1039 lines
34 KiB
C
Executable File
1039 lines
34 KiB
C
Executable File
/*
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* ATRAC3 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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* @file
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* ATRAC3 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 ATRAC3 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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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#include "libavutil/attributes.h"
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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 "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 MIN_CHANNELS 1
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#define MAX_CHANNELS 8
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#define MAX_JS_PAIRS 8 / 2
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#define JOINT_STEREO 0x12
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#define SINGLE 0x2
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#define SAMPLES_PER_FRAME 1024
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#define MDCT_SIZE 512
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typedef struct GainBlock {
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AtracGainInfo 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[MAX_JS_PAIRS][4];
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int matrix_coeff_index_now[MAX_JS_PAIRS][4];
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int matrix_coeff_index_next[MAX_JS_PAIRS][4];
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int weighting_delay[MAX_JS_PAIRS][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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AtracGCContext gainc_ctx;
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FFTContext mdct_ctx;
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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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/**
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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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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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/* 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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if (off)
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c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
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else
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c = av_be2ne32(0x537F6103U);
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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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avpriv_request_sample(NULL, "Offset of %d", off);
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return off;
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}
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static av_cold void init_imdct_window(void)
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{
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int i, j;
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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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av_freep(&q->units);
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av_freep(&q->decoded_bytes_buffer);
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av_freep(&q->fdsp);
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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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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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/* 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 b, j;
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int *level, *loc;
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AtracGainInfo *gain = block->g_block;
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for (b = 0; b <= num_bands; b++) {
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gain[b].num_points = get_bits(gb, 3);
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level = gain[b].lev_code;
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loc = gain[b].loc_code;
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for (j = 0; j < gain[b].num_points; j++) {
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level[j] = get_bits(gb, 4);
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loc[j] = get_bits(gb, 5);
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if (j && loc[j] <= loc[j - 1])
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return AVERROR_INVALIDDATA;
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}
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}
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/* Clear the unused blocks. */
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for (; b < 4 ; b++)
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gain[b].num_points = 0;
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return 0;
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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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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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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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return last_pos;
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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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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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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. */
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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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}
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/* Apply the matrix without interpolation. */
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switch (s2) {
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case 0: /* M/S decoding */
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for (; nsample < band + 256; nsample++) {
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float c1 = su1[nsample];
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float c2 = su2[nsample];
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su1[nsample] = c2 * 2.0;
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su2[nsample] = (c1 - c2) * 2.0;
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}
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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 single channels)
|
|
*/
|
|
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 % 2) == 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 < 0)
|
|
return snd->num_components;
|
|
|
|
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 */
|
|
ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
|
|
&snd->prev_frame[band * 256],
|
|
&gain1->g_block[band], &gain2->g_block[band],
|
|
256, &output[band * 256]);
|
|
}
|
|
|
|
/* 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, ch;
|
|
uint8_t *ptr1;
|
|
|
|
if (q->coding_mode == JOINT_STEREO) {
|
|
/* channel coupling mode */
|
|
|
|
/* Decode sound unit pairs (channels are expected to be even).
|
|
* Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */
|
|
const uint8_t *js_databuf;
|
|
int js_pair, js_block_align;
|
|
|
|
js_block_align = (avctx->block_align / avctx->channels) * 2; /* block pair */
|
|
|
|
for (ch = 0; ch < avctx->channels; ch = ch + 2) {
|
|
js_pair = ch/2;
|
|
js_databuf = databuf + js_pair * js_block_align; /* align to current pair */
|
|
|
|
/* Set the bitstream reader at the start of first channel sound unit. */
|
|
init_get_bits(&q->gb,
|
|
js_databuf, js_block_align * 8);
|
|
|
|
/* decode Sound Unit 1 */
|
|
ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch],
|
|
out_samples[ch], ch, 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 (js_databuf == q->decoded_bytes_buffer) {
|
|
uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1;
|
|
ptr1 = q->decoded_bytes_buffer;
|
|
for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--)
|
|
FFSWAP(uint8_t, *ptr1, *ptr2);
|
|
} else {
|
|
const uint8_t *ptr2 = js_databuf + js_block_align - 1;
|
|
for (i = 0; i < js_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 >= js_block_align)
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
|
|
/* set the bitstream reader at the start of the second Sound Unit */
|
|
ret = init_get_bits8(&q->gb,
|
|
ptr1, q->decoded_bytes_buffer + js_block_align - ptr1);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
/* Fill the Weighting coeffs delay buffer */
|
|
memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2],
|
|
4 * sizeof(*q->weighting_delay[js_pair]));
|
|
q->weighting_delay[js_pair][4] = get_bits1(&q->gb);
|
|
q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3);
|
|
|
|
for (i = 0; i < 4; i++) {
|
|
q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i];
|
|
q->matrix_coeff_index_now[js_pair][i] = q->matrix_coeff_index_next[js_pair][i];
|
|
q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2);
|
|
}
|
|
|
|
/* Decode Sound Unit 2. */
|
|
ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1],
|
|
out_samples[ch+1], ch+1, JOINT_STEREO);
|
|
if (ret != 0)
|
|
return ret;
|
|
|
|
/* Reconstruct the channel coefficients. */
|
|
reverse_matrixing(out_samples[ch], out_samples[ch+1],
|
|
q->matrix_coeff_index_prev[js_pair],
|
|
q->matrix_coeff_index_now[js_pair]);
|
|
|
|
channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]);
|
|
}
|
|
} else {
|
|
/* single channels */
|
|
/* 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 al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
|
|
int size, float **out_samples)
|
|
{
|
|
ATRAC3Context *q = avctx->priv_data;
|
|
int ret, i;
|
|
|
|
/* Set the bitstream reader at the start of a channel sound unit. */
|
|
init_get_bits(&q->gb, databuf, size * 8);
|
|
/* single channels */
|
|
/* Decode the channel sound units. */
|
|
for (i = 0; i < avctx->channels; i++) {
|
|
ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
|
|
out_samples[i], i, q->coding_mode);
|
|
if (ret != 0)
|
|
return ret;
|
|
while (i < avctx->channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) {
|
|
skip_bits(&q->gb, 1);
|
|
}
|
|
}
|
|
|
|
/* 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(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
|
|
return ret;
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return avctx->block_align;
|
|
}
|
|
|
|
static int atrac3al_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
AVFrame *frame = data;
|
|
int ret;
|
|
|
|
frame->nb_samples = SAMPLES_PER_FRAME;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
|
|
ret = al_decode_frame(avctx, avpkt->data, avpkt->size,
|
|
(float **)frame->extended_data);
|
|
if (ret) {
|
|
av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
|
|
return ret;
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return avpkt->size;
|
|
}
|
|
|
|
static av_cold void atrac3_init_static_data(void)
|
|
{
|
|
int i;
|
|
|
|
init_imdct_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);
|
|
}
|
|
}
|
|
|
|
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
|
|
{
|
|
static int static_init_done;
|
|
int i, js_pair, ret;
|
|
int version, delay, samples_per_frame, frame_factor;
|
|
const uint8_t *edata_ptr = avctx->extradata;
|
|
ATRAC3Context *q = avctx->priv_data;
|
|
|
|
if (avctx->channels < MIN_CHANNELS || avctx->channels > MAX_CHANNELS) {
|
|
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->codec_id == AV_CODEC_ID_ATRAC3AL) {
|
|
version = 4;
|
|
samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
|
|
delay = 0x88E;
|
|
q->coding_mode = SINGLE;
|
|
} else 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 : SINGLE;
|
|
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(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
|
|
avctx->extradata_size);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
/* 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 * avctx->channels) {
|
|
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 == SINGLE)
|
|
av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
|
|
else if (q->coding_mode == JOINT_STEREO) {
|
|
if (avctx->channels % 2 == 1) { /* Joint stereo channels must be even */
|
|
av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
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 > 1024 || avctx->block_align <= 0)
|
|
return AVERROR(EINVAL);
|
|
|
|
q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
|
|
AV_INPUT_BUFFER_PADDING_SIZE);
|
|
if (!q->decoded_bytes_buffer)
|
|
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 */
|
|
for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) {
|
|
q->weighting_delay[js_pair][0] = 0;
|
|
q->weighting_delay[js_pair][1] = 7;
|
|
q->weighting_delay[js_pair][2] = 0;
|
|
q->weighting_delay[js_pair][3] = 7;
|
|
q->weighting_delay[js_pair][4] = 0;
|
|
q->weighting_delay[js_pair][5] = 7;
|
|
|
|
for (i = 0; i < 4; i++) {
|
|
q->matrix_coeff_index_prev[js_pair][i] = 3;
|
|
q->matrix_coeff_index_now[js_pair][i] = 3;
|
|
q->matrix_coeff_index_next[js_pair][i] = 3;
|
|
}
|
|
}
|
|
|
|
ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
|
|
q->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
|
|
|
|
q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
|
|
if (!q->units || !q->fdsp) {
|
|
atrac3_decode_close(avctx);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_atrac3_decoder = {
|
|
.name = "atrac3",
|
|
.long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
|
|
.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 = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE },
|
|
};
|
|
|
|
AVCodec ff_atrac3al_decoder = {
|
|
.name = "atrac3al",
|
|
.long_name = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_ATRAC3AL,
|
|
.priv_data_size = sizeof(ATRAC3Context),
|
|
.init = atrac3_decode_init,
|
|
.close = atrac3_decode_close,
|
|
.decode = atrac3al_decode_frame,
|
|
.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE },
|
|
};
|