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Merge pull request #14 from diizy/fix32

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SampleBuffer: fix import of 32-bit floating point samples
This commit is contained in:
tobydox
2014-01-14 00:30:39 -08:00
parent 39f904117f 79119149ef
commit 7105c8ac90
2 changed files with 137 additions and 47 deletions
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2
include/sample_buffer.h
View File

@@ -219,6 +219,8 @@ public slots:
private:
void update( bool _keep_settings = false );
void convertIntToFloat ( int_sample_t * & _ibuf, f_cnt_t _frames, int _channels);
void directFloatWrite ( sample_t * & _fbuf, f_cnt_t _frames, int _channels);
f_cnt_t decodeSampleSF( const char * _f, int_sample_t * & _buf,
ch_cnt_t & _channels,

182
src/core/sample_buffer.cpp
View File

@@ -225,56 +225,26 @@ void sampleBuffer::update( bool _keep_settings )
}
delete[] f;
}
if( m_frames > 0 && buf != NULL )
{
// following code transforms int-samples into
// float-samples and does amplifying & reversing
const float fac = m_amplification /
OUTPUT_SAMPLE_MULTIPLIER;
m_data = new sampleFrame[m_frames];
const int ch = ( channels > 1 ) ? 1 : 0;
// if reversing is on, we also reverse when
// scaling
if( m_reversed )
if ( m_frames == 0 ) // if still no frames, bail
{
int idx = ( m_frames - 1 ) * channels;
for( f_cnt_t frame = 0; frame < m_frames;
++frame )
{
m_data[frame][0] = buf[idx+0] * fac;
m_data[frame][1] = buf[idx+ch] * fac;
idx -= channels;
}
// sample couldn't be decoded, create buffer containing
// one sample-frame
m_data = new sampleFrame[1];
memset( m_data, 0, sizeof( *m_data ) );
m_frames = 1;
m_loopStartFrame = m_startFrame = 0;
m_loopEndFrame = m_endFrame = 1;
}
else
else // otherwise normalize sample rate
{
int idx = 0;
for( f_cnt_t frame = 0; frame < m_frames;
++frame )
{
m_data[frame][0] = buf[idx+0] * fac;
m_data[frame][1] = buf[idx+ch] * fac;
idx += channels;
}
normalizeSampleRate( samplerate, _keep_settings );
}
delete[] buf;
}
normalizeSampleRate( samplerate, _keep_settings );
}
else
{
// sample couldn't be decoded, create buffer containing
// one sample-frame
m_data = new sampleFrame[1];
memset( m_data, 0, sizeof( *m_data ) );
m_frames = 1;
m_loopStartFrame = m_startFrame = 0;
m_loopEndFrame = m_endFrame = 1;
}
}
else
{
@@ -296,6 +266,85 @@ void sampleBuffer::update( bool _keep_settings )
}
void sampleBuffer::convertIntToFloat ( int_sample_t * & _ibuf, f_cnt_t _frames, int _channels)
{
// following code transforms int-samples into
// float-samples and does amplifying & reversing
const float fac = m_amplification /
OUTPUT_SAMPLE_MULTIPLIER;
m_data = new sampleFrame[_frames];
const int ch = ( _channels > 1 ) ? 1 : 0;
// if reversing is on, we also reverse when
// scaling
if( m_reversed )
{
int idx = ( _frames - 1 ) * _channels;
for( f_cnt_t frame = 0; frame < _frames;
++frame )
{
m_data[frame][0] = _ibuf[idx+0] * fac;
m_data[frame][1] = _ibuf[idx+ch] * fac;
idx -= _channels;
}
}
else
{
int idx = 0;
for( f_cnt_t frame = 0; frame < _frames;
++frame )
{
m_data[frame][0] = _ibuf[idx+0] * fac;
m_data[frame][1] = _ibuf[idx+ch] * fac;
idx += _channels;
}
}
delete[] _ibuf;
}
void sampleBuffer::directFloatWrite ( sample_t * & _fbuf, f_cnt_t _frames, int _channels)
{
m_data = new sampleFrame[_frames];
const int ch = ( _channels > 1 ) ? 1 : 0;
// if reversing is on, we also reverse when
// scaling
if( m_reversed )
{
int idx = ( _frames - 1 ) * _channels;
for( f_cnt_t frame = 0; frame < _frames;
++frame )
{
m_data[frame][0] = _fbuf[idx+0];
m_data[frame][1] = _fbuf[idx+ch];
idx -= _channels;
}
}
else
{
int idx = 0;
for( f_cnt_t frame = 0; frame < _frames;
++frame )
{
m_data[frame][0] = _fbuf[idx+0];
m_data[frame][1] = _fbuf[idx+ch];
idx += _channels;
}
}
delete[] _fbuf;
}
void sampleBuffer::normalizeSampleRate( const sample_rate_t _src_sr,
@@ -333,12 +382,26 @@ f_cnt_t sampleBuffer::decodeSampleSF( const char * _f,
SNDFILE * snd_file;
SF_INFO sf_info;
f_cnt_t frames = 0;
bool sf_rr = false;
sample_t * fbuf = 0;
if( ( snd_file = sf_open( _f, SFM_READ, &sf_info ) ) != NULL )
{
frames = sf_info.frames;
_buf = new int_sample_t[sf_info.channels * frames];
if( sf_read_short( snd_file, _buf, sf_info.channels * frames )
< sf_info.channels * frames )
// check if float
if ( (sf_info.format & SF_FORMAT_SUBMASK) == SF_FORMAT_FLOAT ) // if yes, use float format for buffer
{
fbuf = new sample_t[sf_info.channels * frames];
sf_rr = sf_read_float( snd_file, fbuf, sf_info.channels * frames );
}
else // otherwise, use int
{
_buf = new int_sample_t[sf_info.channels * frames];
sf_rr = sf_read_short( snd_file, _buf, sf_info.channels * frames );
}
if( sf_rr < sf_info.channels * frames )
{
#ifdef DEBUG_LMMS
printf( "sampleBuffer::decodeSampleSF(): could not read"
@@ -357,6 +420,17 @@ f_cnt_t sampleBuffer::decodeSampleSF( const char * _f,
"sample %s: %s\n", _f, sf_strerror( NULL ) );
#endif
}
//write down either directly or convert i->f depending on file type
if ( frames > 0 && fbuf != NULL )
{
directFloatWrite ( fbuf, frames, _channels);
}
else if ( frames > 0 && _buf != NULL )
{
convertIntToFloat ( _buf, frames, _channels);
}
return frames;
}
@@ -497,6 +571,12 @@ f_cnt_t sampleBuffer::decodeSampleOGGVorbis( const char * _f,
while( bytes_read != 0 && bitstream == 0 );
ov_clear( &vf );
// if buffer isn't empty, convert it to float and write it down
if ( frames > 0 && _buf != NULL )
{
convertIntToFloat ( _buf, frames, _channels);
}
return frames;
}
@@ -511,7 +591,15 @@ f_cnt_t sampleBuffer::decodeSampleDS( const char * _f,
sample_rate_t & _samplerate )
{
DrumSynth ds;
return ds.GetDSFileSamples( _f, _buf, _channels );
f_cnt_t frames = ds.GetDSFileSamples( _f, _buf, _channels );
if ( frames > 0 && _buf != NULL )
{
convertIntToFloat ( _buf, frames, _channels);
}
return frames;
}