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Diffstat (limited to 'libavcodec/atrac3.c')
| -rw-r--r-- | libavcodec/atrac3.c | 1069 |
1 files changed, 1069 insertions, 0 deletions
diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c new file mode 100644 index 0000000000..4d676c1494 --- /dev/null +++ b/libavcodec/atrac3.c @@ -0,0 +1,1069 @@ +/* + * 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 + */ + +static int decodeSpectrum (GetBitContext *gb, float *pOut) +{ + int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn; + int subband_vlc_index[32], SF_idxs[32]; + int mantissas[128]; + float SF; + + numSubbands = get_bits(gb, 5); // number of coded subbands + codingMode = get_bits(gb, 1); // coding Mode: 0 - VLC/ 1-CLC + + /* Get the VLC selector table for the subbands, 0 means not coded. */ + for (cnt = 0; cnt <= numSubbands; cnt++) + subband_vlc_index[cnt] = get_bits(gb, 3); + + /* Read the scale factor indexes from the stream. */ + for (cnt = 0; cnt <= numSubbands; cnt++) { + if (subband_vlc_index[cnt] != 0) + SF_idxs[cnt] = get_bits(gb, 6); + } + + for (cnt = 0; cnt <= numSubbands; cnt++) { + first = subbandTab[cnt]; + last = subbandTab[cnt+1]; + + subbWidth = last - first; + + if (subband_vlc_index[cnt] != 0) { + /* Decode spectral coefficients for this subband. */ + /* TODO: This can be done faster is several blocks share the + * same VLC selector (subband_vlc_index) */ + readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth); + + /* Decode the scale factor for this subband. */ + SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]]; + + /* Inverse quantize the coefficients. */ + for (pIn=mantissas ; first<last; first++, pIn++) + pOut[first] = *pIn * SF; + } else { + /* This subband was not coded, so zero the entire subband. */ + memset(pOut+first, 0, subbWidth*sizeof(float)); + } + } + + /* Clear the subbands that were not coded. */ + first = subbandTab[cnt]; + memset(pOut+first, 0, (1024 - first) * sizeof(float)); + return numSubbands; +} + +/** + * Restore the quantized tonal components + * + * @param gb the GetBit context + * @param numComponents tonal components to report back + * @param pComponent tone component + * @param numBands amount of coded bands + */ + +static int decodeTonalComponents (GetBitContext *gb, int *numComponents, tonal_component *pComponent, int numBands) +{ + int i,j,k,cnt; + int component_count, components, coding_mode_selector, coding_mode, coded_values_per_component; + int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components; + int band_flags[4], mantissa[8]; + float *pCoef; + float scalefactor; + + component_count = 0; + *numComponents = 0; + + components = get_bits(gb,5); + + /* no tonal components */ + if (components == 0) + return 0; + + coding_mode_selector = get_bits(gb,2); + if (coding_mode_selector == 2) + return -1; + + coding_mode = coding_mode_selector & 1; + + for (i = 0; i < components; i++) { + for (cnt = 0; cnt <= numBands; cnt++) + band_flags[cnt] = get_bits1(gb); + + coded_values_per_component = get_bits(gb,3); + + quant_step_index = get_bits(gb,3); + if (quant_step_index <= 1) + return -1; + + if (coding_mode_selector == 3) + coding_mode = get_bits1(gb); + + for (j = 0; j < (numBands + 1) * 4; j++) { + if (band_flags[j >> 2] == 0) + continue; + + coded_components = get_bits(gb,3); + + for (k=0; k<coded_components; k++) { + sfIndx = get_bits(gb,6); + pComponent[component_count].pos = j * 64 + (get_bits(gb,6)); + max_coded_values = 1024 - pComponent[component_count].pos; + coded_values = coded_values_per_component + 1; + coded_values = FFMIN(max_coded_values,coded_values); + + scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index]; + + readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values); + + pComponent[component_count].numCoefs = coded_values; + + /* inverse quant */ + pCoef = pComponent[k].coef; + for (cnt = 0; cnt < coded_values; cnt++) + pCoef[cnt] = mantissa[cnt] * scalefactor; + + component_count++; + } + } + } + + *numComponents = component_count; + + return 0; +} + +/** + * Decode gain parameters for the coded bands + * + * @param gb the GetBit context + * @param pGb the gainblock for the current band + * @param numBands amount of coded bands + */ + +static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands) +{ + int i, cf, numData; + int *pLevel, *pLoc; + + gain_info *pGain = pGb->gBlock; + + for (i=0 ; i<=numBands; i++) + { + numData = get_bits(gb,3); + pGain[i].num_gain_data = numData; + pLevel = pGain[i].levcode; + pLoc = pGain[i].loccode; + + for (cf = 0; cf < numData; cf++){ + pLevel[cf]= get_bits(gb,4); + pLoc [cf]= get_bits(gb,5); + if(cf && pLoc[cf] <= pLoc[cf-1]) + return -1; + } + } + + /* Clear the unused blocks. */ + for (; i<4 ; i++) + pGain[i].num_gain_data = 0; + + return 0; +} + +/** + * Apply gain parameters and perform the MDCT overlapping part + * + * @param pIn input float buffer + * @param pPrev previous float buffer to perform overlap against + * @param pOut output float buffer + * @param pGain1 current band gain info + * @param pGain2 next band gain info + */ + +static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2) +{ + /* gain compensation function */ + float gain1, gain2, gain_inc; + int cnt, numdata, nsample, startLoc, endLoc; + + + if (pGain2->num_gain_data == 0) + gain1 = 1.0; + else + gain1 = gain_tab1[pGain2->levcode[0]]; + + if (pGain1->num_gain_data == 0) { + for (cnt = 0; cnt < 256; cnt++) + pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt]; + } else { + numdata = pGain1->num_gain_data; + pGain1->loccode[numdata] = 32; + pGain1->levcode[numdata] = 4; + + nsample = 0; // current sample = 0 + + for (cnt = 0; cnt < numdata; cnt++) { + startLoc = pGain1->loccode[cnt] * 8; + endLoc = startLoc + 8; + + gain2 = gain_tab1[pGain1->levcode[cnt]]; + gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15]; + + /* interpolate */ + for (; nsample < startLoc; nsample++) + pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; + + /* interpolation is done over eight samples */ + for (; nsample < endLoc; nsample++) { + pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; + gain2 *= gain_inc; + } + } + + for (; nsample < 256; nsample++) + pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample]; + } + + /* Delay for the overlapping part. */ + memcpy(pPrev, &pIn[256], 256*sizeof(float)); +} + +/** + * Combine the tonal band spectrum and regular band spectrum + * + * @param pSpectrum output spectrum buffer + * @param numComponents amount of tonal components + * @param pComponent tonal components for this band + */ + +static void addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent) +{ + int cnt, i; + float *pIn, *pOut; + + for (cnt = 0; cnt < numComponents; cnt++){ + pIn = pComponent[cnt].coef; + pOut = &(pSpectrum[pComponent[cnt].pos]); + + for (i=0 ; i<pComponent[cnt].numCoefs ; i++) + pOut[i] += pIn[i]; + } +} + + +#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, 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; + + result = decodeTonalComponents (gb, &pSnd->numComponents, pSnd->components, pSnd->bandsCoded); + if (result) return result; + + numSubbands = decodeSpectrum (gb, pSnd->spectrum); + + /* Merge the decoded spectrum and tonal components. */ + addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components); + + + /* Convert number of subbands into number of MLT/QMF bands */ + numBands = (subbandTab[numSubbands] - 1) >> 8; + + + /* 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,q->mdct_tmp); + } 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, uint8_t* databuf) +{ + int result, i; + float *p1, *p2, *p3, *p4; + uint8_t *ptr1, *ptr2; + + 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. */ + ptr1 = databuf; + ptr2 = databuf+q->bytes_per_frame-1; + for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) { + FFSWAP(uint8_t,*ptr1,*ptr2); + } + + /* Skip the sync codes (0xF8). */ + ptr1 = databuf; + 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_bits(&q->gb,1); + 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 avctx pointer to the AVCodecContext + */ + +static int atrac3_decode_frame(AVCodecContext *avctx, + void *data, int *data_size, + uint8_t *buf, int buf_size) { + ATRAC3Context *q = avctx->priv_data; + int result = 0, i; + uint8_t* databuf; + int16_t* samples = data; + + if (buf_size < avctx->block_align) + return buf_size; + + /* Check if we need to descramble and what buffer to pass on. */ + if (q->scrambled_stream) { + decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align); + databuf = q->decoded_bytes_buffer; + } else { + databuf = buf; + } + + result = decodeFrame(q, databuf); + + if (result != 0) { + av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n"); + return -1; + } + + if (q->channels == 1) { + /* mono */ + for (i = 0; i<1024; i++) + samples[i] = av_clip(round(q->outSamples[i]), -32768, 32767); + *data_size = 1024 * sizeof(int16_t); + } else { + /* stereo */ + for (i = 0; i < 1024; i++) { + samples[i*2] = av_clip(round(q->outSamples[i]), -32768, 32767); + samples[i*2+1] = av_clip(round(q->outSamples[1024+i]), -32768, 32767); + } + *data_size = 2048 * sizeof(int16_t); + } + + return avctx->block_align; +} + + +/** + * Atrac3 initialization + * + * @param avctx pointer to the AVCodecContext + */ + +static int atrac3_decode_init(AVCodecContext *avctx) +{ + int i; + uint8_t *edata_ptr = avctx->extradata; + ATRAC3Context *q = avctx->priv_data; + + /* Take data from the AVCodecContext (RM container). */ + q->sample_rate = avctx->sample_rate; + q->channels = avctx->channels; + q->bit_rate = avctx->bit_rate; + q->bits_per_frame = avctx->block_align * 8; + q->bytes_per_frame = avctx->block_align; + + /* Take care of the codec-specific extradata. */ + if (avctx->extradata_size == 14) { + /* Parse the extradata, WAV format */ + av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1 + q->samples_per_channel = bytestream_get_le32(&edata_ptr); + q->codingMode = bytestream_get_le16(&edata_ptr); + av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode + q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1 + av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0 + + /* setup */ + q->samples_per_frame = 1024 * q->channels; + q->atrac3version = 4; + q->delay = 0x88E; + if (q->codingMode) + q->codingMode = JOINT_STEREO; + else + q->codingMode = STEREO; + + q->scrambled_stream = 0; + + if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) { + } else { + av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor); + return -1; + } + + } else if (avctx->extradata_size == 10) { + /* Parse the extradata, RM format. */ + q->atrac3version = bytestream_get_be32(&edata_ptr); + q->samples_per_frame = bytestream_get_be16(&edata_ptr); + q->delay = bytestream_get_be16(&edata_ptr); + q->codingMode = bytestream_get_be16(&edata_ptr); + + q->samples_per_channel = q->samples_per_frame / q->channels; + q->scrambled_stream = 1; + + } else { + av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size); + } + /* Check the extradata. */ + + if (q->atrac3version != 4) { + av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version); + return -1; + } + + if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) { + av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame); + return -1; + } + + if (q->delay != 0x88E) { + av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay); + return -1; + } + + if (q->codingMode == STEREO) { + av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n"); + } else if (q->codingMode == JOINT_STEREO) { + av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n"); + } else { + av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode); + return -1; + } + + if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) { + av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n"); + return -1; + } + + + if(avctx->block_align >= UINT_MAX/2) + return -1; + + /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE, + * this is for the bitstream reader. */ + if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL) + return -1; + + + /* Initialize the VLC tables. */ + for (i=0 ; i<7 ; 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_STATIC); + } + + 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, avctx); + + q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels); + + return 0; +} + + +AVCodec atrac3_decoder = +{ + .name = "atrac 3", + .type = CODEC_TYPE_AUDIO, + .id = CODEC_ID_ATRAC3, + .priv_data_size = sizeof(ATRAC3Context), + .init = atrac3_decode_init, + .close = atrac3_decode_close, + .decode = atrac3_decode_frame, +}; |