diff options
author | Orcan Ogetbil <oget.fedora@gmail.com> | 2011-09-08 02:05:32 +0000 |
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committer | Orcan Ogetbil <oget.fedora@gmail.com> | 2011-09-08 02:05:32 +0000 |
commit | d3e8a1b4c98cb3ba8b73f367ea88ad23f8dbca66 (patch) | |
tree | 921e5193e46287f0c34f4eff1590efb1df18d20f /muse2/muse/audioconvert.cpp | |
parent | ff0c5e9154e7a3d71d2465639b5e0da1ea2c7242 (diff) |
introducing namespaces
Diffstat (limited to 'muse2/muse/audioconvert.cpp')
-rw-r--r-- | muse2/muse/audioconvert.cpp | 32 |
1 files changed, 16 insertions, 16 deletions
diff --git a/muse2/muse/audioconvert.cpp b/muse2/muse/audioconvert.cpp index 95c97883..6b67f30e 100644 --- a/muse2/muse/audioconvert.cpp +++ b/muse2/muse/audioconvert.cpp @@ -142,7 +142,7 @@ off_t AudioConverter::readAudio(SndFileR& f, unsigned offset, float** buffer, in off_t frame = offset; // _spos is added before the call. unsigned fsrate = f.samplerate(); - bool resample = isValid() && ((unsigned)sampleRate != fsrate); + bool resample = isValid() && ((unsigned)MusEGlobal::sampleRate != fsrate); // No resampling needed? if(!resample) @@ -159,7 +159,7 @@ off_t AudioConverter::readAudio(SndFileR& f, unsigned offset, float** buffer, in { // Sample rates are different. Seek to a calculated 'sample rate ratio factored' position. - double srcratio = (double)fsrate / (double)sampleRate; + double srcratio = (double)fsrate / (double)MusEGlobal::sampleRate; //long inSize = long((double)frames * _src_ratio) + 1 // From MusE-2 file converter. off_t newfr = (off_t)floor(((double)frame * srcratio)); // From simplesynth. @@ -343,13 +343,13 @@ off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n // off_t frame = offset; // _spos is added before the call. unsigned fsrate = f.samplerate(); - //bool resample = src_state && ((unsigned)sampleRate != fsrate); -// bool resample = isValid() && ((unsigned)sampleRate != fsrate); + //bool resample = src_state && ((unsigned)MusEGlobal::sampleRate != fsrate); +// bool resample = isValid() && ((unsigned)MusEGlobal::sampleRate != fsrate); - if((sampleRate == 0) || (fsrate == 0)) + if((MusEGlobal::sampleRate == 0) || (fsrate == 0)) { #ifdef AUDIOCONVERT_DEBUG - printf("SRCAudioConverter::process Error: sampleRate or file samplerate is zero!\n"); + printf("SRCAudioConverter::process Error: MusEGlobal::sampleRate or file samplerate is zero!\n"); #endif return _sfCurFrame; } @@ -357,7 +357,7 @@ off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n SRC_DATA srcdata; int fchan = f.channels(); // Ratio is defined as output sample rate over input samplerate. - double srcratio = (double)sampleRate / (double)fsrate; + double srcratio = (double)MusEGlobal::sampleRate / (double)fsrate; // Extra input compensation. long inComp = 1; @@ -369,7 +369,7 @@ off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n //long inSize = (long)floor(((double)outSize / srcratio)); // From simplesynth. //long inFrames = (long)floor(((double)outFrames / srcratio)); // From simplesynth. long inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp; // From simplesynth. - //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(sampleRate)); // From simplesynth. + //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(MusEGlobal::sampleRate)); // From simplesynth. long inSize = inFrames * fchan; //long inSize = inFrames * channel; @@ -583,7 +583,7 @@ RubberBandAudioConverter::RubberBandAudioConverter(int channels, int options) : _rbs = 0; _channels = channels; - _rbs = new RubberBandStretcher(sampleRate, _channels, _options); // , initialTimeRatio = 1.0, initialPitchScale = 1.0 + _rbs = new RubberBandStretcher(MusEGlobal::sampleRate, _channels, _options); // , initialTimeRatio = 1.0, initialPitchScale = 1.0 } RubberBandAudioConverter::~RubberBandAudioConverter() @@ -605,7 +605,7 @@ void RubberBandAudioConverter::setChannels(int ch) _rbs = 0; _channels = ch; - _rbs = new RubberBandStretcher(sampleRate, _channels, _options); // , initialTimeRatio = 1.0, initialPitchScale = 1.0 + _rbs = new RubberBandStretcher(MusEGlobal::sampleRate, _channels, _options); // , initialTimeRatio = 1.0, initialPitchScale = 1.0 } void RubberBandAudioConverter::reset() @@ -639,13 +639,13 @@ off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel // off_t frame = offset; // _spos is added before the call. unsigned fsrate = f.samplerate(); - //bool resample = src_state && ((unsigned)sampleRate != fsrate); -// bool resample = isValid() && ((unsigned)sampleRate != fsrate); + //bool resample = src_state && ((unsigned)MusEGlobal::sampleRate != fsrate); +// bool resample = isValid() && ((unsigned)MusEGlobal::sampleRate != fsrate); - if((sampleRate == 0) || (fsrate == 0)) + if((MusEGlobal::sampleRate == 0) || (fsrate == 0)) { #ifdef AUDIOCONVERT_DEBUG - printf("RubberBandAudioConverter::process Error: sampleRate or file samplerate is zero!\n"); + printf("RubberBandAudioConverter::process Error: MusEGlobal::sampleRate or file samplerate is zero!\n"); #endif return _sfCurFrame; } @@ -653,7 +653,7 @@ off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel // SRC_DATA srcdata; int fchan = f.channels(); // Ratio is defined as output sample rate over input samplerate. - double srcratio = (double)sampleRate / (double)fsrate; + double srcratio = (double)MusEGlobal::sampleRate / (double)fsrate; // Extra input compensation. long inComp = 1; @@ -665,7 +665,7 @@ off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel //long inSize = (long)floor(((double)outSize / srcratio)); // From simplesynth. //long inFrames = (long)floor(((double)outFrames / srcratio)); // From simplesynth. long inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp; // From simplesynth. - //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(sampleRate)); // From simplesynth. + //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(MusEGlobal::sampleRate)); // From simplesynth. long inSize = inFrames * fchan; //long inSize = inFrames * channel; |