diff options
author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-10-17 19:59:40 +0400 |
---|---|---|
committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-10-22 18:10:06 +0400 |
commit | e55d53905f34f8e8747f6d321e9a695dc02ebb2f (patch) | |
tree | 3c220a61440d4304ec055bc51652c7177dc8e698 /libavcodec | |
parent | abdee95224fe6b6ba83212216a7f7f50c3471c42 (diff) |
atrac3: cosmetics: pretty-printing and renaming
also does some minor refactoring.
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/atrac3.c | 1059 | ||||
-rw-r--r-- | libavcodec/atrac3data.h | 98 |
2 files changed, 572 insertions, 585 deletions
diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c index 94b355a362..4223d04521 100644 --- a/libavcodec/atrac3.c +++ b/libavcodec/atrac3.c @@ -38,10 +38,10 @@ #include "libavutil/float_dsp.h" #include "avcodec.h" -#include "get_bits.h" #include "bytestream.h" #include "fft.h" #include "fmtconvert.h" +#include "get_bits.h" #include "atrac.h" #include "atrac3data.h" @@ -52,142 +52,130 @@ #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]; +typedef struct GainInfo { + int num_gain_data; + int lev_code[8]; + int loc_code[8]; +} GainInfo; + +typedef struct GainBlock { + GainInfo g_block[4]; +} GainBlock; + +typedef struct TonalComponent { + int pos; + int num_coefs; + float coef[8]; +} TonalComponent; + +typedef struct ChannelUnit { + int bands_coded; + int num_components; + float prev_frame[SAMPLES_PER_FRAME]; + int gc_blk_switch; + TonalComponent components[64]; + GainBlock gain_block[2]; DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME]; - DECLARE_ALIGNED(32, float, IMDCT_buf)[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; + float delay_buf1[46]; ///<qmf delay buffers + float delay_buf2[46]; + float delay_buf3[46]; +} ChannelUnit; -typedef struct { - AVFrame frame; - GetBitContext gb; +typedef struct ATRAC3Context { + 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; + int channels; + int coding_mode; + int bit_rate; + int sample_rate; + int samples_per_channel; + int samples_per_frame; + + int bits_per_frame; + int bytes_per_frame; + ChannelUnit *units; //@} //@{ /** 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]; + int matrix_coeff_index_prev[4]; + int matrix_coeff_index_now[4]; + int matrix_coeff_index_next[4]; + int weighting_delay[6]; //@} //@{ /** data buffers */ - uint8_t* decoded_bytes_buffer; - float tempBuf[1070]; + uint8_t *decoded_bytes_buffer; + float temp_buf[1070]; //@} //@{ /** extradata */ - int atrac3version; - int delay; - int scrambled_stream; - int frame_factor; + int version; + int delay; + int scrambled_stream; + int frame_factor; //@} - FFTContext mdct_ctx; - FmtConvertContext fmt_conv; - AVFloatDSPContext fdsp; + FFTContext mdct_ctx; + FmtConvertContext fmt_conv; + AVFloatDSPContext fdsp; } 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 VLC spectral_coeff_tab[7]; +static float gain_tab1[16]; +static float gain_tab2[31]; -/** - * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands - * caused by the reverse spectra of the QMF. +/* + * 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) +static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band) { - int i; + 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]); + * 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, input[i], input[255 - i]); } - q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput); + q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input); /* Perform windowing on the output. */ - q->fdsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE); - + q->fdsp.vector_fmul(output, output, 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 +/* + * indata descrambling, only used for data coming from the rm container */ - -static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){ +static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes) +{ int i, off; uint32_t c; - const uint32_t* buf; - uint32_t* obuf = (uint32_t*) out; + const uint32_t *buf; + uint32_t *output = (uint32_t *)out; - off = (intptr_t)inbuffer & 3; - buf = (const uint32_t*) (inbuffer - off); - c = av_be2ne32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8)))); + off = (intptr_t)input & 3; + buf = (const uint32_t *)(input - off); + c = av_be2ne32((0x537F6103 >> (off * 8)) | (0x537F6103 << (32 - (off * 8)))); bytes += 3 + off; - for (i = 0; i < bytes/4; i++) - obuf[i] = c ^ buf[i]; + for (i = 0; i < bytes / 4; i++) + output[i] = c ^ buf[i]; if (off) av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off); @@ -195,35 +183,34 @@ static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){ return off; } - -static av_cold int init_atrac3_transforms(ATRAC3Context *q) { +static av_cold int init_atrac3_transforms(ATRAC3Context *q) +{ float enc_window[256]; int i; - /* Generate the mdct window, for details see + /* generate the mdct window, for details see * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ - for (i=0 ; i<256; i++) + 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]; + 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. */ + /* initialize the MDCT transform */ return ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768); } -/** - * 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->units); av_free(q->decoded_bytes_buffer); ff_mdct_end(&q->mdct_ctx); @@ -231,192 +218,200 @@ static av_cold int atrac3_decode_close(AVCodecContext *avctx) return 0; } -/** -/ * Mantissa decoding +/* + * 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 + * @param selector which table the output values are coded with + * @param coding_flag constant length coding or variable length coding + * @param mantissas mantissa output table + * @param num_codes number of values to get */ - -static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes) +static void read_quant_spectral_coeffs(GetBitContext *gb, int selector, + int coding_flag, int *mantissas, + int num_codes) { - int numBits, cnt, code, huffSymb; + int i, code, huff_symb; if (selector == 1) - numCodes /= 2; + num_codes /= 2; - if (codingFlag != 0) { + if (coding_flag != 0) { /* constant length coding (CLC) */ - numBits = CLCLengthTab[selector]; + int num_bits = clc_length_tab[selector]; if (selector > 1) { - for (cnt = 0; cnt < numCodes; cnt++) { - if (numBits) - code = get_sbits(gb, numBits); + for (i = 0; i < num_codes; i++) { + if (num_bits) + code = get_sbits(gb, num_bits); else code = 0; - mantissas[cnt] = code; + mantissas[i] = code; } } else { - for (cnt = 0; cnt < numCodes; cnt++) { - if (numBits) - code = get_bits(gb, numBits); //numBits is always 4 in this case + for (i = 0; i < num_codes; i++) { + if (num_bits) + code = get_bits(gb, num_bits); // num_bits is always 4 in this case else code = 0; - mantissas[cnt*2] = seTab_0[code >> 2]; - mantissas[cnt*2+1] = seTab_0[code & 3]; + mantissas[i * 2 ] = mantissa_clc_tab[code >> 2]; + mantissas[i * 2 + 1] = mantissa_clc_tab[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) + for (i = 0; i < num_codes; i++) { + huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, + spectral_coeff_tab[selector-1].bits, 3); + huff_symb += 1; + code = huff_symb >> 1; + if (huff_symb & 1) code = -code; - mantissas[cnt] = code; + mantissas[i] = 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]; + for (i = 0; i < num_codes; i++) { + huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table, + spectral_coeff_tab[selector - 1].bits, 3); + mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ]; + mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 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 + * @return subband count, fix for broken specification/files */ - -static int decodeSpectrum (GetBitContext *gb, float *pOut) +static int decode_spectrum(GetBitContext *gb, float *output) { - 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); + int num_subbands, coding_mode, i, j, first, last, subband_size; + int subband_vlc_index[32], sf_index[32]; + int mantissas[128]; + float scale_factor; + + num_subbands = get_bits(gb, 5); // number of coded subbands + coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC + + /* get the VLC selector table for the subbands, 0 means not coded */ + for (i = 0; i <= num_subbands; i++) + subband_vlc_index[i] = get_bits(gb, 3); + + /* read the scale factor indexes from the stream */ + for (i = 0; i <= num_subbands; i++) { + if (subband_vlc_index[i] != 0) + sf_index[i] = get_bits(gb, 6); } - for (cnt = 0; cnt <= numSubbands; cnt++) { - first = subbandTab[cnt]; - last = subbandTab[cnt+1]; + for (i = 0; i <= num_subbands; i++) { + first = subband_tab[i ]; + last = subband_tab[i + 1]; - subbWidth = last - first; + subband_size = last - first; - if (subband_vlc_index[cnt] != 0) { - /* Decode spectral coefficients for this subband. */ + if (subband_vlc_index[i] != 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); + read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode, + mantissas, subband_size); - /* Decode the scale factor for this subband. */ - SF = ff_atrac_sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]]; + /* decode the scale factor for this subband */ + scale_factor = ff_atrac_sf_table[sf_index[i]] * + inv_max_quant[subband_vlc_index[i]]; - /* Inverse quantize the coefficients. */ - for (pIn=mantissas ; first<last; first++, pIn++) - pOut[first] = *pIn * SF; + /* inverse quantize the coefficients */ + for (j = 0; first < last; first++, j++) + output[first] = mantissas[j] * scale_factor; } else { - /* This subband was not coded, so zero the entire subband. */ - memset(pOut+first, 0, subbWidth*sizeof(float)); + /* this subband was not coded, so zero the entire subband */ + memset(output + first, 0, subband_size * sizeof(float)); } } - /* Clear the subbands that were not coded. */ - first = subbandTab[cnt]; - memset(pOut+first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float)); - return numSubbands; + /* clear the subbands that were not coded */ + first = subband_tab[i]; + memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float)); + return num_subbands; } -/** +/* * Restore the quantized tonal components * - * @param gb the GetBit context - * @param pComponent tone component - * @param numBands amount of coded bands + * @param components tonal components + * @param num_bands number of coded bands */ - -static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands) +static int decode_tonal_components(GetBitContext *gb, + TonalComponent *components, int num_bands) { - 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; + int i, b, c, m; + int nb_components, coding_mode_selector, coding_mode; + int band_flags[4], mantissa[8]; + int component_count = 0; - components = get_bits(gb,5); + nb_components = get_bits(gb, 5); /* no tonal components */ - if (components == 0) + if (nb_components == 0) return 0; - coding_mode_selector = get_bits(gb,2); + 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); + for (i = 0; i < nb_components; i++) { + int coded_values_per_component, quant_step_index; + + for (b = 0; b <= num_bands; b++) + band_flags[b] = get_bits1(gb); - coded_values_per_component = get_bits(gb,3); + coded_values_per_component = get_bits(gb, 3); - quant_step_index = 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) + for (b = 0; b < (num_bands + 1) * 4; b++) { + int coded_components; + + if (band_flags[b >> 2] == 0) continue; - coded_components = get_bits(gb,3); + coded_components = get_bits(gb, 3); + + for (c = 0; c < coded_components; c++) { + TonalComponent *cmp = &components[component_count]; + int sf_index, coded_values, max_coded_values; + float scale_factor; - for (k=0; k<coded_components; k++) { - sfIndx = get_bits(gb,6); + sf_index = 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]; + cmp->pos = b * 64 + get_bits(gb, 6); + + max_coded_values = SAMPLES_PER_FRAME - cmp->pos; + coded_values = coded_values_per_component + 1; + coded_values = FFMIN(max_coded_values, coded_values); - readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values); + scale_factor = ff_atrac_sf_table[sf_index] * + inv_max_quant[quant_step_index]; - pComponent[component_count].numCoefs = coded_values; + read_quant_spectral_coeffs(gb, quant_step_index, coding_mode, + mantissa, coded_values); + + cmp->num_coefs = coded_values; /* inverse quant */ - pCoef = pComponent[component_count].coef; - for (cnt = 0; cnt < coded_values; cnt++) - pCoef[cnt] = mantissa[cnt] * scalefactor; + for (m = 0; m < coded_values; m++) + cmp->coef[m] = mantissa[m] * scale_factor; component_count++; } @@ -426,334 +421,326 @@ static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent 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 + * @param block the gainblock for the current band + * @param num_bands amount of coded bands */ - -static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands) +static int decode_gain_control(GetBitContext *gb, GainBlock *block, + int num_bands) { - 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]) + int i, cf, num_data; + int *level, *loc; + + GainInfo *gain = block->g_block; + + for (i = 0; i <= num_bands; i++) { + num_data = get_bits(gb, 3); + gain[i].num_gain_data = num_data; + level = gain[i].lev_code; + loc = gain[i].loc_code; + + for (cf = 0; cf < gain[i].num_gain_data; cf++) { + level[cf] = get_bits(gb, 4); + loc [cf] = get_bits(gb, 5); + if (cf && loc[cf] <= loc[cf - 1]) return AVERROR_INVALIDDATA; } } /* Clear the unused blocks. */ - for (; i<4 ; i++) - pGain[i].num_gain_data = 0; + for (; i < 4 ; i++) + gain[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 + * @param input input buffer + * @param prev previous buffer to perform overlap against + * @param output output buffer + * @param gain1 current band gain info + * @param gain2 next band gain info */ - -static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2) +static void gain_compensate_and_overlap(float *input, float *prev, + float *output, GainInfo *gain1, + GainInfo *gain2) { - /* gain compensation function */ - float gain1, gain2, gain_inc; - int cnt, numdata, nsample, startLoc, endLoc; + float g1, g2, gain_inc; + int i, j, num_data, start_loc, end_loc; - if (pGain2->num_gain_data == 0) - gain1 = 1.0; + if (gain2->num_gain_data == 0) + g1 = 1.0; else - gain1 = gain_tab1[pGain2->levcode[0]]; + g1 = gain_tab1[gain2->lev_code[0]]; - if (pGain1->num_gain_data == 0) { - for (cnt = 0; cnt < 256; cnt++) - pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt]; + if (gain1->num_gain_data == 0) { + for (i = 0; i < 256; i++) + output[i] = input[i] * g1 + prev[i]; } else { - numdata = pGain1->num_gain_data; - pGain1->loccode[numdata] = 32; - pGain1->levcode[numdata] = 4; - - nsample = 0; // current sample = 0 + num_data = gain1->num_gain_data; + gain1->loc_code[num_data] = 32; + gain1->lev_code[num_data] = 4; - for (cnt = 0; cnt < numdata; cnt++) { - startLoc = pGain1->loccode[cnt] * 8; - endLoc = startLoc + 8; + for (i = 0, j = 0; i < num_data; i++) { + start_loc = gain1->loc_code[i] * 8; + end_loc = start_loc + 8; - gain2 = gain_tab1[pGain1->levcode[cnt]]; - gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15]; + g2 = gain_tab1[gain1->lev_code[i]]; + gain_inc = gain_tab2[gain1->lev_code[i + 1] - + gain1->lev_code[i ] + 15]; /* interpolate */ - for (; nsample < startLoc; nsample++) - pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; + for (; j < start_loc; j++) + output[j] = (input[j] * g1 + prev[j]) * g2; /* interpolation is done over eight samples */ - for (; nsample < endLoc; nsample++) { - pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2; - gain2 *= gain_inc; + for (; j < end_loc; j++) { + output[j] = (input[j] * g1 + prev[j]) * g2; + g2 *= gain_inc; } } - for (; nsample < 256; nsample++) - pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample]; + for (; j < 256; j++) + output[j] = input[j] * g1 + prev[j]; } /* Delay for the overlapping part. */ - memcpy(pPrev, &pIn[256], 256*sizeof(float)); + memcpy(prev, &input[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 + * @param spectrum output spectrum buffer + * @param num_components number of tonal components + * @param components tonal components for this band + * @return position of the last tonal coefficient */ - -static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent) +static int add_tonal_components(float *spectrum, int num_components, + TonalComponent *components) { - int cnt, i, lastPos = -1; - float *pIn, *pOut; + int i, j, last_pos = -1; + float *input, *output; - 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 < num_components; i++) { + last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos); + input = components[i].coef; + output = &spectrum[components[i].pos]; - for (i=0 ; i<pComponent[cnt].numCoefs ; i++) - pOut[i] += pIn[i]; + for (j = 0; j < components[i].num_coefs; j++) + output[i] += input[i]; } - return lastPos; + return last_pos; } +#define INTERPOLATE(old, new, nsample) \ + ((old) + (nsample) * 0.125 * ((new) - (old))) -#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old))) - -static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode) +static void reverse_matrixing(float *su1, float *su2, int *prev_code, + int *curr_code) { - int i, band, nsample, s1, s2; - float c1, c2; - float mc1_l, mc1_r, mc2_l, mc2_r; + int i, nsample, band; + 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]; + for (i = 0, band = 0; band < 4 * 256; band += 256, i++) { + int s1 = prev_code[i]; + int s2 = curr_code[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]; + mc1_l = matrix_coeffs[s1 * 2 ]; + mc1_r = matrix_coeffs[s1 * 2 + 1]; + mc2_l = matrix_coeffs[s2 * 2 ]; + mc2_r = matrix_coeffs[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; + for (; nsample < 8; nsample++) { + float c1 = su1[band + nsample]; + float 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); + case 0: /* M/S decoding */ + for (; nsample < 256; nsample++) { + float c1 = su1[band + nsample]; + float c2 = su2[band + nsample]; + su1[band + nsample] = c2 * 2.0; + su2[band + nsample] = (c1 - c2) * 2.0; + } + break; + case 1: + for (; nsample < 256; nsample++) { + float c1 = su1[band + nsample]; + float 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++) { + float c1 = su1[band + nsample]; + float 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) { +static void get_channel_weights(int index, int flag, float ch[2]) +{ + if (index == 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) + ch[0] = (index & 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) +static void channel_weighting(float *su1, float *su2, int *p3) { - int band, nsample; + 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]); + if (p3[1] != 7 || p3[3] != 7) { + get_channel_weights(p3[1], p3[0], w[0]); + get_channel_weights(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 (band = 1; band < 4; band++) { + 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]; + 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) + * @param snd the channel unit to be used + * @param output the decoded samples before IQMF in float representation + * @param channel_num channel number + * @param coding_mode 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) +static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb, + ChannelUnit *snd, float *output, + int channel_num, int coding_mode) { - int band, result=0, numSubbands, lastTonal, numBands; + int band, ret, num_subbands, last_tonal, num_bands; + GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch]; + GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch]; - if (codingMode == JOINT_STEREO && channelNum == 1) { - if (get_bits(gb,2) != 3) { + if (coding_mode == JOINT_STEREO && channel_num == 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) { + 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); + snd->bands_coded = get_bits(gb, 2); - result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded); - if (result) return result; + ret = decode_gain_control(gb, gain2, snd->bands_coded); + if (ret) + return ret; - pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded); - if (pSnd->numComponents == -1) return -1; + snd->num_components = decode_tonal_components(gb, snd->components, + snd->bands_coded); + if (snd->num_components == -1) + return -1; - numSubbands = decodeSpectrum (gb, pSnd->spectrum); + num_subbands = decode_spectrum(gb, snd->spectrum); /* Merge the decoded spectrum and tonal components. */ - lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components); + last_tonal = add_tonal_components(snd->spectrum, snd->num_components, + snd->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); + /* calculate number of used MLT/QMF bands according to the amount of coded + spectral lines */ + num_bands = (subband_tab[num_subbands] - 1) >> 8; + if (last_tonal >= 0) + num_bands = FFMAX((last_tonal + 256) >> 8, num_bands); /* Reconstruct time domain samples. */ - for (band=0; band<4; band++) { + 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)); + if (band <= num_bands) + imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1); + else + memset(snd->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]); + gain_compensate_and_overlap(snd->imdct_buf, + &snd->prev_frame[band * 256], + &output[band * 256], + &gain1->g_block[band], + &gain2->g_block[band]); } /* Swap the gain control buffers for the next frame. */ - pSnd->gcBlkSwitch ^= 1; + snd->gc_blk_switch ^= 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) +static int decode_frame(ATRAC3Context *q, const uint8_t *databuf, + float **out_samples) { - int result, i; - float *p1, *p2, *p3, *p4; + int ret, i; uint8_t *ptr1; - if (q->codingMode == JOINT_STEREO) { - + if (q->coding_mode == 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; + ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0, + JOINT_STEREO); + if (ret != 0) + return ret; /* 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); - } + 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; + 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--; } @@ -767,74 +754,69 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf, /* set the bitstream reader at the start of the second Sound Unit*/ - init_get_bits(&q->gb,ptr1,q->bits_per_frame); + 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)); + 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); + 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); + 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; + ret = decode_channel_sound_unit(q, &q->gb, &q->units[1], + out_samples[1], 1, JOINT_STEREO); + if (ret != 0) + return ret; /* 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); + reverse_matrixing(out_samples[0], out_samples[1], + q->matrix_coeff_index_prev, + q->matrix_coeff_index_now); + channel_weighting(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++) { - + 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; + ret = decode_channel_sound_unit(q, &q->gb, &q->units[i], + out_samples[i], i, q->coding_mode); + if (ret != 0) + return ret; } } /* 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; - ff_atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf); - ff_atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf); - ff_atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf); + for (i = 0; i < q->channels; i++) { + float *p1 = out_samples[i]; + float *p2 = p1 + 256; + float *p3 = p2 + 256; + float *p4 = p3 + 256; + ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf); + ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf); + ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf); } 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; + int ret; + const uint8_t *databuf; if (buf_size < avctx->block_align) { av_log(avctx, AV_LOG_ERROR, @@ -844,9 +826,9 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data, /* get output buffer */ q->frame.nb_samples = SAMPLES_PER_FRAME; - if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) { + if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - return result; + return ret; } /* Check if we need to descramble and what buffer to pass on. */ @@ -857,11 +839,10 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data, databuf = buf; } - result = decodeFrame(q, databuf, (float **)q->frame.extended_data); - - if (result != 0) { - av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n"); - return result; + ret = decode_frame(q, databuf, (float **)q->frame.extended_data); + if (ret) { + av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n"); + return ret; } *got_frame_ptr = 1; @@ -870,13 +851,6 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data, return avctx->block_align; } - -/** - * Atrac3 initialization - * - * @param avctx pointer to the AVCodecContext - */ - static av_cold int atrac3_decode_init(AVCodecContext *avctx) { int i, ret; @@ -886,101 +860,108 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx) 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->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 + 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 + q->coding_mode = 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); + q->version = 4; + q->delay = 0x88E; + q->coding_mode = q->coding_mode ? JOINT_STEREO : 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) { + 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->version = bytestream_get_be32(&edata_ptr); + q->samples_per_frame = bytestream_get_be16(&edata_ptr); + q->delay = bytestream_get_be16(&edata_ptr); + q->coding_mode = bytestream_get_be16(&edata_ptr); q->samples_per_channel = q->samples_per_frame / q->channels; - q->scrambled_stream = 1; + q->scrambled_stream = 1; } else { - av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size); + 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); + /* Check the extradata */ + + if (q->version != 4) { + av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", q->version); 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); + 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); + 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); + if (q->coding_mode == STEREO) + av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n"); + else if (q->coding_mode == 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->coding_mode); 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"); + if (avctx->channels <= 0 || avctx->channels > 2) { + av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n"); return AVERROR(EINVAL); } - - if(avctx->block_align >= UINT_MAX/2) + 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) + q->decoded_bytes_buffer = av_mallocz(avctx->block_align + + (4 - avctx->block_align % 4) + + FF_INPUT_BUFFER_PADDING_SIZE); + if (q->decoded_bytes_buffer == NULL) return AVERROR(ENOMEM); /* Initialize the VLC tables. */ if (!vlcs_initialized) { - for (i=0 ; i<7 ; i++) { + 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); + 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; } @@ -995,12 +976,12 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx) ff_atrac_generate_tables(); - /* Generate gain tables. */ - for (i=0 ; i<16 ; i++) - gain_tab1[i] = powf (2.0, (4 - i)); + /* 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); + 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; @@ -1010,17 +991,17 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx) q->weighting_delay[4] = 0; q->weighting_delay[5] = 7; - for (i=0; i<4; i++) { + for (i = 0; i < 4; i++) { q->matrix_coeff_index_prev[i] = 3; - q->matrix_coeff_index_now[i] = 3; + q->matrix_coeff_index_now[i] = 3; q->matrix_coeff_index_next[i] = 3; } avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); ff_fmt_convert_init(&q->fmt_conv, avctx); - q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels); - if (!q->pUnits) { + q->units = av_mallocz(sizeof(ChannelUnit) * q->channels); + if (!q->units) { atrac3_decode_close(avctx); return AVERROR(ENOMEM); } @@ -1031,18 +1012,16 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx) return 0; } - -AVCodec ff_atrac3_decoder = -{ - .name = "atrac3", - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_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)"), - .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, - AV_SAMPLE_FMT_NONE }, +AVCodec ff_atrac3_decoder = { + .name = "atrac3", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_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)"), + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_NONE }, }; diff --git a/libavcodec/atrac3data.h b/libavcodec/atrac3data.h index 9076d3ae8d..40518ab9da 100644 --- a/libavcodec/atrac3data.h +++ b/libavcodec/atrac3data.h @@ -33,101 +33,109 @@ /* VLC tables */ static const uint8_t huffcode1[9] = { - 0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F, + 0x0, 0x4, 0x5, 0xC, 0xD, 0x1C, 0x1D, 0x1E, 0x1F }; -static const uint8_t huffbits1[9] = { - 1,3,3,4,4,5,5,5,5, -}; +static const uint8_t huffbits1[9] = { 1, 3, 3, 4, 4, 5, 5, 5, 5 }; -static const uint8_t huffcode2[5] = { - 0x0,0x4,0x5,0x6,0x7, -}; +static const uint8_t huffcode2[5] = { 0x0, 0x4, 0x5, 0x6, 0x7 }; -static const uint8_t huffbits2[5] = { - 1,3,3,3,3, -}; +static const uint8_t huffbits2[5] = { 1, 3, 3, 3, 3 }; -static const uint8_t huffcode3[7] = { -0x0,0x4,0x5,0xC,0xD,0xE,0xF, -}; +static const uint8_t huffcode3[7] = { 0x0, 0x4, 0x5, 0xC, 0xD, 0xE, 0xF }; -static const uint8_t huffbits3[7] = { - 1,3,3,4,4,4,4, -}; +static const uint8_t huffbits3[7] = { 1, 3, 3, 4, 4, 4, 4 }; static const uint8_t huffcode4[9] = { - 0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F, + 0x0, 0x4, 0x5, 0xC, 0xD, 0x1C, 0x1D, 0x1E, 0x1F }; -static const uint8_t huffbits4[9] = { - 1,3,3,4,4,5,5,5,5, -}; +static const uint8_t huffbits4[9] = { 1, 3, 3, 4, 4, 5, 5, 5, 5 }; static const uint8_t huffcode5[15] = { - 0x0,0x2,0x3,0x8,0x9,0xA,0xB,0x1C,0x1D,0x3C,0x3D,0x3E,0x3F,0xC,0xD, + 0x00, 0x02, 0x03, 0x08, 0x09, 0x0A, 0x0B, 0x1C, + 0x1D, 0x3C, 0x3D, 0x3E, 0x3F, 0x0C, 0x0D }; static const uint8_t huffbits5[15] = { - 2,3,3,4,4,4,4,5,5,6,6,6,6,4,4 + 2, 3, 3, 4, 4, 4, 4, 5, 5, 6, 6, 6, 6, 4, 4 }; static const uint8_t huffcode6[31] = { - 0x0,0x2,0x3,0x4,0x5,0x6,0x7,0x14,0x15,0x16,0x17,0x18,0x19,0x34,0x35, - 0x36,0x37,0x38,0x39,0x3A,0x3B,0x78,0x79,0x7A,0x7B,0x7C,0x7D,0x7E,0x7F,0x8,0x9, + 0x00, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x14, + 0x15, 0x16, 0x17, 0x18, 0x19, 0x34, 0x35, 0x36, + 0x37, 0x38, 0x39, 0x3A, 0x3B, 0x78, 0x79, 0x7A, + 0x7B, 0x7C, 0x7D, 0x7E, 0x7F, 0x08, 0x09 }; static const uint8_t huffbits6[31] = { - 3,4,4,4,4,4,4,5,5,5,5,5,5,6,6,6,6,6,6,6,6,7,7,7,7,7,7,7,7,4,4 + 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 6, 6, 6, + 6, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 7, 4, 4 }; static const uint8_t huffcode7[63] = { - 0x0,0x8,0x9,0xA,0xB,0xC,0xD,0xE,0xF,0x10,0x11,0x24,0x25,0x26,0x27,0x28, - 0x29,0x2A,0x2B,0x2C,0x2D,0x2E,0x2F,0x30,0x31,0x32,0x33,0x68,0x69,0x6A,0x6B,0x6C, - 0x6D,0x6E,0x6F,0x70,0x71,0x72,0x73,0x74,0x75,0xEC,0xED,0xEE,0xEF,0xF0,0xF1,0xF2, - 0xF3,0xF4,0xF5,0xF6,0xF7,0xF8,0xF9,0xFA,0xFB,0xFC,0xFD,0xFE,0xFF,0x2,0x3, + 0x00, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, + 0x0F, 0x10, 0x11, 0x24, 0x25, 0x26, 0x27, 0x28, + 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F, 0x30, + 0x31, 0x32, 0x33, 0x68, 0x69, 0x6A, 0x6B, 0x6C, + 0x6D, 0x6E, 0x6F, 0x70, 0x71, 0x72, 0x73, 0x74, + 0x75, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, + 0xF3, 0xF4, 0xF5, 0xF6, 0xF7, 0xF8, 0xF9, 0xFA, + 0xFB, 0xFC, 0xFD, 0xFE, 0xFF, 0x02, 0x03 }; static const uint8_t huffbits7[63] = { - 3,5,5,5,5,5,5,5,5,5,5,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,7,7,7,7,7, - 7,7,7,7,7,7,7,7,7,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,4,4 + 3, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, + 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 7, 7, 7, 7, 7, + 7, 7, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 8, + 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 4, 4 }; static const uint8_t huff_tab_sizes[7] = { - 9, 5, 7, 9, 15, 31, 63, + 9, 5, 7, 9, 15, 31, 63, }; static const uint8_t* const huff_codes[7] = { - huffcode1,huffcode2,huffcode3,huffcode4,huffcode5,huffcode6,huffcode7, + huffcode1, huffcode2, huffcode3, huffcode4, huffcode5, huffcode6, huffcode7 }; static const uint8_t* const huff_bits[7] = { - huffbits1,huffbits2,huffbits3,huffbits4,huffbits5,huffbits6,huffbits7, + huffbits1, huffbits2, huffbits3, huffbits4, huffbits5, huffbits6, huffbits7, }; -static const uint16_t atrac3_vlc_offs[] = { - 0,512,1024,1536,2048,2560,3072,3584,4096 +static const uint16_t atrac3_vlc_offs[9] = { + 0, 512, 1024, 1536, 2048, 2560, 3072, 3584, 4096 }; /* selector tables */ -static const uint8_t CLCLengthTab[8] = {0, 4, 3, 3, 4, 4, 5, 6}; -static const int8_t seTab_0[4] = {0, 1, -2, -1}; -static const int8_t decTable1[18] = {0,0, 0,1, 0,-1, 1,0, -1,0, 1,1, 1,-1, -1,1, -1,-1}; +static const uint8_t clc_length_tab[8] = { 0, 4, 3, 3, 4, 4, 5, 6 }; + +static const int8_t mantissa_clc_tab[4] = { 0, 1, -2, -1 }; + +static const int8_t mantissa_vlc_tab[18] = { + 0, 0, 0, 1, 0, -1, 1, 0, -1, 0, 1, 1, 1, -1, -1, 1, -1, -1 +}; /* tables for the scalefactor decoding */ -static const float iMaxQuant[8] = { - 0.0, 1.0/1.5, 1.0/2.5, 1.0/3.5, 1.0/4.5, 1.0/7.5, 1.0/15.5, 1.0/31.5 +static const float inv_max_quant[8] = { + 0.0, 1.0 / 1.5, 1.0 / 2.5, 1.0 / 3.5, + 1.0 / 4.5, 1.0 / 7.5, 1.0 / 15.5, 1.0 / 31.5 }; -static const uint16_t subbandTab[33] = { - 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, - 256, 288, 320, 352, 384, 416, 448, 480, 512, 576, 640, 704, 768, 896, 1024 +static const uint16_t subband_tab[33] = { + 0, 8, 16, 24, 32, 40, 48, 56, + 64, 80, 96, 112, 128, 144, 160, 176, + 192, 224, 256, 288, 320, 352, 384, 416, + 448, 480, 512, 576, 640, 704, 768, 896, + 1024 }; /* joint stereo related tables */ -static const float matrixCoeffs[8] = {0.0, 2.0, 2.0, 2.0, 0.0, 0.0, 1.0, 1.0}; +static const float matrix_coeffs[8] = { + 0.0, 2.0, 2.0, 2.0, 0.0, 0.0, 1.0, 1.0 +}; #endif /* AVCODEC_ATRAC3DATA_H */ |