void StarSeemTest::star_external_ffem()
{
QFETCH(QString, data);
AudioBuffer c;
QVERIFY2(c.read(data), ("Failed reading test wave; " + data).toUtf8().data());
/*=== WORLD による分析 ===*/
int timeLength = GetSamplesForDIO(c.format().sampleRate(), c.length(), msFramePeriod);
int fftLength = GetFFTSizeForStar(c.format().sampleRate());
double *f0 = new double[timeLength];
double *t = new double[timeLength];
double **specgram = new double*[timeLength];
specgram[0] = new double[timeLength * (fftLength / 2 + 1)];
for(int i = 1; i < timeLength; i++)
{
specgram[i] = specgram[0] + i * (fftLength / 2 + 1);
}
Dio(c.data()[0], c.length(), c.format().sampleRate(), msFramePeriod, t, f0);
Star(c.data()[0], c.length(), c.format().sampleRate(), msFramePeriod, f0, timeLength, specgram);
/*=== WORLD による分析ここまで ===*/
/*=== StarSeem による分析 === */
Envelope *e = new Envelope;
QVERIFY2(DioFfem().estimate(e, c.data()[0], c.length(), c.format().sampleRate(), msFramePeriod), "Failed dio FFEM");
// Envelope が正しいか確認する
for(int i = 0; i < e->size(); i++)
{
QVERIFY2(e->value(i) == f0[i], "Error; invalid f0 envelope");
}
Specgram *testset = new Specgram;
QVERIFY2(StarSeem(e).estimate(testset, c.data()[0], c.length(), fftLength, c.format().sampleRate(), msFramePeriod), "Failed STAR SEEM");
/*=== StarSeem による分析ここまで === */
// 比較開始
for(int t = 0; t < timeLength; t++)
{
for(int f = 0; f <= fftLength / 2; f++)
{
if(testset->value(t, f) != specgram[t][f])
{
QString s;
s = "t = " + QString::number(t) + ", f = " + QString::number(f);
s += " actual :" + QString::number(testset->value(t, f)) + " , expceted :" + QString::number(specgram[t][f]);
// WORLD 自体は同じだから値が全く同じでないとおかしい。
QFAIL(("Error ;" + s).toUtf8().data());
}
}
}
delete[] specgram[0];
delete[] specgram;
delete[] t;
delete[] f0;
}
void AudioBuffer::convert(const AudioSpec &_new_spec)
{
if(_new_spec == m_spec) {
return;
}
AudioSpec new_spec = _new_spec;
AudioBuffer dest[2];
unsigned bufidx = 0;
AudioBuffer *source = this;
if(source->rate() != new_spec.rate) {
#if HAVE_LIBSAMPLERATE
if(source->format() != AUDIO_FORMAT_F32) {
dest[1].set_spec({AUDIO_FORMAT_F32, source->channels(), source->rate()});
source->convert_format(dest[1], source->frames());
source = &dest[1];
}
dest[0].set_spec({source->format(), source->channels(), new_spec.rate});
source->convert_rate(dest[0], source->frames(), nullptr);
source = &dest[0];
bufidx = 1;
#else
new_spec.rate = source->rate();
#endif
}
if(source->channels() != new_spec.channels) {
dest[bufidx].set_spec({source->format(),new_spec.channels,source->rate()});
source->convert_channels(dest[bufidx], source->frames());
source = &dest[bufidx];
bufidx = (bufidx + 1) % 2;
}
if(source->format() != new_spec.format) {
dest[bufidx].set_spec({new_spec.format,source->channels(),source->rate()});
source->convert_format(dest[bufidx], source->frames());
source = &dest[bufidx];
}
if(new_spec != m_spec) {
m_data = source->m_data;
m_spec = new_spec;
}
}
void AudioBuffer::convert_format(AudioBuffer &_dest, unsigned _frames_count)
{
AudioSpec destspec{_dest.format(), m_spec.channels, m_spec.rate};
if(_dest.spec() != destspec) {
throw std::logic_error("destination must have same channels and rate");
}
_frames_count = std::min(frames(),_frames_count);
if(m_spec.format == destspec.format) {
_dest.add_frames(*this,_frames_count);
return;
}
const unsigned samples_count = m_spec.frames_to_samples(_frames_count);
std::vector<uint8_t> buffer, *data=&buffer;
switch(m_spec.format) {
case AUDIO_FORMAT_U8:
u8_to_f32(m_data, buffer, samples_count);
break;
case AUDIO_FORMAT_S16:
s16_to_f32(m_data, buffer, samples_count);
break;
case AUDIO_FORMAT_F32:
data = &m_data;
break;
default:
throw std::logic_error("unsupported source format");
}
switch(destspec.format) {
case AUDIO_FORMAT_S16:
f32_to_s16(*data, _dest.m_data, samples_count);
break;
case AUDIO_FORMAT_F32:
_dest.m_data.swap(buffer);
break;
default:
throw std::logic_error("unsupported destination format");
}
}
