ffmpeg/libavformat/astenc.c
James Almer 974ac57e83 astenc: Remove two AVRationals with denominator 1
They are completely superfluous when using av_rescale_q_rnd().
Call av_rescale_rnd() using what used to be the numerators instead.

Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-12-20 05:05:55 +01:00

203 lines
6.1 KiB
C

/*
* 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_rnd(ast->loop ## type, enc->sample_rate, 1000, 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},
};