Whole code base: various cleanups, removed SINGLE_SOURCE_COMPILE relicts

* cleaned up code and improved coding style
* removed old SINGLE_SOURCE_COMPILE macro relicts
* use QString::toUtf8() instead of QString::toAscii() wherever possible
This commit is contained in:
Tobias Doerffel committed 2009-08-05 11:41:59 +02:00
1 parent afbaabed02
commit 0bb54199b0
110 files changed
+132 -416

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+40 -40
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@@ -1,10 +1,8 @@
#ifndef SINGLE_SOURCE_COMPILE
/*
* sample_buffer.cpp - container-class sampleBuffer
*
* Copyright (c) 2005-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
*
* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
*
* This program is free software; you can redistribute it and/or
@@ -66,19 +64,19 @@
#include "lame_library.h"
const char * sampleBuffer::supportedExts[] = {
"wav",
"ogg",
"ds",
"spx",
"au",
"voc",
"aif",
const char * sampleBuffer::supportedExts[] = {
"wav",
"ogg",
"ds",
"spx",
"au",
"voc",
"aif",
"aiff",
"flac",
"raw",
"flac",
"raw",
"mp3",
NULL
NULL
};
sampleBuffer::sampleBuffer( const QString & _audio_file,
@@ -194,10 +192,10 @@ void sampleBuffer::update( bool _keep_settings )
m_loopEndFrame = m_endFrame = m_frames;
}
}
else if( m_audioFile != "" )
else if( !m_audioFile.isEmpty() )
{
QString file = tryToMakeAbsolute( m_audioFile );
char * f = qstrdup( file.toAscii().constData() );
char * f = qstrdup( file.toUtf8().constData() );
int_sample_t * buf = NULL;
ch_cnt_t channels = DEFAULT_CHANNELS;
sample_rate_t samplerate = engine::getMixer()->baseSampleRate();
@@ -215,12 +213,12 @@ void sampleBuffer::update( bool _keep_settings )
m_frames = decodeSampleSF( f, buf, channels, samplerate );
#ifdef LMMS_HAVE_OGGVORBIS
if( m_frames == 0 )
m_frames = decodeSampleOGGVorbis(f, buf, channels, samplerate);
m_frames = decodeSampleOGGVorbis( f, buf, channels, samplerate );
#endif
if( m_frames == 0 )
m_frames = decodeSampleDS( f, buf, channels, samplerate );
// MP3
// MP3
if( m_frames == 0 )
m_frames = decodeSampleMp3( file, buf, channels, samplerate );
@@ -311,20 +309,24 @@ void sampleBuffer::update( bool _keep_settings )
}
QString sampleBuffer::niceListOfExts() const
{
QString ret = "";
QString ret;
for( const char * * i = supportedExts; *i; ++i )
{
ret.append(*i);
ret.append( *i );
if( *(i+1) )
ret.append(", ");
ret.append( ", " );
}
return ret;
}
void sampleBuffer::normalizeSampleRate( const sample_rate_t _src_sr,
bool _keep_settings )
{
@@ -530,7 +532,7 @@ f_cnt_t sampleBuffer::decodeSampleOGGVorbis( const char * _f,
#endif
int lame_decode_fromfile(QFile &in, short pcm_l[], short pcm_r[],
int lame_decode_fromfile(QFile &in, short pcm_l[], short pcm_r[],
mp3data_struct * mp3data, LameLibrary &lame)
{
int ret = 0;
@@ -574,12 +576,12 @@ f_cnt_t sampleBuffer::decodeSampleMp3( QString & file, int_sample_t * & _buf,
{
// create instance of LameLibrary to decode
LameLibrary lame;
// open the file
QFile in(file);
if( ! in.open(QIODevice::ReadOnly) ){
printf("sampleBuffer::decodeSampleMp3: error opening %s for reading\n",
printf("sampleBuffer::decodeSampleMp3: error opening %s for reading\n",
file.toStdString().c_str());
return 0;
}
@@ -599,7 +601,7 @@ f_cnt_t sampleBuffer::decodeSampleMp3( QString & file, int_sample_t * & _buf,
while(1)
{
int ret = lame_decode_fromfile(in, pcm_l, pcm_r, &mp3data, lame);
if( ret == -1 ){
delete[] _buf;
printf("error decoding mp3\n");
@@ -620,11 +622,11 @@ f_cnt_t sampleBuffer::decodeSampleMp3( QString & file, int_sample_t * & _buf,
// process header
_samplerate = mp3data.samplerate;
_channels = mp3data.stereo;
_buf = new int_sample_t[mp3data.totalframes *
_buf = new int_sample_t[mp3data.totalframes *
mp3data.framesize * _channels];
initBuf = true;
}
}
}
// convert the decoded PCM into sample
for(int i = 0; i<ret; ++i)
@@ -634,9 +636,9 @@ f_cnt_t sampleBuffer::decodeSampleMp3( QString & file, int_sample_t * & _buf,
_buf[bufPos++] = pcm_r[i];
}
}
lame.lame_decode_exit();
return bufPos / _channels;
}
@@ -870,12 +872,12 @@ void sampleBuffer::visualize( QPainter & _p, const QRect & _dr,
QString sampleBuffer::openAudioFile( void ) const
QString sampleBuffer::openAudioFile() const
{
QFileDialog ofd( NULL, tr( "Open audio file" ) );
QString dir;
if( m_audioFile != "" )
if( !m_audioFile.isEmpty() )
{
QString f = m_audioFile;
if( QFileInfo( f ).isRelative() )
@@ -915,7 +917,7 @@ QString sampleBuffer::openAudioFile( void ) const
//<< tr( "MOD-Files (*.mod)" )
;
ofd.setFilters( types );
if( m_audioFile != "" )
if( !m_audioFile.isEmpty() )
{
// select previously opened file
ofd.selectFile( QFileInfo( m_audioFile ).fileName() );
@@ -941,7 +943,7 @@ QString sampleBuffer::openAudioFile( void ) const
FLAC__StreamEncoderWriteStatus flacStreamEncoderWriteCallback(
const FLAC__StreamEncoder *
/*_encoder*/,
const FLAC__byte _buffer[],
const FLAC__byte _buffer[],
unsigned int/* _samples*/,
unsigned int _bytes,
unsigned int/* _current_frame*/,
@@ -1152,7 +1154,7 @@ FLAC__StreamDecoderWriteStatus flacStreamDecoderWriteCallback(
static_cast<flacStreamDecoderClientData *>(
_client_data )->write_buffer->write(
(const char *) sframe, sizeof( sframe ) );
}
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
@@ -1220,14 +1222,14 @@ void sampleBuffer::loadFromBase64( const QString & _data )
orig_data = ba_writer.buffer();
printf("%d\n", (int) orig_data.size() );
m_origFrames = orig_data.size() / sizeof( sampleFrame );
m_origFrames = orig_data.size() / sizeof( sampleFrame );
delete[] m_origData;
m_origData = new sampleFrame[m_origFrames];
memcpy( m_origData, orig_data.data(), orig_data.size() );
#else /* LMMS_HAVE_FLAC_STREAM_DECODER_H */
m_origFrames = dsize / sizeof( sampleFrame );
m_origFrames = dsize / sizeof( sampleFrame );
delete[] m_origData;
m_origData = new sampleFrame[m_origFrames];
memcpy( m_origData, dst, dsize );
@@ -1236,7 +1238,7 @@ void sampleBuffer::loadFromBase64( const QString & _data )
delete[] dst;
m_audioFile = "";
m_audioFile = QString();
update();
}
@@ -1357,6 +1359,4 @@ sampleBuffer::handleState::~handleState()
#include "moc_sample_buffer.cxx"
#endif
/* vim: set tw=0 noexpandtab: */