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authorJames Almer <jamrial@gmail.com>2012-11-29 13:14:50 -0300
committerPaul B Mahol <onemda@gmail.com>2012-12-04 11:25:46 +0000
commitb70251c0adcc256899217798ac744b440d41d237 (patch)
tree0e845fc0ef727d04e978d4b12db834d3d04b75ce /libavformat/astenc.c
parent7f0e7fc9e2e668a535dec5a5c70aa95c7788700c (diff)
AST Muxer
Signed-off-by: James Almer <jamrial@gmail.com>
Diffstat (limited to 'libavformat/astenc.c')
-rw-r--r--libavformat/astenc.c202
1 files changed, 202 insertions, 0 deletions
diff --git a/libavformat/astenc.c b/libavformat/astenc.c
new file mode 100644
index 0000000000..2c4afb7afb
--- /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;
+ unsigned int codec_tag;
+
+ if (s->nb_streams = 1) {
+ enc = s->streams[0]->codec;
+ } else {
+ av_log(s, AV_LOG_ERROR, "only one stream is supported\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (enc->codec_id == AV_CODEC_ID_ADPCM_AFC) {
+ av_log(s, AV_LOG_ERROR, "muxing ADPCM AFC is not implemented\n");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ codec_tag = ff_codec_get_tag(ff_codec_ast_tags, enc->codec_id);
+ if (!codec_tag) {
+ av_log(s, AV_LOG_ERROR, "unsupported codec\n");
+ return AVERROR(EINVAL);
+ }
+
+ 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, 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 file_size = avio_tell(pb);
+ int64_t samples = (file_size - 64 - (32 * enc->frame_number)) / enc->block_align; /* PCM_S16BE_PLANAR */
+
+ 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_wb32(pb, ast->fbs);
+ 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},
+};