FFmpeg/libavcodec/mpc.h
Michael Niedermayer d77294c5e4 Merge remote-tracking branch 'qatar/master'
* 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>
2012-02-03 03:51:32 +01:00

79 lines
2.2 KiB
C

/*
* Musepack decoder
* Copyright (c) 2006 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* 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
* Musepack decoder
* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
* divided into 32 subbands.
*/
#ifndef AVCODEC_MPC_H
#define AVCODEC_MPC_H
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#define BANDS 32
#define SAMPLES_PER_BAND 36
#define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
/** Subband structure - hold all variables for each subband */
typedef struct {
int msf; ///< mid-stereo flag
int res[2];
int scfi[2];
int scf_idx[2][3];
int Q[2];
}Band;
typedef struct {
AVFrame frame;
DSPContext dsp;
MPADSPContext mpadsp;
GetBitContext gb;
int IS, MSS, gapless;
int lastframelen;
int maxbands, last_max_band;
int last_bits_used;
int oldDSCF[2][BANDS];
Band bands[BANDS];
int Q[2][MPC_FRAME_SIZE];
int cur_frame, frames;
uint8_t *bits;
int buf_size;
AVLFG rnd;
int frames_to_skip;
/* for synthesis */
DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
int synth_buf_offset[MPA_MAX_CHANNELS];
DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
} MPCContext;
void ff_mpc_init(void);
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, void *dst, int channels);
#endif /* AVCODEC_MPC_H */