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FFmpeg/libavformat/libopenmpt.c

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/*
* Tracked MOD demuxer (libopenmpt)
* Copyright (c) 2016 Josh de Kock
*
* 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 <libopenmpt/libopenmpt.h>
#include <libopenmpt/libopenmpt_stream_callbacks_file.h>
#include <libopenmpt/libopenmpt_version.h>
/* Shims to support libopenmpt < 0.3.0 (as documented by libopenmpt) */
#if !defined(OPENMPT_API_VERSION_MAKE)
#define OPENMPT_API_VERSION_MAKE(major, minor, patch) (((major)<<24)|((minor)<<16)|((patch)<<0))
#endif
#if !defined(OPENMPT_API_VERSION_AT_LEAST)
#define OPENMPT_API_VERSION_AT_LEAST(major, minor, patch) (OPENMPT_API_VERSION >= OPENMPT_API_VERSION_MAKE((major), (minor), (patch)))
#endif
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "avformat.h"
#include "internal.h"
typedef struct OpenMPTContext {
const AVClass *class;
openmpt_module *module;
int channels;
double duration;
/* options */
int sample_rate;
int64_t layout;
int subsong;
} OpenMPTContext;
#define OFFSET(x) offsetof(OpenMPTContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define D AV_OPT_FLAG_DECODING_PARAM
static const AVOption options[] = {
{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1000, INT_MAX, A | D },
{ "layout", "set channel layout", OFFSET(layout), AV_OPT_TYPE_CHANNEL_LAYOUT, { .i64 = AV_CH_LAYOUT_STEREO }, 0, INT64_MAX, A | D },
{ "subsong", "set subsong", OFFSET(subsong), AV_OPT_TYPE_INT, { .i64 = -2 }, -2, INT_MAX, A | D, "subsong"},
{ "all", "all", 0, AV_OPT_TYPE_CONST, { .i64 = -1}, 0, 0, A | D, "subsong" },
{ "auto", "auto", 0, AV_OPT_TYPE_CONST, { .i64 = -2}, 0, 0, A | D, "subsong" },
{ NULL }
};
static void openmpt_logfunc(const char *message, void *userdata)
{
int level = AV_LOG_INFO;
if (strstr(message, "ERROR") != NULL) {
level = AV_LOG_ERROR;
}
av_log(userdata, level, "%s\n", message);
}
#define add_meta(s, name, meta) \
do { \
const char *value = meta; \
if (value && value[0]) \
av_dict_set(&s->metadata, name, value, 0); \
openmpt_free_string(value); \
} while(0)
static int read_header_openmpt(AVFormatContext *s)
{
AVStream *st;
OpenMPTContext *openmpt = s->priv_data;
int64_t size;
char *buf;
#if OPENMPT_API_VERSION_AT_LEAST(0,3,0)
int error;
#endif
int ret;
size = avio_size(s->pb);
if (size <= 0)
return AVERROR_INVALIDDATA;
buf = av_malloc(size);
if (!buf)
return AVERROR(ENOMEM);
size = avio_read(s->pb, buf, size);
if (size < 0) {
av_log(s, AV_LOG_ERROR, "Reading input buffer failed.\n");
av_freep(&buf);
return size;
}
#if OPENMPT_API_VERSION_AT_LEAST(0,3,0)
error = OPENMPT_ERROR_OK;
openmpt->module = openmpt_module_create_from_memory2(buf, size, openmpt_logfunc, s, NULL, NULL, &error, NULL, NULL);
av_freep(&buf);
if (!openmpt->module) {
if (error == OPENMPT_ERROR_OUT_OF_MEMORY)
return AVERROR(ENOMEM);
else if (error >= OPENMPT_ERROR_GENERAL)
return AVERROR_INVALIDDATA;
else
return AVERROR_UNKNOWN;
}
#else
openmpt->module = openmpt_module_create_from_memory(buf, size, openmpt_logfunc, s, NULL);
av_freep(&buf);
if (!openmpt->module)
return AVERROR_INVALIDDATA;
#endif
openmpt->channels = av_get_channel_layout_nb_channels(openmpt->layout);
if (openmpt->subsong >= openmpt_module_get_num_subsongs(openmpt->module)) {
openmpt_module_destroy(openmpt->module);
av_log(s, AV_LOG_ERROR, "Invalid subsong index: %d\n", openmpt->subsong);
return AVERROR(EINVAL);
}
if (openmpt->subsong != -2) {
if (openmpt->subsong >= 0) {
av_dict_set_int(&s->metadata, "track", openmpt->subsong + 1, 0);
}
ret = openmpt_module_select_subsong(openmpt->module, openmpt->subsong);
if (!ret){
openmpt_module_destroy(openmpt->module);
av_log(s, AV_LOG_ERROR, "Could not select requested subsong: %d", openmpt->subsong);
return AVERROR(EINVAL);
}
}
openmpt->duration = openmpt_module_get_duration_seconds(openmpt->module);
add_meta(s, "artist", openmpt_module_get_metadata(openmpt->module, "artist"));
add_meta(s, "title", openmpt_module_get_metadata(openmpt->module, "title"));
add_meta(s, "encoder", openmpt_module_get_metadata(openmpt->module, "tracker"));
add_meta(s, "comment", openmpt_module_get_metadata(openmpt->module, "message"));
add_meta(s, "date", openmpt_module_get_metadata(openmpt->module, "date"));
st = avformat_new_stream(s, NULL);
if (!st) {
openmpt_module_destroy(openmpt->module);
openmpt->module = NULL;
return AVERROR(ENOMEM);
}
avpriv_set_pts_info(st, 64, 1, AV_TIME_BASE);
st->duration = llrint(openmpt->duration*AV_TIME_BASE);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = AV_NE(AV_CODEC_ID_PCM_F32BE, AV_CODEC_ID_PCM_F32LE);
st->codecpar->channels = openmpt->channels;
st->codecpar->sample_rate = openmpt->sample_rate;
return 0;
}
#define AUDIO_PKT_SIZE 2048
static int read_packet_openmpt(AVFormatContext *s, AVPacket *pkt)
{
OpenMPTContext *openmpt = s->priv_data;
int n_samples = AUDIO_PKT_SIZE / (openmpt->channels ? openmpt->channels*4 : 4);
int ret;
if ((ret = av_new_packet(pkt, AUDIO_PKT_SIZE)) < 0)
return ret;
switch (openmpt->channels) {
case 1:
ret = openmpt_module_read_float_mono(openmpt->module, openmpt->sample_rate,
n_samples, (float *)pkt->data);
break;
case 2:
ret = openmpt_module_read_interleaved_float_stereo(openmpt->module, openmpt->sample_rate,
n_samples, (float *)pkt->data);
break;
case 4:
ret = openmpt_module_read_interleaved_float_quad(openmpt->module, openmpt->sample_rate,
n_samples, (float *)pkt->data);
break;
default:
av_log(s, AV_LOG_ERROR, "Unsupported number of channels: %d", openmpt->channels);
return AVERROR(EINVAL);
}
if (ret < 1) {
pkt->size = 0;
return AVERROR_EOF;
}
pkt->size = ret * (openmpt->channels * 4);
return 0;
}
static int read_close_openmpt(AVFormatContext *s)
{
OpenMPTContext *openmpt = s->priv_data;
openmpt_module_destroy(openmpt->module);
openmpt->module = NULL;
return 0;
}
static int read_seek_openmpt(AVFormatContext *s, int stream_idx, int64_t ts, int flags)
{
OpenMPTContext *openmpt = s->priv_data;
openmpt_module_set_position_seconds(openmpt->module, (double)ts/AV_TIME_BASE);
return 0;
}
static const AVClass class_openmpt = {
.class_name = "libopenmpt",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_libopenmpt_demuxer = {
.name = "libopenmpt",
.long_name = NULL_IF_CONFIG_SMALL("Tracker formats (libopenmpt)"),
.priv_data_size = sizeof(OpenMPTContext),
.read_header = read_header_openmpt,
.read_packet = read_packet_openmpt,
.read_close = read_close_openmpt,
.read_seek = read_seek_openmpt,
.priv_class = &class_openmpt,
.extensions = "669,amf,ams,dbm,digi,dmf,dsm,far,gdm,imf,it,j2b,m15,mdl,med,mmcmp,mms,mo3,mod,mptm,mt2,mtm,nst,okt,plm,ppm,psm,pt36,ptm,s3m,sfx,sfx2,stk,stm,ult,umx,wow,xm,xpk",
};