ffmpeg/libavcodec/mpc.h
Michael Niedermayer e4de71677f Merge remote-tracking branch 'qatar/master'
* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-03 03:00:30 +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 */