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use lmmsconfig.h rather than config.h and use prefixed macro-names

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@1097 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2008-06-08 11:30:47 +00:00
parent 6dcedad4f0
commit d6262bb556
81 changed files with 376 additions and 445 deletions
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54
src/core/sample_buffer.cpp
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@@ -28,10 +28,6 @@
#include "sample_buffer.h"
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <QtCore/QBuffer>
#include <QtCore/QFile>
#include <QtCore/QFileInfo>
@@ -42,23 +38,23 @@
#include <cstring>
#ifdef SDL_SDL_SOUND_H
#include SDL_SDL_SOUND_H
#ifdef LMMS_SDL_SDL_SOUND_H
#include LMMS_SDL_SDL_SOUND_H
#endif
#ifdef HAVE_SNDFILE_H
#ifdef LMMS_HAVE_SNDFILE_H
#include <sndfile.h>
#endif
#ifdef HAVE_VORBIS_VORBISFILE_H
#ifdef LMMS_HAVE_VORBIS_VORBISFILE_H
#include <vorbis/vorbisfile.h>
#endif
#ifdef HAVE_FLAC_STREAM_ENCODER_H
#ifdef LMMS_HAVE_FLAC_STREAM_ENCODER_H
#include <FLAC/stream_encoder.h>
#endif
#ifdef HAVE_FLAC_STREAM_DECODER_H
#ifdef LMMS_HAVE_FLAC_STREAM_DECODER_H
#include <FLAC/stream_decoder.h>
#endif
@@ -90,7 +86,7 @@ sampleBuffer::sampleBuffer( const QString & _audio_file,
m_frequency( BaseFreq ),
m_sampleRate( engine::getMixer()->baseSampleRate() )
{
#ifdef SDL_SDL_SOUND_H
#ifdef LMMS_SDL_SDL_SOUND_H
// init sound-file-system of SDL
Sound_Init();
#endif
@@ -125,7 +121,7 @@ sampleBuffer::sampleBuffer( const sampleFrame * _data, const f_cnt_t _frames ) :
memcpy( m_origData, _data, _frames * BYTES_PER_FRAME );
m_origFrames = _frames;
}
#ifdef SDL_SDL_SOUND_H
#ifdef LMMS_SDL_SDL_SOUND_H
// init sound-file-system of SDL
Sound_Init();
#endif
@@ -156,7 +152,7 @@ sampleBuffer::sampleBuffer( const f_cnt_t _frames ) :
memset( m_origData, 0, _frames * BYTES_PER_FRAME );
m_origFrames = _frames;
}
#ifdef SDL_SDL_SOUND_H
#ifdef LMMS_SDL_SDL_SOUND_H
// init sound-file-system of SDL
Sound_Init();
#endif
@@ -209,21 +205,21 @@ void sampleBuffer::update( bool _keep_settings )
m_frames = 0;
#ifdef HAVE_SNDFILE_H
#ifdef LMMS_HAVE_SNDFILE_H
if( m_frames == 0 )
{
m_frames = decodeSampleSF( f, buf, channels,
samplerate );
}
#endif
#ifdef HAVE_VORBIS_VORBISFILE_H
#ifdef LMMS_HAVE_VORBIS_VORBISFILE_H
if( m_frames == 0 )
{
m_frames = decodeSampleOGGVorbis( f, buf, channels,
samplerate );
}
#endif
#ifdef SDL_SDL_SOUND_H
#ifdef LMMS_SDL_SDL_SOUND_H
if( m_frames == 0 )
{
m_frames = decodeSampleSDL( f, buf, channels,
@@ -341,7 +337,7 @@ void sampleBuffer::normalizeSampleRate( const sample_rate_t _src_sr,
#ifdef SDL_SDL_SOUND_H
#ifdef LMMS_SDL_SDL_SOUND_H
f_cnt_t sampleBuffer::decodeSampleSDL( const char * _f,
int_sample_t * & _buf,
ch_cnt_t _channels,
@@ -376,7 +372,7 @@ f_cnt_t sampleBuffer::decodeSampleSDL( const char * _f,
#ifdef HAVE_SNDFILE_H
#ifdef LMMS_HAVE_SNDFILE_H
f_cnt_t sampleBuffer::decodeSampleSF( const char * _f,
int_sample_t * & _buf,
ch_cnt_t & _channels,
@@ -422,7 +418,7 @@ f_cnt_t sampleBuffer::decodeSampleSF( const char * _f,
#ifdef HAVE_VORBIS_VORBISFILE_H
#ifdef LMMS_HAVE_VORBIS_VORBISFILE_H
// callback-functions for reading ogg-file
@@ -576,7 +572,7 @@ f_cnt_t sampleBuffer::decodeSampleDS( const char * _f,
bool FASTCALL sampleBuffer::play( sampleFrame * _ab, handleState * _state,
bool sampleBuffer::play( sampleFrame * _ab, handleState * _state,
const fpp_t _frames,
const float _freq,
const bool _looped ) const
@@ -860,10 +856,10 @@ QString sampleBuffer::openAudioFile( void ) const
}
#undef HAVE_FLAC_STREAM_ENCODER_H /* not yet... */
#undef HAVE_FLAC_STREAM_DECODER_H
#undef LMMS_HAVE_FLAC_STREAM_ENCODER_H /* not yet... */
#undef LMMS_HAVE_FLAC_STREAM_DECODER_H
#ifdef HAVE_FLAC_STREAM_ENCODER_H
#ifdef LMMS_HAVE_FLAC_STREAM_ENCODER_H
FLAC__StreamEncoderWriteStatus flacStreamEncoderWriteCallback(
const FLAC__StreamEncoder *
/*_encoder*/,
@@ -900,7 +896,7 @@ void flacStreamEncoderMetadataCallback( const FLAC__StreamEncoder *,
QString & sampleBuffer::toBase64( QString & _dst ) const
{
#ifdef HAVE_FLAC_STREAM_ENCODER_H
#ifdef LMMS_HAVE_FLAC_STREAM_ENCODER_H
const f_cnt_t FRAMES_PER_BUF = 1152;
FLAC__StreamEncoder * flac_enc = FLAC__stream_encoder_new();
@@ -950,12 +946,12 @@ QString & sampleBuffer::toBase64( QString & _dst ) const
_dst );
#else /* HAVE_FLAC_STREAM_ENCODER_H */
#else /* LMMS_HAVE_FLAC_STREAM_ENCODER_H */
base64::encode( (const char *) m_data,
m_frames * sizeof( sampleFrame ), _dst );
#endif /* HAVE_FLAC_STREAM_ENCODER_H */
#endif /* LMMS_HAVE_FLAC_STREAM_ENCODER_H */
return( _dst );
}
@@ -1013,7 +1009,7 @@ void sampleBuffer::setAudioFile( const QString & _audio_file )
#ifdef HAVE_FLAC_STREAM_DECODER_H
#ifdef LMMS_HAVE_FLAC_STREAM_DECODER_H
struct flacStreamDecoderClientData
{
@@ -1112,7 +1108,7 @@ void sampleBuffer::loadFromBase64( const QString & _data )
int dsize = 0;
base64::decode( _data, &dst, &dsize );
#ifdef HAVE_FLAC_STREAM_DECODER_H
#ifdef LMMS_HAVE_FLAC_STREAM_DECODER_H
QByteArray orig_data = QByteArray::fromRawData( dst, dsize );
QBuffer ba_reader( &orig_data );
@@ -1152,7 +1148,7 @@ void sampleBuffer::loadFromBase64( const QString & _data )
m_origData = new sampleFrame[m_origFrames];
memcpy( m_origData, orig_data.data(), orig_data.size() );
#else /* HAVE_FLAC_STREAM_DECODER_H */
#else /* LMMS_HAVE_FLAC_STREAM_DECODER_H */
m_origFrames = dsize / sizeof( sampleFrame );
delete[] m_origData;