Re: [FFmpeg-devel] [PATCH] avformat/aea: Add aea muxer

[asivery via ffmpeg-devel <ffmpeg-devel@ffmpeg.org>]

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I see, I've changed title_length to size_t, but left the warning as is to inform the user in case truncation needs to take place.

On Saturday, March 9th, 2024 at 6:48 PM, Andreas Rheinhardt <andreas.rheinhardt@outlook.com> wrote:

> asivery via ffmpeg-devel:
> 
> > + 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;
> > + }
> 
> 
> I actually meant that you should use size_t for title_length (so that
> the output of strlen() is never truncated) instead of adding a warning
> that will likely never be triggered in practice.
> 
> - Andreas
> 
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From 512e25ee49e30a0ac2d77cf865185cbfe92a850e Mon Sep 17 00:00:00 2001
From: asivery <asivery@protonmail.com>
Date: Fri, 8 Mar 2024 14:45:02 +0100
Subject: [PATCH] avformat/aea: Add aea muxer

Signed-off-by: asivery <asivery@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..f7b945b382
--- /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;
+    size_t 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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