/* * Limitless Audio Format demuxer * Copyright (c) 2022 Paul B Mahol * * 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 "avformat.h" #include "avio_internal.h" #include "internal.h" #define MAX_STREAMS 4096 typedef struct StreamParams { AVChannelLayout layout; float horizontal; float vertical; int lfe; int stored; } StreamParams; typedef struct LAFContext { uint8_t *data; unsigned nb_stored; unsigned stored_index; unsigned index; unsigned bpp; StreamParams p[MAX_STREAMS]; int header_len; uint8_t header[(MAX_STREAMS + 7) / 8]; } LAFContext; static int laf_probe(const AVProbeData *p) { if (memcmp(p->buf, "LIMITLESS", 9)) return 0; if (memcmp(p->buf + 9, "HEAD", 4)) return 0; return AVPROBE_SCORE_MAX; } static int laf_read_header(AVFormatContext *ctx) { LAFContext *s = ctx->priv_data; AVIOContext *pb = ctx->pb; unsigned st_count, mode; unsigned sample_rate; int64_t duration; int codec_id; int quality; int bpp; avio_skip(pb, 9); if (avio_rb32(pb) != MKBETAG('H','E','A','D')) return AVERROR_INVALIDDATA; quality = avio_r8(pb); if (quality > 3) return AVERROR_INVALIDDATA; mode = avio_r8(pb); if (mode > 1) return AVERROR_INVALIDDATA; st_count = avio_rl32(pb); if (st_count == 0 || st_count > MAX_STREAMS) return AVERROR_INVALIDDATA; for (int i = 0; i < st_count; i++) { StreamParams *stp = &s->p[i]; stp->vertical = av_int2float(avio_rl32(pb)); stp->horizontal = av_int2float(avio_rl32(pb)); stp->lfe = avio_r8(pb); if (stp->lfe) { stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_LOW_FREQUENCY)); } else if (stp->vertical == 0.f && stp->horizontal == 0.f) { stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_CENTER)); } else if (stp->vertical == 0.f && stp->horizontal == -30.f) { stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_LEFT)); } else if (stp->vertical == 0.f && stp->horizontal == 30.f) { stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_RIGHT)); } else if (stp->vertical == 0.f && stp->horizontal == -110.f) { stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_SIDE_LEFT)); } else if (stp->vertical == 0.f && stp->horizontal == 110.f) { stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_SIDE_RIGHT)); } else { stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO; } } sample_rate = avio_rl32(pb); duration = avio_rl64(pb) / st_count; if (avio_feof(pb)) return AVERROR_INVALIDDATA; switch (quality) { case 0: codec_id = AV_CODEC_ID_PCM_U8; bpp = 1; break; case 1: codec_id = AV_CODEC_ID_PCM_S16LE; bpp = 2; break; case 2: codec_id = AV_CODEC_ID_PCM_F32LE; bpp = 4; break; case 3: codec_id = AV_CODEC_ID_PCM_S24LE; bpp = 3; break; default: return AVERROR_INVALIDDATA; } s->index = 0; s->stored_index = 0; s->bpp = bpp; if ((int64_t)bpp * st_count * (int64_t)sample_rate >= INT32_MAX) return AVERROR_INVALIDDATA; s->data = av_calloc(st_count * sample_rate, bpp); if (!s->data) return AVERROR(ENOMEM); for (int st = 0; st < st_count; st++) { StreamParams *stp = &s->p[st]; AVCodecParameters *par; AVStream *st = avformat_new_stream(ctx, NULL); if (!st) return AVERROR(ENOMEM); par = st->codecpar; par->codec_id = codec_id; par->codec_type = AVMEDIA_TYPE_AUDIO; par->ch_layout.nb_channels = 1; par->ch_layout = stp->layout; par->sample_rate = sample_rate; st->duration = duration; avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate); } s->header_len = (ctx->nb_streams + 7) / 8; return 0; } static int laf_read_packet(AVFormatContext *ctx, AVPacket *pkt) { AVIOContext *pb = ctx->pb; LAFContext *s = ctx->priv_data; AVStream *st = ctx->streams[0]; const int bpp = s->bpp; StreamParams *stp; int64_t pos; int ret; pos = avio_tell(pb); again: if (avio_feof(pb)) return AVERROR_EOF; if (s->index >= ctx->nb_streams) { int cur_st = 0, st_count = 0, st_index = 0; ret = ffio_read_size(pb, s->header, s->header_len); if (ret < 0) return ret; for (int i = 0; i < s->header_len; i++) { uint8_t val = s->header[i]; for (int j = 0; j < 8 && cur_st < ctx->nb_streams; j++, cur_st++) { StreamParams *stp = &s->p[st_index]; stp->stored = 0; if (val & 1) { stp->stored = 1; st_count++; } val >>= 1; st_index++; } } s->index = s->stored_index = 0; s->nb_stored = st_count; if (!st_count) return AVERROR_INVALIDDATA; ret = ffio_read_size(pb, s->data, st_count * st->codecpar->sample_rate * bpp); if (ret < 0) return ret; } st = ctx->streams[s->index]; stp = &s->p[s->index]; while (!stp->stored) { s->index++; if (s->index >= ctx->nb_streams) goto again; stp = &s->p[s->index]; } st = ctx->streams[s->index]; ret = av_new_packet(pkt, st->codecpar->sample_rate * bpp); if (ret < 0) return ret; switch (bpp) { case 1: for (int n = 0; n < st->codecpar->sample_rate; n++) pkt->data[n] = s->data[n * s->nb_stored + s->stored_index]; break; case 2: for (int n = 0; n < st->codecpar->sample_rate; n++) AV_WN16(pkt->data + n * 2, AV_RN16(s->data + n * s->nb_stored * 2 + s->stored_index * 2)); break; case 3: for (int n = 0; n < st->codecpar->sample_rate; n++) AV_WL24(pkt->data + n * 3, AV_RL24(s->data + n * s->nb_stored * 3 + s->stored_index * 3)); break; case 4: for (int n = 0; n < st->codecpar->sample_rate; n++) AV_WN32(pkt->data + n * 4, AV_RN32(s->data + n * s->nb_stored * 4 + s->stored_index * 4)); break; } pkt->stream_index = s->index; pkt->pos = pos; s->index++; s->stored_index++; return 0; } static int laf_read_close(AVFormatContext *ctx) { LAFContext *s = ctx->priv_data; av_freep(&s->data); return 0; } static int laf_read_seek(AVFormatContext *ctx, int stream_index, int64_t timestamp, int flags) { LAFContext *s = ctx->priv_data; s->stored_index = s->index = s->nb_stored = 0; return -1; } const AVInputFormat ff_laf_demuxer = { .name = "laf", .long_name = NULL_IF_CONFIG_SMALL("LAF (Limitless Audio Format)"), .priv_data_size = sizeof(LAFContext), .read_probe = laf_probe, .read_header = laf_read_header, .read_packet = laf_read_packet, .read_close = laf_read_close, .read_seek = laf_read_seek, .extensions = "laf", .flags = AVFMT_GENERIC_INDEX, .flags_internal = FF_FMT_INIT_CLEANUP, };