[FFmpeg-devel] [PATCH] AST muxer
jamal
jamrial at gmail.com
Sat Nov 24 00:27:13 CET 2012
On 22/11/12 8:18 PM, Paul B Mahol wrote:
>
> Using double for this is not portable. Please use AVRational and av_rescale*.
Ok.
>> Ok, I'll resend this patch with this included, unless you prefer a separate
>> patch to rename the demuxer.
>
> Separate. (Note that I do not know which solution table or current
> code bloat less; both in source code and in binary).
Ok, made a separate patch then.
>>>> +
>>>> + if(enc->channels != 2 && enc->channels != 4) {
>>>> + av_log(s, AV_LOG_ERROR, "Unsupported channel amount: %d\n",
>>>> enc->channels);
>>>> + return AVERROR_INVALIDDATA;
>>>> + }
>>>
>>> Being able to test on Wii that different channel count are supported
>>> would be welcome.
>>
>> So far every game using the format uses stereo streams and occasionally 4.0
>> streams. The AST format can support more, but since no game seems to use
>> anything beyond stereo and quad i thought about forcing them.
>
> You get that from wiki entry? Wikis are not meant to be definite truth.
>> I can remove the channel limitation if that's better, since the muxer and
>> the demuxers accept and work fine with streams of any amount of channels as
>> far as i tested.
>>
>>>> +
>>>> + ffio_wfourcc(pb, "STRM");
>>>> +
>>>> + ast->size = avio_tell(pb);
>>>> + avio_wb32(pb, 0); /* File size minus header */
>>>> + avio_wb16(pb, codec_id);
>>>> + avio_wb16(pb, enc->bits_per_coded_sample);
>>>
>>> Demuxer does not export this. And it is actually bits per sample for
>>> output which is always 16 (and not input, where it is 4 for APC and 16 for
>>> PCM).
>>
>> avio_wb16(pb, 0x10) then?
>
> 16
Ok.
>>
>>>> +static int ast_write_trailer(AVFormatContext *s)
>>>> +{
>>>> + AVIOContext *pb = s->pb;
>>>> + ASTContext *ast = s->priv_data;
>>>> + AVCodecContext *enc = s->streams[0]->codec;
>>>> + int64_t file_size = avio_tell(pb);
>>>> + int64_t samples = (file_size - 64 - (32 * enc->frame_number)) /
>>>> enc->block_align;
>>>
>>> Does this work for APC too? You could use -c copy and then duration of
>>> output should match one of input.
>>
>> It should, as long as enc->block_align is not zero. To be sure, can you
>> upload the ADPCM AFC sample somewhere so i can check this?
>> In any case, I can add a check for enc->block_align and if it's zero make it
>> (enc->bits_per_coded_sample * enc->channels) >> 3
>
> You can look in source code how samples are stored in ADPCM AFC.
>
> It is 9 bytes for each 16 samples per channel. Doing 9 : 32 compression ratio.
Fixed.
>>
>> And what do you mean with using -c copy?
>
> That is used with ffmpeg to copy packets from input to output - just
> remuxing without any encoding involved.
>>
>>>> + avio_seek(pb, file_size, SEEK_SET);
>>>
>>> SEEK_END
>>
>> Ok.
>>
>>>
>>> That's all (for now).
>>
>> Thanks. I'll send a new patch after you answer the concerns above.
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
See attached patches.
Regards.
-------------- next part --------------
>From 1a59c86e957240a2a41eeb78811202d0cc8c397c Mon Sep 17 00:00:00 2001
From: jamal <jamrial at gmail.com>
Date: Fri, 23 Nov 2012 18:24:42 -0300
Subject: [PATCH 1/2] ast: rename ast.c -> astdec.c
---
libavformat/Makefile | 2 +-
libavformat/{ast.c => astdec.c} | 0
2 files changed, 1 insertion(+), 1 deletion(-)
rename libavformat/{ast.c => astdec.c} (100%)
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 136ada8..fc00811 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -50,7 +50,7 @@ OBJS-$(CONFIG_ASF_DEMUXER) += asfdec.o asf.o asfcrypt.o \
OBJS-$(CONFIG_ASF_MUXER) += asfenc.o asf.o
OBJS-$(CONFIG_ASS_DEMUXER) += assdec.o
OBJS-$(CONFIG_ASS_MUXER) += assenc.o
-OBJS-$(CONFIG_AST_DEMUXER) += ast.o
+OBJS-$(CONFIG_AST_DEMUXER) += astdec.o
OBJS-$(CONFIG_AU_DEMUXER) += au.o pcm.o
OBJS-$(CONFIG_AU_MUXER) += au.o
OBJS-$(CONFIG_AVI_DEMUXER) += avidec.o
diff --git a/libavformat/ast.c b/libavformat/astdec.c
similarity index 100%
rename from libavformat/ast.c
rename to libavformat/astdec.c
--
1.8.0.msysgit.0
-------------- next part --------------
>From 7efc2a0df3d5ad575f096b2e2da4e0f19388a9ac Mon Sep 17 00:00:00 2001
From: jamal <jamrial at gmail.com>
Date: Fri, 23 Nov 2012 20:02:21 -0300
Subject: [PATCH 2/2] AST Muxer
---
Changelog | 2 +-
doc/general.texi | 2 +-
libavformat/Makefile | 3 +-
libavformat/allformats.c | 2 +-
libavformat/ast.c | 30 +++++++
libavformat/ast.h | 30 +++++++
libavformat/astdec.c | 15 ++--
libavformat/astenc.c | 202 +++++++++++++++++++++++++++++++++++++++++++++++
libavformat/version.h | 2 +-
tests/fate/avformat.mak | 1 +
tests/lavf-regression.sh | 4 +
tests/ref/lavf/ast | 3 +
12 files changed, 281 insertions(+), 15 deletions(-)
create mode 100644 libavformat/ast.c
create mode 100644 libavformat/ast.h
create mode 100644 libavformat/astenc.c
create mode 100644 tests/ref/lavf/ast
diff --git a/Changelog b/Changelog
index bca5568..d2e7d5e 100644
--- a/Changelog
+++ b/Changelog
@@ -24,7 +24,7 @@ version <next>:
- AVR demuxer
- geq filter ported from libmpcodecs
- remove ffserver daemon mode
-- AST demuxer
+- AST muxer/demuxer
- new expansion syntax for drawtext
- BRender PIX image decoder
- ffprobe -show_entries option
diff --git a/doc/general.texi b/doc/general.texi
index 4d145a7..2c9aaf7 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -148,7 +148,7 @@ library:
@item Apple HTTP Live Streaming @tab @tab X
@item Artworx Data Format @tab @tab X
@item ASF @tab X @tab X
- at item AST @tab @tab X
+ at item AST @tab X @tab X
@tab Used on the Nintendo Wii.
@item AVI @tab X @tab X
@item AVISynth @tab @tab X
diff --git a/libavformat/Makefile b/libavformat/Makefile
index fc00811..2f83e25 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -50,7 +50,8 @@ OBJS-$(CONFIG_ASF_DEMUXER) += asfdec.o asf.o asfcrypt.o \
OBJS-$(CONFIG_ASF_MUXER) += asfenc.o asf.o
OBJS-$(CONFIG_ASS_DEMUXER) += assdec.o
OBJS-$(CONFIG_ASS_MUXER) += assenc.o
-OBJS-$(CONFIG_AST_DEMUXER) += astdec.o
+OBJS-$(CONFIG_AST_DEMUXER) += ast.o astdec.o
+OBJS-$(CONFIG_AST_MUXER) += ast.o astenc.o
OBJS-$(CONFIG_AU_DEMUXER) += au.o pcm.o
OBJS-$(CONFIG_AU_MUXER) += au.o
OBJS-$(CONFIG_AVI_DEMUXER) += avidec.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index eaeb51a..b73eeae 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -64,7 +64,7 @@ void av_register_all(void)
REGISTER_DEMUXER (APE, ape);
REGISTER_MUXDEMUX (ASF, asf);
REGISTER_MUXDEMUX (ASS, ass);
- REGISTER_DEMUXER (AST, ast);
+ REGISTER_MUXDEMUX (AST, ast);
REGISTER_MUXER (ASF_STREAM, asf_stream);
REGISTER_MUXDEMUX (AU, au);
REGISTER_MUXDEMUX (AVI, avi);
diff --git a/libavformat/ast.c b/libavformat/ast.c
new file mode 100644
index 0000000..ddb6e2c
--- /dev/null
+++ b/libavformat/ast.c
@@ -0,0 +1,30 @@
+/*
+ * AST common code
+ * Copyright (c) 2012 James Almer
+ *
+ * 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 "internal.h"
+#include "ast.h"
+
+const AVCodecTag ff_codec_ast_tags[] = {
+ { AV_CODEC_ID_ADPCM_AFC, 0 },
+ { AV_CODEC_ID_PCM_S16BE_PLANAR, 1 },
+ { AV_CODEC_ID_NONE, 0 },
+};
diff --git a/libavformat/ast.h b/libavformat/ast.h
new file mode 100644
index 0000000..4a399ea
--- /dev/null
+++ b/libavformat/ast.h
@@ -0,0 +1,30 @@
+/*
+ * AST common code
+ * Copyright (c) 2012 James Almer
+ *
+ * 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
+ */
+
+#ifndef AVFORMAT_AST_H
+#define AVFORMAT_AST_H
+
+#include "avformat.h"
+#include "internal.h"
+
+extern const AVCodecTag ff_codec_ast_tags[];
+
+#endif /* AVFORMAT_AST_H */
diff --git a/libavformat/astdec.c b/libavformat/astdec.c
index 4f83540..4d74d4c 100644
--- a/libavformat/astdec.c
+++ b/libavformat/astdec.c
@@ -23,6 +23,8 @@
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
+#include "riff.h"
+#include "ast.h"
static int ast_probe(AVProbeData *p)
{
@@ -42,16 +44,6 @@ static int ast_read_header(AVFormatContext *s)
avio_skip(s->pb, 8);
codec = avio_rb16(s->pb);
- switch (codec) {
- case 0:
- st->codec->codec_id = AV_CODEC_ID_ADPCM_AFC;
- break;
- case 1:
- st->codec->codec_id = AV_CODEC_ID_PCM_S16BE_PLANAR;
- break;
- default:
- av_log(s, AV_LOG_ERROR, "unsupported codec %d\n", codec);
- }
depth = avio_rb16(s->pb);
if (depth != 16) {
@@ -60,6 +52,8 @@ static int ast_read_header(AVFormatContext *s)
}
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codec->codec_id = ff_codec_get_id(ff_codec_ast_tags, codec);
+ st->codec->codec_tag = codec;
st->codec->channels = avio_rb16(s->pb);
if (!st->codec->channels)
return AVERROR_INVALIDDATA;
@@ -121,4 +115,5 @@ AVInputFormat ff_ast_demuxer = {
.read_packet = ast_read_packet,
.extensions = "ast",
.flags = AVFMT_GENERIC_INDEX,
+ .codec_tag = (const AVCodecTag* const []){ff_codec_ast_tags, 0},
};
diff --git a/libavformat/astenc.c b/libavformat/astenc.c
new file mode 100644
index 0000000..0fa45c2
--- /dev/null
+++ b/libavformat/astenc.c
@@ -0,0 +1,202 @@
+/*
+ * AST muxer
+ * Copyright (c) 2012 James Almer
+ *
+ * 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 "internal.h"
+#include "ast.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
+
+typedef struct ASTMuxContext {
+ AVClass *class;
+ int64_t size;
+ int64_t samples;
+ int64_t loopstart;
+ int64_t loopend;
+ int fbs;
+} ASTMuxContext;
+
+#define CHECK_LOOP(type) \
+ if(ast->loop ## type) { \
+ ast->loop ## type = av_rescale_q_rnd(ast->loop ## type, (AVRational){enc->sample_rate, 1}, (AVRational){1000, 1}, AV_ROUND_DOWN); \
+ if(ast->loop ## type < 0 || ast->loop ## type > UINT_MAX) { \
+ av_log(s, AV_LOG_ERROR, "Invalid loop" #type " value\n"); \
+ return AVERROR(EINVAL); \
+ } \
+ }
+
+static int ast_write_header(AVFormatContext *s)
+{
+ ASTMuxContext *ast = s->priv_data;
+ AVIOContext *pb = s->pb;
+ AVCodecContext *enc = NULL;
+
+ if(s->nb_streams = 1) {
+ enc = s->streams[0]->codec;
+ } else {
+ av_log(s, AV_LOG_ERROR, "Muxer supports only one stream\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if(!enc->codec_tag && enc->codec_id != AV_CODEC_ID_ADPCM_AFC) {
+ av_log(s, AV_LOG_ERROR, "Unsupported codec\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if(ast->loopstart && ast->loopend && ast->loopstart >= ast->loopend) {
+ av_log(s, AV_LOG_ERROR, "Loopend can't be less or equal to loopstart\n");
+ return AVERROR(EINVAL);
+ }
+
+ /* Convert milliseconds to samples */
+ CHECK_LOOP(start)
+ CHECK_LOOP(end)
+
+ ffio_wfourcc(pb, "STRM");
+
+ ast->size = avio_tell(pb);
+ avio_wb32(pb, 0); /* File size minus header */
+ avio_wb16(pb, enc->codec_tag);
+ avio_wb16(pb, 16); /* Bit depth */
+ avio_wb16(pb, enc->channels);
+ avio_wb16(pb, 0xFFFF);
+ avio_wb32(pb, enc->sample_rate);
+
+ ast->samples = avio_tell(pb);
+ avio_wb32(pb, 0); /* Number of samples */
+ avio_wb32(pb, 0); /* Loopstart */
+ avio_wb32(pb, 0); /* Loopend */
+ avio_wb32(pb, 0); /* Size of first block */
+
+ /* Unknown */
+ avio_wb32(pb, 0);
+ avio_wl32(pb, 0x7F);
+ avio_wb64(pb, 0);
+ avio_wb64(pb, 0);
+ avio_wb32(pb, 0);
+
+ avio_flush(pb);
+
+ return 0;
+}
+
+static int ast_write_packet(AVFormatContext *s, AVPacket *pkt)
+{
+ AVIOContext *pb = s->pb;
+ ASTMuxContext *ast = s->priv_data;
+ AVCodecContext *enc = s->streams[0]->codec;
+ int size = pkt->size / enc->channels;
+
+ if(enc->frame_number == 1)
+ ast->fbs = size;
+
+ ffio_wfourcc(pb, "BLCK");
+ avio_wb32(pb, size); /* Block size */
+
+ /* padding */
+ avio_wb64(pb, 0);
+ avio_wb64(pb, 0);
+ avio_wb64(pb, 0);
+
+ avio_write(pb, pkt->data, pkt->size);
+
+ return 0;
+}
+
+static int ast_write_trailer(AVFormatContext *s)
+{
+ AVIOContext *pb = s->pb;
+ ASTMuxContext *ast = s->priv_data;
+ AVCodecContext *enc = s->streams[0]->codec;
+ int64_t samples, file_size = avio_tell(pb);
+
+ if(enc->codec_tag)
+ samples = (file_size - 64 - (32 * enc->frame_number)) / enc->block_align;
+ else
+ samples = enc->frame_number * ((ast->fbs * 2) / (9 * enc->channels) * 16);
+
+ av_log(s, AV_LOG_DEBUG, "total samples: %"PRId64"\n", samples);
+
+ if (s->pb->seekable) {
+ /* File size minus header */
+ avio_seek(pb, ast->size, SEEK_SET);
+ avio_wb32(pb, file_size - 64);
+
+ /* Number of samples */
+ avio_seek(pb, ast->samples, SEEK_SET);
+ avio_wb32(pb, samples);
+
+ /* Loopstart if provided */
+ if(ast->loopstart && ast->loopstart >= samples) {
+ av_log(s, AV_LOG_WARNING, "Loopstart value is out of range and will be ignored\n");
+ ast->loopstart = 0;
+ }
+ avio_wb32(pb, ast->loopstart);
+
+ /* Loopend if provided. Otherwise number of samples again */
+ if(ast->loopend) {
+ if(ast->loopend > samples) {
+ av_log(s, AV_LOG_WARNING, "Loopend value is out of range and will be ignored\n");
+ ast->loopend = samples;
+ }
+ avio_wb32(pb, ast->loopend);
+ } else {
+ avio_wb32(pb, samples);
+ }
+
+ /* Size of first block */
+ avio_seek(pb, ast->samples + 12, SEEK_SET);
+ avio_wb32(pb, ast->fbs);
+
+ avio_seek(pb, file_size, SEEK_END);
+ avio_flush(pb);
+ }
+ return 0;
+}
+
+#define OFFSET(obj) offsetof(ASTMuxContext, obj)
+static const AVOption options[] = {
+ { "loopstart", "Loopstart position in milliseconds.", OFFSET(loopstart), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
+ { "loopend", "Loopend position in milliseconds.", OFFSET(loopend), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
+ { NULL },
+};
+
+static const AVClass ast_muxer_class = {
+ .class_name = "AST muxer",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVOutputFormat ff_ast_muxer = {
+ .name = "ast",
+ .long_name = NULL_IF_CONFIG_SMALL("AST (Audio Stream)"),
+ .extensions = "ast",
+ .priv_data_size = sizeof(ASTMuxContext),
+ .audio_codec = AV_CODEC_ID_PCM_S16BE_PLANAR,
+ .video_codec = AV_CODEC_ID_NONE,
+ .write_header = ast_write_header,
+ .write_packet = ast_write_packet,
+ .write_trailer = ast_write_trailer,
+ .priv_class = &ast_muxer_class,
+ .codec_tag = (const AVCodecTag* const []){ff_codec_ast_tags, 0},
+};
diff --git a/libavformat/version.h b/libavformat/version.h
index 02ebb8c..10391b3 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -30,7 +30,7 @@
#include "libavutil/avutil.h"
#define LIBAVFORMAT_VERSION_MAJOR 54
-#define LIBAVFORMAT_VERSION_MINOR 37
+#define LIBAVFORMAT_VERSION_MINOR 38
#define LIBAVFORMAT_VERSION_MICRO 100
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
diff --git a/tests/fate/avformat.mak b/tests/fate/avformat.mak
index 77c6a2f..4c99707 100644
--- a/tests/fate/avformat.mak
+++ b/tests/fate/avformat.mak
@@ -1,6 +1,7 @@
FATE_LAVF-$(call ENCDEC, PCM_S16BE, AIFF) += aiff
FATE_LAVF-$(call ENCDEC, PCM_ALAW, PCM_ALAW) += alaw
FATE_LAVF-$(call ENCDEC2, MSMPEG4V3, MP2, ASF) += asf
+FATE_LAVF-$(call ENCDEC, PCM_S16BE_PLANAR, AST) += ast
FATE_LAVF-$(call ENCDEC, PCM_S16BE, AU) += au
FATE_LAVF-$(call ENCDEC2, MPEG4, MP2, AVI) += avi
FATE_LAVF-$(call ENCDEC, BMP, IMAGE2) += bmp
diff --git a/tests/lavf-regression.sh b/tests/lavf-regression.sh
index 64ebc0a..473c235 100755
--- a/tests/lavf-regression.sh
+++ b/tests/lavf-regression.sh
@@ -326,6 +326,10 @@ if [ -n "$do_caf" ] ; then
do_audio_only caf
fi
+if [ -n "$do_ast" ] ; then
+do_audio_only ast "-ac 2" "-loopstart 1 -loopend 10"
+fi
+
# pix_fmt conversions
if [ -n "$do_pixfmt" ] ; then
diff --git a/tests/ref/lavf/ast b/tests/ref/lavf/ast
new file mode 100644
index 0000000..72a9824
--- /dev/null
+++ b/tests/ref/lavf/ast
@@ -0,0 +1,3 @@
+7fa8cd2dd7453428e71930a7c65f7b62 *./tests/data/lavf/lavf.ast
+181696 ./tests/data/lavf/lavf.ast
+./tests/data/lavf/lavf.ast CRC=0x7bd585ff
--
1.8.0.msysgit.0
More information about the ffmpeg-devel
mailing list