atrac3.c

ffmpeg的完整源代码和作者自己写的文档。不但有在Linux的工程哦

/*
 * Atrac 3 compatible decoder
 * Copyright (c) 2006-2007 Maxim Poliakovski
 * Copyright (c) 2006-2007 Benjamin Larsson
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file atrac3.c
 * Atrac 3 compatible decoder.
 * This decoder handles RealNetworks, RealAudio atrc data.
 * Atrac 3 is identified by the codec name atrc in RealMedia files.
 *
 * To use this decoder, a calling application must supply the extradata
 * bytes provided from the RealMedia container: 10 bytes or 14 bytes
 * from the WAV container.
 */

#include <math.h>
#include <stddef.h>
#include <stdio.h>

#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "bytestream.h"

#include "atrac3data.h"

#define JOINT_STEREO    0x12
#define STEREO          0x2


/* These structures are needed to store the parsed gain control data. */
typedef struct {
    int   num_gain_data;
    int   levcode[8];
    int   loccode[8];
} gain_info;

typedef struct {
    gain_info   gBlock[4];
} gain_block;

typedef struct {
    int     pos;
    int     numCoefs;
    float   coef[8];
} tonal_component;

typedef struct {
    int               bandsCoded;
    int               numComponents;
    tonal_component   components[64];
    float             prevFrame[1024];
    int               gcBlkSwitch;
    gain_block        gainBlock[2];

    DECLARE_ALIGNED_16(float, spectrum[1024]);
    DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);

    float             delayBuf1[46]; ///<qmf delay buffers
    float             delayBuf2[46];
    float             delayBuf3[46];
} channel_unit;

typedef struct {
    GetBitContext       gb;
    //@{
    /** stream data */
    int                 channels;
    int                 codingMode;
    int                 bit_rate;
    int                 sample_rate;
    int                 samples_per_channel;
    int                 samples_per_frame;

    int                 bits_per_frame;
    int                 bytes_per_frame;
    int                 pBs;
    channel_unit*       pUnits;
    //@}
    //@{
    /** joint-stereo related variables */
    int                 matrix_coeff_index_prev[4];
    int                 matrix_coeff_index_now[4];
    int                 matrix_coeff_index_next[4];
    int                 weighting_delay[6];
    //@}
    //@{
    /** data buffers */
    float               outSamples[2048];
    uint8_t*            decoded_bytes_buffer;
    float               tempBuf[1070];
    DECLARE_ALIGNED_16(float,mdct_tmp[512]);
    //@}
    //@{
    /** extradata */
    int                 atrac3version;
    int                 delay;
    int                 scrambled_stream;
    int                 frame_factor;
    //@}
} ATRAC3Context;

static DECLARE_ALIGNED_16(float,mdct_window[512]);
static float            qmf_window[48];
static VLC              spectral_coeff_tab[7];
static float            SFTable[64];
static float            gain_tab1[16];
static float            gain_tab2[31];
static MDCTContext      mdct_ctx;
static DSPContext       dsp;


/* quadrature mirror synthesis filter */

/**
 * Quadrature mirror synthesis filter.
 *
 * @param inlo      lower part of spectrum
 * @param inhi      higher part of spectrum
 * @param nIn       size of spectrum buffer
 * @param pOut      out buffer
 * @param delayBuf  delayBuf buffer
 * @param temp      temp buffer
 */


static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
{
    int   i, j;
    float   *p1, *p3;

    memcpy(temp, delayBuf, 46*sizeof(float));

    p3 = temp + 46;

    /* loop1 */
    for(i=0; i<nIn; i+=2){
        p3[2*i+0] = inlo[i  ] + inhi[i  ];
        p3[2*i+1] = inlo[i  ] - inhi[i  ];
        p3[2*i+2] = inlo[i+1] + inhi[i+1];
        p3[2*i+3] = inlo[i+1] - inhi[i+1];
    }

    /* loop2 */
    p1 = temp;
    for (j = nIn; j != 0; j--) {
        float s1 = 0.0;
        float s2 = 0.0;

        for (i = 0; i < 48; i += 2) {
            s1 += p1[i] * qmf_window[i];
            s2 += p1[i+1] * qmf_window[i+1];
        }

        pOut[0] = s2;
        pOut[1] = s1;

        p1 += 2;
        pOut += 2;
    }

    /* Update the delay buffer. */
    memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
}

/**
 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
 * caused by the reverse spectra of the QMF.
 *
 * @param pInput    float input
 * @param pOutput   float output
 * @param odd_band  1 if the band is an odd band
 * @param mdct_tmp  aligned temporary buffer for the mdct
 */

static void IMLT(float *pInput, float *pOutput, int odd_band, float* mdct_tmp)
{
    int     i;

    if (odd_band) {
        /**
        * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
        * or it gives better compression to do it this way.
        * FIXME: It should be possible to handle this in ff_imdct_calc
        * for that to happen a modification of the prerotation step of
        * all SIMD code and C code is needed.
        * Or fix the functions before so they generate a pre reversed spectrum.
        */

        for (i=0; i<128; i++)
            FFSWAP(float, pInput[i], pInput[255-i]);
    }

    mdct_ctx.fft.imdct_calc(&mdct_ctx,pOutput,pInput,mdct_tmp);

    /* Perform windowing on the output. */
    dsp.vector_fmul(pOutput,mdct_window,512);

}


/**
 * Atrac 3 indata descrambling, only used for data coming from the rm container
 *
 * @param in        pointer to 8 bit array of indata
 * @param bits      amount of bits
 * @param out       pointer to 8 bit array of outdata
 */

static int decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){
    int i, off;
    uint32_t c;
    uint32_t* buf;
    uint32_t* obuf = (uint32_t*) out;

    off = (int)((long)inbuffer & 3);
    buf = (uint32_t*) (inbuffer - off);
    c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
    bytes += 3 + off;
    for (i = 0; i < bytes/4; i++)
        obuf[i] = c ^ buf[i];

    if (off)
        av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);

    return off;
}


static void init_atrac3_transforms(ATRAC3Context *q) {
    float enc_window[256];
    float s;
    int i;

    /* Generate the mdct window, for details see
     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
    for (i=0 ; i<256; i++)
        enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;

    if (!mdct_window[0])
        for (i=0 ; i<256; i++) {
            mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
            mdct_window[511-i] = mdct_window[i];
        }

    /* Generate the QMF window. */
    for (i=0 ; i<24; i++) {
        s = qmf_48tap_half[i] * 2.0;
        qmf_window[i] = s;
        qmf_window[47 - i] = s;
    }

    /* Initialize the MDCT transform. */
    ff_mdct_init(&mdct_ctx, 9, 1);
}

/**
 * Atrac3 uninit, free all allocated memory
 */

static int atrac3_decode_close(AVCodecContext *avctx)
{
    ATRAC3Context *q = avctx->priv_data;

    av_free(q->pUnits);
    av_free(q->decoded_bytes_buffer);

    return 0;
}

/**
/ * Mantissa decoding
 *
 * @param gb            the GetBit context
 * @param selector      what table is the output values coded with
 * @param codingFlag    constant length coding or variable length coding
 * @param mantissas     mantissa output table
 * @param numCodes      amount of values to get
 */

static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
{
    int   numBits, cnt, code, huffSymb;

    if (selector == 1)
        numCodes /= 2;

    if (codingFlag != 0) {
        /* constant length coding (CLC) */
        //FIXME we don't have any samples coded in CLC mode
        numBits = CLCLengthTab[selector];

        if (selector > 1) {
            for (cnt = 0; cnt < numCodes; cnt++) {
                if (numBits)
                    code = get_sbits(gb, numBits);
                else
                    code = 0;
                mantissas[cnt] = code;
            }
        } else {
            for (cnt = 0; cnt < numCodes; cnt++) {
                if (numBits)
                    code = get_bits(gb, numBits); //numBits is always 4 in this case
                else
                    code = 0;
                mantissas[cnt*2] = seTab_0[code >> 2];
                mantissas[cnt*2+1] = seTab_0[code & 3];
            }
        }
    } else {
        /* variable length coding (VLC) */
        if (selector != 1) {
            for (cnt = 0; cnt < numCodes; cnt++) {
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
                huffSymb += 1;
                code = huffSymb >> 1;
                if (huffSymb & 1)
                    code = -code;
                mantissas[cnt] = code;
            }
        } else {
            for (cnt = 0; cnt < numCodes; cnt++) {
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
                mantissas[cnt*2] = decTable1[huffSymb*2];
                mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
            }
        }
    }
}

/**
 * Restore the quantized band spectrum coefficients
 *
 * @param gb            the GetBit context
 * @param pOut          decoded band spectrum
 * @return outSubbands   subband counter, fix for broken specification/files
 */