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Diffstat (limited to 'attic/muse2-oom/muse2/muse/audioconvert.cpp')
-rw-r--r--attic/muse2-oom/muse2/muse/audioconvert.cpp886
1 files changed, 0 insertions, 886 deletions
diff --git a/attic/muse2-oom/muse2/muse/audioconvert.cpp b/attic/muse2-oom/muse2/muse/audioconvert.cpp
deleted file mode 100644
index 552b5e95..00000000
--- a/attic/muse2-oom/muse2/muse/audioconvert.cpp
+++ /dev/null
@@ -1,886 +0,0 @@
-//=========================================================
-// MusE
-// Linux Music Editor
-// $Id: audioconvert.cpp,v 1.1.1.1 2009/12/28 16:07:33 terminator356 Exp $
-//
-// (C) Copyright 1999-2009 Werner Schweer (ws@seh.de)
-//
-// Audio converter module created by Tim terminator356
-//=========================================================
-
-#include <math.h>
-
-#include "wave.h"
-#include "globals.h"
-#include "audioconvert.h"
-#include "eventbase.h"
-
-//#define AUDIOCONVERT_DEBUG
-//#define AUDIOCONVERT_DEBUG_PRC
-
-//---------------------------------------------------------
-// AudioConvertMap
-//---------------------------------------------------------
-
-void AudioConvertMap::remapEvents(const EventList* /*el*/)
-{
-
-}
-
-iAudioConvertMap AudioConvertMap::addEvent(EventBase* eb)
-{
- iAudioConvertMap iacm = getConverter(eb);
- if(iacm == end())
- {
- AudioConverter* cv = 0;
- if(!eb->sndFile().isNull())
- cv = new SRCAudioConverter(eb->sndFile().channels(), SRC_SINC_MEDIUM_QUALITY);
-
- // Use insert with hint for speed.
- return insert(iacm, std::pair<EventBase*, AudioConverter*> (eb, cv));
- }
- else
- // Adopt a policy of returning an already existing item to enforce no-duplicates.
- return iacm;
-}
-
-void AudioConvertMap::removeEvent(EventBase* eb)
-{
- iAudioConvertMap iacm = find(eb);
- if(iacm != end())
- {
- AudioConverter* cv = iacm->second;
- if(cv)
- delete cv;
- erase(iacm);
- }
-}
-
-iAudioConvertMap AudioConvertMap::getConverter(EventBase* eb)
-{
- return find(eb);
-}
-
-//---------------------------------------------------------
-// AudioConverter
-//---------------------------------------------------------
-
-AudioConverter::AudioConverter()
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("AudioConverter::AudioConverter this:%p\n", this);
- #endif
-
- _refCount = 1;
- _sfCurFrame = 0;
-}
-
-AudioConverter::~AudioConverter()
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("AudioConverter::~AudioConverter this:%p\n", this);
- #endif
-}
-
-AudioConverter* AudioConverter::reference()
-{
- _refCount += 1;
- #ifdef AUDIOCONVERT_DEBUG
- printf("AudioConverter::reference this:%p current refcount:%d\n", this, _refCount);
- #endif
- return this;
-}
-
-AudioConverter* AudioConverter::release(AudioConverter* cv)
-{
- if(!cv)
- return 0;
- //if(cv->incRefCount(-1) <= 0)
- cv->_refCount -= 1;
- #ifdef AUDIOCONVERT_DEBUG
- printf("AudioConverter::release converter:%p current refcount:%d\n", cv, cv->_refCount);
- #endif
- if(cv->_refCount <= 0)
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("AudioConverter::release deleting converter:%p\n", cv);
- #endif
- delete cv;
- cv = 0;
- }
- return cv;
-}
-
-//off_t AudioConverter::readAudio(SndFileR& f, off_t sfCurFrame, unsigned offset, float** buffer, int channel, int n, bool doSeek, bool overwrite)
-off_t AudioConverter::readAudio(SndFileR& f, unsigned offset, float** buffer, int channel, int n, bool doSeek, bool overwrite)
-{
- if(f.isNull())
- return _sfCurFrame;
-
- // Added by Tim. p3.3.17
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf("AudioConverter::process %s audConv:%p sfCurFrame:%ld offset:%u channel:%d fchan:%d n:%d\n",
- // f.name().toLatin1(), this, sfCurFrame, offset, channel, f.channels(), n);
- //#endif
-
- off_t frame = offset; // _spos is added before the call.
- unsigned fsrate = f.samplerate();
- bool resample = isValid() && ((unsigned)sampleRate != fsrate);
-
- // No resampling needed?
- if(!resample)
- {
- // Sample rates are the same. Just a regular seek + read, no conversion.
- _sfCurFrame = f.seek(frame, 0);
- return _sfCurFrame + f.read(channel, buffer, n, overwrite);
- }
-
- // Is a 'transport' seek requested? (Not to be requested with every read! Should only be for 'first read' seeks, or positional 'transport' seeks.)
- // Due to the support of sound file references in MusE, seek must ALWAYS be done before read, as before,
- // except now we alter the seek position if sample rate conversion is being used and remember the seek positions.
- if(doSeek)
- {
- // Sample rates are different. Seek to a calculated 'sample rate ratio factored' position.
-
- double srcratio = (double)fsrate / (double)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.
-
- _sfCurFrame = f.seek(newfr, 0);
-
- // Added by Tim. p3.3.17
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf("AudioConverter::process Seek frame:%ld converted to frame:%ld sfCurFrame:%ld\n", frame, newfr, sfCurFrame);
- //#endif
-
- // Reset the converter. Its current state is meaningless now.
- reset();
- }
- else
- {
- // No seek requested.
- // Added by Tim. p3.3.17
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf("AudioConverter::process No 'transport' seek, rates different. Seeking to sfCurFrame:%ld\n", sfCurFrame);
- //#endif
-
- // Sample rates are different. We can't just tell seek to go to an absolute calculated position,
- // since the last position can vary - it might not be what the calculated position is.
- // We must use the last position left by SRC conversion, ie. let the file position progress on its own.
- _sfCurFrame = f.seek(_sfCurFrame, 0);
- }
-
- /*
- int fchan = f.channels();
- long outFrames = n;
- long outSize = outFrames * fchan;
- float outbuffer[outSize];
- */
-
- //sfCurFrame = process(f, sfCurFrame, offset, &outbuffer[0], channel, n);
-// sfCurFrame = process(f, sfCurFrame, outbuffer, channel, n);
- //sfCurFrame = process(f, sfCurFrame, buffer, channel, n, overwrite);
- _sfCurFrame = process(f, buffer, channel, n, overwrite);
-
- /*
- float* poutbuf = &outbuffer[0];
- if(fchan == channel)
- {
- if(overwrite)
- //for (size_t i = 0; i < rn; ++i)
- for (int i = 0; i < n; ++i)
- {
- for(int ch = 0; ch < channel; ++ch)
- *(buffer[ch] + i) = *poutbuf++;
- }
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- for(int ch = 0; ch < channel; ++ch)
- *(buffer[ch] + i) += *poutbuf++;
- }
- }
- else if((fchan == 2) && (channel == 1))
- {
- // stereo to mono
- if(overwrite)
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- *(buffer[0] + i) = poutbuf[i + i] + poutbuf[i + i + 1];
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- *(buffer[0] + i) += poutbuf[i + i] + poutbuf[i + i + 1];
- }
- else if((fchan == 1) && (channel == 2))
- {
- // mono to stereo
- if(overwrite)
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- float data = *poutbuf++;
- *(buffer[0]+i) = data;
- *(buffer[1]+i) = data;
- }
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- float data = *poutbuf++;
- *(buffer[0]+i) += data;
- *(buffer[1]+i) += data;
- }
- }
- else
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("AudioConverter::readAudio Channel mismatch: source chans:%d -> dst chans:%d\n", fchan, channel);
- #endif
- }
- */
-
- return _sfCurFrame;
-}
-
-//---------------------------------------------------------
-// SRCAudioConverter
-//---------------------------------------------------------
-
-SRCAudioConverter::SRCAudioConverter(int channels, int type) : AudioConverter()
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::SRCAudioConverter this:%p channels:%d type:%d\n", this, channels, type);
- #endif
-
- _type = type;
- _src_state = 0;
- _channels = channels;
-
- int srcerr;
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::SRCaudioConverter Creating samplerate converter type:%d with %d channels\n", _type, _channels);
- #endif
- _src_state = src_new(_type, _channels, &srcerr);
- if(!_src_state)
- printf("SRCAudioConverter::SRCaudioConverter Creation of samplerate converter type:%d with %d channels failed:%s\n", _type, _channels, src_strerror(srcerr));
-}
-
-SRCAudioConverter::~SRCAudioConverter()
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::~SRCAudioConverter this:%p\n", this);
- #endif
- if(_src_state)
- src_delete(_src_state);
-}
-
-void SRCAudioConverter::setChannels(int ch)
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::setChannels this:%p channels:%d\n", this, ch);
- #endif
- if(_src_state)
- src_delete(_src_state);
- _src_state = 0;
-
- _channels = ch;
- int srcerr;
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::setChannels Creating samplerate converter type:%d with %d channels\n", _type, ch);
- #endif
- _src_state = src_new(_type, ch, &srcerr);
- if(!_src_state)
- printf("SRCAudioConverter::setChannels of samplerate converter type:%d with %d channels failed:%s\n", _type, ch, src_strerror(srcerr));
- return;
-}
-
-void SRCAudioConverter::reset()
-{
- if(!_src_state)
- return;
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::reset this:%p\n", this);
- #endif
- int srcerr = src_reset(_src_state);
- if(srcerr != 0)
- printf("SRCAudioConverter::reset Converter reset failed: %s\n", src_strerror(srcerr));
- return;
-}
-
-//off_t SRCAudioConverter::process(SndFileR& f, off_t sfCurFrame, float** buffer, int channel, int n, bool overwrite)
-off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n, bool overwrite)
-{
- //return src_process(_src_state, sd);
-
- if(f.isNull())
- //return;
- return _sfCurFrame;
-
- // Added by Tim. p3.3.17
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf("AudioConverter::process %s audConv:%p sfCurFrame:%ld offset:%u channel:%d fchan:%d n:%d\n",
- // f.name().toLatin1(), this, sfCurFrame, offset, channel, f.channels(), n);
- //#endif
-
-// 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);
-
- if((sampleRate == 0) || (fsrate == 0))
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::process Error: sampleRate or file samplerate is zero!\n");
- #endif
- return _sfCurFrame;
- }
-
- SRC_DATA srcdata;
- int fchan = f.channels();
- // Ratio is defined as output sample rate over input samplerate.
- double srcratio = (double)sampleRate / (double)fsrate;
- // Extra input compensation.
- long inComp = 1;
-
- long outFrames = n;
- //long outSize = outFrames * channel;
- long outSize = outFrames * fchan;
-
- //long inSize = long(outSize * srcratio) + 1 // From MusE-2 file converter.
- //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 inSize = inFrames * fchan;
- //long inSize = inFrames * channel;
-
- // Start with buffers at expected sizes. We won't need anything larger than this, but add 4 for good luck.
- float inbuffer[inSize + 4];
- float outbuffer[outSize];
-
- //size_t sfTotalRead = 0;
- size_t rn = 0;
- long totalOutFrames = 0;
-
- srcdata.data_in = inbuffer;
- srcdata.data_out = outbuffer;
-// srcdata.data_out = buffer;
-
- // Set some kind of limit on the number of attempts to completely fill the output buffer,
- // in case something is really screwed up - we don't want to get stuck in a loop here.
- int attempts = 10;
- for(int attempt = 0; attempt < attempts; ++attempt)
- {
- rn = f.readDirect(inbuffer, inFrames);
- //sfTotalRead += rn;
-
- // convert
- //srcdata.data_in = inbuffer;
- //srcdata.data_out = outbuffer;
- //srcdata.data_out = poutbuf;
- //srcdata.input_frames = inSize;
- srcdata.input_frames = rn;
- srcdata.output_frames = outFrames;
- srcdata.end_of_input = ((long)rn != inFrames);
- srcdata.src_ratio = srcratio;
-
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf("AudioConverter::process attempt:%d inFrames:%ld outFrames:%ld rn:%d data in:%p out:%p",
- // attempt, inFrames, outFrames, rn, srcdata.data_in, srcdata.data_out);
- //#endif
-
- int srcerr = src_process(_src_state, &srcdata);
- if(srcerr != 0)
- {
- printf("\nSRCAudioConverter::process SampleRate converter process failed: %s\n", src_strerror(srcerr));
- return _sfCurFrame += rn;
- }
-
- totalOutFrames += srcdata.output_frames_gen;
-
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf(" frames used in:%ld out:%ld totalOutFrames:%ld data in:%p out:%p\n", srcdata.input_frames_used, srcdata.output_frames_gen, totalOutFrames, srcdata.data_in, srcdata.data_out);
- //#endif
-
- #ifdef AUDIOCONVERT_DEBUG
- if(srcdata.output_frames_gen != outFrames)
- printf("SRCAudioConverter::process %s output_frames_gen:%ld != outFrames:%ld inFrames:%ld srcdata.input_frames_used:%ld rn:%d\n",
- f.name().toLatin1(), srcdata.output_frames_gen, outFrames, inFrames, srcdata.input_frames_used, rn);
- #endif
-
- // If the number of frames read by the soundfile equals the input frames, go back.
- // Otherwise we have reached the end of the file, so going back is useless since
- // there shouldn't be any further calls.
- if((long)rn == inFrames)
- {
- // Go back by the amount of unused frames.
- sf_count_t seekn = inFrames - srcdata.input_frames_used;
- if(seekn != 0)
- {
- #ifdef AUDIOCONVERT_DEBUG_PRC
- printf("SRCAudioConverter::process Seek-back by:%d\n", seekn);
- #endif
- _sfCurFrame = f.seek(-seekn, SEEK_CUR);
- }
- else
- _sfCurFrame += rn;
-
- if(totalOutFrames == n)
- {
- // We got our desired number of output frames. Stop attempting.
- break;
- }
- else
- {
- // No point in continuing if on last attempt.
- if(attempt == (attempts - 1))
- break;
-
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::process %s attempt:%d totalOutFrames:%ld != n:%d try again\n", f.name().toLatin1(), attempt, totalOutFrames, n);
- #endif
-
- // SRC didn't give us the number of frames we requested.
- // This can occasionally be radically different from the requested frames, or zero,
- // even when ample excess input frames are supplied.
- // Move the src output pointer to a new position.
- srcdata.data_out += srcdata.output_frames_gen * channel;
- // Set new number of maximum out frames.
- outFrames -= srcdata.output_frames_gen;
- // Calculate the new number of file input frames required.
- inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp;
- // Keep trying.
- continue;
- }
- }
- else
- {
- _sfCurFrame += rn;
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::process %s rn:%zd != inFrames:%ld output_frames_gen:%ld outFrames:%ld srcdata.input_frames_used:%ld\n",
- f.name().toLatin1(), rn, inFrames, srcdata.output_frames_gen, outFrames, srcdata.input_frames_used);
- #endif
-
- // We've reached the end of the file. Convert the number of frames read.
- //rn = (double)rn * srcratio + 1;
- //rn = (long)floor((double)rn * srcratio);
- //if(rn > (size_t)outFrames)
- // rn = outFrames;
- // Stop attempting.
- break;
- }
- }
-
- // If we still didn't get the desired number of output frames.
- if(totalOutFrames != n)
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::process %s totalOutFrames:%ld != n:%d\n", f.name().toLatin1(), totalOutFrames, n);
- #endif
-
- // Let's zero the rest of it.
- long b = totalOutFrames * channel;
- long e = n * channel;
- for(long i = b; i < e; ++i)
- outbuffer[i] = 0.0f;
- //buffer[i] = 0.0f;
- }
-
- //float* poutbuf = &outbuffer[0];
- float* poutbuf = outbuffer;
- if(fchan == channel)
- {
- if(overwrite)
- //for (size_t i = 0; i < rn; ++i)
- for (int i = 0; i < n; ++i)
- {
- for(int ch = 0; ch < channel; ++ch)
- *(buffer[ch] + i) = *poutbuf++;
- }
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- for(int ch = 0; ch < channel; ++ch)
- *(buffer[ch] + i) += *poutbuf++;
- }
- }
- else if((fchan == 2) && (channel == 1))
- {
- // stereo to mono
- if(overwrite)
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- *(buffer[0] + i) = poutbuf[i + i] + poutbuf[i + i + 1];
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- *(buffer[0] + i) += poutbuf[i + i] + poutbuf[i + i + 1];
- }
- else if((fchan == 1) && (channel == 2))
- {
- // mono to stereo
- if(overwrite)
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- float data = *poutbuf++;
- *(buffer[0]+i) = data;
- *(buffer[1]+i) = data;
- }
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- float data = *poutbuf++;
- *(buffer[0]+i) += data;
- *(buffer[1]+i) += data;
- }
- }
- else
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("SRCAudioConverter::process Channel mismatch: source chans:%d -> dst chans:%d\n", fchan, channel);
- #endif
- }
-
- return _sfCurFrame;
-}
-
-#ifdef RUBBERBAND_SUPPORT
-
-//---------------------------------------------------------
-// RubberBandAudioConverter
-//---------------------------------------------------------
-
-RubberBandAudioConverter::RubberBandAudioConverter(int channels, int options) : AudioConverter()
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::RubberBandAudioConverter this:%p channels:%d options:%x\n", this, channels, options);
- #endif
-
- _options = options;
- _rbs = 0;
- _channels = channels;
-
- _rbs = new RubberBandStretcher(sampleRate, _channels, _options); // , initialTimeRatio = 1.0, initialPitchScale = 1.0
-}
-
-RubberBandAudioConverter::~RubberBandAudioConverter()
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::~RubberBandAudioConverter this:%p\n", this);
- #endif
- if(_rbs)
- delete _rbs;
-}
-
-void RubberBandAudioConverter::setChannels(int ch)
-{
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::setChannels this:%p channels:%d\n", this, ch);
- #endif
- if(_rbs)
- delete _rbs;
- _rbs = 0;
-
- _channels = ch;
- _rbs = new RubberBandStretcher(sampleRate, _channels, _options); // , initialTimeRatio = 1.0, initialPitchScale = 1.0
-}
-
-void RubberBandAudioConverter::reset()
-{
- if(!_rbs)
- return;
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::reset this:%p\n", this);
- #endif
- _rbs->reset();
- return;
-}
-
-/////////////////////////////////
-// TODO: Not finished yet..
-////////////////////////////////
-//off_t RubberBandAudioConverter::process(SndFileR& f, off_t sfCurFrame, float** buffer, int channel, int n, bool overwrite)
-off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel, int n, bool overwrite)
-{
- //return src_process(_src_state, sd);
-
- if(f.isNull())
- //return;
- return _sfCurFrame;
-
- // Added by Tim. p3.3.17
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf("AudioConverter::process %s audConv:%p sfCurFrame:%ld offset:%u channel:%d fchan:%d n:%d\n",
- // f.name().toLatin1(), this, sfCurFrame, offset, channel, f.channels(), n);
- //#endif
-
-// 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);
-
- if((sampleRate == 0) || (fsrate == 0))
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::process Error: sampleRate or file samplerate is zero!\n");
- #endif
- return _sfCurFrame;
- }
-
-// SRC_DATA srcdata;
- int fchan = f.channels();
- // Ratio is defined as output sample rate over input samplerate.
- double srcratio = (double)sampleRate / (double)fsrate;
- // Extra input compensation.
- long inComp = 1;
-
- long outFrames = n;
- //long outSize = outFrames * channel;
- long outSize = outFrames * fchan;
-
- //long inSize = long(outSize * srcratio) + 1 // From MusE-2 file converter.
- //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 inSize = inFrames * fchan;
- //long inSize = inFrames * channel;
-
- // Start with buffers at expected sizes. We won't need anything larger than this, but add 4 for good luck.
- float inbuffer[inSize]; // +4
-// float outbuffer[outSize];
-
- //float* rbinbuffer[fchan];
- //float rbindata[inSize];
- //for (int i = 0; i < fchan; ++i)
- // rbinbuffer[i] = rbindata + i * inFrames;
-
- float* rboutbuffer[fchan];
- float rboutdata[outSize];
- for (int i = 0; i < fchan; ++i)
- rboutbuffer[i] = rboutdata + i * outFrames;
-
- //size_t sfTotalRead = 0;
- size_t rn = 0;
- long totalOutFrames = 0;
-
-// srcdata.data_in = inbuffer;
- //srcdata.data_out = outbuffer;
-// srcdata.data_out = buffer;
- float** data_out = rboutbuffer;
-
- // For just sample rate conversion, apply same ratio to both time and pitch.
- _rbs->setTimeRatio(srcratio);
- _rbs->setPitchScale(srcratio);
-
- // Set some kind of limit on the number of attempts to completely fill the output buffer,
- // in case something is really screwed up - we don't want to get stuck in a loop here.
- int attempts = 10;
- for(int attempt = 0; attempt < attempts; ++attempt)
- {
- size_t sreq = _rbs->getSamplesRequired();
-
- size_t rbinSize = sreq * fchan;
- float* rbinbuffer[fchan];
- float rbindata[rbinSize];
- for(int i = 0; i < fchan; ++i)
- rbinbuffer[i] = rbindata + i * sreq;
-
-// rn = f.readDirect(inbuffer, inFrames);
- rn = f.readDirect(inbuffer, sreq);
- //sfTotalRead += rn;
-
- // Must de-interleave soundfile data to feed to rubberband.
- for(size_t i = 0; i < rn; ++i)
- {
- for(int ch = 0; ch < fchan; ++ch)
- *(rbinbuffer[ch] + i) = *inbuffer++;
- }
-
- _rbs->process(rbinbuffer, rn, (long)rn != inFrames);
-
- // "This function returns -1 if all data has been fully processed and all output read, and the stretch process is now finished."
- int savail = _rbs->available();
-
-
- // convert
- //srcdata.data_in = inbuffer;
- //srcdata.data_out = outbuffer;
- //srcdata.data_out = poutbuf;
- //srcdata.input_frames = inSize;
- srcdata.input_frames = rn;
- srcdata.output_frames = outFrames;
- srcdata.end_of_input = ((long)rn != inFrames);
- srcdata.src_ratio = srcratio;
-
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf("AudioConverter::process attempt:%d inFrames:%ld outFrames:%ld rn:%d data in:%p out:%p",
- // attempt, inFrames, outFrames, rn, srcdata.data_in, srcdata.data_out);
- //#endif
-
- int srcerr = src_process(_src_state, &srcdata);
- if(srcerr != 0)
- {
- printf("\RubberBandAudioConverter::process SampleRate converter process failed: %s\n", src_strerror(srcerr));
- return _sfCurFrame += rn;
- }
-
- totalOutFrames += srcdata.output_frames_gen;
-
- //#ifdef AUDIOCONVERT_DEBUG_PRC
- //printf(" frames used in:%ld out:%ld totalOutFrames:%ld data in:%p out:%p\n", srcdata.input_frames_used, srcdata.output_frames_gen, totalOutFrames, srcdata.data_in, srcdata.data_out);
- //#endif
-
- #ifdef AUDIOCONVERT_DEBUG
- if(srcdata.output_frames_gen != outFrames)
- printf("RubberBandAudioConverter::process %s output_frames_gen:%ld != outFrames:%ld inFrames:%ld srcdata.input_frames_used:%ld rn:%d\n",
- f.name().toLatin1(), srcdata.output_frames_gen, outFrames, inFrames, srcdata.input_frames_used, rn);
- #endif
-
- // If the number of frames read by the soundfile equals the input frames, go back.
- // Otherwise we have reached the end of the file, so going back is useless since
- // there shouldn't be any further calls.
- if((long)rn == inFrames)
- {
- // Go back by the amount of unused frames.
- sf_count_t seekn = inFrames - srcdata.input_frames_used;
- if(seekn != 0)
- {
- #ifdef AUDIOCONVERT_DEBUG_PRC
- printf("RubberBandAudioConverter::process Seek-back by:%d\n", seekn);
- #endif
- _sfCurFrame = f.seek(-seekn, SEEK_CUR);
- }
- else
- _sfCurFrame += rn;
-
- if(totalOutFrames == n)
- {
- // We got our desired number of output frames. Stop attempting.
- break;
- }
- else
- {
- // No point in continuing if on last attempt.
- if(attempt == (attempts - 1))
- break;
-
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::process %s attempt:%d totalOutFrames:%ld != n:%d try again\n", f.name().toLatin1(), attempt, totalOutFrames, n);
- #endif
-
- // We didn't get the number of frames we requested.
- // This can occasionally be radically different from the requested frames, or zero,
- // even when ample excess input frames are supplied.
- // Move the src output pointer to a new position.
- srcdata.data_out += srcdata.output_frames_gen * channel;
- // Set new number of maximum out frames.
- outFrames -= srcdata.output_frames_gen;
- // Calculate the new number of file input frames required.
- inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp;
- // Keep trying.
- continue;
- }
- }
- else
- {
- _sfCurFrame += rn;
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::process %s rn:%zd != inFrames:%ld output_frames_gen:%ld outFrames:%ld srcdata.input_frames_used:%ld\n",
- f.name().toLatin1(), rn, inFrames, srcdata.output_frames_gen, outFrames, srcdata.input_frames_used);
- #endif
-
- // We've reached the end of the file. Convert the number of frames read.
- //rn = (double)rn * srcratio + 1;
- //rn = (long)floor((double)rn * srcratio);
- //if(rn > (size_t)outFrames)
- // rn = outFrames;
- // Stop attempting.
- break;
- }
- }
-
- // If we still didn't get the desired number of output frames.
- if(totalOutFrames != n)
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::process %s totalOutFrames:%ld != n:%d\n", f.name().toLatin1(), totalOutFrames, n);
- #endif
-
- // Let's zero the rest of it.
- long b = totalOutFrames * channel;
- long e = n * channel;
- for(long i = b; i < e; ++i)
- //outbuffer[i] = 0.0f;
- buffer[i] = 0.0f;
- }
-
- //float* poutbuf = &outbuffer[0];
- float* poutbuf = outbuffer;
- if(fchan == channel)
- {
- if(overwrite)
- //for (size_t i = 0; i < rn; ++i)
- for (int i = 0; i < n; ++i)
- {
- for(int ch = 0; ch < channel; ++ch)
- *(buffer[ch] + i) = *poutbuf++;
- }
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- for(int ch = 0; ch < channel; ++ch)
- *(buffer[ch] + i) += *poutbuf++;
- }
- }
- else if((fchan == 2) && (channel == 1))
- {
- // stereo to mono
- if(overwrite)
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- *(buffer[0] + i) = poutbuf[i + i] + poutbuf[i + i + 1];
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- *(buffer[0] + i) += poutbuf[i + i] + poutbuf[i + i + 1];
- }
- else if((fchan == 1) && (channel == 2))
- {
- // mono to stereo
- if(overwrite)
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- float data = *poutbuf++;
- *(buffer[0]+i) = data;
- *(buffer[1]+i) = data;
- }
- else
- //for(size_t i = 0; i < rn; ++i)
- for(int i = 0; i < n; ++i)
- {
- float data = *poutbuf++;
- *(buffer[0]+i) += data;
- *(buffer[1]+i) += data;
- }
- }
- else
- {
- #ifdef AUDIOCONVERT_DEBUG
- printf("RubberBandAudioConverter::process Channel mismatch: source chans:%d -> dst chans:%d\n", fchan, channel);
- #endif
- }
-
- return _sfCurFrame;
-}
-
-#endif // RUBBERBAND_SUPPORT
generated by cgit v1.2.3 (git 2.39.1) at 2024-09-06 22:07:51 +0000