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1012 lines
32 KiB
1012 lines
32 KiB
/* |
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* Atrac 3 compatible decoder |
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* Copyright (c) 2006-2008 Maxim Poliakovski |
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* Copyright (c) 2006-2008 Benjamin Larsson |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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|
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/** |
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* @file |
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* Atrac 3 compatible decoder. |
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* This decoder handles Sony's ATRAC3 data. |
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* |
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* Container formats used to store atrac 3 data: |
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* RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3). |
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* |
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* To use this decoder, a calling application must supply the extradata |
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* bytes provided in the containers above. |
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*/ |
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|
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#include <math.h> |
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#include <stddef.h> |
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#include <stdio.h> |
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|
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#include "libavutil/float_dsp.h" |
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#include "libavutil/libm.h" |
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#include "avcodec.h" |
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#include "bytestream.h" |
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#include "fft.h" |
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#include "fmtconvert.h" |
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#include "get_bits.h" |
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#include "internal.h" |
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#include "atrac.h" |
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#include "atrac3data.h" |
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#define JOINT_STEREO 0x12 |
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#define STEREO 0x2 |
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#define SAMPLES_PER_FRAME 1024 |
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#define MDCT_SIZE 512 |
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typedef struct GainInfo { |
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int num_gain_data; |
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int lev_code[8]; |
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int loc_code[8]; |
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} GainInfo; |
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typedef struct GainBlock { |
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GainInfo g_block[4]; |
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} GainBlock; |
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typedef struct TonalComponent { |
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int pos; |
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int num_coefs; |
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float coef[8]; |
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} TonalComponent; |
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typedef struct ChannelUnit { |
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int bands_coded; |
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int num_components; |
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float prev_frame[SAMPLES_PER_FRAME]; |
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int gc_blk_switch; |
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TonalComponent components[64]; |
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GainBlock gain_block[2]; |
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DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME]; |
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DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME]; |
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float delay_buf1[46]; ///<qmf delay buffers |
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float delay_buf2[46]; |
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float delay_buf3[46]; |
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} ChannelUnit; |
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typedef struct ATRAC3Context { |
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GetBitContext gb; |
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//@{ |
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/** stream data */ |
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int coding_mode; |
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ChannelUnit *units; |
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//@} |
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//@{ |
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/** joint-stereo related variables */ |
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int matrix_coeff_index_prev[4]; |
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int matrix_coeff_index_now[4]; |
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int matrix_coeff_index_next[4]; |
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int weighting_delay[6]; |
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//@} |
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//@{ |
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/** data buffers */ |
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uint8_t *decoded_bytes_buffer; |
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float temp_buf[1070]; |
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//@} |
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//@{ |
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/** extradata */ |
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int scrambled_stream; |
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//@} |
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FFTContext mdct_ctx; |
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FmtConvertContext fmt_conv; |
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AVFloatDSPContext fdsp; |
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} ATRAC3Context; |
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static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE]; |
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static VLC_TYPE atrac3_vlc_table[4096][2]; |
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static VLC spectral_coeff_tab[7]; |
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static float gain_tab1[16]; |
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static float gain_tab2[31]; |
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/** |
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* Regular 512 points IMDCT without overlapping, with the exception of the |
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* swapping of odd bands caused by the reverse spectra of the QMF. |
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* |
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* @param odd_band 1 if the band is an odd band |
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*/ |
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static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band) |
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{ |
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int i; |
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|
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if (odd_band) { |
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/** |
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* Reverse the odd bands before IMDCT, this is an effect of the QMF |
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* transform or it gives better compression to do it this way. |
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* FIXME: It should be possible to handle this in imdct_calc |
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* for that to happen a modification of the prerotation step of |
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* all SIMD code and C code is needed. |
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* Or fix the functions before so they generate a pre reversed spectrum. |
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*/ |
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for (i = 0; i < 128; i++) |
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FFSWAP(float, input[i], input[255 - i]); |
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} |
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q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input); |
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|
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/* Perform windowing on the output. */ |
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q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE); |
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} |
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/* |
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* indata descrambling, only used for data coming from the rm container |
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*/ |
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static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes) |
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{ |
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int i, off; |
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uint32_t c; |
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const uint32_t *buf; |
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uint32_t *output = (uint32_t *)out; |
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off = (intptr_t)input & 3; |
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buf = (const uint32_t *)(input - off); |
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c = av_be2ne32((0x537F6103 >> (off * 8)) | (0x537F6103 << (32 - (off * 8)))); |
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bytes += 3 + off; |
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for (i = 0; i < bytes / 4; i++) |
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output[i] = c ^ buf[i]; |
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if (off) |
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av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off); |
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return off; |
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} |
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static av_cold void init_atrac3_window(void) |
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{ |
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int i, j; |
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|
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/* generate the mdct window, for details see |
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* http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ |
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for (i = 0, j = 255; i < 128; i++, j--) { |
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float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; |
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float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0; |
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float w = 0.5 * (wi * wi + wj * wj); |
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mdct_window[i] = mdct_window[511 - i] = wi / w; |
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mdct_window[j] = mdct_window[511 - j] = wj / w; |
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} |
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} |
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static av_cold int atrac3_decode_close(AVCodecContext *avctx) |
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{ |
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ATRAC3Context *q = avctx->priv_data; |
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|
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av_free(q->units); |
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av_free(q->decoded_bytes_buffer); |
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ff_mdct_end(&q->mdct_ctx); |
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return 0; |
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} |
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/** |
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* Mantissa decoding |
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* |
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* @param selector which table the output values are coded with |
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* @param coding_flag constant length coding or variable length coding |
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* @param mantissas mantissa output table |
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* @param num_codes number of values to get |
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*/ |
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static void read_quant_spectral_coeffs(GetBitContext *gb, int selector, |
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int coding_flag, int *mantissas, |
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int num_codes) |
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{ |
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int i, code, huff_symb; |
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|
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if (selector == 1) |
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num_codes /= 2; |
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if (coding_flag != 0) { |
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/* constant length coding (CLC) */ |
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int num_bits = clc_length_tab[selector]; |
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if (selector > 1) { |
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for (i = 0; i < num_codes; i++) { |
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if (num_bits) |
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code = get_sbits(gb, num_bits); |
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else |
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code = 0; |
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mantissas[i] = code; |
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} |
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} else { |
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for (i = 0; i < num_codes; i++) { |
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if (num_bits) |
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code = get_bits(gb, num_bits); // num_bits is always 4 in this case |
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else |
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code = 0; |
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mantissas[i * 2 ] = mantissa_clc_tab[code >> 2]; |
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mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3]; |
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} |
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} |
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} else { |
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/* variable length coding (VLC) */ |
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if (selector != 1) { |
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for (i = 0; i < num_codes; i++) { |
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huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, |
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spectral_coeff_tab[selector-1].bits, 3); |
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huff_symb += 1; |
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code = huff_symb >> 1; |
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if (huff_symb & 1) |
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code = -code; |
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mantissas[i] = code; |
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} |
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} else { |
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for (i = 0; i < num_codes; i++) { |
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huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table, |
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spectral_coeff_tab[selector - 1].bits, 3); |
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mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ]; |
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mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1]; |
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} |
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} |
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} |
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} |
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/** |
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* Restore the quantized band spectrum coefficients |
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* |
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* @return subband count, fix for broken specification/files |
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*/ |
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static int decode_spectrum(GetBitContext *gb, float *output) |
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{ |
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int num_subbands, coding_mode, i, j, first, last, subband_size; |
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int subband_vlc_index[32], sf_index[32]; |
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int mantissas[128]; |
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float scale_factor; |
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num_subbands = get_bits(gb, 5); // number of coded subbands |
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coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC |
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|
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/* get the VLC selector table for the subbands, 0 means not coded */ |
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for (i = 0; i <= num_subbands; i++) |
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subband_vlc_index[i] = get_bits(gb, 3); |
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/* read the scale factor indexes from the stream */ |
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for (i = 0; i <= num_subbands; i++) { |
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if (subband_vlc_index[i] != 0) |
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sf_index[i] = get_bits(gb, 6); |
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} |
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for (i = 0; i <= num_subbands; i++) { |
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first = subband_tab[i ]; |
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last = subband_tab[i + 1]; |
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subband_size = last - first; |
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if (subband_vlc_index[i] != 0) { |
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/* decode spectral coefficients for this subband */ |
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/* TODO: This can be done faster is several blocks share the |
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* same VLC selector (subband_vlc_index) */ |
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read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode, |
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mantissas, subband_size); |
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/* decode the scale factor for this subband */ |
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scale_factor = ff_atrac_sf_table[sf_index[i]] * |
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inv_max_quant[subband_vlc_index[i]]; |
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/* inverse quantize the coefficients */ |
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for (j = 0; first < last; first++, j++) |
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output[first] = mantissas[j] * scale_factor; |
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} else { |
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/* this subband was not coded, so zero the entire subband */ |
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memset(output + first, 0, subband_size * sizeof(*output)); |
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} |
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} |
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/* clear the subbands that were not coded */ |
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first = subband_tab[i]; |
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memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output)); |
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return num_subbands; |
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} |
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/** |
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* Restore the quantized tonal components |
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* |
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* @param components tonal components |
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* @param num_bands number of coded bands |
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*/ |
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static int decode_tonal_components(GetBitContext *gb, |
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TonalComponent *components, int num_bands) |
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{ |
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int i, b, c, m; |
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int nb_components, coding_mode_selector, coding_mode; |
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int band_flags[4], mantissa[8]; |
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int component_count = 0; |
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nb_components = get_bits(gb, 5); |
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/* no tonal components */ |
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if (nb_components == 0) |
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return 0; |
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coding_mode_selector = get_bits(gb, 2); |
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if (coding_mode_selector == 2) |
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return AVERROR_INVALIDDATA; |
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coding_mode = coding_mode_selector & 1; |
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for (i = 0; i < nb_components; i++) { |
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int coded_values_per_component, quant_step_index; |
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for (b = 0; b <= num_bands; b++) |
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band_flags[b] = get_bits1(gb); |
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coded_values_per_component = get_bits(gb, 3); |
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quant_step_index = get_bits(gb, 3); |
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if (quant_step_index <= 1) |
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return AVERROR_INVALIDDATA; |
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if (coding_mode_selector == 3) |
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coding_mode = get_bits1(gb); |
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for (b = 0; b < (num_bands + 1) * 4; b++) { |
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int coded_components; |
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if (band_flags[b >> 2] == 0) |
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continue; |
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coded_components = get_bits(gb, 3); |
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for (c = 0; c < coded_components; c++) { |
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TonalComponent *cmp = &components[component_count]; |
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int sf_index, coded_values, max_coded_values; |
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float scale_factor; |
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sf_index = get_bits(gb, 6); |
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if (component_count >= 64) |
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return AVERROR_INVALIDDATA; |
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cmp->pos = b * 64 + get_bits(gb, 6); |
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max_coded_values = SAMPLES_PER_FRAME - cmp->pos; |
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coded_values = coded_values_per_component + 1; |
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coded_values = FFMIN(max_coded_values, coded_values); |
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scale_factor = ff_atrac_sf_table[sf_index] * |
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inv_max_quant[quant_step_index]; |
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read_quant_spectral_coeffs(gb, quant_step_index, coding_mode, |
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mantissa, coded_values); |
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cmp->num_coefs = coded_values; |
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/* inverse quant */ |
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for (m = 0; m < coded_values; m++) |
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cmp->coef[m] = mantissa[m] * scale_factor; |
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component_count++; |
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} |
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} |
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} |
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return component_count; |
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} |
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/** |
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* Decode gain parameters for the coded bands |
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* |
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* @param block the gainblock for the current band |
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* @param num_bands amount of coded bands |
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*/ |
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static int decode_gain_control(GetBitContext *gb, GainBlock *block, |
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int num_bands) |
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{ |
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int i, cf, num_data; |
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int *level, *loc; |
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GainInfo *gain = block->g_block; |
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for (i = 0; i <= num_bands; i++) { |
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num_data = get_bits(gb, 3); |
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gain[i].num_gain_data = num_data; |
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level = gain[i].lev_code; |
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loc = gain[i].loc_code; |
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for (cf = 0; cf < gain[i].num_gain_data; cf++) { |
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level[cf] = get_bits(gb, 4); |
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loc [cf] = get_bits(gb, 5); |
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if (cf && loc[cf] <= loc[cf - 1]) |
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return AVERROR_INVALIDDATA; |
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} |
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} |
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|
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/* Clear the unused blocks. */ |
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for (; i < 4 ; i++) |
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gain[i].num_gain_data = 0; |
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|
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return 0; |
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} |
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/** |
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* Apply gain parameters and perform the MDCT overlapping part |
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* |
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* @param input input buffer |
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* @param prev previous buffer to perform overlap against |
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* @param output output buffer |
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* @param gain1 current band gain info |
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* @param gain2 next band gain info |
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*/ |
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static void gain_compensate_and_overlap(float *input, float *prev, |
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float *output, GainInfo *gain1, |
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GainInfo *gain2) |
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{ |
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float g1, g2, gain_inc; |
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int i, j, num_data, start_loc, end_loc; |
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if (gain2->num_gain_data == 0) |
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g1 = 1.0; |
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else |
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g1 = gain_tab1[gain2->lev_code[0]]; |
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|
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if (gain1->num_gain_data == 0) { |
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for (i = 0; i < 256; i++) |
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output[i] = input[i] * g1 + prev[i]; |
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} else { |
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num_data = gain1->num_gain_data; |
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gain1->loc_code[num_data] = 32; |
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gain1->lev_code[num_data] = 4; |
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|
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for (i = 0, j = 0; i < num_data; i++) { |
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start_loc = gain1->loc_code[i] * 8; |
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end_loc = start_loc + 8; |
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|
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g2 = gain_tab1[gain1->lev_code[i]]; |
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gain_inc = gain_tab2[gain1->lev_code[i + 1] - |
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gain1->lev_code[i ] + 15]; |
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|
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/* interpolate */ |
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for (; j < start_loc; j++) |
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output[j] = (input[j] * g1 + prev[j]) * g2; |
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|
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/* interpolation is done over eight samples */ |
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for (; j < end_loc; j++) { |
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output[j] = (input[j] * g1 + prev[j]) * g2; |
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g2 *= gain_inc; |
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} |
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} |
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|
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for (; j < 256; j++) |
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output[j] = input[j] * g1 + prev[j]; |
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} |
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|
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/* Delay for the overlapping part. */ |
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memcpy(prev, &input[256], 256 * sizeof(*prev)); |
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} |
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|
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/** |
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* Combine the tonal band spectrum and regular band spectrum |
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* |
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* @param spectrum output spectrum buffer |
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* @param num_components number of tonal components |
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* @param components tonal components for this band |
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* @return position of the last tonal coefficient |
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*/ |
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static int add_tonal_components(float *spectrum, int num_components, |
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TonalComponent *components) |
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{ |
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int i, j, last_pos = -1; |
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float *input, *output; |
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|
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for (i = 0; i < num_components; i++) { |
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last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos); |
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input = components[i].coef; |
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output = &spectrum[components[i].pos]; |
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|
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for (j = 0; j < components[i].num_coefs; j++) |
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output[j] += input[j]; |
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} |
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|
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return last_pos; |
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} |
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|
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#define INTERPOLATE(old, new, nsample) \ |
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((old) + (nsample) * 0.125 * ((new) - (old))) |
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|
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static void reverse_matrixing(float *su1, float *su2, int *prev_code, |
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int *curr_code) |
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{ |
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int i, nsample, band; |
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float mc1_l, mc1_r, mc2_l, mc2_r; |
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|
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for (i = 0, band = 0; band < 4 * 256; band += 256, i++) { |
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int s1 = prev_code[i]; |
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int s2 = curr_code[i]; |
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nsample = band; |
|
|
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if (s1 != s2) { |
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/* Selector value changed, interpolation needed. */ |
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mc1_l = matrix_coeffs[s1 * 2 ]; |
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mc1_r = matrix_coeffs[s1 * 2 + 1]; |
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mc2_l = matrix_coeffs[s2 * 2 ]; |
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mc2_r = matrix_coeffs[s2 * 2 + 1]; |
|
|
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/* Interpolation is done over the first eight samples. */ |
|
for (; nsample < band + 8; nsample++) { |
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float c1 = su1[nsample]; |
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float c2 = su2[nsample]; |
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c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) + |
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c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band); |
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su1[nsample] = c2; |
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su2[nsample] = c1 * 2.0 - c2; |
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} |
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} |
|
|
|
/* Apply the matrix without interpolation. */ |
|
switch (s2) { |
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case 0: /* M/S decoding */ |
|
for (; nsample < band + 256; nsample++) { |
|
float c1 = su1[nsample]; |
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float c2 = su2[nsample]; |
|
su1[nsample] = c2 * 2.0; |
|
su2[nsample] = (c1 - c2) * 2.0; |
|
} |
|
break; |
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case 1: |
|
for (; nsample < band + 256; nsample++) { |
|
float c1 = su1[nsample]; |
|
float c2 = su2[nsample]; |
|
su1[nsample] = (c1 + c2) * 2.0; |
|
su2[nsample] = c2 * -2.0; |
|
} |
|
break; |
|
case 2: |
|
case 3: |
|
for (; nsample < band + 256; nsample++) { |
|
float c1 = su1[nsample]; |
|
float c2 = su2[nsample]; |
|
su1[nsample] = c1 + c2; |
|
su2[nsample] = c1 - c2; |
|
} |
|
break; |
|
default: |
|
av_assert1(0); |
|
} |
|
} |
|
} |
|
|
|
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] = (index & 7) / 7.0; |
|
ch[1] = sqrt(2 - ch[0] * ch[0]); |
|
if (flag) |
|
FFSWAP(float, ch[0], ch[1]); |
|
} |
|
} |
|
|
|
static void channel_weighting(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) { |
|
get_channel_weights(p3[1], p3[0], w[0]); |
|
get_channel_weights(p3[3], p3[2], w[1]); |
|
|
|
for (band = 256; band < 4 * 256; band += 256) { |
|
for (nsample = band; nsample < band + 8; nsample++) { |
|
su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band); |
|
su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band); |
|
} |
|
for(; nsample < band + 256; nsample++) { |
|
su1[nsample] *= w[1][0]; |
|
su2[nsample] *= w[1][1]; |
|
} |
|
} |
|
} |
|
} |
|
|
|
/** |
|
* Decode a Sound Unit |
|
* |
|
* @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 decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb, |
|
ChannelUnit *snd, float *output, |
|
int channel_num, int coding_mode) |
|
{ |
|
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 (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) { |
|
av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n"); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
} |
|
|
|
/* number of coded QMF bands */ |
|
snd->bands_coded = get_bits(gb, 2); |
|
|
|
ret = decode_gain_control(gb, gain2, snd->bands_coded); |
|
if (ret) |
|
return ret; |
|
|
|
snd->num_components = decode_tonal_components(gb, snd->components, |
|
snd->bands_coded); |
|
if (snd->num_components == -1) |
|
return -1; |
|
|
|
num_subbands = decode_spectrum(gb, snd->spectrum); |
|
|
|
/* Merge the decoded spectrum and tonal 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 */ |
|
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++) { |
|
/* Perform the IMDCT step without overlapping. */ |
|
if (band <= num_bands) |
|
imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1); |
|
else |
|
memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf)); |
|
|
|
/* gain compensation and overlapping */ |
|
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. */ |
|
snd->gc_blk_switch ^= 1; |
|
|
|
return 0; |
|
} |
|
|
|
static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf, |
|
float **out_samples) |
|
{ |
|
ATRAC3Context *q = avctx->priv_data; |
|
int ret, i; |
|
uint8_t *ptr1; |
|
|
|
if (q->coding_mode == JOINT_STEREO) { |
|
/* channel coupling mode */ |
|
/* decode Sound Unit 1 */ |
|
init_get_bits(&q->gb, databuf, avctx->block_align * 8); |
|
|
|
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 + avctx->block_align - 1; |
|
ptr1 = q->decoded_bytes_buffer; |
|
for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--) |
|
FFSWAP(uint8_t, *ptr1, *ptr2); |
|
} else { |
|
const uint8_t *ptr2 = databuf + avctx->block_align - 1; |
|
for (i = 0; i < avctx->block_align; 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 >= avctx->block_align) |
|
return AVERROR_INVALIDDATA; |
|
} |
|
|
|
|
|
/* set the bitstream reader at the start of the second Sound Unit*/ |
|
init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1); |
|
|
|
/* Fill the Weighting coeffs delay buffer */ |
|
memmove(q->weighting_delay, &q->weighting_delay[2], |
|
4 * sizeof(*q->weighting_delay)); |
|
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. */ |
|
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. */ |
|
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 < avctx->channels; i++) { |
|
/* Set the bitstream reader at the start of a channel sound unit. */ |
|
init_get_bits(&q->gb, |
|
databuf + i * avctx->block_align / avctx->channels, |
|
avctx->block_align * 8 / avctx->channels); |
|
|
|
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 < avctx->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; |
|
} |
|
|
|
static int atrac3_decode_frame(AVCodecContext *avctx, void *data, |
|
int *got_frame_ptr, AVPacket *avpkt) |
|
{ |
|
AVFrame *frame = data; |
|
const uint8_t *buf = avpkt->data; |
|
int buf_size = avpkt->size; |
|
ATRAC3Context *q = avctx->priv_data; |
|
int ret; |
|
const uint8_t *databuf; |
|
|
|
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 */ |
|
frame->nb_samples = SAMPLES_PER_FRAME; |
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
|
return ret; |
|
|
|
/* 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; |
|
} |
|
|
|
ret = decode_frame(avctx, databuf, (float **)frame->extended_data); |
|
if (ret) { |
|
av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n"); |
|
return ret; |
|
} |
|
|
|
*got_frame_ptr = 1; |
|
|
|
return avctx->block_align; |
|
} |
|
|
|
static void atrac3_init_static_data(void) |
|
{ |
|
int i; |
|
|
|
init_atrac3_window(); |
|
ff_atrac_generate_tables(); |
|
|
|
/* Initialize the VLC tables. */ |
|
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); |
|
} |
|
|
|
/* Generate gain tables */ |
|
for (i = 0; i < 16; i++) |
|
gain_tab1[i] = exp2f (4 - i); |
|
|
|
for (i = -15; i < 16; i++) |
|
gain_tab2[i + 15] = exp2f (i * -0.125); |
|
} |
|
|
|
static av_cold int atrac3_decode_init(AVCodecContext *avctx) |
|
{ |
|
static int static_init_done; |
|
int i, ret; |
|
int version, delay, samples_per_frame, frame_factor; |
|
const uint8_t *edata_ptr = avctx->extradata; |
|
ATRAC3Context *q = avctx->priv_data; |
|
|
|
if (avctx->channels <= 0 || avctx->channels > 2) { |
|
av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n"); |
|
return AVERROR(EINVAL); |
|
} |
|
|
|
if (!static_init_done) |
|
atrac3_init_static_data(); |
|
static_init_done = 1; |
|
|
|
/* 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 |
|
edata_ptr += 4; // samples per channel |
|
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 |
|
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 */ |
|
samples_per_frame = SAMPLES_PER_FRAME * avctx->channels; |
|
version = 4; |
|
delay = 0x88E; |
|
q->coding_mode = q->coding_mode ? JOINT_STEREO : STEREO; |
|
q->scrambled_stream = 0; |
|
|
|
if (avctx->block_align != 96 * avctx->channels * frame_factor && |
|
avctx->block_align != 152 * avctx->channels * frame_factor && |
|
avctx->block_align != 192 * avctx->channels * frame_factor) { |
|
av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor " |
|
"configuration %d/%d/%d\n", avctx->block_align, |
|
avctx->channels, frame_factor); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
} else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) { |
|
/* Parse the extradata, RM format. */ |
|
version = bytestream_get_be32(&edata_ptr); |
|
samples_per_frame = bytestream_get_be16(&edata_ptr); |
|
delay = bytestream_get_be16(&edata_ptr); |
|
q->coding_mode = bytestream_get_be16(&edata_ptr); |
|
q->scrambled_stream = 1; |
|
|
|
} else { |
|
av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n", |
|
avctx->extradata_size); |
|
return AVERROR(EINVAL); |
|
} |
|
|
|
if (q->coding_mode == JOINT_STEREO && avctx->channels < 2) { |
|
av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n"); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
|
|
/* Check the extradata */ |
|
|
|
if (version != 4) { |
|
av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
|
|
if (samples_per_frame != SAMPLES_PER_FRAME && |
|
samples_per_frame != SAMPLES_PER_FRAME * 2) { |
|
av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n", |
|
samples_per_frame); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
|
|
if (delay != 0x88E) { |
|
av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n", |
|
delay); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
|
|
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->block_align >= UINT_MAX / 2) |
|
return AVERROR(EINVAL); |
|
|
|
q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) + |
|
FF_INPUT_BUFFER_PADDING_SIZE); |
|
if (q->decoded_bytes_buffer == NULL) |
|
return AVERROR(ENOMEM); |
|
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; |
|
|
|
/* initialize the MDCT transform */ |
|
if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) { |
|
av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); |
|
av_freep(&q->decoded_bytes_buffer); |
|
return ret; |
|
} |
|
|
|
/* 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; |
|
} |
|
|
|
avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); |
|
ff_fmt_convert_init(&q->fmt_conv, avctx); |
|
|
|
q->units = av_mallocz(sizeof(*q->units) * avctx->channels); |
|
if (!q->units) { |
|
atrac3_decode_close(avctx); |
|
return AVERROR(ENOMEM); |
|
} |
|
|
|
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 }, |
|
};
|
|
|