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>
56 lines
2.4 KiB
C
56 lines
2.4 KiB
C
/*
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* audio interleaving prototypes and declarations
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*
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* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot 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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#ifndef AVFORMAT_AUDIOINTERLEAVE_H
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#define AVFORMAT_AUDIOINTERLEAVE_H
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#include "libavutil/fifo.h"
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#include "avformat.h"
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typedef struct AudioInterleaveContext {
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AVFifoBuffer *fifo;
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unsigned fifo_size; ///< size of currently allocated FIFO
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uint64_t dts; ///< current dts
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int sample_size; ///< size of one sample all channels included
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const int *samples_per_frame; ///< must be 0-terminated
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const int *samples; ///< current samples per frame, pointer to samples_per_frame
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AVRational time_base; ///< time base of output audio packets
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} AudioInterleaveContext;
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int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base);
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void ff_audio_interleave_close(AVFormatContext *s);
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/**
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* Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame
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* and interleave them correctly.
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* The first element of AVStream->priv_data must be AudioInterleaveContext
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* when using this function.
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*
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* @param get_packet function will output a packet when streams are correctly interleaved.
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* @param compare_ts function will compare AVPackets and decide interleaving order.
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*/
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int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
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int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
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int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *));
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#endif /* AVFORMAT_AUDIOINTERLEAVE_H */
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