[FFmpeg-devel] [PATCH] AST muxer
jamal
jamrial at gmail.com
Thu Nov 22 09:58:20 CET 2012
Hello all. The attached patch introduces an AST muxer (an audio container used on some Wii games).
It needs Mahol's ADPCM AFC patch for the AV_CODEC_ID_ADPCM_AFC codec id, which hasn't been applied yet.
Regards.
-------------- next part --------------
>From 65cc4c2f4e61bedb4fb5d8e850d2b4d1b8d4808e Mon Sep 17 00:00:00 2001
From: jamal <jamrial at gmail.com>
Date: Thu, 22 Nov 2012 05:28:02 -0300
Subject: [PATCH] AST muxer
---
Changelog | 2 +-
doc/general.texi | 2 +-
libavformat/Makefile | 1 +
libavformat/allformats.c | 2 +-
libavformat/astenc.c | 202 +++++++++++++++++++++++++++++++++++++++++++++++
tests/fate/avformat.mak | 1 +
tests/lavf-regression.sh | 4 +
tests/ref/lavf/ast | 3 +
8 files changed, 214 insertions(+), 3 deletions(-)
create mode 100644 libavformat/astenc.c
create mode 100644 tests/ref/lavf/ast
diff --git a/Changelog b/Changelog
index 2ff7921..b719f12 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
diff --git a/doc/general.texi b/doc/general.texi
index 4bc0b78..dd98857 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 136ada8..fbc5c9d 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -51,6 +51,7 @@ 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-$(CONFIG_AST_MUXER) += astenc.o
OBJS-$(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/astenc.c b/libavformat/astenc.c
new file mode 100644
index 0000000..2121e0c
--- /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 "libavutil/opt.h"
+
+typedef struct ASTContext {
+ AVClass *class;
+ int64_t size;
+ int64_t samples;
+ int64_t loopstart;
+ int64_t loopend;
+ int fbs;
+} ASTContext;
+
+#define CHECK_LOOP(type) \
+ if(ast->loop ## type) { \
+ ast->loop ## type = ast->loop ## type * sample; \
+ av_log(s, AV_LOG_DEBUG, "loop ## type: %"PRId64"\n", ast->loop ## type); \
+ if(ast->loop ## type < 0 || ast->loop ## type > UINT_MAX) { \
+ av_log(s, AV_LOG_ERROR, "Invalid loop" #type " value\n"); \
+ return -1; \
+ } \
+ }
+
+static int ast_write_header(AVFormatContext *s)
+{
+ ASTContext *ast = s->priv_data;
+ AVIOContext *pb = s->pb;
+ AVCodecContext *enc = s->streams[0]->codec;
+ int codec_id;
+ double sample = (double)enc->sample_rate / 1000;
+ av_log(s, AV_LOG_DEBUG, "(double) sample_rate / 1000: %f\n", sample);
+
+ CHECK_LOOP(start)
+ CHECK_LOOP(end)
+
+ 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 -1;
+ }
+
+ switch(enc->codec_id) {
+ case AV_CODEC_ID_ADPCM_AFC:
+ codec_id = 0;
+ break;
+ case AV_CODEC_ID_PCM_S16BE_PLANAR:
+ codec_id = 1;
+ break;
+ default:
+ av_log(s, AV_LOG_ERROR, "Unsupported codec\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if(enc->channels != 2 && enc->channels != 4) {
+ av_log(s, AV_LOG_ERROR, "Unsupported channel amount: %d\n", enc->channels);
+ return AVERROR_INVALIDDATA;
+ }
+
+ 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);
+ 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;
+ ASTContext *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;
+ 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;
+
+ 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_SET);
+ avio_flush(pb);
+ }
+ return 0;
+}
+
+#define OFFSET(obj) offsetof(ASTContext, obj)
+static const AVOption options[] = {
+ { "loopstart", "Loopstart position in miliseconds.", OFFSET(loopstart), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
+ { "loopend", "Loopend position in miliseconds.", 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(ASTContext),
+ .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,
+};
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
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