diff options
author | Paul B Mahol <onemda@gmail.com> | 2022-09-11 20:10:27 +0200 |
---|---|---|
committer | Paul B Mahol <onemda@gmail.com> | 2022-09-16 10:02:19 +0200 |
commit | dd2a01ef5cad08347ecbbcba7afd5e5a0810f504 (patch) | |
tree | af1576d6418c43337922906beabb102a9c9f1b90 | |
parent | e91bc521afff17a73012af153f219c3e5f3f2ce5 (diff) | |
download | ffmpeg-dd2a01ef5cad08347ecbbcba7afd5e5a0810f504.tar.gz |
avformat: add LAF demuxer
-rw-r--r-- | Changelog | 1 | ||||
-rw-r--r-- | doc/general_contents.texi | 2 | ||||
-rw-r--r-- | libavformat/Makefile | 1 | ||||
-rw-r--r-- | libavformat/allformats.c | 1 | ||||
-rw-r--r-- | libavformat/lafdec.c | 271 | ||||
-rw-r--r-- | libavformat/version.h | 2 |
6 files changed, 277 insertions, 1 deletions
@@ -13,6 +13,7 @@ version <next>: - a3dscope filter - bonk decoder and demuxer - Micronas SC-4 audio decoder +- LAF demuxer version 5.1: diff --git a/doc/general_contents.texi b/doc/general_contents.texi index 150b7944a8..a632b23f6f 100644 --- a/doc/general_contents.texi +++ b/doc/general_contents.texi @@ -510,6 +510,8 @@ library: @tab A format used by libvpx @item Internet Video Recording @tab @tab X @item IRCAM @tab X @tab X +@item LAF @tab @tab X + @tab Limitless Audio Format @item LATM @tab X @tab X @item LMLM4 @tab @tab X @tab Used by Linux Media Labs MPEG-4 PCI boards diff --git a/libavformat/Makefile b/libavformat/Makefile index 5cdcda3239..19a4ba2a8f 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -319,6 +319,7 @@ OBJS-$(CONFIG_JV_DEMUXER) += jvdec.o OBJS-$(CONFIG_KUX_DEMUXER) += flvdec.o OBJS-$(CONFIG_KVAG_DEMUXER) += kvag.o OBJS-$(CONFIG_KVAG_MUXER) += kvag.o rawenc.o +OBJS-$(CONFIG_LAF_DEMUXER) += lafdec.o OBJS-$(CONFIG_LATM_MUXER) += latmenc.o rawenc.o OBJS-$(CONFIG_LMLM4_DEMUXER) += lmlm4.o OBJS-$(CONFIG_LOAS_DEMUXER) += loasdec.o rawdec.o diff --git a/libavformat/allformats.c b/libavformat/allformats.c index cebd5e0c67..a545b5ff45 100644 --- a/libavformat/allformats.c +++ b/libavformat/allformats.c @@ -236,6 +236,7 @@ extern const AVInputFormat ff_jv_demuxer; extern const AVInputFormat ff_kux_demuxer; extern const AVInputFormat ff_kvag_demuxer; extern const AVOutputFormat ff_kvag_muxer; +extern const AVInputFormat ff_laf_demuxer; extern const AVOutputFormat ff_latm_muxer; extern const AVInputFormat ff_lmlm4_demuxer; extern const AVInputFormat ff_loas_demuxer; diff --git a/libavformat/lafdec.c b/libavformat/lafdec.c new file mode 100644 index 0000000000..12b0d8540b --- /dev/null +++ b/libavformat/lafdec.c @@ -0,0 +1,271 @@ +/* + * 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 "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; + + 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; + } + + 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; + + avio_read(pb, s->header, s->header_len); + 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 = avio_read(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_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_seek = laf_read_seek, + .extensions = "laf", + .flags = AVFMT_GENERIC_INDEX, +}; diff --git a/libavformat/version.h b/libavformat/version.h index 36f22982d8..ede3f46428 100644 --- a/libavformat/version.h +++ b/libavformat/version.h @@ -31,7 +31,7 @@ #include "version_major.h" -#define LIBAVFORMAT_VERSION_MINOR 31 +#define LIBAVFORMAT_VERSION_MINOR 32 #define LIBAVFORMAT_VERSION_MICRO 100 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \ |