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made quality-settings (internal processing samplerate, interpolation and so on) independent of audio-devices (final output) and added new qualitySettings-structure to mixer - might be still buggy and HQ-mode is currently not working, this is going to be fixed tomorrow, anyways important preparation for new project-export-dialog

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@957 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2008-05-05 23:14:48 +00:00
parent 2b7b813f2f
commit 5cfb1b2f96
57 changed files with 441 additions and 325 deletions
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20
src/core/sample_buffer.cpp
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@@ -88,7 +88,7 @@ sampleBuffer::sampleBuffer( const QString & _audio_file,
m_amplification( 1.0f ),
m_reversed( FALSE ),
m_frequency( BaseFreq ),
m_sampleRate( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
m_sampleRate( engine::getMixer()->baseSampleRate() )
{
#ifdef SDL_SDL_SOUND_H
// init sound-file-system of SDL
@@ -117,7 +117,7 @@ sampleBuffer::sampleBuffer( const sampleFrame * _data, const f_cnt_t _frames ) :
m_amplification( 1.0f ),
m_reversed( FALSE ),
m_frequency( BaseFreq ),
m_sampleRate( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
m_sampleRate( engine::getMixer()->baseSampleRate() )
{
if( _frames > 0 )
{
@@ -148,7 +148,7 @@ sampleBuffer::sampleBuffer( const f_cnt_t _frames ) :
m_amplification( 1.0f ),
m_reversed( FALSE ),
m_frequency( BaseFreq ),
m_sampleRate( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
m_sampleRate( engine::getMixer()->baseSampleRate() )
{
if( _frames > 0 )
{
@@ -205,7 +205,7 @@ void sampleBuffer::update( bool _keep_settings )
char * f = qstrdup( file.toAscii().constData() );
int_sample_t * buf = NULL;
ch_cnt_t channels = DEFAULT_CHANNELS;
sample_rate_t samplerate = SAMPLE_RATES[DEFAULT_QUALITY_LEVEL];
sample_rate_t samplerate = engine::getMixer()->baseSampleRate();
m_frames = 0;
@@ -279,7 +279,7 @@ m_data[frame][chnl] = buf[idx] * fac;
delete[] buf;
normalize_sample_rate( samplerate, _keep_settings );
normalizeSampleRate( samplerate, _keep_settings );
}
else
{
@@ -314,14 +314,14 @@ m_data[frame][chnl] = buf[idx] * fac;
void sampleBuffer::normalize_sample_rate( const sample_rate_t _src_sr,
void sampleBuffer::normalizeSampleRate( const sample_rate_t _src_sr,
bool _keep_settings )
{
// do samplerate-conversion to our default-samplerate
if( _src_sr != SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
if( _src_sr != engine::getMixer()->baseSampleRate() )
{
sampleBuffer * resampled = resample( this, _src_sr,
SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] );
engine::getMixer()->baseSampleRate() );
delete[] m_data;
m_frames = resampled->frames();
m_data = new sampleFrame[m_frames];
@@ -588,8 +588,8 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, handleState * _state,
return( FALSE );
}
const double freq_factor = (double) _freq / (double) m_frequency
* m_sampleRate / engine::getMixer()->sampleRate();
const double freq_factor = (double) _freq / (double) m_frequency *
m_sampleRate / engine::getMixer()->processingSampleRate();
// calculate how many frames we have in requested pitch
const f_cnt_t total_frames_for_current_pitch = static_cast<f_cnt_t>( (