FFmpeg 7.1 + cuframes input demuxer

Snapshot FFmpeg n7.1 (release tag) с применённым patch'ем для cuframes input format. Используется как FFMPEG_REPO_OVERRIDE в NickM-27/FFmpeg-Builds fork для статической сборки patched binary под Frigate (Debian 12 / glibc 2.36). Apply changes: + libavformat/cuframesdec.c (новый — реализация демуксера) M libavformat/Makefile (CONFIG_CUFRAMES_DEMUXER target) M libavformat/allformats.c (extern declaration) M configure (--enable-libcuframes option + dep check) Patch source: https://git.goldix.org/gx/cuframes (filter/ffmpeg-7.1-cuframes-demuxer.patch) History сброшена (snapshot вместо fork) потому что upstream shallow clone не позволял push в gitea. Полная история FFmpeg — на github.com/FFmpeg/FFmpeg n7.1.
2026-05-17 11:43:10 +01:00
commit dcecd42de4
8554 changed files with 2122707 additions and 0 deletions
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+/*
+ * LOAS AudioSyncStream demuxer
+ * Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "libavutil/internal.h"
+#include "avformat.h"
+#include "demux.h"
+#include "internal.h"
+#include "rawdec.h"
+
+#define LOAS_SYNC_WORD 0x2b7
+
+static int loas_probe(const AVProbeData *p)
+{
+    int max_frames = 0, first_frames = 0;
+    int fsize, frames;
+    const uint8_t *buf0 = p->buf;
+    const uint8_t *buf2;
+    const uint8_t *buf;
+    const uint8_t *end = buf0 + p->buf_size - 3;
+    buf = buf0;
+
+    for (; buf < end; buf = buf2 + 1) {
+        buf2 = buf;
+
+        for (frames = 0; buf2 < end; frames++) {
+            uint32_t header = AV_RB24(buf2);
+            if ((header >> 13) != LOAS_SYNC_WORD)
+                break;
+            fsize = (header & 0x1FFF) + 3;
+            if (fsize < 7)
+                break;
+            fsize = FFMIN(fsize, end - buf2);
+            buf2 += fsize;
+        }
+        max_frames = FFMAX(max_frames, frames);
+        if (buf == buf0)
+            first_frames = frames;
+    }
+
+    if (first_frames >= 3)
+        return AVPROBE_SCORE_EXTENSION + 1;
+    else if (max_frames > 100)
+        return AVPROBE_SCORE_EXTENSION;
+    else if (max_frames >= 3)
+        return AVPROBE_SCORE_EXTENSION / 2;
+    else
+        return 0;
+}
+
+static int loas_read_header(AVFormatContext *s)
+{
+    AVStream *st;
+
+    st = avformat_new_stream(s, NULL);
+    if (!st)
+        return AVERROR(ENOMEM);
+
+    st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+    st->codecpar->codec_id   = AV_CODEC_ID_AAC_LATM;
+    ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
+
+    //LCM of all possible AAC sample rates
+    avpriv_set_pts_info(st, 64, 1, 28224000);
+
+    return 0;
+}
+
+const FFInputFormat ff_loas_demuxer = {
+    .p.name         = "loas",
+    .p.long_name    = NULL_IF_CONFIG_SMALL("LOAS AudioSyncStream"),
+    .p.flags        = AVFMT_GENERIC_INDEX,
+    .p.priv_class   = &ff_raw_demuxer_class,
+    .read_probe     = loas_probe,
+    .read_header    = loas_read_header,
+    .read_packet    = ff_raw_read_partial_packet,
+    .raw_codec_id = AV_CODEC_ID_AAC_LATM,
+    .priv_data_size = sizeof(FFRawDemuxerContext),
+};