mirror/lmms
1
0
Fork 0
mirror of https://github.com/LMMS/lmms.git synced 2026-03-21 07:23:19 -04:00
Code Issues Packages Projects Releases Wiki Activity

small bug-fixes

Browse Source 
git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@42 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2005-12-22 12:57:51 +00:00
parent f543cd8f4e
commit 181f9b56f6
11 changed files with 95 additions and 46 deletions
Download Patch File Download Diff File Expand all files Collapse all files

48
src/lib/sample_buffer.cpp
View File

@@ -144,7 +144,7 @@ sampleBuffer::sampleBuffer( const sampleFrame * _data, Uint32 _frames ) :
m_dataMutex()
{
m_origData = new sampleFrame[_frames];
memcpy( m_origData, _data, _frames*BYTES_PER_FRAME );
memcpy( m_origData, _data, _frames * BYTES_PER_FRAME );
m_origFrames = _frames;
#ifdef SDL_SDL_SOUND_H
// init sound-file-system of SDL
@@ -225,18 +225,18 @@ void sampleBuffer::update( bool _keep_settings )
Sint16 * buf = NULL;
Uint8 channels;
#ifdef SDL_SDL_SOUND_H
if( m_frames == 0 )
{
m_frames = decodeSampleSDL( f, buf, channels );
}
#endif
#ifdef HAVE_SNDFILE_H
if( m_frames == 0 )
{
m_frames = decodeSampleSF( f, buf, channels );
}
#endif
#ifdef SDL_SDL_SOUND_H
if( m_frames == 0 )
{
m_frames = decodeSampleSDL( f, buf, channels );
}
#endif
#ifdef HAVE_VORBIS_VORBISFILE_H
if( m_frames == 0 )
{
@@ -338,7 +338,7 @@ Uint32 sampleBuffer::decodeSampleSDL( const char * _f, Sint16 * & _buf,
Sound_Sample * snd_sample = Sound_NewSampleFromFile( _f,
&STD_AUDIO_INFO, 16384 );
// file not found?
// file found?
if( snd_sample != NULL )
{
// let SDL_sound decode our file to requested format
@@ -561,12 +561,10 @@ SRC_STATE * sampleBuffer::createResamplingContext( void )
{
int error;
SRC_STATE * state;
if( ( state = src_new(
#ifdef HQ_SINC
SRC_SINC_MEDIUM_QUALITY,
#else
SRC_ZERO_ORDER_HOLD,
#endif
if( ( state = src_new(/*
( mixer::inst()->highQuality() == TRUE ) ?
SRC_SINC_FASTEST :*/
SRC_LINEAR,
DEFAULT_CHANNELS, &error ) ) == NULL )
{
printf( "Error: src_new() failed in sample_buffer.cpp!\n" );
@@ -598,7 +596,7 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
return( FALSE );
}
const float freq_factor = 1.0f / (BASE_FREQ / _freq);
const double freq_factor = (double) _freq / (double) BASE_FREQ;
const Sint16 freq_diff = static_cast<Sint16>( BASE_FREQ - _freq );
Uint32 frames_to_process = _frames;
@@ -635,11 +633,21 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
{
frames_to_process = frames_for_loop;
}
const Uint32 f1 = static_cast<Uint32>( play_frame * freq_factor );
/* Uint32 f2 = 0;
while( f2 < f1 )
{
f2 += frames_to_process * freq_factor;
}
if( f2 > f1 && f2 >= frames_to_process )
{
f2 -= frames_to_process * freq_factor;
}*/
// static int foo = 0;
// calc pointer of first frame
sampleFrame * start_frame = (sampleFrame *) m_data +
static_cast<Uint32>( play_frame *
freq_factor );
sampleFrame * start_frame = (sampleFrame *) m_data + f1;
//printf("diff:%d %f %d f2: %d input: %d\n", f2 -foo, play_frame * freq_factor, static_cast<Uint32>( play_frame * freq_factor ), f2, (Uint32)( frames_for_loop * freq_factor ) );
// foo = f2;
sampleFrame * loop_start = start_frame;
// check whether we have to change pitch...
@@ -670,7 +678,7 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, Uint32 _start_frame,
#else
Uint32 src_frame_base = 0;
// check whether we're in high-quality-mode
if( mixer::inst()->highQuality() )
if( mixer::inst()->highQuality() == TRUE )
{
// we are, so let's use cubic interpolation...
for( Uint32 frame = 0; frame < frames_to_process;