mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
e4de71677f
* 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>
79 lines
2.2 KiB
C
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 */
|