mirror of
https://github.com/hrydgard/ppsspp.git
synced 2026-11-01 20:18:15 +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>
111 lines
5.8 KiB
C
111 lines
5.8 KiB
C
/*
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* MPEG-4 Audio common header
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* Copyright (c) 2008 Baptiste Coudurier <baptiste.coudurier@free.fr>
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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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#ifndef AVCODEC_MPEG4AUDIO_H
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#define AVCODEC_MPEG4AUDIO_H
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#include <stdint.h>
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#include "get_bits.h"
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#include "put_bits.h"
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typedef struct MPEG4AudioConfig {
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int object_type;
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int sampling_index;
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int sample_rate;
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int chan_config;
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int sbr; ///< -1 implicit, 1 presence
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int ext_object_type;
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int ext_sampling_index;
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int ext_sample_rate;
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int ext_chan_config;
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int channels;
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int ps; ///< -1 implicit, 1 presence
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int frame_length_short;
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} MPEG4AudioConfig;
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extern av_export const int avpriv_mpeg4audio_sample_rates[16];
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extern const uint8_t ff_mpeg4audio_channels[8];
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/**
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* Parse MPEG-4 systems extradata to retrieve audio configuration.
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* @param[in] c MPEG4AudioConfig structure to fill.
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* @param[in] buf Extradata from container.
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* @param[in] bit_size Extradata size in bits.
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* @param[in] sync_extension look for a sync extension after config if true.
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* @return On error -1 is returned, on success AudioSpecificConfig bit index in extradata.
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*/
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int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf,
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int bit_size, int sync_extension);
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enum AudioObjectType {
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AOT_NULL,
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// Support? Name
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AOT_AAC_MAIN, ///< Y Main
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AOT_AAC_LC, ///< Y Low Complexity
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AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
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AOT_AAC_LTP, ///< Y Long Term Prediction
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AOT_SBR, ///< Y Spectral Band Replication
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AOT_AAC_SCALABLE, ///< N Scalable
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AOT_TWINVQ, ///< N Twin Vector Quantizer
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AOT_CELP, ///< N Code Excited Linear Prediction
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AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
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AOT_TTSI = 12, ///< N Text-To-Speech Interface
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AOT_MAINSYNTH, ///< N Main Synthesis
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AOT_WAVESYNTH, ///< N Wavetable Synthesis
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AOT_MIDI, ///< N General MIDI
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AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
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AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
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AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
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AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
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AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
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AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
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AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
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AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
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AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
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AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
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AOT_ER_PARAM, ///< N Error Resilient Parametric
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AOT_SSC, ///< N SinuSoidal Coding
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AOT_PS, ///< N Parametric Stereo
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AOT_SURROUND, ///< N MPEG Surround
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AOT_ESCAPE, ///< Y Escape Value
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AOT_L1, ///< Y Layer 1
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AOT_L2, ///< Y Layer 2
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AOT_L3, ///< Y Layer 3
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AOT_DST, ///< N Direct Stream Transfer
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AOT_ALS, ///< Y Audio LosslesS
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AOT_SLS, ///< N Scalable LosslesS
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AOT_SLS_NON_CORE, ///< N Scalable LosslesS (non core)
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AOT_ER_AAC_ELD, ///< N Error Resilient Enhanced Low Delay
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AOT_SMR_SIMPLE, ///< N Symbolic Music Representation Simple
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AOT_SMR_MAIN, ///< N Symbolic Music Representation Main
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AOT_USAC_NOSBR, ///< N Unified Speech and Audio Coding (no SBR)
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AOT_SAOC, ///< N Spatial Audio Object Coding
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AOT_LD_SURROUND, ///< N Low Delay MPEG Surround
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AOT_USAC, ///< N Unified Speech and Audio Coding
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};
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#define MAX_PCE_SIZE 320 ///<Maximum size of a PCE including the 3-bit ID_PCE
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///<marker and the comment
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int avpriv_copy_pce_data(PutBitContext *pb, GetBitContext *gb);
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#endif /* AVCODEC_MPEG4AUDIO_H */
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