mirror of
https://github.com/hrydgard/ppsspp.git
synced 2026-11-02 20:42:29 +01:00
Henrik RydgårdandClaude Opus 5.5
d0e3141f43
The ffmpeg submodule held prebuilt libraries for every platform next to the sources (about 1.5 GB checked out), and nothing uses those any more. Only the needed sources are copied in, from the submodule's commit, along with every header and template they include. The path is new, so checking out commits on either side of this works. It's also cut down to what PPSSPP uses: the H.264, AAC, MP3, ATRAC3 and ATRAC3+ decoders, the H.264 parser, the demuxers as before, and the FFV1 encoder, AVI muxer and file protocol for AVI dumping. config.h takes its CONFIG_ values from FFmpeg 3.0.2's own configure run with just those enabled. The raw H.264 demuxer stays even though nothing opens raw H.264: probing PSMF data uses it to identify the video stream's codec, which otherwise comes out as none. On top of that: - The files nothing links (configure would build them, but they're never pulled out of the static library, going by the GCC link map) are left out: mostly libavutil's crypto and hashing and the alternative DCT/IDCT/FFT versions. - libswresample isn't built, now that nothing uses it. - The H.264 decoder only does 8-bit video, which is all the PSP has. The SPS parser rejects any other bit depth, and the DSP files no longer build their 9- to 16-bit variants, which the linker kept because the init functions pick one by bit depth at run time. That's 160 files to compile instead of 277, and 391 of FFmpeg's vendored, about 5.8 MB. The README in ext/ffmpeg has what was done, the PPSSPP patches in the copy, how to add a component, and the tradeoffs. Co-Authored-By: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
181 lines
6.0 KiB
C
181 lines
6.0 KiB
C
/*
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* AAC data declarations
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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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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* AAC data declarations
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* @author Oded Shimon ( ods15 ods15 dyndns org )
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* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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*/
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#ifndef AVCODEC_AACTAB_H
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#define AVCODEC_AACTAB_H
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#include "libavutil/mem.h"
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#include "aac.h"
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#include <stdint.h>
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/* NOTE:
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* Tables in this file are shared by the AAC decoders and encoder
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*/
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extern float ff_aac_pow2sf_tab[428];
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extern float ff_aac_pow34sf_tab[428];
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static inline void ff_aac_tableinit(void)
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{
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int i;
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/* 2^(i/16) for 0 <= i <= 15 */
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static const float exp2_lut[] = {
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1.00000000000000000000,
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1.04427378242741384032,
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1.09050773266525765921,
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1.13878863475669165370,
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1.18920711500272106672,
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1.24185781207348404859,
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1.29683955465100966593,
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1.35425554693689272830,
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1.41421356237309504880,
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1.47682614593949931139,
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1.54221082540794082361,
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1.61049033194925430818,
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1.68179283050742908606,
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1.75625216037329948311,
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1.83400808640934246349,
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1.91520656139714729387,
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};
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float t1 = 8.8817841970012523233890533447265625e-16; // 2^(-50)
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float t2 = 3.63797880709171295166015625e-12; // 2^(-38)
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int t1_inc_cur, t2_inc_cur;
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int t1_inc_prev = 0;
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int t2_inc_prev = 8;
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for (i = 0; i < 428; i++) {
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t1_inc_cur = 4 * (i % 4);
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t2_inc_cur = (8 + 3*i) % 16;
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if (t1_inc_cur < t1_inc_prev)
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t1 *= 2;
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if (t2_inc_cur < t2_inc_prev)
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t2 *= 2;
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// A much more efficient and accurate way of doing:
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// ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
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// ff_aac_pow34sf_tab[i] = pow(ff_aac_pow2sf_tab[i], 3.0/4.0);
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ff_aac_pow2sf_tab[i] = t1 * exp2_lut[t1_inc_cur];
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ff_aac_pow34sf_tab[i] = t2 * exp2_lut[t2_inc_cur];
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t1_inc_prev = t1_inc_cur;
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t2_inc_prev = t2_inc_cur;
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}
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}
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/* @name ltp_coef
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* Table of the LTP coefficients
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*/
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static const INTFLOAT ltp_coef[8] = {
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Q30(0.570829f), Q30(0.696616f), Q30(0.813004f), Q30(0.911304f),
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Q30(0.984900f), Q30(1.067894f), Q30(1.194601f), Q30(1.369533f),
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};
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/* @name tns_tmp2_map
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* Tables of the tmp2[] arrays of LPC coefficients used for TNS.
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* The suffix _M_N[] indicate the values of coef_compress and coef_res
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* respectively.
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* @{
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*/
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static const INTFLOAT tns_tmp2_map_1_3[4] = {
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Q31(0.00000000f), Q31(-0.43388373f), Q31(0.64278758f), Q31(0.34202015f),
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};
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static const INTFLOAT tns_tmp2_map_0_3[8] = {
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Q31(0.00000000f), Q31(-0.43388373f), Q31(-0.78183150f), Q31(-0.97492790f),
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Q31(0.98480773f), Q31( 0.86602539f), Q31( 0.64278758f), Q31( 0.34202015f),
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};
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static const INTFLOAT tns_tmp2_map_1_4[8] = {
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Q31(0.00000000f), Q31(-0.20791170f), Q31(-0.40673664f), Q31(-0.58778524f),
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Q31(0.67369562f), Q31( 0.52643216f), Q31( 0.36124167f), Q31( 0.18374951f),
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};
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static const INTFLOAT tns_tmp2_map_0_4[16] = {
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Q31( 0.00000000f), Q31(-0.20791170f), Q31(-0.40673664f), Q31(-0.58778524f),
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Q31(-0.74314481f), Q31(-0.86602539f), Q31(-0.95105654f), Q31(-0.99452192f),
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Q31( 0.99573416f), Q31( 0.96182561f), Q31( 0.89516330f), Q31( 0.79801720f),
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Q31( 0.67369562f), Q31( 0.52643216f), Q31( 0.36124167f), Q31( 0.18374951f),
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};
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static const INTFLOAT * const tns_tmp2_map[4] = {
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tns_tmp2_map_0_3,
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tns_tmp2_map_0_4,
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tns_tmp2_map_1_3,
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tns_tmp2_map_1_4
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};
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// @}
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/* @name window coefficients
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* @{
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*/
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DECLARE_ALIGNED(32, extern float, ff_aac_kbd_long_1024)[1024];
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DECLARE_ALIGNED(32, extern float, ff_aac_kbd_short_128)[128];
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DECLARE_ALIGNED(32, extern int, ff_aac_kbd_long_1024_fixed)[1024];
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DECLARE_ALIGNED(32, extern int, ff_aac_kbd_long_512_fixed)[512];
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DECLARE_ALIGNED(32, extern int, ff_aac_kbd_short_128_fixed)[128];
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const DECLARE_ALIGNED(32, extern float, ff_aac_eld_window_512)[1920];
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const DECLARE_ALIGNED(32, extern int, ff_aac_eld_window_512_fixed)[1920];
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const DECLARE_ALIGNED(32, extern float, ff_aac_eld_window_480)[1800];
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const DECLARE_ALIGNED(32, extern int, ff_aac_eld_window_480_fixed)[1800];
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// @}
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/* @name number of scalefactor window bands for long and short transform windows respectively
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* @{
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*/
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extern const uint8_t ff_aac_num_swb_1024[];
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extern const uint8_t ff_aac_num_swb_512 [];
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extern const uint8_t ff_aac_num_swb_480 [];
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extern const uint8_t ff_aac_num_swb_128 [];
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// @}
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extern const uint8_t ff_aac_pred_sfb_max [];
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extern const uint32_t ff_aac_scalefactor_code[121];
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extern const uint8_t ff_aac_scalefactor_bits[121];
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extern const uint16_t * const ff_aac_spectral_codes[11];
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extern const uint8_t * const ff_aac_spectral_bits [11];
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extern const uint16_t ff_aac_spectral_sizes[11];
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extern const float *ff_aac_codebook_vectors[];
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extern const float *ff_aac_codebook_vector_vals[];
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extern const uint16_t *ff_aac_codebook_vector_idx[];
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extern const uint16_t * const ff_swb_offset_1024[13];
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extern const uint16_t * const ff_swb_offset_512 [13];
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extern const uint16_t * const ff_swb_offset_480 [13];
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extern const uint16_t * const ff_swb_offset_128 [13];
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extern const uint8_t ff_tns_max_bands_1024[13];
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extern const uint8_t ff_tns_max_bands_512 [13];
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extern const uint8_t ff_tns_max_bands_480 [13];
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extern const uint8_t ff_tns_max_bands_128 [13];
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#endif /* AVCODEC_AACTAB_H */
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