519adfbaae
The decoder is still in floating point. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22235 a1c6a512-1295-4272-9138-f99709370657
1249 lines
38 KiB
C
1249 lines
38 KiB
C
/*
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* Atrac 3 compatible decoder
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* Copyright (c) 2006-2008 Maxim Poliakovski
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* Copyright (c) 2006-2008 Benjamin Larsson
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*
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* This file is part of FFmpeg.
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*
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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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*
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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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*
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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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*/
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/**
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* @file libavcodec/atrac3.c
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* Atrac 3 compatible decoder.
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* This decoder handles Sony's ATRAC3 data.
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*
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* Container formats used to store atrac 3 data:
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* RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
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*
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* To use this decoder, a calling application must supply the extradata
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* bytes provided in the containers above.
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*/
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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#include "avcodec.h"
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#include "bitstream.h"
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#include "dsputil.h"
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#include "bytestream.h"
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#include <stdint.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <fcntl.h>
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#include <unistd.h>
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#include <string.h>
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#include "../librm/rm.h"
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#include "atrac3data.h"
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#define JOINT_STEREO 0x12
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#define STEREO 0x2
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#define AVERROR(...) -1
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/* These structures are needed to store the parsed gain control data. */
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typedef struct {
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int num_gain_data;
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int levcode[8];
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int loccode[8];
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} gain_info;
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typedef struct {
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gain_info gBlock[4];
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} gain_block;
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typedef struct {
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int pos;
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int numCoefs;
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float coef[8];
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} tonal_component;
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typedef struct {
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int bandsCoded;
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int numComponents;
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tonal_component components[64];
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float prevFrame[1024];
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int gcBlkSwitch;
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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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float delayBuf2[46];
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float delayBuf3[46];
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} channel_unit;
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typedef struct {
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GetBitContext gb;
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//@{
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/** stream data */
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int channels;
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int codingMode;
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int bit_rate;
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int sample_rate;
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int samples_per_channel;
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int samples_per_frame;
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int bits_per_frame;
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int bytes_per_frame;
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int pBs;
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channel_unit* pUnits;
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//@}
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//@{
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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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//@}
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//@{
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/** data buffers */
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float outSamples[2048];
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uint8_t* decoded_bytes_buffer;
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float tempBuf[1070];
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//@}
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//@{
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/** extradata */
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int atrac3version;
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int delay;
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int scrambled_stream;
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int frame_factor;
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//@}
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} ATRAC3Context;
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static DECLARE_ALIGNED_16(float,mdct_window[512]);
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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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/**
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* Quadrature mirror synthesis filter.
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*
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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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*/
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static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
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{
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int i, j;
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float *p1, *p3;
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memcpy(temp, delayBuf, 46*sizeof(float));
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p3 = temp + 46;
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/* loop1 */
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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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}
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/* loop2 */
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p1 = temp;
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for (j = nIn; j != 0; j--) {
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float s1 = 0.0;
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float s2 = 0.0;
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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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}
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pOut[0] = s2;
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pOut[1] = s1;
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p1 += 2;
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pOut += 2;
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}
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/* Update the delay buffer. */
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memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
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}
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/**
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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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*
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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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*/
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static void IMLT(float *pInput, float *pOutput, int odd_band)
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{
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int i;
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if (odd_band) {
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/**
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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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*/
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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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}
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ff_imdct_calc(&mdct_ctx,pOutput,pInput);
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/* Perform windowing on the output. */
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dsp.vector_fmul(pOutput,mdct_window,512);
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}
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/**
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* Atrac 3 indata descrambling, only used for data coming from the rm container
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*
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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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*/
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static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
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int i, off;
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uint32_t c;
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const uint32_t* buf;
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uint32_t* obuf = (uint32_t*) out;
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#ifdef TEST
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off = 0; //no check for memory alignment of inbuffer
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#else
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off = (intptr_t)inbuffer & 3;
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#endif /* TEST */
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buf = (const uint32_t*) (inbuffer - off);
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c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
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bytes += 3 + off;
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for (i = 0; i < bytes/4; i++)
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obuf[i] = c ^ buf[i];
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if (off)
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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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return off;
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}
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static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
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float enc_window[256];
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float s;
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int i;
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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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if (!mdct_window[0])
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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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}
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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[i] = s;
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qmf_window[47 - i] = s;
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}
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/* Initialize the MDCT transform. */
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ff_mdct_init(&mdct_ctx, 9, 1);
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}
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/**
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* Atrac3 uninit, free all allocated memory
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*/
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static av_cold int atrac3_decode_close(ATRAC3Context *q)
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{
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//ATRAC3Context *q = rmctx->priv_data;
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av_free(q->pUnits);
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av_free(q->decoded_bytes_buffer);
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return 0;
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}
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/**
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/ * Mantissa decoding
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*
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* @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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*/
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static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
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{
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int numBits, cnt, code, huffSymb;
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if (selector == 1)
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numCodes /= 2;
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if (codingFlag != 0) {
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/* constant length coding (CLC) */
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numBits = CLCLengthTab[selector];
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if (selector > 1) {
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for (cnt = 0; cnt < numCodes; cnt++) {
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if (numBits)
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code = get_sbits(gb, numBits);
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else
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code = 0;
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mantissas[cnt] = code;
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}
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} else {
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for (cnt = 0; cnt < numCodes; cnt++) {
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if (numBits)
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code = get_bits(gb, numBits); //numBits is always 4 in this case
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else
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code = 0;
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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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}
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}
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} else {
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/* variable length coding (VLC) */
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if (selector != 1) {
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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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huffSymb += 1;
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code = huffSymb >> 1;
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if (huffSymb & 1)
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code = -code;
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mantissas[cnt] = code;
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}
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} else {
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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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}
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}
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}
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}
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/**
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* Restore the quantized band spectrum coefficients
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*
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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
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*/
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static int decodeSpectrum (GetBitContext *gb, float *pOut)
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{
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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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int mantissas[128];
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float SF;
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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. */
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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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}
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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) */
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readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
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/* Decode the scale factor for this subband. */
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SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
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/* Inverse quantize the coefficients. */
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for (pIn=mantissas ; first<last; first++, pIn++)
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pOut[first] = *pIn * SF;
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} else {
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/* This subband was not coded, so zero the entire subband. */
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memset(pOut+first, 0, subbWidth*sizeof(float));
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}
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}
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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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return numSubbands;
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}
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/**
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* Restore the quantized tonal components
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*
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* @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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*/
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static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
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{
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int i,j,k,cnt;
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int components, coding_mode_selector, coding_mode, coded_values_per_component;
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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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float *pCoef;
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float scalefactor;
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int component_count = 0;
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components = get_bits(gb,5);
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/* no tonal components */
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if (components == 0)
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return 0;
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coding_mode_selector = get_bits(gb,2);
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if (coding_mode_selector == 2)
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return -1;
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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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return -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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continue;
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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);
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pComponent[component_count].numCoefs = coded_values;
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/* inverse quant */
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pCoef = pComponent[component_count].coef;
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for (cnt = 0; cnt < coded_values; cnt++)
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pCoef[cnt] = mantissa[cnt] * scalefactor;
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component_count++;
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}
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}
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}
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return component_count;
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}
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/**
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* Decode gain parameters for the coded bands
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*
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* @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
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*/
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static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
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{
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int i, cf, numData;
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int *pLevel, *pLoc;
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gain_info *pGain = pGb->gBlock;
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for (i=0 ; i<=numBands; i++)
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{
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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])
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|
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;
|
|
iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
|
|
iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
|
|
iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
|
|
p1 +=1024;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Atrac frame decoding
|
|
*
|
|
* @param rmctx pointer to the AVCodecContext
|
|
*/
|
|
|
|
static int atrac3_decode_frame(RMContext *rmctx, ATRAC3Context *q,
|
|
void *data, int *data_size,
|
|
const uint8_t *buf, int buf_size) {
|
|
//ATRAC3Context *q = rmctx->priv_data;
|
|
int result = 0, i;
|
|
const uint8_t* databuf;
|
|
int16_t* samples = data;
|
|
|
|
if (buf_size < rmctx->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, rmctx->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 rmctx->block_align;
|
|
}
|
|
|
|
|
|
/**
|
|
* Atrac3 initialization
|
|
*
|
|
* @param rmctx pointer to the RMContext
|
|
*/
|
|
|
|
static av_cold int atrac3_decode_init(ATRAC3Context *q, RMContext *rmctx)
|
|
{
|
|
int i;
|
|
const uint8_t *edata_ptr = rmctx->codec_extradata;
|
|
//ATRAC3Context *q = rmctx->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 = rmctx->sample_rate;
|
|
q->channels = rmctx->nb_channels;
|
|
q->bit_rate = rmctx->bit_rate;
|
|
q->bits_per_frame = rmctx->block_align * 8;
|
|
q->bytes_per_frame = rmctx->block_align;
|
|
|
|
/* Take care of the codec-specific extradata. */
|
|
if (rmctx->extradata_size == 14) {
|
|
/* Parse the extradata, WAV format */
|
|
av_log(rmctx,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(rmctx,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(rmctx,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(rmctx,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 (rmctx->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",rmctx->extradata_size);
|
|
}
|
|
/* Check the extradata. */
|
|
|
|
if (q->atrac3version != 4) {
|
|
av_log(rmctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
|
|
return -1;
|
|
}
|
|
|
|
if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
|
|
av_log(rmctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
|
|
return -1;
|
|
}
|
|
|
|
if (q->delay != 0x88E) {
|
|
av_log(rmctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
|
|
return -1;
|
|
}
|
|
|
|
if (q->codingMode == STEREO) {
|
|
av_log(rmctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
|
|
} else if (q->codingMode == JOINT_STEREO) {
|
|
av_log(rmctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
|
|
} else {
|
|
av_log(rmctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
|
|
return -1;
|
|
}
|
|
|
|
if (rmctx->nb_channels <= 0 || rmctx->nb_channels > 2 /*|| ((rmctx->channels * 1024) != q->samples_per_frame)*/) {
|
|
av_log(rmctx,AV_LOG_ERROR,"Channel configuration error!\n");
|
|
return -1;
|
|
}
|
|
|
|
|
|
if(rmctx->block_align >= UINT16_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((rmctx->block_align+(4-rmctx->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);
|
|
|
|
/* Generate the scale factors. */
|
|
for (i=0 ; i<64 ; i++)
|
|
SFTable[i] = pow(2.0, (i - 15) / 3.0);
|
|
|
|
/* 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);
|
|
|
|
q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
|
|
if (!q->pUnits) {
|
|
av_free(q->decoded_bytes_buffer);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/***************************************************************
|
|
* Following is a test program to convert from atrac/rm to wav *
|
|
***************************************************************/
|
|
static unsigned char wav_header[44]={
|
|
'R','I','F','F',// 0 - ChunkID
|
|
0,0,0,0, // 4 - ChunkSize (filesize-8)
|
|
'W','A','V','E',// 8 - Format
|
|
'f','m','t',' ',// 12 - SubChunkID
|
|
16,0,0,0, // 16 - SubChunk1ID // 16 for PCM
|
|
1,0, // 20 - AudioFormat (1=Uncompressed)
|
|
2,0, // 22 - NumChannels
|
|
0,0,0,0, // 24 - SampleRate in Hz
|
|
0,0,0,0, // 28 - Byte Rate (SampleRate*NumChannels*(BitsPerSample/8)
|
|
4,0, // 32 - BlockAlign (== NumChannels * BitsPerSample/8)
|
|
16,0, // 34 - BitsPerSample
|
|
'd','a','t','a',// 36 - Subchunk2ID
|
|
0,0,0,0 // 40 - Subchunk2Size
|
|
};
|
|
|
|
int open_wav(char* filename) {
|
|
int fd,res;
|
|
|
|
fd=open(filename,O_CREAT|O_WRONLY|O_TRUNC,S_IRUSR|S_IWUSR);
|
|
if (fd >= 0) {
|
|
res = write(fd,wav_header,sizeof(wav_header));
|
|
}
|
|
|
|
return(fd);
|
|
}
|
|
|
|
void close_wav(int fd, RMContext *rmctx, ATRAC3Context *q) {
|
|
int x,res;
|
|
int filesize;
|
|
int bytes_per_sample = 2;
|
|
int samples_per_frame = q->samples_per_frame;
|
|
int nb_channels = rmctx->nb_channels;
|
|
int sample_rate = rmctx->sample_rate;
|
|
int nb_frames = rmctx->audio_framesize/rmctx->block_align * rmctx->nb_packets - 2; // first 2 frames have no valid audio; skipped in output
|
|
|
|
filesize= samples_per_frame*bytes_per_sample*nb_frames +44;
|
|
printf("Filesize = %d\n",filesize);
|
|
|
|
// ChunkSize
|
|
x=filesize-8;
|
|
wav_header[4]=(x&0xff);
|
|
wav_header[5]=(x&0xff00)>>8;
|
|
wav_header[6]=(x&0xff0000)>>16;
|
|
wav_header[7]=(x&0xff000000)>>24;
|
|
|
|
// Number of channels
|
|
wav_header[22]=nb_channels;
|
|
|
|
// Samplerate
|
|
wav_header[24]=sample_rate&0xff;
|
|
wav_header[25]=(sample_rate&0xff00)>>8;
|
|
wav_header[26]=(sample_rate&0xff0000)>>16;
|
|
wav_header[27]=(sample_rate&0xff000000)>>24;
|
|
|
|
// ByteRate
|
|
x=sample_rate*bytes_per_sample*nb_channels;
|
|
wav_header[28]=(x&0xff);
|
|
wav_header[29]=(x&0xff00)>>8;
|
|
wav_header[30]=(x&0xff0000)>>16;
|
|
wav_header[31]=(x&0xff000000)>>24;
|
|
|
|
// BlockAlign
|
|
wav_header[32]=rmctx->block_align;//2*rmctx->nb_channels;
|
|
|
|
// Bits per sample
|
|
wav_header[34]=16;
|
|
|
|
// Subchunk2Size
|
|
x=filesize-44;
|
|
wav_header[40]=(x&0xff);
|
|
wav_header[41]=(x&0xff00)>>8;
|
|
wav_header[42]=(x&0xff0000)>>16;
|
|
wav_header[43]=(x&0xff000000)>>24;
|
|
|
|
lseek(fd,0,SEEK_SET);
|
|
res = write(fd,wav_header,sizeof(wav_header));
|
|
close(fd);
|
|
}
|
|
|
|
int main(int argc, char *argv[])
|
|
{
|
|
int fd, fd_dec;
|
|
int res, i, datasize = 0;
|
|
|
|
#ifdef DUMP_RAW_FRAMES
|
|
char filename[15];
|
|
int fd_out;
|
|
#endif
|
|
int16_t outbuf[2048];
|
|
uint16_t fs,sps,h;
|
|
uint32_t packet_count;
|
|
ATRAC3Context q;
|
|
RMContext rmctx;
|
|
RMPacket pkt;
|
|
|
|
memset(&q,0,sizeof(ATRAC3Context));
|
|
memset(&rmctx,0,sizeof(RMContext));
|
|
memset(&pkt,0,sizeof(RMPacket));
|
|
|
|
if (argc != 2) {
|
|
DEBUGF("Incorrect number of arguments\n");
|
|
return -1;
|
|
}
|
|
|
|
fd = open(argv[1],O_RDONLY);
|
|
if (fd < 0) {
|
|
DEBUGF("Error opening file %s\n", argv[1]);
|
|
return -1;
|
|
}
|
|
|
|
/* copy the input rm file to a memory buffer */
|
|
uint8_t * filebuf = (uint8_t *)calloc((int)filesize(fd),sizeof(uint8_t));
|
|
res = read(fd,filebuf,filesize(fd));
|
|
|
|
fd_dec = open_wav("output.wav");
|
|
if (fd_dec < 0) {
|
|
DEBUGF("Error creating output file\n");
|
|
return -1;
|
|
}
|
|
res = real_parse_header(fd, &rmctx);
|
|
packet_count = rmctx.nb_packets;
|
|
rmctx.audio_framesize = rmctx.block_align;
|
|
rmctx.block_align = rmctx.sub_packet_size;
|
|
fs = rmctx.audio_framesize;
|
|
sps= rmctx.block_align;
|
|
h = rmctx.sub_packet_h;
|
|
atrac3_decode_init(&q,&rmctx);
|
|
|
|
/* change the buffer pointer to point at the first audio frame */
|
|
advance_buffer(&filebuf, rmctx.data_offset + DATA_HEADER_SIZE);
|
|
while(packet_count)
|
|
{
|
|
rm_get_packet(&filebuf, &rmctx, &pkt);
|
|
for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
|
|
{
|
|
/* output raw audio frames that are sent to the decoder into separate files */
|
|
#ifdef DUMP_RAW_FRAMES
|
|
snprintf(filename,sizeof(filename),"dump%d.raw",++x);
|
|
fd_out = open(filename,O_WRONLY|O_CREAT|O_APPEND);
|
|
write(fd_out,pkt.frames[i],sps);
|
|
close(fd_out);
|
|
#endif
|
|
if(pkt.length > 0)
|
|
res = atrac3_decode_frame(&rmctx,&q, outbuf, &datasize, pkt.frames[i] , rmctx.block_align);
|
|
rmctx.frame_number++;
|
|
res = write(fd_dec,outbuf,datasize);
|
|
}
|
|
packet_count -= rmctx.audio_pkt_cnt;
|
|
rmctx.audio_pkt_cnt = 0;
|
|
}
|
|
atrac3_decode_close(&q);
|
|
close_wav(fd_dec, &rmctx, &q);
|
|
close(fd);
|
|
|
|
return 0;
|
|
}
|