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* 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
/*
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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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