Files
ppsspp/ext/ffmpeg/libavcodec/sbr.h
T
Henrik RydgårdandClaude Opus 5.5 d0e3141f43 FFmpeg: Vendor what PPSSPP uses into ext/ffmpeg, drop the submodule
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>
2026-10-25 22:27:54 +01:00

218 lines
7.5 KiB
C

/*
* Spectral Band Replication definitions and structures
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
* Spectral Band Replication definitions and structures
* @author Robert Swain ( rob opendot cl )
*/
#ifndef AVCODEC_SBR_H
#define AVCODEC_SBR_H
#include <stdint.h>
#include "fft.h"
#include "aacps.h"
#include "sbrdsp.h"
typedef struct AACContext AACContext;
/**
* Spectral Band Replication header - spectrum parameters that invoke a reset if they differ from the previous header.
*/
typedef struct SpectrumParameters {
uint8_t bs_start_freq;
uint8_t bs_stop_freq;
uint8_t bs_xover_band;
/**
* @name Variables associated with bs_header_extra_1
* @{
*/
uint8_t bs_freq_scale;
uint8_t bs_alter_scale;
uint8_t bs_noise_bands;
/** @} */
} SpectrumParameters;
#define SBR_SYNTHESIS_BUF_SIZE ((1280-128)*2)
/**
* Spectral Band Replication per channel data
*/
typedef struct SBRData {
/**
* @name Main bitstream data variables
* @{
*/
unsigned bs_frame_class;
unsigned bs_add_harmonic_flag;
AAC_SIGNE bs_num_env;
uint8_t bs_freq_res[7];
AAC_SIGNE bs_num_noise;
uint8_t bs_df_env[5];
uint8_t bs_df_noise[2];
uint8_t bs_invf_mode[2][5];
uint8_t bs_add_harmonic[48];
unsigned bs_amp_res;
/** @} */
/**
* @name State variables
* @{
*/
DECLARE_ALIGNED(32, INTFLOAT, synthesis_filterbank_samples)[SBR_SYNTHESIS_BUF_SIZE];
DECLARE_ALIGNED(32, INTFLOAT, analysis_filterbank_samples) [1312];
int synthesis_filterbank_samples_offset;
///l_APrev and l_A
int e_a[2];
///Chirp factors
INTFLOAT bw_array[5];
///QMF values of the original signal
INTFLOAT W[2][32][32][2];
///QMF output of the HF adjustor
int Ypos;
DECLARE_ALIGNED(16, INTFLOAT, Y)[2][38][64][2];
DECLARE_ALIGNED(16, AAC_FLOAT, g_temp)[42][48];
AAC_FLOAT q_temp[42][48];
uint8_t s_indexmapped[8][48];
///Envelope scalefactors
uint8_t env_facs_q[6][48];
AAC_FLOAT env_facs[6][48];
///Noise scalefactors
uint8_t noise_facs_q[3][5];
AAC_FLOAT noise_facs[3][5];
///Envelope time borders
uint8_t t_env[8];
///Envelope time border of the last envelope of the previous frame
uint8_t t_env_num_env_old;
///Noise time borders
uint8_t t_q[3];
unsigned f_indexnoise;
unsigned f_indexsine;
/** @} */
} SBRData;
typedef struct SpectralBandReplication SpectralBandReplication;
/**
* aacsbr functions pointers
*/
typedef struct AACSBRContext {
int (*sbr_lf_gen)(AACContext *ac, SpectralBandReplication *sbr,
INTFLOAT X_low[32][40][2], const INTFLOAT W[2][32][32][2],
int buf_idx);
void (*sbr_hf_assemble)(INTFLOAT Y1[38][64][2],
const INTFLOAT X_high[64][40][2],
SpectralBandReplication *sbr, SBRData *ch_data,
const int e_a[2]);
int (*sbr_x_gen)(SpectralBandReplication *sbr, INTFLOAT X[2][38][64],
const INTFLOAT Y0[38][64][2], const INTFLOAT Y1[38][64][2],
const INTFLOAT X_low[32][40][2], int ch);
void (*sbr_hf_inverse_filter)(SBRDSPContext *dsp,
INTFLOAT (*alpha0)[2], INTFLOAT (*alpha1)[2],
const INTFLOAT X_low[32][40][2], int k0);
} AACSBRContext;
/**
* Spectral Band Replication
*/
struct SpectralBandReplication {
int sample_rate;
int start;
int ready_for_dequant;
int id_aac;
int reset;
SpectrumParameters spectrum_params;
int bs_amp_res_header;
/**
* @name Variables associated with bs_header_extra_2
* @{
*/
unsigned bs_limiter_bands;
unsigned bs_limiter_gains;
unsigned bs_interpol_freq;
unsigned bs_smoothing_mode;
/** @} */
unsigned bs_coupling;
AAC_SIGNE k[5]; ///< k0, k1, k2
///kx', and kx respectively, kx is the first QMF subband where SBR is used.
///kx' is its value from the previous frame
AAC_SIGNE kx[2];
///M' and M respectively, M is the number of QMF subbands that use SBR.
AAC_SIGNE m[2];
unsigned kx_and_m_pushed;
///The number of frequency bands in f_master
AAC_SIGNE n_master;
SBRData data[2];
PSContext ps;
///N_Low and N_High respectively, the number of frequency bands for low and high resolution
AAC_SIGNE n[2];
///Number of noise floor bands
AAC_SIGNE n_q;
///Number of limiter bands
AAC_SIGNE n_lim;
///The master QMF frequency grouping
uint16_t f_master[49];
///Frequency borders for low resolution SBR
uint16_t f_tablelow[25];
///Frequency borders for high resolution SBR
uint16_t f_tablehigh[49];
///Frequency borders for noise floors
uint16_t f_tablenoise[6];
///Frequency borders for the limiter
uint16_t f_tablelim[30];
AAC_SIGNE num_patches;
uint8_t patch_num_subbands[6];
uint8_t patch_start_subband[6];
///QMF low frequency input to the HF generator
DECLARE_ALIGNED(16, INTFLOAT, X_low)[32][40][2];
///QMF output of the HF generator
DECLARE_ALIGNED(16, INTFLOAT, X_high)[64][40][2];
///QMF values of the reconstructed signal
DECLARE_ALIGNED(16, INTFLOAT, X)[2][2][38][64];
///Zeroth coefficient used to filter the subband signals
DECLARE_ALIGNED(16, INTFLOAT, alpha0)[64][2];
///First coefficient used to filter the subband signals
DECLARE_ALIGNED(16, INTFLOAT, alpha1)[64][2];
///Dequantized envelope scalefactors, remapped
AAC_FLOAT e_origmapped[7][48];
///Dequantized noise scalefactors, remapped
AAC_FLOAT q_mapped[7][48];
///Sinusoidal presence, remapped
uint8_t s_mapped[7][48];
///Estimated envelope
AAC_FLOAT e_curr[7][48];
///Amplitude adjusted noise scalefactors
AAC_FLOAT q_m[7][48];
///Sinusoidal levels
AAC_FLOAT s_m[7][48];
AAC_FLOAT gain[7][48];
DECLARE_ALIGNED(32, INTFLOAT, qmf_filter_scratch)[5][64];
FFTContext mdct_ana;
FFTContext mdct;
SBRDSPContext dsp;
AACSBRContext c;
};
#endif /* AVCODEC_SBR_H */