Thank you both for the suggestions. I've updated the code as requested, and I
apologize for the AV_LOG_WARNING instead of AV_LOG_ERROR - it was an oversight
on my part.
I have also added the stream codec check, and it did get triggered when I tried
to feed it audio that was not ATRAC1, so it seems it is required.
Regarding titles being truncated - that was my intention. However I've now
added a warning if it was going to happen.
As for the block count in the header - none of the modern software which uses
AEA reads that field, but for the older software, it will now be truncated to
UINT32_MAX if needed.
Is there anything else that needs changes?
- asivery
On Saturday, March 9th, 2024 at 1:06 PM, Andreas Rheinhardt
<andreas.rheinha...@outlook.com> wrote:
> asivery via ffmpeg-devel:
>
> > +#include "libavutil/intreadwrite.h"
> > +#include "libavutil/avstring.h"
>
>
> These two headers seem unused.
>
> > +#include "avformat.h"
> > +#include "avio_internal.h"
> > +#include "rawenc.h"
> > +#include "mux.h"
> > +
> > +static int aea_write_header(AVFormatContext *s)
> > +{
> > + const AVDictionaryEntry title_entry;
> > + int title_length = 0;
> > + char title_contents;
>
>
> const please. Also we put the * to the variable (because "char* foo,
> bar" still only declares one pointer to char). Furthermore, the scope
> for this should be the block below.
>
> > + AVStream st;
> > +
> > + if (s->nb_streams > 1) {
> > + av_log(s, AV_LOG_WARNING, "Got more than one stream to encode. This is
> > not supported.\n");
> > + return AVERROR(EINVAL);
> > + }
> > +
> > + st = s->streams[0];
> > + if (st->codecpar->ch_layout.nb_channels != 1 &&
> > st->codecpar->ch_layout.nb_channels != 2) {
> > + av_log(s, AV_LOG_ERROR, "Invalid amount of channels to mux (%d).\n",
> > st->codecpar->ch_layout.nb_channels);
> > + return AVERROR(EINVAL);
> > + }
> > +
> > + if (st->codecpar->sample_rate != 44100) {
> > + av_log(s, AV_LOG_ERROR, "Invalid sample rate (%d) AEA only supports
> > 44.1kHz.\n", st->codecpar->sample_rate);
> > + return AVERROR(EINVAL);
> > + }
> > +
> > + / Write magic /
> > + avio_wl32(s->pb, 2048);
> > +
> > + / Write AEA title */
> > + title_entry = av_dict_get(st->metadata, "title", NULL, 0);
> > + if (title_entry) {
> > + title_contents = title_entry->value;
> > + title_length = strlen(title_contents);
>
>
> Possible truncation here.
>
> > + title_length = FFMIN(256, title_length);
> > + avio_write(s->pb, title_contents, title_length);
> > + }
> > +
> > + ffio_fill(s->pb, 0, 256 - title_length);
> > +
> > + /* Write number of frames (zero at header-writing time, will seek later),
> > number of channels /
> > + avio_wl32(s->pb, 0);
> > + avio_w8(s->pb, st->codecpar->ch_layout.nb_channels);
> > + avio_w8(s->pb, 0);
> > +
> > + / Pad the header to 2048 bytes */
> > + ffio_fill(s->pb, 0, 1782);
> > +
> > + return 0;
> > +}
> > +
> > +static int aea_write_trailer(struct AVFormatContext *s)
> > +{
> > + AVIOContext *pb = s->pb;
> > + AVStream st = s->streams[0];
> > + if (pb->seekable & AVIO_SEEKABLE_NORMAL) {
> > + / Seek to rewrite the block count. */
> > + avio_seek(pb, 260, SEEK_SET);
> > + avio_wl32(pb, st->nb_frames * st->codecpar->ch_layout.nb_channels);
>
>
> I don't see anything guaranteeing that the result fits into 32 bits.
>
> > + } else {
> > + av_log(s, AV_LOG_WARNING, "unable to rewrite AEA header.\n");
> > + }
> > +
> > + return 0;
> > +}
> > +
> > +const FFOutputFormat ff_aea_muxer = {
> > + .p.name = "aea",
> > + .p.long_name = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
> > + .p.extensions = "aea",
> > + .p.audio_codec = AV_CODEC_ID_ATRAC1,
> > +
> > + .write_header = aea_write_header,
> > + .write_packet = ff_raw_write_packet,
> > + .write_trailer = aea_write_trailer,
> > +};
>
>
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> ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".From ee1d4c93c66e729d9d0456b2e8e789f3f98389e3 Mon Sep 17 00:00:00 2001
From: asivery <asiv...@protonmail.com>
Date: Fri, 8 Mar 2024 14:45:02 +0100
Subject: [PATCH] avformat/aea: Add aea muxer
Signed-off-by: asivery <asiv...@protonmail.com>
---
doc/muxers.texi | 10 +++
libavformat/Makefile | 3 +-
libavformat/{aea.c => aeadec.c} | 0
libavformat/aeaenc.c | 115 ++++++++++++++++++++++++++++++++
libavformat/allformats.c | 1 +
5 files changed, 128 insertions(+), 1 deletion(-)
rename libavformat/{aea.c => aeadec.c} (100%)
create mode 100644 libavformat/aeaenc.c
diff --git a/doc/muxers.texi b/doc/muxers.texi
index 2104cc4a95..a4df8f736d 100644
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -663,6 +663,16 @@ when enabled, write a CRC checksum for each packet to the output,
default is @code{false}
@end table
+@anchor{aea}
+@section aea
+MD STUDIO audio muxer.
+
+This muxer accepts a single ATRAC1 audio stream with either one or two channels
+and a sample rate of 44100Hz.
+
+As AEA supports storing the track title, this muxer will also write
+the title from stream's metadata to the container.
+
@anchor{adts}
@section adts
Audio Data Transport Stream muxer.
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 8811a0ffc9..70d56f391f 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -91,7 +91,8 @@ OBJS-$(CONFIG_ADTS_MUXER) += adtsenc.o apetag.o img2.o \
id3v2enc.o
OBJS-$(CONFIG_ADX_DEMUXER) += adxdec.o
OBJS-$(CONFIG_ADX_MUXER) += rawenc.o
-OBJS-$(CONFIG_AEA_DEMUXER) += aea.o pcm.o
+OBJS-$(CONFIG_AEA_DEMUXER) += aeadec.o pcm.o
+OBJS-$(CONFIG_AEA_MUXER) += aeaenc.o rawenc.o
OBJS-$(CONFIG_AFC_DEMUXER) += afc.o
OBJS-$(CONFIG_AIFF_DEMUXER) += aiffdec.o aiff.o pcm.o \
mov_chan.o replaygain.o
diff --git a/libavformat/aea.c b/libavformat/aeadec.c
similarity index 100%
rename from libavformat/aea.c
rename to libavformat/aeadec.c
diff --git a/libavformat/aeaenc.c b/libavformat/aeaenc.c
new file mode 100644
index 0000000000..12aed72a13
--- /dev/null
+++ b/libavformat/aeaenc.c
@@ -0,0 +1,115 @@
+/*
+ * MD STUDIO audio muxer
+ *
+ * Copyright (c) 2024 asivery
+ *
+ * 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 "avformat.h"
+#include "avio_internal.h"
+#include "rawenc.h"
+#include "mux.h"
+
+static int aea_write_header(AVFormatContext *s)
+{
+ const AVDictionaryEntry *title_entry;
+ int title_length = 0;
+ AVStream *st;
+
+ if (s->nb_streams > 1) {
+ av_log(s, AV_LOG_ERROR, "Got more than one stream to encode. This is not supported.\n");
+ return AVERROR(EINVAL);
+ }
+
+ st = s->streams[0];
+ if (st->codecpar->ch_layout.nb_channels != 1 && st->codecpar->ch_layout.nb_channels != 2) {
+ av_log(s, AV_LOG_ERROR, "Only maximum 2 channels are supported in the audio"
+ " stream, %d channels were found.\n", st->codecpar->ch_layout.nb_channels);
+ return AVERROR(EINVAL);
+ }
+
+ if (st->codecpar->codec_id != AV_CODEC_ID_ATRAC1) {
+ av_log(s, AV_LOG_ERROR, "AEA can only store ATRAC1 streams, %s was found.\n", avcodec_get_name(st->codecpar->codec_id));
+ return AVERROR(EINVAL);
+ }
+
+ if (st->codecpar->sample_rate != 44100) {
+ av_log(s, AV_LOG_ERROR, "Invalid sample rate (%d) AEA only supports 44.1kHz.\n", st->codecpar->sample_rate);
+ return AVERROR(EINVAL);
+ }
+
+ /* Write magic */
+ avio_wl32(s->pb, 2048);
+
+ /* Write AEA title */
+ title_entry = av_dict_get(st->metadata, "title", NULL, 0);
+ if (title_entry) {
+ const char *title_contents = title_entry->value;
+ title_length = strlen(title_contents);
+ if (title_length > 256) {
+ av_log(s, AV_LOG_WARNING, "Title too long, truncated to 256 bytes.\n");
+ title_length = 256;
+ }
+ avio_write(s->pb, title_contents, title_length);
+ }
+
+ ffio_fill(s->pb, 0, 256 - title_length);
+
+ /* Write number of frames (zero at header-writing time, will seek later), number of channels */
+ avio_wl32(s->pb, 0);
+ avio_w8(s->pb, st->codecpar->ch_layout.nb_channels);
+ avio_w8(s->pb, 0);
+
+ /* Pad the header to 2048 bytes */
+ ffio_fill(s->pb, 0, 1782);
+
+ return 0;
+}
+
+static int aea_write_trailer(struct AVFormatContext *s)
+{
+ int64_t total_blocks;
+ AVIOContext *pb = s->pb;
+ AVStream *st = s->streams[0];
+ if (pb->seekable & AVIO_SEEKABLE_NORMAL) {
+ /* Seek to rewrite the block count. */
+ avio_seek(pb, 260, SEEK_SET);
+ total_blocks = st->nb_frames * st->codecpar->ch_layout.nb_channels;
+ if (total_blocks > UINT32_MAX) {
+ av_log(s, AV_LOG_WARNING, "Too many frames in the file to properly encode the header (%ld)."
+ " Block count in the header will be truncated.\n", total_blocks);
+ total_blocks = UINT32_MAX;
+ }
+ avio_wl32(pb, total_blocks);
+ } else {
+ av_log(s, AV_LOG_WARNING, "Unable to rewrite AEA header.\n");
+ }
+
+ return 0;
+}
+
+const FFOutputFormat ff_aea_muxer = {
+ .p.name = "aea",
+ .p.long_name = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
+ .p.extensions = "aea",
+ .p.audio_codec = AV_CODEC_ID_ATRAC1,
+
+ .write_header = aea_write_header,
+ .write_packet = ff_raw_write_packet,
+ .write_trailer = aea_write_trailer,
+};
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 0a0e76138f..5639715104 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -47,6 +47,7 @@ extern const FFOutputFormat ff_adts_muxer;
extern const FFInputFormat ff_adx_demuxer;
extern const FFOutputFormat ff_adx_muxer;
extern const FFInputFormat ff_aea_demuxer;
+extern const FFOutputFormat ff_aea_muxer;
extern const FFInputFormat ff_afc_demuxer;
extern const FFInputFormat ff_aiff_demuxer;
extern const FFOutputFormat ff_aiff_muxer;
--
2.34.1
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