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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>
231 lines
8.0 KiB
C
231 lines
8.0 KiB
C
/*
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* Copyright (c) 2010 Alex Converse <alex.converse@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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* Note: Rounding-to-nearest used unless otherwise stated
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*
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*/
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#include <stdint.h>
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#include "config.h"
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#include "libavutil/attributes.h"
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#include "aacpsdsp.h"
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static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n)
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{
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int i;
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for (i = 0; i < n; i++)
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dst[i] += AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]);
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}
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static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
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int n)
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{
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int i;
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for (i = 0; i < n; i++) {
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dst[i][0] = AAC_MUL16(src0[i][0], src1[i]);
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dst[i][1] = AAC_MUL16(src0[i][1], src1[i]);
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}
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}
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static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
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const INTFLOAT (*filter)[8][2],
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int stride, int n)
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{
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int i, j;
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for (i = 0; i < n; i++) {
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INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0];
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INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
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for (j = 0; j < 6; j++) {
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INTFLOAT in0_re = in[j][0];
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INTFLOAT in0_im = in[j][1];
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INTFLOAT in1_re = in[12-j][0];
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INTFLOAT in1_im = in[12-j][1];
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sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
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(INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
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sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
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(INT64FLOAT)filter[i][j][1] * (in0_re - in1_re);
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}
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#if USE_FIXED
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out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31);
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out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31);
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#else
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out[i * stride][0] = sum_re;
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out[i * stride][1] = sum_im;
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#endif /* USE_FIXED */
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}
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}
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static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
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int i, int len)
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{
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int j;
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for (; i < 64; i++) {
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for (j = 0; j < len; j++) {
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out[i][j][0] = L[0][j][i];
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out[i][j][1] = L[1][j][i];
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}
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}
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}
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static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64],
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INTFLOAT (*in)[32][2],
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int i, int len)
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{
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int n;
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for (; i < 64; i++) {
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for (n = 0; n < len; n++) {
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out[0][n][i] = in[i][n][0];
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out[1][n][i] = in[i][n][1];
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}
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}
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}
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static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
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INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
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const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
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const INTFLOAT *transient_gain,
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INTFLOAT g_decay_slope,
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int len)
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{
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static const INTFLOAT a[] = { Q31(0.65143905753106f),
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Q31(0.56471812200776f),
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Q31(0.48954165955695f) };
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INTFLOAT ag[PS_AP_LINKS];
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int m, n;
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for (m = 0; m < PS_AP_LINKS; m++)
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ag[m] = AAC_MUL30(a[m], g_decay_slope);
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for (n = 0; n < len; n++) {
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INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]);
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INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]);
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for (m = 0; m < PS_AP_LINKS; m++) {
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INTFLOAT a_re = AAC_MUL31(ag[m], in_re);
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INTFLOAT a_im = AAC_MUL31(ag[m], in_im);
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INTFLOAT link_delay_re = ap_delay[m][n+2-m][0];
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INTFLOAT link_delay_im = ap_delay[m][n+2-m][1];
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INTFLOAT fractional_delay_re = Q_fract[m][0];
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INTFLOAT fractional_delay_im = Q_fract[m][1];
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INTFLOAT apd_re = in_re;
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INTFLOAT apd_im = in_im;
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in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
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link_delay_im, fractional_delay_im);
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in_re -= a_re;
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in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
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link_delay_im, fractional_delay_re);
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in_im -= a_im;
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ap_delay[m][n+5][0] = apd_re + AAC_MUL31(ag[m], in_re);
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ap_delay[m][n+5][1] = apd_im + AAC_MUL31(ag[m], in_im);
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}
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out[n][0] = AAC_MUL16(transient_gain[n], in_re);
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out[n][1] = AAC_MUL16(transient_gain[n], in_im);
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}
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}
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static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
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INTFLOAT h[2][4], INTFLOAT h_step[2][4],
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int len)
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{
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INTFLOAT h0 = h[0][0];
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INTFLOAT h1 = h[0][1];
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INTFLOAT h2 = h[0][2];
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INTFLOAT h3 = h[0][3];
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INTFLOAT hs0 = h_step[0][0];
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INTFLOAT hs1 = h_step[0][1];
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INTFLOAT hs2 = h_step[0][2];
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INTFLOAT hs3 = h_step[0][3];
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int n;
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for (n = 0; n < len; n++) {
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//l is s, r is d
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INTFLOAT l_re = l[n][0];
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INTFLOAT l_im = l[n][1];
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INTFLOAT r_re = r[n][0];
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INTFLOAT r_im = r[n][1];
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h0 += hs0;
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h1 += hs1;
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h2 += hs2;
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h3 += hs3;
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l[n][0] = AAC_MADD30(h0, l_re, h2, r_re);
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l[n][1] = AAC_MADD30(h0, l_im, h2, r_im);
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r[n][0] = AAC_MADD30(h1, l_re, h3, r_re);
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r[n][1] = AAC_MADD30(h1, l_im, h3, r_im);
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}
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}
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static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
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INTFLOAT h[2][4], INTFLOAT h_step[2][4],
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int len)
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{
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INTFLOAT h00 = h[0][0], h10 = h[1][0];
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INTFLOAT h01 = h[0][1], h11 = h[1][1];
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INTFLOAT h02 = h[0][2], h12 = h[1][2];
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INTFLOAT h03 = h[0][3], h13 = h[1][3];
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INTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
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INTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
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INTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
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INTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
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int n;
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for (n = 0; n < len; n++) {
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//l is s, r is d
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INTFLOAT l_re = l[n][0];
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INTFLOAT l_im = l[n][1];
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INTFLOAT r_re = r[n][0];
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INTFLOAT r_im = r[n][1];
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h00 += hs00;
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h01 += hs01;
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h02 += hs02;
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h03 += hs03;
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h10 += hs10;
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h11 += hs11;
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h12 += hs12;
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h13 += hs13;
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l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im);
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l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re);
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r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im);
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r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re);
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}
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}
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av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s)
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{
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s->add_squares = ps_add_squares_c;
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s->mul_pair_single = ps_mul_pair_single_c;
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s->hybrid_analysis = ps_hybrid_analysis_c;
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s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
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s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
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s->decorrelate = ps_decorrelate_c;
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s->stereo_interpolate[0] = ps_stereo_interpolate_c;
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s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c;
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#if !USE_FIXED
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if (ARCH_ARM)
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ff_psdsp_init_arm(s);
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if (ARCH_MIPS)
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ff_psdsp_init_mips(s);
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if (ARCH_X86)
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ff_psdsp_init_x86(s);
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#endif /* !USE_FIXED */
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}
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