| /* |
| * Atrac 3 compatible decoder |
| * Copyright (c) 2006-2008 Maxim Poliakovski |
| * Copyright (c) 2006-2008 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 |
| * Atrac 3 compatible decoder. |
| * This decoder handles Sony's ATRAC3 data. |
| * |
| * Container formats used to store atrac 3 data: |
| * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3). |
| * |
| * To use this decoder, a calling application must supply the extradata |
| * bytes provided in the containers above. |
| */ |
| |
| #include <math.h> |
| #include <stddef.h> |
| #include <stdio.h> |
| |
| #include "avcodec.h" |
| #include "get_bits.h" |
| #include "dsputil.h" |
| #include "bytestream.h" |
| #include "fft.h" |
| #include "fmtconvert.h" |
| |
| #include "atrac.h" |
| #include "atrac3data.h" |
| |
| #define JOINT_STEREO 0x12 |
| #define STEREO 0x2 |
| |
| #define SAMPLES_PER_FRAME 1024 |
| #define MDCT_SIZE 512 |
| |
| /* 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[SAMPLES_PER_FRAME]; |
| int gcBlkSwitch; |
| gain_block gainBlock[2]; |
| |
| DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME]; |
| DECLARE_ALIGNED(32, float, IMDCT_buf)[SAMPLES_PER_FRAME]; |
| |
| float delayBuf1[46]; ///<qmf delay buffers |
| float delayBuf2[46]; |
| float delayBuf3[46]; |
| } channel_unit; |
| |
| typedef struct { |
| AVFrame frame; |
| 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[2]; |
| uint8_t* decoded_bytes_buffer; |
| float tempBuf[1070]; |
| //@} |
| //@{ |
| /** extradata */ |
| int atrac3version; |
| int delay; |
| int scrambled_stream; |
| int frame_factor; |
| //@} |
| |
| FFTContext mdct_ctx; |
| FmtConvertContext fmt_conv; |
| } ATRAC3Context; |
| |
| static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE]; |
| static VLC spectral_coeff_tab[7]; |
| static float gain_tab1[16]; |
| static float gain_tab2[31]; |
| static DSPContext dsp; |
| |
| |
| /** |
| * 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 |
| */ |
| |
| static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band) |
| { |
| 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 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]); |
| } |
| |
| q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput); |
| |
| /* Perform windowing on the output. */ |
| dsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE); |
| |
| } |
| |
| |
| /** |
| * Atrac 3 indata descrambling, only used for data coming from the rm container |
| * |
| * @param inbuffer pointer to 8 bit array of indata |
| * @param out pointer to 8 bit array of outdata |
| * @param bytes amount of bytes |
| */ |
| |
| 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 = (intptr_t)inbuffer & 3; |
| buf = (const uint32_t *)(inbuffer - off); |
| if (off) |
| c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8)))); |
| else |
| c = av_be2ne32(0x537F6103U); |
| bytes += 3 + off; |
| for (i = 0; i < bytes/4; i++) |
| obuf[i] = c ^ buf[i]; |
| |
| if (off) |
| av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off); |
| |
| return off; |
| } |
| |
| |
| static av_cold int init_atrac3_transforms(ATRAC3Context *q, int is_float) { |
| float enc_window[256]; |
| 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]; |
| } |
| |
| /* Initialize the MDCT transform. */ |
| return ff_mdct_init(&q->mdct_ctx, 9, 1, is_float ? 1.0 / 32768 : 1.0); |
| } |
| |
| /** |
| * Atrac3 uninit, free all allocated memory |
| */ |
| |
| static av_cold int atrac3_decode_close(AVCodecContext *avctx) |
| { |
| ATRAC3Context *q = avctx->priv_data; |
| |
| av_free(q->pUnits); |
| av_free(q->decoded_bytes_buffer); |
| av_freep(&q->outSamples[0]); |
| |
| ff_mdct_end(&q->mdct_ctx); |
| |
| 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) */ |
| 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 = ff_atrac_sf_table[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, (SAMPLES_PER_FRAME - 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 AVERROR_INVALIDDATA; |
| |
| 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 AVERROR_INVALIDDATA; |
| |
| 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); |
| if (component_count >= 64) |
| return AVERROR_INVALIDDATA; |
| pComponent[component_count].pos = j * 64 + (get_bits(gb,6)); |
| max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos; |
| coded_values = coded_values_per_component + 1; |
| coded_values = FFMIN(max_coded_values,coded_values); |
| |
| scalefactor = ff_atrac_sf_table[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[component_count].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 AVERROR_INVALIDDATA; |
| } |
| } |
| |
| /* 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 AVERROR_INVALIDDATA; |
| } |
| } else { |
| if (get_bits(gb,6) != 0x28) { |
| av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| } |
| |
| /* 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(q, &(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, |
| float **out_samples) |
| { |
| 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, out_samples[0], 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 AVERROR_INVALIDDATA; |
| } |
| |
| |
| /* 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], out_samples[1], 1, JOINT_STEREO); |
| if (result != 0) |
| return result; |
| |
| /* Reconstruct the channel coefficients. */ |
| reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now); |
| |
| channelWeighting(out_samples[0], out_samples[1], 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], out_samples[i], i, q->codingMode); |
| if (result != 0) |
| return result; |
| } |
| } |
| |
| /* Apply the iQMF synthesis filter. */ |
| for (i=0 ; i<q->channels ; i++) { |
| p1 = out_samples[i]; |
| p2= p1+256; |
| p3= p2+256; |
| p4= p3+256; |
| atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf); |
| atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf); |
| atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf); |
| } |
| |
| return 0; |
| } |
| |
| |
| /** |
| * Atrac frame decoding |
| * |
| * @param avctx pointer to the AVCodecContext |
| */ |
| |
| static int atrac3_decode_frame(AVCodecContext *avctx, void *data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| const uint8_t *buf = avpkt->data; |
| int buf_size = avpkt->size; |
| ATRAC3Context *q = avctx->priv_data; |
| int result; |
| const uint8_t* databuf; |
| float *samples_flt; |
| int16_t *samples_s16; |
| |
| if (buf_size < avctx->block_align) { |
| av_log(avctx, AV_LOG_ERROR, |
| "Frame too small (%d bytes). Truncated file?\n", buf_size); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| /* get output buffer */ |
| q->frame.nb_samples = SAMPLES_PER_FRAME; |
| if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) { |
| av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); |
| return result; |
| } |
| samples_flt = (float *)q->frame.data[0]; |
| samples_s16 = (int16_t *)q->frame.data[0]; |
| |
| /* 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; |
| } |
| |
| if (q->channels == 1 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) |
| result = decodeFrame(q, databuf, &samples_flt); |
| else |
| result = decodeFrame(q, databuf, q->outSamples); |
| |
| if (result != 0) { |
| av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n"); |
| return result; |
| } |
| |
| /* interleave */ |
| if (q->channels == 2 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) { |
| q->fmt_conv.float_interleave(samples_flt, |
| (const float **)q->outSamples, |
| SAMPLES_PER_FRAME, 2); |
| } else if (avctx->sample_fmt == AV_SAMPLE_FMT_S16) { |
| q->fmt_conv.float_to_int16_interleave(samples_s16, |
| (const float **)q->outSamples, |
| SAMPLES_PER_FRAME, q->channels); |
| } |
| |
| *got_frame_ptr = 1; |
| *(AVFrame *)data = q->frame; |
| |
| return avctx->block_align; |
| } |
| |
| |
| /** |
| * Atrac3 initialization |
| * |
| * @param avctx pointer to the AVCodecContext |
| */ |
| |
| static av_cold int atrac3_decode_init(AVCodecContext *avctx) |
| { |
| int i, ret; |
| const uint8_t *edata_ptr = avctx->extradata; |
| ATRAC3Context *q = avctx->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 = 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 = SAMPLES_PER_FRAME * 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 AVERROR_INVALIDDATA; |
| } |
| |
| } 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 AVERROR_INVALIDDATA; |
| } |
| |
| if (q->samples_per_frame != SAMPLES_PER_FRAME && q->samples_per_frame != SAMPLES_PER_FRAME*2) { |
| av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (q->delay != 0x88E) { |
| av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| 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 AVERROR_INVALIDDATA; |
| } |
| |
| 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 AVERROR(EINVAL); |
| } |
| |
| |
| if(avctx->block_align >= UINT_MAX/2) |
| return AVERROR(EINVAL); |
| |
| /* 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. */ |
| 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; |
| } |
| |
| if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) |
| avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
| else |
| avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
| |
| if ((ret = init_atrac3_transforms(q, avctx->sample_fmt == AV_SAMPLE_FMT_FLT))) { |
| av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); |
| av_freep(&q->decoded_bytes_buffer); |
| return ret; |
| } |
| |
| atrac_generate_tables(); |
| |
| /* 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); |
| ff_fmt_convert_init(&q->fmt_conv, avctx); |
| |
| q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels); |
| if (!q->pUnits) { |
| atrac3_decode_close(avctx); |
| return AVERROR(ENOMEM); |
| } |
| |
| if (avctx->channels > 1 || avctx->sample_fmt == AV_SAMPLE_FMT_S16) { |
| q->outSamples[0] = av_mallocz(SAMPLES_PER_FRAME * avctx->channels * sizeof(*q->outSamples[0])); |
| q->outSamples[1] = q->outSamples[0] + SAMPLES_PER_FRAME; |
| if (!q->outSamples[0]) { |
| atrac3_decode_close(avctx); |
| return AVERROR(ENOMEM); |
| } |
| } |
| |
| avcodec_get_frame_defaults(&q->frame); |
| avctx->coded_frame = &q->frame; |
| |
| return 0; |
| } |
| |
| |
| AVCodec ff_atrac3_decoder = |
| { |
| .name = "atrac3", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = CODEC_ID_ATRAC3, |
| .priv_data_size = sizeof(ATRAC3Context), |
| .init = atrac3_decode_init, |
| .close = atrac3_decode_close, |
| .decode = atrac3_decode_frame, |
| .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, |
| .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"), |
| }; |
