[FFmpeg-cvslog] avformat: add LAF demuxer
Paul B Mahol
git at videolan.org
Fri Sep 16 11:01:48 EEST 2022
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun Sep 11 20:10:27 2022 +0200| [dd2a01ef5cad08347ecbbcba7afd5e5a0810f504] | committer: Paul B Mahol
avformat: add LAF demuxer
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=dd2a01ef5cad08347ecbbcba7afd5e5a0810f504
---
Changelog | 1 +
doc/general_contents.texi | 2 +
libavformat/Makefile | 1 +
libavformat/allformats.c | 1 +
libavformat/lafdec.c | 271 ++++++++++++++++++++++++++++++++++++++++++++++
libavformat/version.h | 2 +-
6 files changed, 277 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 9e1f705539..720a092659 100644
--- a/Changelog
+++ b/Changelog
@@ -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
+ at 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, \
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