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d77294c5e4
* qatar/master: libx264: fix indentation. vorbis: fix overflows in floor1[] vector and inverse db table index. win64: add a XMM clobber test configure option. movdec: Parse the dvc1 atom ARM: ac3: fix ac3_bit_alloc_calc_bap_armv6 swscale: K&R formatting cosmetics for Blackfin code frwu: lowercase the FRWU codec name movdec: fix dts generation in fragmented files fate: make acodec-ac3_fixed test output raw AC3 APIchanges: add missing commit hashes swscale: implement MMX, SSE2 and AVX functions for RGB32 input. ra144enc: drop pointless "encoder" from .long_name bethsoftvideo: fix palette reading. mpc7: use av_fast_padded_malloc() mpc7: simplify handling of packet sizes that are not a multiple of 4 bytes doc: decoding Forward Uncompressed is supported Fix a typo in the x86 asm version of ff_vector_clip_int32() pcmenc: Do not set avpkt->size. ff_alloc_packet: modify the size of the packet to match the requested size Conflicts: doc/APIchanges libavcodec/libx264.c libavcodec/mpc7.c libavformat/isom.h libswscale/Makefile libswscale/bfin/yuv2rgb_bfin.c tests/ref/fate/bethsoft-vid tests/ref/seek/ac3_ac3 Merged-by: Michael Niedermayer <michaelni@gmx.at>
79 lines
2.2 KiB
C
79 lines
2.2 KiB
C
/*
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* Musepack decoder
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* Copyright (c) 2006 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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* Musepack decoder
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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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#ifndef AVCODEC_MPC_H
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#define AVCODEC_MPC_H
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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 "mpegaudio.h"
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#include "mpegaudiodsp.h"
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#define BANDS 32
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#define SAMPLES_PER_BAND 36
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#define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
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/** Subband structure - hold all variables for each subband */
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typedef struct {
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int msf; ///< mid-stereo flag
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int res[2];
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int scfi[2];
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int scf_idx[2][3];
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int Q[2];
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}Band;
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typedef struct {
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AVFrame frame;
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DSPContext dsp;
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MPADSPContext mpadsp;
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GetBitContext gb;
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int IS, MSS, gapless;
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int lastframelen;
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int maxbands, last_max_band;
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int last_bits_used;
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int oldDSCF[2][BANDS];
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Band bands[BANDS];
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int Q[2][MPC_FRAME_SIZE];
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int cur_frame, frames;
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uint8_t *bits;
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int buf_size;
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AVLFG rnd;
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int frames_to_skip;
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/* for synthesis */
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DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
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int synth_buf_offset[MPA_MAX_CHANNELS];
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DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
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} MPCContext;
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void ff_mpc_init(void);
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void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, void *dst, int channels);
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#endif /* AVCODEC_MPC_H */
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