Chris@148: /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
Chris@148: 
Chris@148: /*
Chris@148:     Sonic Visualiser
Chris@148:     An audio file viewer and annotation editor.
Chris@148:     Centre for Digital Music, Queen Mary, University of London.
Chris@297:     This file copyright 2006-2007 Chris Cannam and QMUL.
Chris@148:     
Chris@148:     This program is free software; you can redistribute it and/or
Chris@148:     modify it under the terms of the GNU General Public License as
Chris@148:     published by the Free Software Foundation; either version 2 of the
Chris@148:     License, or (at your option) any later version.  See the file
Chris@148:     COPYING included with this distribution for more information.
Chris@148: */
Chris@148: 
Chris@148: #include "CodedAudioFileReader.h"
Chris@148: 
Chris@148: #include "WavFileReader.h"
Chris@148: #include "base/TempDirectory.h"
Chris@148: #include "base/Exceptions.h"
Chris@192: #include "base/Profiler.h"
Chris@297: #include "base/Serialiser.h"
Chris@1098: #include "base/StorageAdviser.h"
Chris@148: 
Chris@1318: #include <bqresample/Resampler.h>
Chris@1318: 
Chris@723: #include <stdint.h>
Chris@148: #include <iostream>
Chris@148: #include <QDir>
Chris@263: #include <QMutexLocker>
Chris@148: 
Chris@1096: using namespace std;
Chris@1096: 
Chris@297: CodedAudioFileReader::CodedAudioFileReader(CacheMode cacheMode,
Chris@1040:                                            sv_samplerate_t targetRate,
Chris@920:                                            bool normalised) :
Chris@148:     m_cacheMode(cacheMode),
Chris@148:     m_initialised(false),
Chris@297:     m_serialiser(0),
Chris@297:     m_fileRate(0),
Chris@148:     m_cacheFileWritePtr(0),
Chris@148:     m_cacheFileReader(0),
Chris@148:     m_cacheWriteBuffer(0),
Chris@148:     m_cacheWriteBufferIndex(0),
Chris@1320:     m_cacheWriteBufferFrames(65536),
Chris@297:     m_resampler(0),
Chris@757:     m_resampleBuffer(0),
Chris@1320:     m_resampleBufferFrames(0),
Chris@920:     m_fileFrameCount(0),
Chris@920:     m_normalised(normalised),
Chris@920:     m_max(0.f),
Chris@1285:     m_gain(1.f),
Chris@1305:     m_trimFromStart(0),
Chris@1305:     m_trimFromEnd(0),
Chris@1285:     m_clippedCount(0),
Chris@1286:     m_firstNonzero(0),
Chris@1286:     m_lastNonzero(0)
Chris@148: {
Chris@1279:     SVDEBUG << "CodedAudioFileReader:: cache mode: " << cacheMode
Chris@1279:             << " (" << (cacheMode == CacheInTemporaryFile
Chris@1279:                         ? "CacheInTemporaryFile" : "CacheInMemory") << ")"
Chris@1279:             << ", rate: " << targetRate
Chris@1279:             << (targetRate == 0 ? " (use source rate)" : "")
Chris@1279:             << ", normalised: " << normalised << endl;
Chris@297: 
Chris@297:     m_frameCount = 0;
Chris@297:     m_sampleRate = targetRate;
Chris@148: }
Chris@148: 
Chris@148: CodedAudioFileReader::~CodedAudioFileReader()
Chris@148: {
Chris@263:     QMutexLocker locker(&m_cacheMutex);
Chris@263: 
Chris@1279:     if (m_serialiser) endSerialised();
Chris@1098:     
Chris@148:     if (m_cacheFileWritePtr) sf_close(m_cacheFileWritePtr);
Chris@297: 
Chris@742:     SVDEBUG << "CodedAudioFileReader::~CodedAudioFileReader: deleting cache file reader" << endl;
Chris@532: 
Chris@297:     delete m_cacheFileReader;
Chris@297:     delete[] m_cacheWriteBuffer;
Chris@1279:     
Chris@148:     if (m_cacheFileName != "") {
Chris@1279:         SVDEBUG << "CodedAudioFileReader::~CodedAudioFileReader: deleting cache file " << m_cacheFileName << endl;
Chris@290:         if (!QFile(m_cacheFileName).remove()) {
Chris@1279:             SVDEBUG << "WARNING: CodedAudioFileReader::~CodedAudioFileReader: Failed to delete cache file \"" << m_cacheFileName << "\"" << endl;
Chris@148:         }
Chris@148:     }
Chris@297: 
Chris@297:     delete m_resampler;
Chris@297:     delete[] m_resampleBuffer;
Chris@1098: 
Chris@1098:     if (!m_data.empty()) {
Chris@1098:         StorageAdviser::notifyDoneAllocation
Chris@1098:             (StorageAdviser::MemoryAllocation,
Chris@1098:              (m_data.size() * sizeof(float)) / 1024);
Chris@1098:     }
Chris@297: }
Chris@297: 
Chris@297: void
Chris@1307: CodedAudioFileReader::setFramesToTrim(sv_frame_t fromStart, sv_frame_t fromEnd)
Chris@1305: {
Chris@1305:     m_trimFromStart = fromStart;
Chris@1305:     m_trimFromEnd = fromEnd;
Chris@1305: }
Chris@1305: 
Chris@1305: void
Chris@297: CodedAudioFileReader::startSerialised(QString id)
Chris@297: {
Chris@1279:     SVDEBUG << "CodedAudioFileReader(" << this << ")::startSerialised: id = " << id << endl;
Chris@297: 
Chris@297:     delete m_serialiser;
Chris@297:     m_serialiser = new Serialiser(id);
Chris@297: }
Chris@297: 
Chris@297: void
Chris@297: CodedAudioFileReader::endSerialised()
Chris@297: {
Chris@844:     SVDEBUG << "CodedAudioFileReader(" << this << ")::endSerialised: id = " << (m_serialiser ? m_serialiser->getId() : "(none)") << endl;
Chris@297: 
Chris@297:     delete m_serialiser;
Chris@297:     m_serialiser = 0;
Chris@148: }
Chris@148: 
Chris@148: void
Chris@148: CodedAudioFileReader::initialiseDecodeCache()
Chris@148: {
Chris@263:     QMutexLocker locker(&m_cacheMutex);
Chris@263: 
Chris@742:     SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: file rate = " << m_fileRate << endl;
Chris@297: 
Chris@1307:     if (m_channelCount == 0) {
Chris@1307:         SVCERR << "CodedAudioFileReader::initialiseDecodeCache: No channel count set!" << endl;
Chris@1307:         throw std::logic_error("No channel count set");
Chris@1307:     }
Chris@1307:     
Chris@297:     if (m_fileRate == 0) {
Chris@1279:         SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: ERROR: File sample rate unknown (bug in subclass implementation?)" << endl;
Chris@1354:         throw FileOperationFailed("(coded file)", "sample rate unknown (bug in subclass implementation?)");
Chris@297:     }
Chris@297:     if (m_sampleRate == 0) {
Chris@297:         m_sampleRate = m_fileRate;
Chris@690:         SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: rate (from file) = " << m_fileRate << endl;
Chris@297:     }
Chris@297:     if (m_fileRate != m_sampleRate) {
Chris@757:         SVDEBUG << "CodedAudioFileReader: resampling " << m_fileRate << " -> " <<  m_sampleRate << endl;
Chris@1329: 
Chris@1329:         breakfastquay::Resampler::Parameters params;
Chris@1329:         params.quality = breakfastquay::Resampler::FastestTolerable;
Chris@1329:         params.maxBufferSize = int(m_cacheWriteBufferFrames);
Chris@1329:         params.initialSampleRate = m_fileRate;
Chris@1329:         m_resampler = new breakfastquay::Resampler(params, m_channelCount);
Chris@1329: 
Chris@1040:         double ratio = m_sampleRate / m_fileRate;
Chris@1320:         m_resampleBufferFrames = int(ceil(double(m_cacheWriteBufferFrames) *
Chris@1320:                                           ratio + 1));
Chris@1320:         m_resampleBuffer = new float[m_resampleBufferFrames * m_channelCount];
Chris@297:     }
Chris@297: 
Chris@1320:     m_cacheWriteBuffer = new float[m_cacheWriteBufferFrames * m_channelCount];
Chris@297:     m_cacheWriteBufferIndex = 0;
Chris@297: 
Chris@148:     if (m_cacheMode == CacheInTemporaryFile) {
Chris@148: 
Chris@148:         try {
Chris@148:             QDir dir(TempDirectory::getInstance()->getPath());
Chris@1359:             m_cacheFileName = dir.filePath(QString("decoded_%1.w64")
Chris@290:                                            .arg((intptr_t)this));
Chris@148: 
Chris@148:             SF_INFO fileInfo;
Chris@1040:             int fileRate = int(round(m_sampleRate));
Chris@1040:             if (m_sampleRate != sv_samplerate_t(fileRate)) {
Chris@1279:                 SVDEBUG << "CodedAudioFileReader: WARNING: Non-integer sample rate "
Chris@1040:                      << m_sampleRate << " presented for writing, rounding to " << fileRate
Chris@1040:                      << endl;
Chris@1040:             }
Chris@1040:             fileInfo.samplerate = fileRate;
Chris@148:             fileInfo.channels = m_channelCount;
Chris@1161: 
Chris@1161:             // Previously we were writing SF_FORMAT_PCM_16 and in a
Chris@1161:             // comment I wrote: "No point in writing 24-bit or float;
Chris@1161:             // generally this class is used for decoding files that
Chris@1161:             // have come from a 16 bit source or that decode to only
Chris@1161:             // 16 bits anyway." That was naive -- we want to preserve
Chris@1161:             // the original values to the same float precision that we
Chris@1161:             // use internally. Saving PCM_16 obviously doesn't
Chris@1161:             // preserve values for sources at bit depths greater than
Chris@1161:             // 16, but it also doesn't always do so for sources at bit
Chris@1161:             // depths less than 16.
Chris@1161:             //
Chris@1161:             // (This came to light with a bug in libsndfile 1.0.26,
Chris@1161:             // which always reports every file as non-seekable, so
Chris@1161:             // that coded readers were being used even for WAV
Chris@1161:             // files. This changed the values that came from PCM_8 WAV
Chris@1161:             // sources, breaking Sonic Annotator's output comparison
Chris@1161:             // tests.)
Chris@1161:             //
Chris@1161:             // So: now we write floats.
Chris@1359:             fileInfo.format = SF_FORMAT_W64 | SF_FORMAT_FLOAT;
Chris@1359: 
Chris@1359: #ifdef Q_OS_WIN
Chris@1359:             m_cacheFileWritePtr = sf_wchar_open
Chris@1361:                 ((LPCWSTR)m_cacheFileName.utf16(), SFM_WRITE, &fileInfo);
Chris@1359: #else
Chris@1359:             m_cacheFileWritePtr = sf_open
Chris@1359:                 (m_cacheFileName.toLocal8Bit(), SFM_WRITE, &fileInfo);
Chris@1359: #endif
Chris@148: 
Chris@265:             if (m_cacheFileWritePtr) {
Chris@265: 
Chris@297:                 // Ideally we would do this now only if we were in a
Chris@297:                 // threaded mode -- creating the reader later if we're
Chris@297:                 // not threaded -- but we don't have access to that
Chris@297:                 // information here
Chris@265: 
Chris@265:                 m_cacheFileReader = new WavFileReader(m_cacheFileName);
Chris@265: 
Chris@265:                 if (!m_cacheFileReader->isOK()) {
Chris@1279:                     SVDEBUG << "ERROR: CodedAudioFileReader::initialiseDecodeCache: Failed to construct WAV file reader for temporary file: " << m_cacheFileReader->getError() << endl;
Chris@265:                     delete m_cacheFileReader;
Chris@265:                     m_cacheFileReader = 0;
Chris@265:                     m_cacheMode = CacheInMemory;
Chris@265:                     sf_close(m_cacheFileWritePtr);
Chris@265:                 }
Chris@297: 
Chris@265:             } else {
Chris@1279:                 SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: failed to open cache file \"" << m_cacheFileName << "\" (" << m_channelCount << " channels, sample rate " << m_sampleRate << " for writing, falling back to in-memory cache" << endl;
Chris@148:                 m_cacheMode = CacheInMemory;
Chris@148:             }
Chris@265: 
Chris@148:         } catch (DirectoryCreationFailed f) {
Chris@1279:             SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: failed to create temporary directory! Falling back to in-memory cache" << endl;
Chris@148:             m_cacheMode = CacheInMemory;
Chris@148:         }
Chris@148:     }
Chris@148: 
Chris@148:     if (m_cacheMode == CacheInMemory) {
Chris@148:         m_data.clear();
Chris@148:     }
Chris@148: 
Chris@1320:     if (m_trimFromEnd >= (m_cacheWriteBufferFrames * m_channelCount)) {
Chris@1320:         SVCERR << "WARNING: CodedAudioFileReader::setSamplesToTrim: Can't handle trimming more frames from end (" << m_trimFromEnd << ") than can be stored in cache-write buffer (" << (m_cacheWriteBufferFrames * m_channelCount) << "), won't trim anything from the end after all";
Chris@1307:         m_trimFromEnd = 0;
Chris@1307:     }
Chris@1307: 
Chris@148:     m_initialised = true;
Chris@148: }
Chris@148: 
Chris@148: void
Chris@1038: CodedAudioFileReader::addSamplesToDecodeCache(float **samples, sv_frame_t nframes)
Chris@148: {
Chris@263:     QMutexLocker locker(&m_cacheMutex);
Chris@263: 
Chris@148:     if (!m_initialised) return;
Chris@148: 
Chris@1038:     for (sv_frame_t i = 0; i < nframes; ++i) {
Chris@1305: 
Chris@1305:         if (m_trimFromStart > 0) {
Chris@1305:             --m_trimFromStart;
Chris@1305:             continue;
Chris@1305:         }
Chris@297:         
Chris@929:         for (int c = 0; c < m_channelCount; ++c) {
Chris@148: 
Chris@297:             float sample = samples[c][i];
Chris@297:             m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;
Chris@148: 
Chris@1306:         }
Chris@297: 
Chris@1306:         pushCacheWriteBufferMaybe(false);
Chris@297:     }
Chris@297: }
Chris@297: 
Chris@297: void
Chris@1038: CodedAudioFileReader::addSamplesToDecodeCache(float *samples, sv_frame_t nframes)
Chris@297: {
Chris@297:     QMutexLocker locker(&m_cacheMutex);
Chris@297: 
Chris@297:     if (!m_initialised) return;
Chris@297: 
Chris@1038:     for (sv_frame_t i = 0; i < nframes; ++i) {
Chris@1305: 
Chris@1305:         if (m_trimFromStart > 0) {
Chris@1305:             --m_trimFromStart;
Chris@1305:             continue;
Chris@1305:         }
Chris@297:         
Chris@929:         for (int c = 0; c < m_channelCount; ++c) {
Chris@297: 
Chris@297:             float sample = samples[i * m_channelCount + c];
Chris@297:         
Chris@297:             m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;
Chris@1306:         }
Chris@297: 
Chris@1306:         pushCacheWriteBufferMaybe(false);
Chris@297:     }
Chris@297: }
Chris@297: 
Chris@297: void
Chris@1326: CodedAudioFileReader::addSamplesToDecodeCache(const floatvec_t &samples)
Chris@297: {
Chris@297:     QMutexLocker locker(&m_cacheMutex);
Chris@297: 
Chris@297:     if (!m_initialised) return;
Chris@297: 
Chris@1038:     for (float sample: samples) {
Chris@1305: 
Chris@1305:         if (m_trimFromStart > 0) {
Chris@1305:             --m_trimFromStart;
Chris@1305:             continue;
Chris@1305:         }
Chris@297:         
Chris@148:         m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;
Chris@148: 
Chris@1306:         pushCacheWriteBufferMaybe(false);
Chris@148:     }
Chris@148: }
Chris@148: 
Chris@148: void
Chris@148: CodedAudioFileReader::finishDecodeCache()
Chris@148: {
Chris@263:     QMutexLocker locker(&m_cacheMutex);
Chris@263: 
Chris@1295:     Profiler profiler("CodedAudioFileReader::finishDecodeCache");
Chris@192: 
Chris@148:     if (!m_initialised) {
Chris@1279:         SVDEBUG << "WARNING: CodedAudioFileReader::finishDecodeCache: Cache was never initialised!" << endl;
Chris@148:         return;
Chris@148:     }
Chris@148: 
Chris@1306:     pushCacheWriteBufferMaybe(true);
Chris@297: 
Chris@297:     delete[] m_cacheWriteBuffer;
Chris@297:     m_cacheWriteBuffer = 0;
Chris@297: 
Chris@297:     delete[] m_resampleBuffer;
Chris@297:     m_resampleBuffer = 0;
Chris@297: 
Chris@297:     delete m_resampler;
Chris@297:     m_resampler = 0;
Chris@297: 
Chris@297:     if (m_cacheMode == CacheInTemporaryFile) {
Chris@1098: 
Chris@297:         sf_close(m_cacheFileWritePtr);
Chris@297:         m_cacheFileWritePtr = 0;
Chris@297:         if (m_cacheFileReader) m_cacheFileReader->updateFrameCount();
Chris@1098: 
Chris@1098:     } else {
Chris@1098:         // I know, I know, we already allocated it...
Chris@1098:         StorageAdviser::notifyPlannedAllocation
Chris@1098:             (StorageAdviser::MemoryAllocation,
Chris@1098:              (m_data.size() * sizeof(float)) / 1024);
Chris@297:     }
Chris@1285: 
Chris@1285:     SVDEBUG << "CodedAudioFileReader: File decodes to " << m_fileFrameCount
Chris@1285:             << " frames" << endl;
Chris@1285:     if (m_fileFrameCount != m_frameCount) {
Chris@1285:         SVDEBUG << "CodedAudioFileReader: Resampled to " << m_frameCount
Chris@1285:                 << " frames" << endl;
Chris@1285:     }
Chris@1285:     SVDEBUG << "CodedAudioFileReader: Signal abs max is " << m_max
Chris@1285:             << ", " << m_clippedCount
Chris@1285:             << " samples clipped, first non-zero frame is at "
Chris@1286:             << m_firstNonzero << ", last at " << m_lastNonzero << endl;
Chris@1285:     if (m_normalised) {
Chris@1285:         SVDEBUG << "CodedAudioFileReader: Normalising, gain is " << m_gain << endl;
Chris@1285:     }
Chris@297: }
Chris@297: 
Chris@297: void
Chris@1306: CodedAudioFileReader::pushCacheWriteBufferMaybe(bool final)
Chris@1306: {
Chris@1306:     if (final ||
Chris@1306:         (m_cacheWriteBufferIndex ==
Chris@1320:          m_cacheWriteBufferFrames * m_channelCount)) {
Chris@1307: 
Chris@1307:         if (m_trimFromEnd > 0) {
Chris@1306:         
Chris@1307:             sv_frame_t framesToPush =
Chris@1307:                 (m_cacheWriteBufferIndex / m_channelCount) - m_trimFromEnd;
Chris@1307: 
Chris@1307:             if (framesToPush <= 0 && !final) {
Chris@1307:                 // This won't do, the buffer is full so we have to push
Chris@1307:                 // something. Should have checked for this earlier
Chris@1307:                 throw std::logic_error("Buffer full but nothing to push");
Chris@1307:             }
Chris@1307: 
Chris@1307:             pushBuffer(m_cacheWriteBuffer, framesToPush, final);
Chris@1307:             
Chris@1307:             m_cacheWriteBufferIndex -= framesToPush * m_channelCount;
Chris@1307: 
Chris@1307:             for (sv_frame_t i = 0; i < m_cacheWriteBufferIndex; ++i) {
Chris@1307:                 m_cacheWriteBuffer[i] =
Chris@1307:                     m_cacheWriteBuffer[framesToPush * m_channelCount + i];
Chris@1307:             }
Chris@1307: 
Chris@1307:         } else {
Chris@1307: 
Chris@1307:             pushBuffer(m_cacheWriteBuffer,
Chris@1307:                        m_cacheWriteBufferIndex / m_channelCount,
Chris@1307:                        final);
Chris@1307: 
Chris@1307:             m_cacheWriteBufferIndex = 0;
Chris@1307:         }
Chris@1306: 
Chris@1306:         if (m_cacheFileReader) {
Chris@1306:             m_cacheFileReader->updateFrameCount();
Chris@1306:         }
Chris@1306:     }
Chris@1306: }
Chris@1306: 
Chris@1306: sv_frame_t
Chris@1038: CodedAudioFileReader::pushBuffer(float *buffer, sv_frame_t sz, bool final)
Chris@297: {
Chris@757:     m_fileFrameCount += sz;
Chris@757: 
Chris@1040:     double ratio = 1.0;
Chris@758:     if (m_resampler && m_fileRate != 0) {
Chris@1040:         ratio = m_sampleRate / m_fileRate;
Chris@758:     }
Chris@758:         
Chris@1040:     if (ratio != 1.0) {
Chris@758:         pushBufferResampling(buffer, sz, ratio, final);
Chris@758:     } else {
Chris@758:         pushBufferNonResampling(buffer, sz);
Chris@758:     }
Chris@1306: 
Chris@1306:     return sz;
Chris@758: }
Chris@757: 
Chris@758: void
Chris@1038: CodedAudioFileReader::pushBufferNonResampling(float *buffer, sv_frame_t sz)
Chris@758: {
Chris@920:     float clip = 1.0;
Chris@1038:     sv_frame_t count = sz * m_channelCount;
Chris@318: 
Chris@1305:     // statistics
Chris@1286:     for (sv_frame_t j = 0; j < sz; ++j) {
Chris@1286:         for (int c = 0; c < m_channelCount; ++c) {
Chris@1286:             sv_frame_t i = j * m_channelCount + c;
Chris@1286:             float v = buffer[i];
Chris@1286:             if (!m_normalised) {
Chris@1286:                 if (v > clip) {
Chris@1286:                     buffer[i] = clip;
Chris@1286:                     ++m_clippedCount;
Chris@1286:                 } else if (v < -clip) {
Chris@1286:                     buffer[i] = -clip;
Chris@1286:                     ++m_clippedCount;
Chris@1286:                 }
Chris@1285:             }
Chris@1286:             v = fabsf(v);
Chris@1286:             if (v != 0.f) {
Chris@1286:                 if (m_firstNonzero == 0) {
Chris@1286:                     m_firstNonzero = m_frameCount;
Chris@1286:                 }
Chris@1286:                 m_lastNonzero = m_frameCount;
Chris@1286:                 if (v > m_max) {
Chris@1286:                     m_max = v;
Chris@1286:                 }
Chris@920:             }
Chris@920:         }
Chris@1286:         ++m_frameCount;
Chris@297:     }
Chris@297: 
Chris@1286:     if (m_max > 0.f) {
Chris@1286:         m_gain = 1.f / m_max; // used when normalising only
Chris@1286:     }
Chris@297: 
Chris@148:     switch (m_cacheMode) {
Chris@148: 
Chris@148:     case CacheInTemporaryFile:
Chris@1038:         if (sf_writef_float(m_cacheFileWritePtr, buffer, sz) < sz) {
Chris@544:             sf_close(m_cacheFileWritePtr);
Chris@544:             m_cacheFileWritePtr = 0;
Chris@544:             throw InsufficientDiscSpace(TempDirectory::getInstance()->getPath());
Chris@544:         }
Chris@148:         break;
Chris@148: 
Chris@148:     case CacheInMemory:
Chris@1100:         m_dataLock.lock();
Chris@1401:         try {
Chris@1401:             m_data.insert(m_data.end(), buffer, buffer + count);
Chris@1401:         } catch (const std::bad_alloc &e) {
Chris@1401:             m_data.clear();
Chris@1403:             SVCERR << "CodedAudioFileReader: Caught bad_alloc when trying to add " << count << " elements to buffer" << endl;
Chris@1401:             m_dataLock.unlock();
Chris@1401:             throw e;
Chris@1401:         }
Chris@543:         m_dataLock.unlock();
Chris@148:         break;
Chris@148:     }
Chris@758: }
Chris@757: 
Chris@758: void
Chris@1038: CodedAudioFileReader::pushBufferResampling(float *buffer, sv_frame_t sz,
Chris@1038:                                            double ratio, bool final)
Chris@758: {
Chris@1306: //    SVDEBUG << "pushBufferResampling: ratio = " << ratio << ", sz = " << sz << ", final = " << final << endl;
Chris@757: 
Chris@759:     if (sz > 0) {
Chris@759: 
Chris@1038:         sv_frame_t out = m_resampler->resampleInterleaved
Chris@1320:             (m_resampleBuffer,
Chris@1320:              m_resampleBufferFrames,
Chris@1320:              buffer,
Chris@1323:              int(sz),
Chris@759:              ratio,
Chris@759:              false);
Chris@759: 
Chris@759:         pushBufferNonResampling(m_resampleBuffer, out);
Chris@759:     }
Chris@757: 
Chris@758:     if (final) {
Chris@758: 
Chris@1038:         sv_frame_t padFrames = 1;
Chris@1038:         if (double(m_frameCount) / ratio < double(m_fileFrameCount)) {
Chris@1038:             padFrames = m_fileFrameCount - sv_frame_t(double(m_frameCount) / ratio) + 1;
Chris@757:         }
Chris@758: 
Chris@1038:         sv_frame_t padSamples = padFrames * m_channelCount;
Chris@758: 
Chris@1307:         SVDEBUG << "CodedAudioFileReader::pushBufferResampling: frameCount = " << m_frameCount << ", equivFileFrames = " << double(m_frameCount) / ratio << ", m_fileFrameCount = " << m_fileFrameCount << ", padFrames = " << padFrames << ", padSamples = " << padSamples << endl;
Chris@758: 
Chris@758:         float *padding = new float[padSamples];
Chris@1038:         for (sv_frame_t i = 0; i < padSamples; ++i) padding[i] = 0.f;
Chris@758: 
Chris@1038:         sv_frame_t out = m_resampler->resampleInterleaved
Chris@1320:             (m_resampleBuffer,
Chris@1320:              m_resampleBufferFrames,
Chris@1320:              padding,
Chris@1323:              int(padFrames),
Chris@758:              ratio,
Chris@758:              true);
Chris@758: 
Chris@1379:         SVDEBUG << "CodedAudioFileReader::pushBufferResampling: resampled padFrames to " << out << " frames" << endl;
Chris@1379: 
Chris@1379:         sv_frame_t expected = sv_frame_t(round(double(m_fileFrameCount) * ratio));
Chris@1379:         if (m_frameCount + out > expected) {
Chris@1379:             out = expected - m_frameCount;
Chris@1379:             SVDEBUG << "CodedAudioFileReader::pushBufferResampling: clipping that to " << out << " to avoid producing more samples than desired" << endl;
Chris@759:         }
Chris@759: 
Chris@758:         pushBufferNonResampling(m_resampleBuffer, out);
Chris@758:         delete[] padding;
Chris@757:     }
Chris@148: }
Chris@148: 
Chris@1326: floatvec_t
Chris@1041: CodedAudioFileReader::getInterleavedFrames(sv_frame_t start, sv_frame_t count) const
Chris@148: {
Chris@543:     // Lock is only required in CacheInMemory mode (the cache file
Chris@543:     // reader is expected to be thread safe and manage its own
Chris@543:     // locking)
Chris@263: 
Chris@265:     if (!m_initialised) {
Chris@690:         SVDEBUG << "CodedAudioFileReader::getInterleavedFrames: not initialised" << endl;
Chris@1096:         return {};
Chris@265:     }
Chris@148: 
Chris@1326:     floatvec_t frames;
Chris@1041:     
Chris@148:     switch (m_cacheMode) {
Chris@148: 
Chris@148:     case CacheInTemporaryFile:
Chris@148:         if (m_cacheFileReader) {
Chris@1041:             frames = m_cacheFileReader->getInterleavedFrames(start, count);
Chris@148:         }
Chris@148:         break;
Chris@148: 
Chris@148:     case CacheInMemory:
Chris@148:     {
Chris@1096:         if (!isOK()) return {};
Chris@1096:         if (count == 0) return {};
Chris@148: 
Chris@1100:         sv_frame_t ix0 = start * m_channelCount;
Chris@1100:         sv_frame_t ix1 = ix0 + (count * m_channelCount);
Chris@148: 
Chris@1100:         // This lock used to be a QReadWriteLock, but it appears that
Chris@1100:         // its lock mechanism is significantly slower than QMutex so
Chris@1100:         // it's not a good idea in cases like this where we don't
Chris@1100:         // really have threads taking a long time to read concurrently
Chris@1100:         m_dataLock.lock();
Chris@1100:         sv_frame_t n = sv_frame_t(m_data.size());
Chris@1282:         if (ix0 > n) ix0 = n;
Chris@1100:         if (ix1 > n) ix1 = n;
Chris@1326:         frames = floatvec_t(m_data.begin() + ix0, m_data.begin() + ix1);
Chris@543:         m_dataLock.unlock();
Chris@1282:         break;
Chris@148:     }
Chris@148:     }
Chris@920: 
Chris@920:     if (m_normalised) {
Chris@1052:         for (auto &f: frames) f *= m_gain;
Chris@920:     }
Chris@1041: 
Chris@1041:     return frames;
Chris@148: }
Chris@148: