Mercurial > libavcodec.hg
view mpegaudiodec_float.c @ 12085:8454bb880008 libavcodec
PPC: add _interleave versions of fft{4,6,16}_altivec
This removes the need for a post-swizzle with the small FFTs.
author | mru |
---|---|
date | Sat, 03 Jul 2010 18:36:10 +0000 |
parents | 49c528a39187 |
children | fb3fcaf3c1b6 |
line wrap: on
line source
/* * 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; } } #if CONFIG_MP1FLOAT_DECODER AVCodec mp1float_decoder = { "mp1float", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP1, sizeof(MPADecodeContext), decode_init, NULL, NULL, 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, NULL, 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, NULL, 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, NULL, 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