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https://github.com/FFmpeg/FFmpeg.git
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80d156d7fd
* qatar/master: qdm2: Use floating point synthesis filter. h264: correct border check. h264: fix loopfilter with threading at slice boundaries. Fix ff_mpa_synth_filter_fixed() prototype Rename costablegen.c ---> cos_tablegen.c. Collapse tableprint.c into tableprint.h. Simplify trig table rules Remove potentially unstable filenames from comments in generated files. Ignore generated tables and generated table generator programs. Simplify CLEANFILES make variable by using wildcards. Remove silly insults from avformat_version() Doxygen documentation. mpegaudiodsp: fix x86 and ppc makefiles configure: Adjust AVX assembler check. mpegaudio: remove unused version of SAME_HEADER_MASK mpegaudio: remove useless #undef at end of file asfdec: add missing #include for av_bswap32() mpegaudio: merge two #if CONFIG_FLOAT blocks mpegaudio: move some struct definitions from mpegaudio.h Move some mpegaudio functions to new mpegaudiodsp subsystem Conflicts: libavcodec/h264.c libavcodec/x86/Makefile Merged-by: Michael Niedermayer <michaelni@gmx.at>
420 lines
14 KiB
C
420 lines
14 KiB
C
/*
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* Musepack SV8 decoder
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* Copyright (c) 2007 Konstantin Shishkov
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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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* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
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* divided into 32 subbands.
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*/
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#include "libavutil/lfg.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 "mpegaudiodsp.h"
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#include "libavutil/audioconvert.h"
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#include "mpc.h"
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#include "mpcdata.h"
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#include "mpc8data.h"
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#include "mpc8huff.h"
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static VLC band_vlc, scfi_vlc[2], dscf_vlc[2], res_vlc[2];
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static VLC q1_vlc, q2_vlc[2], q3_vlc[2], quant_vlc[4][2], q9up_vlc;
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static const int q3_offsets[2] = { MPC8_Q3_OFFSET, MPC8_Q4_OFFSET };
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static const int quant_offsets[6] = { MPC8_Q5_OFFSET, MPC8_Q6_OFFSET, MPC8_Q7_OFFSET, MPC8_Q8_OFFSET };
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static inline int mpc8_dec_base(GetBitContext *gb, int k, int n)
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{
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int len = mpc8_cnk_len[k-1][n-1] - 1;
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int code = len ? get_bits_long(gb, len) : 0;
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if (code >= mpc8_cnk_lost[k-1][n-1])
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code = ((code << 1) | get_bits1(gb)) - mpc8_cnk_lost[k-1][n-1];
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return code;
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}
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static inline int mpc8_dec_enum(GetBitContext *gb, int k, int n)
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{
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int bits = 0;
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const uint32_t * C = mpc8_cnk[k-1];
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int code = mpc8_dec_base(gb, k, n);
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do {
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n--;
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if (code >= C[n]) {
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bits |= 1 << n;
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code -= C[n];
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C -= 32;
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k--;
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}
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} while(k > 0);
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return bits;
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}
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static inline int mpc8_get_mod_golomb(GetBitContext *gb, int m)
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{
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if(mpc8_cnk_len[0][m] < 1) return 0;
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return mpc8_dec_base(gb, 1, m+1);
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}
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static int mpc8_get_mask(GetBitContext *gb, int size, int t)
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{
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int mask = 0;
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if(t && t != size)
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mask = mpc8_dec_enum(gb, FFMIN(t, size - t), size);
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if((t << 1) > size) mask = ~mask;
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return mask;
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}
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static const uint16_t vlc_offsets[13] = {
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0, 640, 1184, 1748, 2298, 2426, 2554, 3066, 3578, 4106, 4618, 5196, 5708
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};
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static av_cold int mpc8_decode_init(AVCodecContext * avctx)
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{
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int i;
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MPCContext *c = avctx->priv_data;
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GetBitContext gb;
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static int vlc_initialized = 0;
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int channels;
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static VLC_TYPE band_table[542][2];
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static VLC_TYPE q1_table[520][2];
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static VLC_TYPE q9up_table[524][2];
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static VLC_TYPE scfi0_table[1 << MPC8_SCFI0_BITS][2];
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static VLC_TYPE scfi1_table[1 << MPC8_SCFI1_BITS][2];
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static VLC_TYPE dscf0_table[560][2];
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static VLC_TYPE dscf1_table[598][2];
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static VLC_TYPE q3_0_table[512][2];
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static VLC_TYPE q3_1_table[516][2];
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static VLC_TYPE codes_table[5708][2];
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if(avctx->extradata_size < 2){
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av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
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return -1;
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}
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memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
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av_lfg_init(&c->rnd, 0xDEADBEEF);
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dsputil_init(&c->dsp, avctx);
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ff_mpadsp_init(&c->mpadsp);
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ff_mpc_init();
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init_get_bits(&gb, avctx->extradata, 16);
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skip_bits(&gb, 3);//sample rate
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c->maxbands = get_bits(&gb, 5) + 1;
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channels = get_bits(&gb, 4) + 1;
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if (channels > 2) {
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av_log_missing_feature(avctx, "Multichannel MPC SV8", 1);
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return -1;
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}
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c->MSS = get_bits1(&gb);
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c->frames = 1 << (get_bits(&gb, 3) * 2);
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
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if(vlc_initialized) return 0;
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av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
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band_vlc.table = band_table;
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band_vlc.table_allocated = 542;
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init_vlc(&band_vlc, MPC8_BANDS_BITS, MPC8_BANDS_SIZE,
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mpc8_bands_bits, 1, 1,
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mpc8_bands_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
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q1_vlc.table = q1_table;
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q1_vlc.table_allocated = 520;
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init_vlc(&q1_vlc, MPC8_Q1_BITS, MPC8_Q1_SIZE,
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mpc8_q1_bits, 1, 1,
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mpc8_q1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
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q9up_vlc.table = q9up_table;
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q9up_vlc.table_allocated = 524;
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init_vlc(&q9up_vlc, MPC8_Q9UP_BITS, MPC8_Q9UP_SIZE,
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mpc8_q9up_bits, 1, 1,
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mpc8_q9up_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
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scfi_vlc[0].table = scfi0_table;
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scfi_vlc[0].table_allocated = 1 << MPC8_SCFI0_BITS;
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init_vlc(&scfi_vlc[0], MPC8_SCFI0_BITS, MPC8_SCFI0_SIZE,
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mpc8_scfi0_bits, 1, 1,
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mpc8_scfi0_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
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scfi_vlc[1].table = scfi1_table;
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scfi_vlc[1].table_allocated = 1 << MPC8_SCFI1_BITS;
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init_vlc(&scfi_vlc[1], MPC8_SCFI1_BITS, MPC8_SCFI1_SIZE,
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mpc8_scfi1_bits, 1, 1,
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mpc8_scfi1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
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dscf_vlc[0].table = dscf0_table;
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dscf_vlc[0].table_allocated = 560;
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init_vlc(&dscf_vlc[0], MPC8_DSCF0_BITS, MPC8_DSCF0_SIZE,
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mpc8_dscf0_bits, 1, 1,
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mpc8_dscf0_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
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dscf_vlc[1].table = dscf1_table;
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dscf_vlc[1].table_allocated = 598;
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init_vlc(&dscf_vlc[1], MPC8_DSCF1_BITS, MPC8_DSCF1_SIZE,
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mpc8_dscf1_bits, 1, 1,
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mpc8_dscf1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
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q3_vlc[0].table = q3_0_table;
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q3_vlc[0].table_allocated = 512;
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init_vlc_sparse(&q3_vlc[0], MPC8_Q3_BITS, MPC8_Q3_SIZE,
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mpc8_q3_bits, 1, 1,
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mpc8_q3_codes, 1, 1,
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mpc8_q3_syms, 1, 1, INIT_VLC_USE_NEW_STATIC);
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q3_vlc[1].table = q3_1_table;
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q3_vlc[1].table_allocated = 516;
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init_vlc_sparse(&q3_vlc[1], MPC8_Q4_BITS, MPC8_Q4_SIZE,
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mpc8_q4_bits, 1, 1,
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mpc8_q4_codes, 1, 1,
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mpc8_q4_syms, 1, 1, INIT_VLC_USE_NEW_STATIC);
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for(i = 0; i < 2; i++){
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res_vlc[i].table = &codes_table[vlc_offsets[0+i]];
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res_vlc[i].table_allocated = vlc_offsets[1+i] - vlc_offsets[0+i];
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init_vlc(&res_vlc[i], MPC8_RES_BITS, MPC8_RES_SIZE,
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&mpc8_res_bits[i], 1, 1,
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&mpc8_res_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
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q2_vlc[i].table = &codes_table[vlc_offsets[2+i]];
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q2_vlc[i].table_allocated = vlc_offsets[3+i] - vlc_offsets[2+i];
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init_vlc(&q2_vlc[i], MPC8_Q2_BITS, MPC8_Q2_SIZE,
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&mpc8_q2_bits[i], 1, 1,
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&mpc8_q2_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
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quant_vlc[0][i].table = &codes_table[vlc_offsets[4+i]];
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quant_vlc[0][i].table_allocated = vlc_offsets[5+i] - vlc_offsets[4+i];
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init_vlc(&quant_vlc[0][i], MPC8_Q5_BITS, MPC8_Q5_SIZE,
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&mpc8_q5_bits[i], 1, 1,
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&mpc8_q5_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
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quant_vlc[1][i].table = &codes_table[vlc_offsets[6+i]];
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quant_vlc[1][i].table_allocated = vlc_offsets[7+i] - vlc_offsets[6+i];
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init_vlc(&quant_vlc[1][i], MPC8_Q6_BITS, MPC8_Q6_SIZE,
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&mpc8_q6_bits[i], 1, 1,
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&mpc8_q6_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
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quant_vlc[2][i].table = &codes_table[vlc_offsets[8+i]];
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quant_vlc[2][i].table_allocated = vlc_offsets[9+i] - vlc_offsets[8+i];
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init_vlc(&quant_vlc[2][i], MPC8_Q7_BITS, MPC8_Q7_SIZE,
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&mpc8_q7_bits[i], 1, 1,
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&mpc8_q7_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
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quant_vlc[3][i].table = &codes_table[vlc_offsets[10+i]];
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quant_vlc[3][i].table_allocated = vlc_offsets[11+i] - vlc_offsets[10+i];
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init_vlc(&quant_vlc[3][i], MPC8_Q8_BITS, MPC8_Q8_SIZE,
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&mpc8_q8_bits[i], 1, 1,
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&mpc8_q8_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
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}
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vlc_initialized = 1;
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return 0;
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}
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static int mpc8_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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MPCContext *c = avctx->priv_data;
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GetBitContext gb2, *gb = &gb2;
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int i, j, k, ch, cnt, res, t;
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Band *bands = c->bands;
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int off;
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int maxband, keyframe;
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int last[2];
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keyframe = c->cur_frame == 0;
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if(keyframe){
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memset(c->Q, 0, sizeof(c->Q));
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c->last_bits_used = 0;
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}
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init_get_bits(gb, buf, buf_size * 8);
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skip_bits(gb, c->last_bits_used & 7);
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if(keyframe)
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maxband = mpc8_get_mod_golomb(gb, c->maxbands + 1);
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else{
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maxband = c->last_max_band + get_vlc2(gb, band_vlc.table, MPC8_BANDS_BITS, 2);
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if(maxband > 32) maxband -= 33;
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}
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c->last_max_band = maxband;
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/* read subband indexes */
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if(maxband){
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last[0] = last[1] = 0;
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for(i = maxband - 1; i >= 0; i--){
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for(ch = 0; ch < 2; ch++){
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last[ch] = get_vlc2(gb, res_vlc[last[ch] > 2].table, MPC8_RES_BITS, 2) + last[ch];
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if(last[ch] > 15) last[ch] -= 17;
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bands[i].res[ch] = last[ch];
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}
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}
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if(c->MSS){
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int mask;
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cnt = 0;
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for(i = 0; i < maxband; i++)
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if(bands[i].res[0] || bands[i].res[1])
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cnt++;
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t = mpc8_get_mod_golomb(gb, cnt);
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mask = mpc8_get_mask(gb, cnt, t);
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for(i = maxband - 1; i >= 0; i--)
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if(bands[i].res[0] || bands[i].res[1]){
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bands[i].msf = mask & 1;
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mask >>= 1;
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}
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}
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}
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for(i = maxband; i < c->maxbands; i++)
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bands[i].res[0] = bands[i].res[1] = 0;
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if(keyframe){
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for(i = 0; i < 32; i++)
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c->oldDSCF[0][i] = c->oldDSCF[1][i] = 1;
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}
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for(i = 0; i < maxband; i++){
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if(bands[i].res[0] || bands[i].res[1]){
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cnt = !!bands[i].res[0] + !!bands[i].res[1] - 1;
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if(cnt >= 0){
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t = get_vlc2(gb, scfi_vlc[cnt].table, scfi_vlc[cnt].bits, 1);
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if(bands[i].res[0]) bands[i].scfi[0] = t >> (2 * cnt);
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if(bands[i].res[1]) bands[i].scfi[1] = t & 3;
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}
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}
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}
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for(i = 0; i < maxband; i++){
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for(ch = 0; ch < 2; ch++){
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if(!bands[i].res[ch]) continue;
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if(c->oldDSCF[ch][i]){
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bands[i].scf_idx[ch][0] = get_bits(gb, 7) - 6;
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c->oldDSCF[ch][i] = 0;
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}else{
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t = get_vlc2(gb, dscf_vlc[1].table, MPC8_DSCF1_BITS, 2);
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if(t == 64)
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t += get_bits(gb, 6);
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bands[i].scf_idx[ch][0] = ((bands[i].scf_idx[ch][2] + t - 25) & 0x7F) - 6;
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}
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for(j = 0; j < 2; j++){
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if((bands[i].scfi[ch] << j) & 2)
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bands[i].scf_idx[ch][j + 1] = bands[i].scf_idx[ch][j];
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else{
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t = get_vlc2(gb, dscf_vlc[0].table, MPC8_DSCF0_BITS, 2);
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if(t == 31)
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t = 64 + get_bits(gb, 6);
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bands[i].scf_idx[ch][j + 1] = ((bands[i].scf_idx[ch][j] + t - 25) & 0x7F) - 6;
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}
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}
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}
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}
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for(i = 0, off = 0; i < maxband; i++, off += SAMPLES_PER_BAND){
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for(ch = 0; ch < 2; ch++){
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res = bands[i].res[ch];
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switch(res){
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case -1:
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for(j = 0; j < SAMPLES_PER_BAND; j++)
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c->Q[ch][off + j] = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
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break;
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case 0:
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break;
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case 1:
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for(j = 0; j < SAMPLES_PER_BAND; j += SAMPLES_PER_BAND / 2){
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cnt = get_vlc2(gb, q1_vlc.table, MPC8_Q1_BITS, 2);
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t = mpc8_get_mask(gb, 18, cnt);
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for(k = 0; k < SAMPLES_PER_BAND / 2; k++, t <<= 1)
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c->Q[ch][off + j + k] = (t & 0x20000) ? (get_bits1(gb) << 1) - 1 : 0;
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}
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break;
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case 2:
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cnt = 6;//2*mpc8_thres[res]
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for(j = 0; j < SAMPLES_PER_BAND; j += 3){
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t = get_vlc2(gb, q2_vlc[cnt > 3].table, MPC8_Q2_BITS, 2);
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c->Q[ch][off + j + 0] = mpc8_idx50[t];
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c->Q[ch][off + j + 1] = mpc8_idx51[t];
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c->Q[ch][off + j + 2] = mpc8_idx52[t];
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cnt = (cnt >> 1) + mpc8_huffq2[t];
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}
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break;
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case 3:
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case 4:
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for(j = 0; j < SAMPLES_PER_BAND; j += 2){
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t = get_vlc2(gb, q3_vlc[res - 3].table, MPC8_Q3_BITS, 2) + q3_offsets[res - 3];
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c->Q[ch][off + j + 1] = t >> 4;
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c->Q[ch][off + j + 0] = (t & 8) ? (t & 0xF) - 16 : (t & 0xF);
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}
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break;
|
|
case 5:
|
|
case 6:
|
|
case 7:
|
|
case 8:
|
|
cnt = 2 * mpc8_thres[res];
|
|
for(j = 0; j < SAMPLES_PER_BAND; j++){
|
|
t = get_vlc2(gb, quant_vlc[res - 5][cnt > mpc8_thres[res]].table, quant_vlc[res - 5][cnt > mpc8_thres[res]].bits, 2) + quant_offsets[res - 5];
|
|
c->Q[ch][off + j] = t;
|
|
cnt = (cnt >> 1) + FFABS(c->Q[ch][off + j]);
|
|
}
|
|
break;
|
|
default:
|
|
for(j = 0; j < SAMPLES_PER_BAND; j++){
|
|
c->Q[ch][off + j] = get_vlc2(gb, q9up_vlc.table, MPC8_Q9UP_BITS, 2);
|
|
if(res != 9){
|
|
c->Q[ch][off + j] <<= res - 9;
|
|
c->Q[ch][off + j] |= get_bits(gb, res - 9);
|
|
}
|
|
c->Q[ch][off + j] -= (1 << (res - 2)) - 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
ff_mpc_dequantize_and_synth(c, maxband, data, avctx->channels);
|
|
|
|
c->cur_frame++;
|
|
|
|
c->last_bits_used = get_bits_count(gb);
|
|
if(c->cur_frame >= c->frames)
|
|
c->cur_frame = 0;
|
|
*data_size = MPC_FRAME_SIZE * 2 * avctx->channels;
|
|
|
|
return c->cur_frame ? c->last_bits_used >> 3 : buf_size;
|
|
}
|
|
|
|
AVCodec ff_mpc8_decoder = {
|
|
"mpc8",
|
|
AVMEDIA_TYPE_AUDIO,
|
|
CODEC_ID_MUSEPACK8,
|
|
sizeof(MPCContext),
|
|
mpc8_decode_init,
|
|
NULL,
|
|
NULL,
|
|
mpc8_decode_frame,
|
|
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"),
|
|
};
|