da0ac0ee64
These functions are not used outside their respective files, and they lack a prototype in a header. Originally committed as revision 22259 to svn://svn.ffmpeg.org/ffmpeg/trunk
383 lines
12 KiB
C
383 lines
12 KiB
C
/*
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* Atrac 1 compatible decoder
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* Copyright (c) 2009 Maxim Poliakovski
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* Copyright (c) 2009 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 libavcodec/atrac1.c
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* Atrac 1 compatible decoder.
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* This decoder handles raw ATRAC1 data and probably SDDS data.
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*/
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/* Many thanks to Tim Craig for all the help! */
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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 "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "fft.h"
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#include "atrac.h"
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#include "atrac1data.h"
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#define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
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#define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
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#define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
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#define AT1_FRAME_SIZE AT1_SU_SIZE * 2
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#define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
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#define AT1_MAX_CHANNELS 2
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#define AT1_QMF_BANDS 3
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#define IDX_LOW_BAND 0
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#define IDX_MID_BAND 1
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#define IDX_HIGH_BAND 2
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/**
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* Sound unit struct, one unit is used per channel
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*/
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typedef struct {
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int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
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int num_bfus; ///< number of Block Floating Units
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float* spectrum[2];
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DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer
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DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer
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DECLARE_ALIGNED(16, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter
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DECLARE_ALIGNED(16, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter
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DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter
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} AT1SUCtx;
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/**
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* The atrac1 context, holds all needed parameters for decoding
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*/
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typedef struct {
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AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
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DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
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DECLARE_ALIGNED(16, float, low)[256];
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DECLARE_ALIGNED(16, float, mid)[256];
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DECLARE_ALIGNED(16, float, high)[512];
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float* bands[3];
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DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
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FFTContext mdct_ctx[3];
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int channels;
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DSPContext dsp;
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} AT1Ctx;
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/** size of the transform in samples in the long mode for each QMF band */
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static const uint16_t samples_per_band[3] = {128, 128, 256};
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static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
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static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
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int rev_spec)
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{
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FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
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int transf_size = 1 << nbits;
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if (rev_spec) {
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int i;
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for (i = 0; i < transf_size / 2; i++)
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FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
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}
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ff_imdct_half(mdct_context, out, spec);
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}
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static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
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{
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int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
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unsigned int start_pos, ref_pos = 0, pos = 0;
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for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
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float *prev_buf;
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int j;
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band_samples = samples_per_band[band_num];
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log2_block_count = su->log2_block_count[band_num];
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/* number of mdct blocks in the current QMF band: 1 - for long mode */
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/* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
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num_blocks = 1 << log2_block_count;
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if (num_blocks == 1) {
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/* mdct block size in samples: 128 (long mode, low & mid bands), */
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/* 256 (long mode, high band) and 32 (short mode, all bands) */
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block_size = band_samples >> log2_block_count;
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/* calc transform size in bits according to the block_size_mode */
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nbits = mdct_long_nbits[band_num] - log2_block_count;
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if (nbits != 5 && nbits != 7 && nbits != 8)
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return -1;
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} else {
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block_size = 32;
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nbits = 5;
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}
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start_pos = 0;
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prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
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for (j=0; j < num_blocks; j++) {
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at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
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/* overlap and window */
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q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
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&su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
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prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
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start_pos += block_size;
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pos += block_size;
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}
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if (num_blocks == 1)
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memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
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ref_pos += band_samples;
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}
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/* Swap buffers so the mdct overlap works */
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FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
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return 0;
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}
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/**
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* Parse the block size mode byte
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*/
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static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
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{
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int log2_block_count_tmp, i;
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for (i = 0; i < 2; i++) {
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/* low and mid band */
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log2_block_count_tmp = get_bits(gb, 2);
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if (log2_block_count_tmp & 1)
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return -1;
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log2_block_cnt[i] = 2 - log2_block_count_tmp;
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}
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/* high band */
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log2_block_count_tmp = get_bits(gb, 2);
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if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
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return -1;
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log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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skip_bits(gb, 2);
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return 0;
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}
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static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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float spec[AT1_SU_SAMPLES])
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{
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int bits_used, band_num, bfu_num, i;
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uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
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uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
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/* parse the info byte (2nd byte) telling how much BFUs were coded */
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su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
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/* calc number of consumed bits:
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num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
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+ info_byte_copy(8bits) + log2_block_count_copy(8bits) */
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bits_used = su->num_bfus * 10 + 32 +
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bfu_amount_tab2[get_bits(gb, 2)] +
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(bfu_amount_tab3[get_bits(gb, 3)] << 1);
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/* get word length index (idwl) for each BFU */
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for (i = 0; i < su->num_bfus; i++)
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idwls[i] = get_bits(gb, 4);
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/* get scalefactor index (idsf) for each BFU */
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for (i = 0; i < su->num_bfus; i++)
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idsfs[i] = get_bits(gb, 6);
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/* zero idwl/idsf for empty BFUs */
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for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
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idwls[i] = idsfs[i] = 0;
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/* read in the spectral data and reconstruct MDCT spectrum of this channel */
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for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
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for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
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int pos;
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int num_specs = specs_per_bfu[bfu_num];
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int word_len = !!idwls[bfu_num] + idwls[bfu_num];
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float scale_factor = sf_table[idsfs[bfu_num]];
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bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
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/* check for bitstream overflow */
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if (bits_used > AT1_SU_MAX_BITS)
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return -1;
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/* get the position of the 1st spec according to the block size mode */
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pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
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if (word_len) {
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float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
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for (i = 0; i < num_specs; i++) {
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/* read in a quantized spec and convert it to
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* signed int and then inverse quantization
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*/
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spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
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}
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} else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
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memset(&spec[pos], 0, num_specs * sizeof(float));
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}
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}
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}
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return 0;
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}
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static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
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{
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float temp[256];
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float iqmf_temp[512 + 46];
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/* combine low and middle bands */
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atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
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/* delay the signal of the high band by 23 samples */
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memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
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memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
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/* combine (low + middle) and high bands */
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atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
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}
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static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
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int *data_size, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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AT1Ctx *q = avctx->priv_data;
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int ch, ret, i;
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GetBitContext gb;
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float* samples = data;
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if (buf_size < 212 * q->channels) {
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av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
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return -1;
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}
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for (ch = 0; ch < q->channels; ch++) {
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AT1SUCtx* su = &q->SUs[ch];
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init_get_bits(&gb, &buf[212 * ch], 212 * 8);
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/* parse block_size_mode, 1st byte */
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ret = at1_parse_bsm(&gb, su->log2_block_count);
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if (ret < 0)
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return ret;
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ret = at1_unpack_dequant(&gb, su, q->spec);
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if (ret < 0)
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return ret;
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ret = at1_imdct_block(su, q);
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if (ret < 0)
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return ret;
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at1_subband_synthesis(q, su, q->out_samples[ch]);
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}
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/* round, convert to 16bit and interleave */
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if (q->channels == 1) {
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/* mono */
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q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
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32700.0 / (1 << 15), AT1_SU_SAMPLES);
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} else {
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/* stereo */
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for (i = 0; i < AT1_SU_SAMPLES; i++) {
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samples[i * 2] = av_clipf(q->out_samples[0][i],
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-32700.0 / (1 << 15),
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32700.0 / (1 << 15));
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samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
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-32700.0 / (1 << 15),
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32700.0 / (1 << 15));
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}
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}
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*data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
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return avctx->block_align;
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}
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static av_cold int atrac1_decode_init(AVCodecContext *avctx)
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{
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AT1Ctx *q = avctx->priv_data;
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avctx->sample_fmt = SAMPLE_FMT_FLT;
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q->channels = avctx->channels;
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/* Init the mdct transforms */
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ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
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ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
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ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
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ff_init_ff_sine_windows(5);
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atrac_generate_tables();
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dsputil_init(&q->dsp, avctx);
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q->bands[0] = q->low;
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q->bands[1] = q->mid;
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q->bands[2] = q->high;
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/* Prepare the mdct overlap buffers */
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q->SUs[0].spectrum[0] = q->SUs[0].spec1;
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q->SUs[0].spectrum[1] = q->SUs[0].spec2;
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q->SUs[1].spectrum[0] = q->SUs[1].spec1;
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q->SUs[1].spectrum[1] = q->SUs[1].spec2;
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return 0;
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}
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static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
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AT1Ctx *q = avctx->priv_data;
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ff_mdct_end(&q->mdct_ctx[0]);
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ff_mdct_end(&q->mdct_ctx[1]);
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ff_mdct_end(&q->mdct_ctx[2]);
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return 0;
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}
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AVCodec atrac1_decoder = {
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.name = "atrac1",
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.type = CODEC_TYPE_AUDIO,
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.id = CODEC_ID_ATRAC1,
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.priv_data_size = sizeof(AT1Ctx),
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.init = atrac1_decode_init,
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.close = atrac1_decode_end,
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.decode = atrac1_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
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};
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