Mercurial > libavcodec.hg
view mpc.h @ 12492:58a960d6e34c libavcodec
Rename h264_idct_sse2.asm to h264_idct.asm; move inline IDCT asm from
h264dsp_mmx.c to h264_idct.asm (as yasm code). Because the loops are now
coded in asm instead of C, this is (depending on the function) up to 50%
faster for cases where gcc didn't do a great job at looping.
Since h264_idct_add8() is now faster than the manual loop setup in h264.c,
in-asm idct calling can now be enabled for chroma as well (see r16207). For
MMX, this is 5% faster. For SSE2 (which isn't done for chroma if h264.c does
the looping), this makes it up to 50% faster. Speed gain overall is ~0.5-1.0%.
author | rbultje |
---|---|
date | Tue, 14 Sep 2010 13:36:26 +0000 |
parents | 7dd2a45249a9 |
children |
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/* * 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 "mpcdata.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 { DSPContext dsp; 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); #endif /* AVCODEC_MPC_H */