Mercurial > libavcodec.hg
view mpegaudiodec_float.c @ 12266:48d6738904a9 libavcodec
Fix SPLATB_REG mess. Used to be a if/elseif/elseif/elseif spaghetti, so this
splits it into small optimization-specific macros which are selected for each
DSP function. The advantage of this approach is that the sse4 functions now
use the ssse3 codepath also without needing an explicit sse4 codepath.
| author | rbultje |
|---|---|
| date | Sat, 24 Jul 2010 19:33:05 +0000 |
| parents | fb3fcaf3c1b6 |
| children |
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/* * Float MPEG Audio decoder * Copyright (c) 2010 Michael Niedermayer * * 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 */ #define CONFIG_FLOAT 1 #include "mpegaudiodec.c" void ff_mpa_synth_filter_float(MPADecodeContext *s, float *synth_buf_ptr, int *synth_buf_offset, float *window, int *dither_state, float *samples, int incr, float sb_samples[SBLIMIT]) { float *synth_buf; int offset; offset = *synth_buf_offset; synth_buf = synth_buf_ptr + offset; s->dct.dct32(synth_buf, sb_samples); s->apply_window_mp3(synth_buf, window, dither_state, samples, incr); offset = (offset - 32) & 511; *synth_buf_offset = offset; } static void compute_antialias_float(MPADecodeContext *s, GranuleDef *g) { float *ptr; int n, i; /* we antialias only "long" bands */ if (g->block_type == 2) { if (!g->switch_point) return; /* XXX: check this for 8000Hz case */ n = 1; } else { n = SBLIMIT - 1; } ptr = g->sb_hybrid + 18; for(i = n;i > 0;i--) { float tmp0, tmp1; float *csa = &csa_table_float[0][0]; #define FLOAT_AA(j)\ tmp0= ptr[-1-j];\ tmp1= ptr[ j];\ ptr[-1-j] = tmp0 * csa[0+4*j] - tmp1 * csa[1+4*j];\ ptr[ j] = tmp0 * csa[1+4*j] + tmp1 * csa[0+4*j]; FLOAT_AA(0) FLOAT_AA(1) FLOAT_AA(2) FLOAT_AA(3) FLOAT_AA(4) FLOAT_AA(5) FLOAT_AA(6) FLOAT_AA(7) ptr += 18; } } static av_cold int decode_end(AVCodecContext * avctx) { MPADecodeContext *s = avctx->priv_data; ff_dct_end(&s->dct); return 0; } #if CONFIG_MP1FLOAT_DECODER AVCodec mp1float_decoder = { "mp1float", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP1, sizeof(MPADecodeContext), decode_init, NULL, decode_end, decode_frame, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), }; #endif #if CONFIG_MP2FLOAT_DECODER AVCodec mp2float_decoder = { "mp2float", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP2, sizeof(MPADecodeContext), decode_init, NULL, decode_end, decode_frame, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), }; #endif #if CONFIG_MP3FLOAT_DECODER AVCodec mp3float_decoder = { "mp3float", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP3, sizeof(MPADecodeContext), decode_init, NULL, decode_end, decode_frame, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), }; #endif #if CONFIG_MP3ADUFLOAT_DECODER AVCodec mp3adufloat_decoder = { "mp3adufloat", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP3ADU, sizeof(MPADecodeContext), decode_init, NULL, decode_end, decode_frame_adu, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), }; #endif #if CONFIG_MP3ON4FLOAT_DECODER AVCodec mp3on4float_decoder = { "mp3on4float", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP3ON4, sizeof(MP3On4DecodeContext), decode_init_mp3on4, NULL, decode_close_mp3on4, decode_frame_mp3on4, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"), }; #endif
