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//=============================================================================
// MuseScore
// Music Composition & Notation
//
// Copyright (C) 2007-2011 Werner Schweer
//
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License version 2
// as published by the Free Software Foundation and appearing in
// the file LICENCE.GPL
//=============================================================================
#include "audiofile.h"
static sf_count_t getFileLen(void* f) {
return ((AudioFile*)f)->getFileLen();
}
static sf_count_t seek(sf_count_t offset, int whence, void* f) {
return ((AudioFile*)f)->seek(offset, whence);
}
static sf_count_t read(void* ptr, sf_count_t count, void* f) {
return ((AudioFile*)f)->read(ptr, count);
}
static sf_count_t write(const void* ptr, sf_count_t count, void* f) {
return ((AudioFile*)f)->write(ptr, count);
}
static sf_count_t tell(void* f) {
return ((AudioFile*)f)->tell();
}
static SF_VIRTUAL_IO sfio = {
getFileLen,
seek,
read,
write,
tell
};
//---------------------------------------------------------
// AudioFile
//---------------------------------------------------------
AudioFile::AudioFile()
{
memset(&info, 0, sizeof(info));
memset(&inst, 0, sizeof(inst));
sf = 0;
}
AudioFile::~AudioFile()
{
if (sf)
sf_close(sf);
}
//---------------------------------------------------------
// open
//---------------------------------------------------------
bool AudioFile::open(const QByteArray& b)
{
buf = b;
idx = 0;
sf = sf_open_virtual(&sfio, SFM_READ, &info, this);
hasInstrument = sf_command(sf, SFC_GET_INSTRUMENT, &inst, sizeof(inst)) == SF_TRUE;
_type = info.format & SF_FORMAT_OGG ? fltp : s16p;
return sf != 0;
}
//---------------------------------------------------------
// readData
//---------------------------------------------------------
sf_count_t AudioFile::readData(short* data, sf_count_t frames)
{
//see https://musescore.org/en/node/22086#comment-83671
//see https://github.com/erikd/libsndfile/issues/16
//this code fixes the bug in libsndfile: float values are not normalized when reading .ogg
//this leads to overflowing signed short values, reverting a sign and clicking noise
sf_count_t resFrames = 0;
if (s16p == _type)
resFrames = sf_readf_short(sf, data, frames);
else {
//read native float values
int totalFrames = frames * channels();
std::vector<float> dataF;
dataF.resize(totalFrames);
resFrames = sf_readf_float(sf, dataF.data(), frames);
//find the maximum signal value
float maxSignal = 0.f;
for (int i = 0; i < totalFrames; ++i) {
if (fabs(dataF[i]) > maxSignal)
maxSignal = dataF[i] > 0 ? dataF[i] : -dataF[i];
}
//normalize values if and only if sample values range is incorrect
//which means having at least one sample value more than 1.0
float adjScale = maxSignal > 1.f ? 1.f/maxSignal : 1.f;
//convert normalized floats to signed short values
for (int i = 0; i < totalFrames; ++i)
data[i] = adjScale * lrintf(dataF[i] * (dataF[i] > 0 ? SHRT_MAX : -SHRT_MIN));
}
return resFrames;
}
//---------------------------------------------------------
// seek
//---------------------------------------------------------
sf_count_t AudioFile::seek(sf_count_t offset, int whence)
{
switch(whence) {
case SEEK_SET:
idx = offset;
break;
case SEEK_CUR:
idx += offset;
break;
case SEEK_END:
idx = buf.size() + offset;
break;
}
return idx;
}
//---------------------------------------------------------
// read
//---------------------------------------------------------
sf_count_t AudioFile::read(void* ptr, sf_count_t count)
{
count = qMin(count, (sf_count_t)(buf.size() - idx));
memcpy(ptr, buf.data() + idx, count);
idx += count;
return count;
}
//---------------------------------------------------------
// write
//---------------------------------------------------------
sf_count_t AudioFile::write(const void* /*ptr*/, sf_count_t /*count*/)
{
//printf("write\n");
return 0;
}
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