* [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer
@ 2022-09-12 9:40 Paul B Mahol
2022-09-12 10:56 ` Andreas Rheinhardt
2022-09-12 19:16 ` Paul B Mahol
0 siblings, 2 replies; 6+ messages in thread
From: Paul B Mahol @ 2022-09-12 9:40 UTC (permalink / raw)
To: FFmpeg development discussions and patches
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Patch attached.
[-- Attachment #2: 0001-avformat-add-LAF-demuxer.patch --]
[-- Type: text/x-patch, Size: 9389 bytes --]
From d867b825507b5f38a051dd0ccf4612b7570a2088 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Sun, 11 Sep 2022 20:10:27 +0200
Subject: [PATCH] avformat: add LAF demuxer
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
libavformat/Makefile | 1 +
libavformat/allformats.c | 1 +
libavformat/lafdec.c | 253 +++++++++++++++++++++++++++++++++++++++
3 files changed, 255 insertions(+)
create mode 100644 libavformat/lafdec.c
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..35bce2b327
--- /dev/null
+++ b/libavformat/lafdec.c
@@ -0,0 +1,253 @@
+/*
+ * 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"
+
+typedef struct StreamParams {
+ float horizontal;
+ float vertical;
+ int lfe;
+ AVChannelLayout layout;
+} StreamParams;
+
+typedef struct LAFContext {
+ uint8_t *data;
+ unsigned nb_stored;
+ unsigned stored_index;
+ unsigned index;
+ unsigned bpp;
+
+ StreamParams p[1024];
+} LAFContext;
+
+typedef struct LAFStream {
+ unsigned stored;
+} LAFStream;
+
+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 > 1024)
+ 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;
+ s->data = av_mallocz(st_count * sample_rate * bpp);
+ if (!s->data)
+ return AVERROR(ENOMEM);
+
+ for (int st = 0; st < st_count; st++) {
+ StreamParams *stp = &s->p[st];
+ LAFStream *lafst;
+ 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;
+ st->priv_data = lafst = av_mallocz(sizeof(LAFStream));
+ if (!st->priv_data)
+ return AVERROR(ENOMEM);
+
+ avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
+ }
+
+ 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];
+ LAFStream *lafst = st->priv_data;
+ const int bpp = s->bpp;
+ int header_len = (ctx->nb_streams / 8) + !!(ctx->nb_streams & 7);
+ int64_t pos;
+ int ret;
+
+again:
+ if (avio_feof(pb))
+ return AVERROR_EOF;
+
+ pos = avio_tell(pb);
+
+ if (s->index >= ctx->nb_streams) {
+ int cur_st = 0, st_count = 0, st_index = 0;
+
+ for (int i = 0; i < header_len; i++) {
+ uint8_t val = avio_r8(pb);
+
+ for (int j = 0; j < 8 && cur_st < ctx->nb_streams; j++, cur_st++) {
+ AVStream *st = ctx->streams[st_index];
+ LAFStream *lafst = st->priv_data;
+
+ lafst->stored = 0;
+ if (val & 1) {
+ lafst->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];
+ lafst = st->priv_data;
+ while (!lafst->stored) {
+ s->index++;
+ if (s->index >= ctx->nb_streams)
+ goto again;
+ lafst = ctx->streams[s->index]->priv_data;
+ }
+ st = ctx->streams[s->index];
+
+ ret = av_new_packet(pkt, st->codecpar->sample_rate * bpp);
+ if (ret < 0)
+ return ret;
+
+ for (int n = 0; n < st->codecpar->sample_rate; n++)
+ memcpy(pkt->data + n * bpp, s->data + n * s->nb_stored * bpp + s->stored_index * bpp, bpp);
+
+ pkt->stream_index = s->index;
+ pkt->pos = pos;
+ s->index++;
+ s->stored_index++;
+
+ return ret;
+}
+
+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 = 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,
+};
--
2.37.2
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^ permalink raw reply [flat|nested] 6+ messages in thread* Re: [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer
2022-09-12 9:40 [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer Paul B Mahol
@ 2022-09-12 10:56 ` Andreas Rheinhardt
2022-09-12 19:16 ` Paul B Mahol
1 sibling, 0 replies; 6+ messages in thread
From: Andreas Rheinhardt @ 2022-09-12 10:56 UTC (permalink / raw)
To: ffmpeg-devel
Paul B Mahol:
> +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 > 1024)
I don't know whether the limit of 1024 is arbitrary or something from
some spec. If it is the latter, you should use a #define for it and also
for the size of the StreamParams array in the ctx. If it is the former,
you might just use FF_ARRAY_ELEMS(s->p) instead of 1024 here. Or a
define, as you prefer.
> + 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;
> + s->data = av_mallocz(st_count * sample_rate * bpp);
sample_rate is read via avio_rl32() and therefore the multiplication on
the right can overflow (it's performed in 32bits, so this can happen
even on 64bit systems). Maybe use av_calloc(sample_rate, st_count *
bpp). But you also need to ensure that sample_rate actually fits into an
int and that st_count * sample_rate * bpp performed in the avio_read()
below also fits into an int, so you should probably just ensure this here.
> + if (!s->data)
> + return AVERROR(ENOMEM);
> +
> + for (int st = 0; st < st_count; st++) {
> + StreamParams *stp = &s->p[st];
> + LAFStream *lafst;
> + 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;
> + st->priv_data = lafst = av_mallocz(sizeof(LAFStream));
lafst is set-but-unused. And given that you are already imposing a
hardcoded limit on the number of streams you could just add an array of
1024 uint8_t to your context.
> + if (!st->priv_data)
> + return AVERROR(ENOMEM);
> +
> + avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
> + }
> +
> + 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];
> + LAFStream *lafst = st->priv_data;
> + const int bpp = s->bpp;
> + int header_len = (ctx->nb_streams / 8) + !!(ctx->nb_streams & 7);
(ctx->nb_streams + 7) / 8.
> + int64_t pos;
> + int ret;
> +
> +again:
> + if (avio_feof(pb))
> + return AVERROR_EOF;
> +
> + pos = avio_tell(pb);
> +
> + if (s->index >= ctx->nb_streams) {
> + int cur_st = 0, st_count = 0, st_index = 0;
> +
> + for (int i = 0; i < header_len; i++) {
> + uint8_t val = avio_r8(pb);
Given that you impose a limit of 1024 for the number of streams, you can
actually put an uint8_t [128] on the stack in this loop and read all the
values at once. This would allow to remove the outer loop.
(If you used an array of uint8_t instead of the st->priv_data for
stored, you could also use that array.)
> +
> + for (int j = 0; j < 8 && cur_st < ctx->nb_streams; j++, cur_st++) {
> + AVStream *st = ctx->streams[st_index];
> + LAFStream *lafst = st->priv_data;
> +
> + lafst->stored = 0;
> + if (val & 1) {
> + lafst->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];
> + lafst = st->priv_data;
> + while (!lafst->stored) {
> + s->index++;
> + if (s->index >= ctx->nb_streams)
> + goto again;
> + lafst = ctx->streams[s->index]->priv_data;
> + }
> + st = ctx->streams[s->index];
> +
> + ret = av_new_packet(pkt, st->codecpar->sample_rate * bpp);
> + if (ret < 0)
> + return ret;
> +
> + for (int n = 0; n < st->codecpar->sample_rate; n++)
> + memcpy(pkt->data + n * bpp, s->data + n * s->nb_stored * bpp + s->stored_index * bpp, bpp);
This looks like something that can easily trigger a timeout.
> +
> + pkt->stream_index = s->index;
> + pkt->pos = pos;
If you have data from multiple streams interleaved, then the first
stream will get the position from before reading header_len bytes, but
all the other streams will get the position from after reading the
common data. IMO all packets should get the position of the common data.
> + s->index++;
> + s->stored_index++;
> +
> + return ret;
return 0 -- it is not really defined what happens in case read_packet
callbacks return positive values (it is currently ignored and some
demuxers return the size of the packet, but that is a remnant of an
earlier API) which could happen if av_new_packet() were changed to allow
to return positive values.
> +}
> +
> +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 = 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,
> +};
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^ permalink raw reply [flat|nested] 6+ messages in thread* Re: [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer
2022-09-12 9:40 [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer Paul B Mahol
2022-09-12 10:56 ` Andreas Rheinhardt
@ 2022-09-12 19:16 ` Paul B Mahol
2022-09-12 22:01 ` Andreas Rheinhardt
2022-09-15 17:14 ` Paul B Mahol
1 sibling, 2 replies; 6+ messages in thread
From: Paul B Mahol @ 2022-09-12 19:16 UTC (permalink / raw)
To: FFmpeg development discussions and patches
[-- Attachment #1: Type: text/plain, Size: 96 bytes --]
On 9/12/22, Paul B Mahol <onemda@gmail.com> wrote:
> Patch attached.
>
Updated patch attached.
[-- Attachment #2: 0001-avformat-add-LAF-demuxer.patch --]
[-- Type: text/x-patch, Size: 9965 bytes --]
From 33efa252db96d9eac7f162f17b22c1cd8b3b1c14 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Sun, 11 Sep 2022 20:10:27 +0200
Subject: [PATCH] avformat: add LAF demuxer
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
libavformat/Makefile | 1 +
libavformat/allformats.c | 1 +
libavformat/lafdec.c | 271 +++++++++++++++++++++++++++++++++++++++
3 files changed, 273 insertions(+)
create mode 100644 libavformat/lafdec.c
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,
+};
--
2.37.2
[-- Attachment #3: Type: text/plain, Size: 251 bytes --]
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^ permalink raw reply [flat|nested] 6+ messages in thread* Re: [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer
2022-09-12 19:16 ` Paul B Mahol
@ 2022-09-12 22:01 ` Andreas Rheinhardt
2022-09-12 22:12 ` Paul B Mahol
2022-09-15 17:14 ` Paul B Mahol
1 sibling, 1 reply; 6+ messages in thread
From: Andreas Rheinhardt @ 2022-09-12 22:01 UTC (permalink / raw)
To: ffmpeg-devel
Paul B Mahol:
> + 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;
Looking at intreadwrite.h shows that we actually have AV_RN24 and AV_WN24.
- Andreas
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^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer
2022-09-12 22:01 ` Andreas Rheinhardt
@ 2022-09-12 22:12 ` Paul B Mahol
0 siblings, 0 replies; 6+ messages in thread
From: Paul B Mahol @ 2022-09-12 22:12 UTC (permalink / raw)
To: FFmpeg development discussions and patches
On 9/13/22, Andreas Rheinhardt <andreas.rheinhardt@outlook.com> wrote:
> Paul B Mahol:
>> + 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;
>
> Looking at intreadwrite.h shows that we actually have AV_RN24 and AV_WN24.
Didnt compile.
>
> - Andreas
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
> To unsubscribe, visit link above, or email
> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
>
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^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer
2022-09-12 19:16 ` Paul B Mahol
2022-09-12 22:01 ` Andreas Rheinhardt
@ 2022-09-15 17:14 ` Paul B Mahol
1 sibling, 0 replies; 6+ messages in thread
From: Paul B Mahol @ 2022-09-15 17:14 UTC (permalink / raw)
To: FFmpeg development discussions and patches
On 9/12/22, Paul B Mahol <onemda@gmail.com> wrote:
> On 9/12/22, Paul B Mahol <onemda@gmail.com> wrote:
>> Patch attached.
>>
>
> Updated patch attached.
>
Will apply soon.
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^ permalink raw reply [flat|nested] 6+ messages in thread
end of thread, other threads:[~2022-09-15 17:14 UTC | newest]
Thread overview: 6+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2022-09-12 9:40 [FFmpeg-devel] [PATCH] avformat: add Limitless Audio Format demuxer Paul B Mahol
2022-09-12 10:56 ` Andreas Rheinhardt
2022-09-12 19:16 ` Paul B Mahol
2022-09-12 22:01 ` Andreas Rheinhardt
2022-09-12 22:12 ` Paul B Mahol
2022-09-15 17:14 ` Paul B Mahol
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