here is the "NOT working" code :
CODE:
void diywt::writeWavFile(const std::string& filename) {realSize = m_maxWrite;sdkTraceInt(realSize);sdkTraceInt(m_lastMarkerNumber);// Structures pour les en-têtesstruct RiffHeader {char chunkId[4] = { 'R', 'I', 'F', 'F' };uint32_t chunkSize;char format[4] = { 'W', 'A', 'V', 'E' };};struct FmtChunk {char chunkId[4] = { 'f', 'm', 't', ' ' };uint32_t chunkSize = 16;uint16_t audioFormat = 3; // 32-bit floatuint16_t numChannels;uint32_t sampleRate;uint32_t byteRate;uint16_t blockAlign;uint16_t bitsPerSample;};struct DataChunk {char chunkId[4] = { 'd', 'a', 't', 'a' };uint32_t chunkSize;};std::ofstream outFile(filename, std::ios::binary);if (!outFile) {sdkTraceErrorChar("can't open file");return;}// Paramètres audioint numChannels = 1;int bitsPerSample = 32;int numSamples = m_markersStart[m_lastMarkerNumber];//realSize;// Calculer la taille des données audiouint32_t dataSize = numSamples * numChannels * bitsPerSample / 8;// Remplir les en-têtesRiffHeader riffHeader;FmtChunk fmtChunk;fmtChunk.numChannels = numChannels;fmtChunk.sampleRate = sdkGetSampleRate();fmtChunk.byteRate = fmtChunk.sampleRate * numChannels * bitsPerSample / 8;fmtChunk.blockAlign = numChannels * bitsPerSample / 8;fmtChunk.bitsPerSample = bitsPerSample;DataChunk dataChunk;dataChunk.chunkSize = dataSize;// Calculer la taille totale du fichieruint32_t totalFileSize = sizeof(RiffHeader) + sizeof(FmtChunk) + sizeof(DataChunk) + dataSize;// +sizeof(CueChunk) + cueChunkSize + sizeof(AdtlChunk) + adtlChunkSize;riffHeader.chunkSize = totalFileSize - 8; // Soustraire les 8 octets du chunkId et du chunkSize// Écrire les en-têtes dans le fichieroutFile.write(reinterpret_cast<char*>(&riffHeader), sizeof(RiffHeader));outFile.write(reinterpret_cast<char*>(&fmtChunk), sizeof(FmtChunk));outFile.write(reinterpret_cast<char*>(&dataChunk), sizeof(DataChunk));// Écrire les données audiofor (f = 0; f < numSamples; ++f) {outFile.write(reinterpret_cast<const char*>(&m_audioBuffer[f]), sizeof(float));}// Ajouter un padding si nécessaire pour aligner le chunk suivant sur un nombre pair d'octetsif (dataSize % 2 != 0) {outFile.put(0);}outFile.close();/*//mettre à jour la sortie markerOutint tempSize = m_lastMarkerNumber;markersOut.setSize(tempSize);for (int m = 0; m < tempSize; m++) {markersOut.setArrayData(m, float(m_markersStart[m])/float(numSamples));}*/if (!outFile) {sdkTraceErrorChar("Error closing file");return;}std::string s = fdrFileName.getPChar();s = s + " successfully created";sdkTraceChar(s.c_str());m_signalisation = 2;}Statistics: Posted by oli_lab — 21 Apr 2026, 19:41
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But I think Usine should have read and write sample modules that support wav and aif at least and the metadata for looping
It could be written with the SDK ...
Statistics: Posted by sm_jamieson — 21 Apr 2026, 19:06
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But I did an addon called "player with saved markers" that can save and retrieved markers.
It is not possible to do loops at sample level but.
for this king of thing, I rather use REAPER and make regionss to be render or a specialized sample editor.
Otherwise, I pretty much like the idea of using Usine as a modular studio to produce processed samples (just like with an old school hardware studio)
Statistics: Posted by oli_lab — 21 Apr 2026, 17:29
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Can it or a patch do the following:
1. Set and play a loop
2. Add loop start and end cue points to metadata
3. Snap cue to zero crossing point
4. Modify sample to cross fade across loop points.
5. Save the sample in the same or different format with loop cue points in the metadata that can be used by another sampler ?
Thanks.
Statistics: Posted by sm_jamieson — 21 Apr 2026, 10:46
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