diff options
Diffstat (limited to 'contrib/ffmpeg/libavcodec/atrac3.c')
| -rw-r--r-- | contrib/ffmpeg/libavcodec/atrac3.c | 1068 |
1 files changed, 0 insertions, 1068 deletions
diff --git a/contrib/ffmpeg/libavcodec/atrac3.c b/contrib/ffmpeg/libavcodec/atrac3.c deleted file mode 100644 index de6b32836..000000000 --- a/contrib/ffmpeg/libavcodec/atrac3.c +++ /dev/null @@ -1,1068 +0,0 @@ -/* - * 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(const uint8_t* inbuffer, uint8_t* out, int bytes){ - int i, off; - uint32_t c; - const uint32_t* buf; - uint32_t* obuf = (uint32_t*) out; - - off = (int)((long)inbuffer & 3); - buf = (const uint32_t*) (inbuffer - off); - c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8)))); - bytes += 3 + off; - for (i = 0; i < bytes/4; i++) - obuf[i] = c ^ buf[i]; - - if (off) - av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off); - - return off; -} - - -static 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_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC - - /* Get the VLC selector table for the subbands, 0 means not coded. */ - for (cnt = 0; cnt <= numSubbands; cnt++) - subband_vlc_index[cnt] = get_bits(gb, 3); - - /* Read the scale factor indexes from the stream. */ - for (cnt = 0; cnt <= numSubbands; cnt++) { - if (subband_vlc_index[cnt] != 0) - SF_idxs[cnt] = get_bits(gb, 6); - } - - for (cnt = 0; cnt <= numSubbands; cnt++) { - first = subbandTab[cnt]; - last = subbandTab[cnt+1]; - - subbWidth = last - first; - - if (subband_vlc_index[cnt] != 0) { - /* Decode spectral coefficients for this subband. */ - /* TODO: This can be done faster is several blocks share the - * same VLC selector (subband_vlc_index) */ - readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth); - - /* Decode the scale factor for this subband. */ - SF = 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 pComponent tone component - * @param numBands amount of coded bands - */ - -static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands) -{ - int i,j,k,cnt; - int components, coding_mode_selector, coding_mode, coded_values_per_component; - int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components; - int band_flags[4], mantissa[8]; - float *pCoef; - float scalefactor; - int component_count = 0; - - components = get_bits(gb,5); - - /* no tonal components */ - if (components == 0) - return 0; - - coding_mode_selector = get_bits(gb,2); - if (coding_mode_selector == 2) - return -1; - - coding_mode = coding_mode_selector & 1; - - for (i = 0; i < components; i++) { - for (cnt = 0; cnt <= numBands; cnt++) - band_flags[cnt] = get_bits1(gb); - - coded_values_per_component = get_bits(gb,3); - - quant_step_index = get_bits(gb,3); - if (quant_step_index <= 1) - return -1; - - if (coding_mode_selector == 3) - coding_mode = get_bits1(gb); - - for (j = 0; j < (numBands + 1) * 4; j++) { - if (band_flags[j >> 2] == 0) - continue; - - coded_components = get_bits(gb,3); - - for (k=0; k<coded_components; k++) { - sfIndx = get_bits(gb,6); - pComponent[component_count].pos = j * 64 + (get_bits(gb,6)); - max_coded_values = 1024 - pComponent[component_count].pos; - coded_values = coded_values_per_component + 1; - coded_values = FFMIN(max_coded_values,coded_values); - - scalefactor = 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++; - } - } - } - - return component_count; -} - -/** - * Decode gain parameters for the coded bands - * - * @param gb the GetBit context - * @param pGb the gainblock for the current band - * @param numBands amount of coded bands - */ - -static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands) -{ - int i, cf, numData; - int *pLevel, *pLoc; - - gain_info *pGain = pGb->gBlock; - - for (i=0 ; i<=numBands; i++) - { - numData = get_bits(gb,3); - pGain[i].num_gain_data = numData; - pLevel = pGain[i].levcode; - pLoc = pGain[i].loccode; - - for (cf = 0; cf < numData; cf++){ - pLevel[cf]= get_bits(gb,4); - pLoc [cf]= get_bits(gb,5); - if(cf && pLoc[cf] <= pLoc[cf-1]) - return -1; - } - } - - /* Clear the unused blocks. */ - for (; i<4 ; i++) - pGain[i].num_gain_data = 0; - - return 0; -} - -/** - * Apply gain parameters and perform the MDCT overlapping part - * - * @param pIn input float buffer - * @param pPrev previous float buffer to perform overlap against - * @param pOut output float buffer - * @param pGain1 current band gain info - * @param pGain2 next band gain info - */ - -static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2) -{ - /* gain compensation function */ - float gain1, gain2, gain_inc; - int cnt, numdata, nsample, startLoc, endLoc; - - - if (pGain2->num_gain_data == 0) - gain1 = 1.0; - else - gain1 = gain_tab1[pGain2->levcode[0]]; - - if (pGain1->num_gain_data == 0) { - for (cnt = 0; cnt < 256; cnt++) - pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt]; - } else { - numdata = pGain1->num_gain_data; - pGain1->loccode[numdata] = 32; - pGain1->levcode[numdata] = 4; - - nsample = 0; // current sample = 0 - - for (cnt = 0; cnt < numdata; cnt++) { - startLoc = pGain1->loccode[cnt] * 8; - endLoc = startLoc + 8; - - gain2 = gain_tab1[pGain1->levcode[cnt]]; - gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15]; - - /* interpolate */ - for (; nsample < startLoc; nsample++) - pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; - - /* interpolation is done over eight samples */ - for (; nsample < endLoc; nsample++) { - pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; - gain2 *= gain_inc; - } - } - - for (; nsample < 256; nsample++) - pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample]; - } - - /* Delay for the overlapping part. */ - memcpy(pPrev, &pIn[256], 256*sizeof(float)); -} - -/** - * Combine the tonal band spectrum and regular band spectrum - * - * @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; - - 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. */ - 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_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 avctx pointer to the AVCodecContext - */ - -static int atrac3_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - const 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_int16(round(q->outSamples[i])); - *data_size = 1024 * sizeof(int16_t); - } else { - /* stereo */ - for (i = 0; i < 1024; i++) { - samples[i*2] = av_clip_int16(round(q->outSamples[i])); - samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i])); - } - *data_size = 2048 * sizeof(int16_t); - } - - return avctx->block_align; -} - - -/** - * Atrac3 initialization - * - * @param avctx pointer to the AVCodecContext - */ - -static int atrac3_decode_init(AVCodecContext *avctx) -{ - int i; - const 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 AVERROR(ENOMEM); - - - /* 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); - if (!q->pUnits) { - av_free(q->decoded_bytes_buffer); - return AVERROR(ENOMEM); - } - - 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, -}; |