mirror of https://git.videolan.org/git/ffmpeg.git
Initial commit of the atrac1 decoder, not hooked up yet
Originally committed as revision 19811 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Benjamin Larsson
parent
76ec34a5c3
commit
dbb0f96f0f
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@ -0,0 +1,402 @@
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/*
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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.
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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 "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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int idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
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int idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
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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,short_buf[64]); ///< buffer for the short mode
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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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float out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
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MDCTContext mdct_ctx[3];
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int channels;
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DSPContext dsp;
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} AT1Ctx;
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static float *short_window;
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static float *mid_window;
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DECLARE_ALIGNED_16(static float, long_window[256]);
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static float *window_per_band[3];
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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, int rev_spec)
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{
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MDCTContext* mdct_context;
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int transf_size = 1 << nbits;
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mdct_context = &q->mdct_ctx[nbits - 5 - (nbits>6)];
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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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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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/* 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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if (num_blocks == 1) {
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at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num);
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pos += block_size; // move to the next mdct block in the spectrum
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} else {
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/* calc start position for the 1st short block: 96(128) or 112(256) */
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start_pos = (band_samples * (num_blocks - 1)) >> (log2_block_count + 1);
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memset(&su->spectrum[0][ref_pos], 0, sizeof(float) * (band_samples * 2));
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for (; num_blocks!=0 ; num_blocks--) {
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/* use hardcoded nbits for the short mode */
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at1_imdct(q, &q->spec[pos], q->short_buf, 5, band_num);
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/* overlap and window between short blocks */
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q->dsp.vector_fmul_window(&su->spectrum[0][ref_pos+start_pos],
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&su->spectrum[0][ref_pos+start_pos],q->short_buf,short_window, 0, 16);
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start_pos += 32; // use hardcoded block_size
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pos += 32;
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}
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}
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/* overlap and window with the previous frame and output the result */
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q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos+band_samples/2],
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&su->spectrum[0][ref_pos], window_per_band[band_num], 0, band_samples/2);
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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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static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[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_count[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_count[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, 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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/* 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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su->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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su->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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su->idwls[i] = su->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 = !!su->idwls[bfu_num] + su->idwls[bfu_num];
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float scale_factor = sf_table[su->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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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,
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void *data, int *data_size,
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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_block_size_mode(&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./(1<<15), 32700./(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], -32700./(1<<15), 32700./(1<<15));
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samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700./(1<<15), 32700./(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 void init_mdct_windows(void)
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{
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int i;
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/** The mid and long windows uses the same sine window splitted
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* in the middle and wrapped into zero/one regions as follows:
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*
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* region of "ones"
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* -------------
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* /
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* / 1st half
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* / of the sine
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* / window
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* ---------/
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* zero region
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*
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* The mid and short windows are subsets of the long window.
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*/
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/* Build "zero" region */
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memset(long_window, 0, sizeof(long_window));
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/* Build sine window region */
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short_window = &long_window[112];
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ff_sine_window_init(short_window,32);
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/* Build "ones" region */
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for (i = 0; i < 112; i++)
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long_window[144 + i] = 1.0f;
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/* Save the mid window subset start */
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mid_window = &long_window[64];
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/* Prepare the window table */
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window_per_band[0] = mid_window;
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window_per_band[1] = mid_window;
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window_per_band[2] = long_window;
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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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init_mdct_windows();
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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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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 = NULL,
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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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@ -0,0 +1,62 @@
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/*
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* Atrac 1 compatible decoder data
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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
|
||||
* 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,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with FFmpeg; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
/**
|
||||
* @file libavcodec/atrac1data.h
|
||||
* Atrac 1 compatible decoder data
|
||||
*/
|
||||
|
||||
#ifndef AVCODEC_ATRAC1DATA_H
|
||||
#define AVCODEC_ATRAC1DATA_H
|
||||
|
||||
static const uint8_t bfu_amount_tab1[8] = {20, 28, 32, 36, 40, 44, 48, 52};
|
||||
static const uint8_t bfu_amount_tab2[4] = { 0, 112, 176, 208};
|
||||
static const uint8_t bfu_amount_tab3[8] = { 0, 24, 36, 48, 72, 108, 132, 156};
|
||||
|
||||
/** number of BFUs in each QMF band */
|
||||
static const uint8_t bfu_bands_t[4] = {0, 20, 36, 52};
|
||||
|
||||
/** number of spectral lines in each BFU
|
||||
* block floating unit = group of spectral frequencies having the
|
||||
* same quantization parameters like word length and scale factor
|
||||
*/
|
||||
static const uint8_t specs_per_bfu[52] = {
|
||||
8, 8, 8, 8, 4, 4, 4, 4, 8, 8, 8, 8, 6, 6, 6, 6, 6, 6, 6, 6, // low band
|
||||
6, 6, 6, 6, 7, 7, 7, 7, 9, 9, 9, 9, 10, 10, 10, 10, // midle band
|
||||
12, 12, 12, 12, 12, 12, 12, 12, 20, 20, 20, 20, 20, 20, 20, 20 // high band
|
||||
};
|
||||
|
||||
/** start position of each BFU in the MDCT spectrum for the long mode */
|
||||
static const uint16_t bfu_start_long[52] = {
|
||||
0, 8, 16, 24, 32, 36, 40, 44, 48, 56, 64, 72, 80, 86, 92, 98, 104, 110, 116, 122,
|
||||
128, 134, 140, 146, 152, 159, 166, 173, 180, 189, 198, 207, 216, 226, 236, 246,
|
||||
256, 268, 280, 292, 304, 316, 328, 340, 352, 372, 392, 412, 432, 452, 472, 492,
|
||||
};
|
||||
|
||||
/** start position of each BFU in the MDCT spectrum for the short mode */
|
||||
static const uint16_t bfu_start_short[52] = {
|
||||
0, 32, 64, 96, 8, 40, 72, 104, 12, 44, 76, 108, 20, 52, 84, 116, 26, 58, 90, 122,
|
||||
128, 160, 192, 224, 134, 166, 198, 230, 141, 173, 205, 237, 150, 182, 214, 246,
|
||||
256, 288, 320, 352, 384, 416, 448, 480, 268, 300, 332, 364, 396, 428, 460, 492
|
||||
};
|
||||
|
||||
#endif /* AVCODEC_ATRAC1DATA_H */
|
||||
Loading…
Reference in New Issue