF_idxs[32];
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numSubbands = get_bits(gb, 5); // number of coded subbands
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codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
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/* Get the VLC selector table for the subbands, 0 means not coded. */
369
for (cnt = 0; cnt <= numSubbands; cnt++)
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subband_vlc_index[cnt] = get_bits(gb, 3);
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/* Read the scale factor indexes from the stream. */
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for (cnt = 0; cnt <= numSubbands; cnt++) {
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if (subband_vlc_index[cnt] != 0)
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SF_idxs[cnt] = get_bits(gb, 6);
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for (cnt = 0; cnt <= numSubbands; cnt++) {
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first = subbandTab[cnt];
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last = subbandTab[cnt+1];
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subbWidth = last - first;
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if (subband_vlc_index[cnt] != 0) {
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/* Decode spectral coefficients for this subband. */
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/* TODO: This can be done faster is several blocks share the
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* same VLC selector (subband_vlc_index) */
388
readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
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/* Decode the scale factor for this subband. */
391
SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
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/* Inverse quantize the coefficients. */
394
for (pIn=mantissas ; first<last; first++, pIn++)
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pOut[first] = *pIn * SF;
397
/* This subband was not coded, so zero the entire subband. */
398
memset(pOut+first, 0, subbWidth*sizeof(float));
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/* Clear the subbands that were not coded. */
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first = subbandTab[cnt];
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memset(pOut+first, 0, (1024 - first) * sizeof(float));
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* Restore the quantized tonal components
411
* @param gb the GetBit context
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* @param pComponent tone component
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* @param numBands amount of coded bands
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static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
419
int components, coding_mode_selector, coding_mode, coded_values_per_component;
420
int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
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int band_flags[4], mantissa[8];
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int component_count = 0;
426
components = get_bits(gb,5);
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/* no tonal components */
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coding_mode_selector = get_bits(gb,2);
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if (coding_mode_selector == 2)
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coding_mode = coding_mode_selector & 1;
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for (i = 0; i < components; i++) {
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for (cnt = 0; cnt <= numBands; cnt++)
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band_flags[cnt] = get_bits1(gb);
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coded_values_per_component = get_bits(gb,3);
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quant_step_index = get_bits(gb,3);
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if (quant_step_index <= 1)
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if (coding_mode_selector == 3)
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coding_mode = get_bits1(gb);
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for (j = 0; j < (numBands + 1) * 4; j++) {
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if (band_flags[j >> 2] == 0)
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coded_components = get_bits(gb,3);
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for (k=0; k<coded_components; k++) {
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sfIndx = get_bits(gb,6);
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pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
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max_coded_values = 1024 - pComponent[component_count].pos;
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coded_values = coded_values_per_component + 1;
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coded_values = FFMIN(max_coded_values,coded_values);
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scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];
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readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
468
pComponent[component_count].numCoefs = coded_values;
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pCoef = pComponent[k].coef;
472
for (cnt = 0; cnt < coded_values; cnt++)
473
pCoef[cnt] = mantissa[cnt] * scalefactor;
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return component_count;
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* Decode gain parameters for the coded bands
486
* @param gb the GetBit context
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* @param pGb the gainblock for the current band
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* @param numBands amount of coded bands
491
static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
496
gain_info *pGain = pGb->gBlock;
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for (i=0 ; i<=numBands; i++)
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numData = get_bits(gb,3);
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pGain[i].num_gain_data = numData;
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pLevel = pGain[i].levcode;
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pLoc = pGain[i].loccode;
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for (cf = 0; cf < numData; cf++){
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pLevel[cf]= get_bits(gb,4);
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pLoc [cf]= get_bits(gb,5);
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if(cf && pLoc[cf] <= pLoc[cf-1])
513
/* Clear the unused blocks. */
515
pGain[i].num_gain_data = 0;
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* Apply gain parameters and perform the MDCT overlapping part
523
* @param pIn input float buffer
524
* @param pPrev previous float buffer to perform overlap against
525
* @param pOut output float buffer
526
* @param pGain1 current band gain info
527
* @param pGain2 next band gain info
530
static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
532
/* gain compensation function */
533
float gain1, gain2, gain_inc;
534
int cnt, numdata, nsample, startLoc, endLoc;
537
if (pGain2->num_gain_data == 0)
540
gain1 = gain_tab1[pGain2->levcode[0]];
542
if (pGain1->num_gain_data == 0) {
543
for (cnt = 0; cnt < 256; cnt++)
544
pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
546
numdata = pGain1->num_gain_data;
547
pGain1->loccode[numdata] = 32;
548
pGain1->levcode[numdata] = 4;
550
nsample = 0; // current sample = 0
552
for (cnt = 0; cnt < numdata; cnt++) {
553
startLoc = pGain1->loccode[cnt] * 8;
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endLoc = startLoc + 8;
556
gain2 = gain_tab1[pGain1->levcode[cnt]];
557
gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
560
for (; nsample < startLoc; nsample++)
561
pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
563
/* interpolation is done over eight samples */
564
for (; nsample < endLoc; nsample++) {
565
pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
570
for (; nsample < 256; nsample++)
571
pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
574
/* Delay for the overlapping part. */
575
memcpy(pPrev, &pIn[256], 256*sizeof(float));
579
* Combine the tonal band spectrum and regular band spectrum
581
* @param pSpectrum output spectrum buffer
582
* @param numComponents amount of tonal components
583
* @param pComponent tonal components for this band
586
static void addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
591
for (cnt = 0; cnt < numComponents; cnt++){
592
pIn = pComponent[cnt].coef;
593
pOut = &(pSpectrum[pComponent[cnt].pos]);
595
for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
601
#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
603
static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
605
int i, band, nsample, s1, s2;
607
float mc1_l, mc1_r, mc2_l, mc2_r;
609
for (i=0,band = 0; band < 4*256; band+=256,i++) {
615
/* Selector value changed, interpolation needed. */
616
mc1_l = matrixCoeffs[s1*2];
617
mc1_r = matrixCoeffs[s1*2+1];
618
mc2_l = matrixCoeffs[s2*2];
619
mc2_r = matrixCoeffs[s2*2+1];
621
/* Interpolation is done over the first eight samples. */
622
for(; nsample < 8; nsample++) {
623
c1 = su1[band+nsample];
624
c2 = su2[band+nsample];
625
c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
626
su1[band+nsample] = c2;
627
su2[band+nsample] = c1 * 2.0 - c2;
631
/* Apply the matrix without interpolation. */
633
case 0: /* M/S decoding */
634
for (; nsample < 256; nsample++) {
635
c1 = su1[band+nsample];
636
c2 = su2[band+nsample];
637
su1[band+nsample] = c2 * 2.0;
638
su2[band+nsample] = (c1 - c2) * 2.0;
643
for (; nsample < 256; nsample++) {
644
c1 = su1[band+nsample];
645
c2 = su2[band+nsample];
646
su1[band+nsample] = (c1 + c2) * 2.0;
647
su2[band+nsample] = c2 * -2.0;
652
for (; nsample < 256; nsample++) {
653
c1 = su1[band+nsample];
654
c2 = su2[band+nsample];
655
su1[band+nsample] = c1 + c2;
656
su2[band+nsample] = c1 - c2;
665
static void getChannelWeights (int indx, int flag, float ch[2]){
671
ch[0] = (float)(indx & 7) / 7.0;
672
ch[1] = sqrt(2 - ch[0]*ch[0]);
674
FFSWAP(float, ch[0], ch[1]);
678
static void channelWeighting (float *su1, float *su2, int *p3)
681
/* w[x][y] y=0 is left y=1 is right */
684
if (p3[1] != 7 || p3[3] != 7){
685
getChannelWeights(p3[1], p3[0], w[0]);
686
getChannelWeights(p3[3], p3[2], w[1]);
688
for(band = 1; band < 4; band++) {
689
/* scale the channels by the weights */
690
for(nsample = 0; nsample < 8; nsample++) {
691
su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
692
su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
695
for(; nsample < 256; nsample++) {
696
su1[band*256+nsample] *= w[1][0];
697
su2[band*256+nsample] *= w[1][1];
705
* Decode a Sound Unit
707
* @param gb the GetBit context
708
* @param pSnd the channel unit to be used
709
* @param pOut the decoded samples before IQMF in float representation
710
* @param channelNum channel number
711
* @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono)
715
static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
717
int band, result=0, numSubbands, numBands;
719
if (codingMode == JOINT_STEREO && channelNum == 1) {
720
if (get_bits(gb,2) != 3) {
721
av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
725
if (get_bits(gb,6) != 0x28) {
726
av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
731
/* number of coded QMF bands */
732
pSnd->bandsCoded = get_bits(gb,2);
734
result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
735
if (result) return result;
737
pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
738
if (pSnd->numComponents == -1) return -1;
740
numSubbands = decodeSpectrum (gb, pSnd->spectrum);
742
/* Merge the decoded spectrum and tonal components. */
743
addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
746
/* Convert number of subbands into number of MLT/QMF bands */
747
numBands = (subbandTab[numSubbands] - 1) >> 8;
750
/* Reconstruct time domain samples. */
751
for (band=0; band<4; band++) {
752
/* Perform the IMDCT step without overlapping. */
753
if (band <= numBands) {
754
IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1,q->mdct_tmp);
756
memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
758
/* gain compensation and overlapping */
759
gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
760
&((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
761
&((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
764
/* Swap the gain control buffers for the next frame. */
765
pSnd->gcBlkSwitch ^= 1;
773
* @param q Atrac3 private context
774
* @param databuf the input data
777
static int decodeFrame(ATRAC3Context *q, uint8_t* databuf)
780
float *p1, *p2, *p3, *p4;
781
uint8_t *ptr1, *ptr2;
783
if (q->codingMode == JOINT_STEREO) {
785
/* channel coupling mode */
786
/* decode Sound Unit 1 */
787
init_get_bits(&q->gb,databuf,q->bits_per_frame);
789
result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
793
/* Framedata of the su2 in the joint-stereo mode is encoded in
794
* reverse byte order so we need to swap it first. */
796
ptr2 = databuf+q->bytes_per_frame-1;
797
for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
798
FFSWAP(uint8_t,*ptr1,*ptr2);
801
/* Skip the sync codes (0xF8). */
803
for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
804
if (i >= q->bytes_per_frame)
809
/* set the bitstream reader at the start of the second Sound Unit*/
810
init_get_bits(&q->gb,ptr1,q->bits_per_frame);
812
/* Fill the Weighting coeffs delay buffer */
813
memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
814
q->weighting_delay[4] = get_bits1(&q->gb);
815
q->weighting_delay[5] = get_bits(&q->gb,3);
817
for (i = 0; i < 4; i++) {
818
q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
819
q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
820
q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
823
/* Decode Sound Unit 2. */
824
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
828
/* Reconstruct the channel coefficients. */
829
reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
831
channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
834
/* normal stereo mode or mono */
835
/* Decode the channel sound units. */
836
for (i=0 ; i<q->channels ; i++) {
838
/* Set the bitstream reader at the start of a channel sound unit. */
839
init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
841
result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
847
/* Apply the iQMF synthesis filter. */
849
for (i=0 ; i<q->channels ; i++) {
853
iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
854
iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
855
iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
864
* Atrac frame decoding
866
* @param avctx pointer to the AVCodecContext
869
static int atrac3_decode_frame(AVCodecContext *avctx,
870
void *data, int *data_size,
871
uint8_t *buf, int buf_size) {
872
ATRAC3Context *q = avctx->priv_data;
875
int16_t* samples = data;
877
if (buf_size < avctx->block_align)
880
/* Check if we need to descramble and what buffer to pass on. */
881
if (q->scrambled_stream) {
882
decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
883
databuf = q->decoded_bytes_buffer;
888
result = decodeFrame(q, databuf);
891
av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
895
if (q->channels == 1) {
897
for (i = 0; i<1024; i++)
898
samples[i] = av_clip_int16(round(q->outSamples[i]));
899
*data_size = 1024 * sizeof(int16_t);
902
for (i = 0; i < 1024; i++) {
903
samples[i*2] = av_clip_int16(round(q->outSamples[i]));
904
samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
906
*data_size = 2048 * sizeof(int16_t);
909
return avctx->block_align;
914
* Atrac3 initialization
916
* @param avctx pointer to the AVCodecContext
919
static int atrac3_decode_init(AVCodecContext *avctx)
922
uint8_t *edata_ptr = avctx->extradata;
923
ATRAC3Context *q = avctx->priv_data;
925
/* Take data from the AVCodecContext (RM container). */
926
q->sample_rate = avctx->sample_rate;
927
q->channels = avctx->channels;
928
q->bit_rate = avctx->bit_rate;
929
q->bits_per_frame = avctx->block_align * 8;
930
q->bytes_per_frame = avctx->block_align;
932
/* Take care of the codec-specific extradata. */
933
if (avctx->extradata_size == 14) {
934
/* Parse the extradata, WAV format */
935
av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1
936
q->samples_per_channel = bytestream_get_le32(&edata_ptr);
937
q->codingMode = bytestream_get_le16(&edata_ptr);
938
av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
939
q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1
940
av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0
943
q->samples_per_frame = 1024 * q->channels;
944
q->atrac3version = 4;
947
q->codingMode = JOINT_STEREO;
949
q->codingMode = STEREO;
951
q->scrambled_stream = 0;
953
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)) {
955
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);
959
} else if (avctx->extradata_size == 10) {
960
/* Parse the extradata, RM format. */
961
q->atrac3version = bytestream_get_be32(&edata_ptr);
962
q->samples_per_frame = bytestream_get_be16(&edata_ptr);
963
q->delay = bytestream_get_be16(&edata_ptr);
964
q->codingMode = bytestream_get_be16(&edata_ptr);
966
q->samples_per_channel = q->samples_per_frame / q->channels;
967
q->scrambled_stream = 1;
970
av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
972
/* Check the extradata. */
974
if (q->atrac3version != 4) {
975
av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
979
if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
980
av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
984
if (q->delay != 0x88E) {
985
av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
989
if (q->codingMode == STEREO) {
990
av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
991
} else if (q->codingMode == JOINT_STEREO) {
992
av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
994
av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
998
if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
999
av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
1004
if(avctx->block_align >= UINT_MAX/2)
1007
/* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
1008
* this is for the bitstream reader. */
1009
if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL)
1010
return AVERROR(ENOMEM);
1013
/* Initialize the VLC tables. */
1014
for (i=0 ; i<7 ; i++) {
1015
init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
1017
huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);
1020
init_atrac3_transforms(q);
1022
/* Generate the scale factors. */
1023
for (i=0 ; i<64 ; i++)
1024
SFTable[i] = pow(2.0, (i - 15) / 3.0);
1026
/* Generate gain tables. */
1027
for (i=0 ; i<16 ; i++)
1028
gain_tab1[i] = powf (2.0, (4 - i));
1030
for (i=-15 ; i<16 ; i++)
1031
gain_tab2[i+15] = powf (2.0, i * -0.125);
1033
/* init the joint-stereo decoding data */
1034
q->weighting_delay[0] = 0;
1035
q->weighting_delay[1] = 7;
1036
q->weighting_delay[2] = 0;
1037
q->weighting_delay[3] = 7;
1038
q->weighting_delay[4] = 0;
1039
q->weighting_delay[5] = 7;
1041
for (i=0; i<4; i++) {
1042
q->matrix_coeff_index_prev[i] = 3;
1043
q->matrix_coeff_index_now[i] = 3;
1044
q->matrix_coeff_index_next[i] = 3;
1047
dsputil_init(&dsp, avctx);
1049
q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
1051
av_free(q->decoded_bytes_buffer);
1052
return AVERROR(ENOMEM);
1059
AVCodec atrac3_decoder =
1062
.type = CODEC_TYPE_AUDIO,
1063
.id = CODEC_ID_ATRAC3,
1064
.priv_data_size = sizeof(ATRAC3Context),
1065
.init = atrac3_decode_init,
1066
.close = atrac3_decode_close,
1067
.decode = atrac3_decode_frame,