annotate data/fileio/CodedAudioFileReader.cpp @ 1881:b504df98c3be

Ensure completion on output model is started at zero, so if it's checked before the input model has become ready and the transform has begun, it is not accidentally reported as complete (affected re-aligning models in Sonic Lineup when replacing the session)

/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */

/*
    Sonic Visualiser
    An audio file viewer and annotation editor.
    Centre for Digital Music, Queen Mary, University of London.
    This file copyright 2006-2007 Chris Cannam and QMUL.
    
    This program is free software; you can redistribute it and/or
    modify it under the terms of the GNU General Public License as
    published by the Free Software Foundation; either version 2 of the
    License, or (at your option) any later version.  See the file
    COPYING included with this distribution for more information.
*/

#include "CodedAudioFileReader.h"

#include "WavFileReader.h"
#include "base/TempDirectory.h"
#include "base/Exceptions.h"
#include "base/Profiler.h"
#include "base/Serialiser.h"
#include "base/StorageAdviser.h"

#include <bqresample/Resampler.h>

#include <stdint.h>
#include <iostream>
#include <QDir>
#include <QMutexLocker>

using namespace std;

CodedAudioFileReader::CodedAudioFileReader(CacheMode cacheMode,
                                           sv_samplerate_t targetRate,
                                           bool normalised) :
    m_cacheMode(cacheMode),
    m_initialised(false),
    m_serialiser(nullptr),
    m_fileRate(0),
    m_cacheFileWritePtr(nullptr),
    m_cacheFileReader(nullptr),
    m_cacheWriteBuffer(nullptr),
    m_cacheWriteBufferIndex(0),
    m_cacheWriteBufferFrames(65536),
    m_resampler(nullptr),
    m_resampleBuffer(nullptr),
    m_resampleBufferFrames(0),
    m_fileFrameCount(0),
    m_normalised(normalised),
    m_max(0.f),
    m_gain(1.f),
    m_trimFromStart(0),
    m_trimFromEnd(0),
    m_clippedCount(0),
    m_firstNonzero(0),
    m_lastNonzero(0)
{
    SVDEBUG << "CodedAudioFileReader:: cache mode: " << cacheMode
            << " (" << (cacheMode == CacheInTemporaryFile
                        ? "CacheInTemporaryFile" : "CacheInMemory") << ")"
            << ", rate: " << targetRate
            << (targetRate == 0 ? " (use source rate)" : "")
            << ", normalised: " << normalised << endl;

    m_frameCount = 0;
    m_sampleRate = targetRate;
}

CodedAudioFileReader::~CodedAudioFileReader()
{
    QMutexLocker locker(&m_cacheMutex);

    if (m_serialiser) endSerialised();
    
    if (m_cacheFileWritePtr) sf_close(m_cacheFileWritePtr);

    SVDEBUG << "CodedAudioFileReader::~CodedAudioFileReader: deleting cache file reader" << endl;

    delete m_cacheFileReader;
    delete[] m_cacheWriteBuffer;
    
    if (m_cacheFileName != "") {
        SVDEBUG << "CodedAudioFileReader::~CodedAudioFileReader: deleting cache file " << m_cacheFileName << endl;
        if (!QFile(m_cacheFileName).remove()) {
            SVDEBUG << "WARNING: CodedAudioFileReader::~CodedAudioFileReader: Failed to delete cache file \"" << m_cacheFileName << "\"" << endl;
        }
    }

    delete m_resampler;
    delete[] m_resampleBuffer;

    if (!m_data.empty()) {
        StorageAdviser::notifyDoneAllocation
            (StorageAdviser::MemoryAllocation,
             (m_data.size() * sizeof(float)) / 1024);
    }
}

void
CodedAudioFileReader::setFramesToTrim(sv_frame_t fromStart, sv_frame_t fromEnd)
{
    m_trimFromStart = fromStart;
    m_trimFromEnd = fromEnd;
}

void
CodedAudioFileReader::startSerialised(QString id,
                                      const std::atomic<bool> *cancelled)
{
//    SVCERR << "CodedAudioFileReader(" << this << ")::startSerialised: id = " << id << endl;
    
    delete m_serialiser;
    m_serialiser = new Serialiser(id, cancelled);
}

void
CodedAudioFileReader::endSerialised()
{
//    SVCERR << "CodedAudioFileReader(" << this << ")::endSerialised: id = " << (m_serialiser ? m_serialiser->getId() : "(none)") << endl;

    delete m_serialiser;
    m_serialiser = nullptr;
}

void
CodedAudioFileReader::initialiseDecodeCache()
{
    QMutexLocker locker(&m_cacheMutex);

    SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: file rate = " << m_fileRate << endl;

    if (m_channelCount == 0) {
        SVCERR << "CodedAudioFileReader::initialiseDecodeCache: No channel count set!" << endl;
        throw std::logic_error("No channel count set");
    }
    
    if (m_fileRate == 0) {
        SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: ERROR: File sample rate unknown (bug in subclass implementation?)" << endl;
        throw FileOperationFailed("(coded file)", "sample rate unknown (bug in subclass implementation?)");
    }
    if (m_sampleRate == 0) {
        m_sampleRate = m_fileRate;
        SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: rate (from file) = " << m_fileRate << endl;
    }
    if (m_fileRate != m_sampleRate) {
        SVDEBUG << "CodedAudioFileReader: resampling " << m_fileRate << " -> " <<  m_sampleRate << endl;

        breakfastquay::Resampler::Parameters params;
        params.quality = breakfastquay::Resampler::FastestTolerable;
        params.maxBufferSize = int(m_cacheWriteBufferFrames);
        params.initialSampleRate = m_fileRate;
        m_resampler = new breakfastquay::Resampler(params, m_channelCount);

        double ratio = m_sampleRate / m_fileRate;
        m_resampleBufferFrames = int(ceil(double(m_cacheWriteBufferFrames) *
                                          ratio + 1));
        m_resampleBuffer = new float[m_resampleBufferFrames * m_channelCount];
    }

    m_cacheWriteBuffer = new float[m_cacheWriteBufferFrames * m_channelCount];
    m_cacheWriteBufferIndex = 0;

    if (m_cacheMode == CacheInTemporaryFile) {

        try {
            QDir dir(TempDirectory::getInstance()->getPath());
            m_cacheFileName = dir.filePath(QString("decoded_%1.w64")
                                           .arg((intptr_t)this));

            SF_INFO fileInfo;
            int fileRate = int(round(m_sampleRate));
            if (m_sampleRate != sv_samplerate_t(fileRate)) {
                SVDEBUG << "CodedAudioFileReader: WARNING: Non-integer sample rate "
                     << m_sampleRate << " presented for writing, rounding to " << fileRate
                     << endl;
            }
            fileInfo.samplerate = fileRate;
            fileInfo.channels = m_channelCount;

            // Previously we were writing SF_FORMAT_PCM_16 and in a
            // comment I wrote: "No point in writing 24-bit or float;
            // generally this class is used for decoding files that
            // have come from a 16 bit source or that decode to only
            // 16 bits anyway." That was naive -- we want to preserve
            // the original values to the same float precision that we
            // use internally. Saving PCM_16 obviously doesn't
            // preserve values for sources at bit depths greater than
            // 16, but it also doesn't always do so for sources at bit
            // depths less than 16.
            //
            // (This came to light with a bug in libsndfile 1.0.26,
            // which always reports every file as non-seekable, so
            // that coded readers were being used even for WAV
            // files. This changed the values that came from PCM_8 WAV
            // sources, breaking Sonic Annotator's output comparison
            // tests.)
            //
            // So: now we write floats.
            fileInfo.format = SF_FORMAT_W64 | SF_FORMAT_FLOAT;

#ifdef Q_OS_WIN
            m_cacheFileWritePtr = sf_wchar_open
                ((LPCWSTR)m_cacheFileName.utf16(), SFM_WRITE, &fileInfo);
#else
            m_cacheFileWritePtr = sf_open
                (m_cacheFileName.toLocal8Bit(), SFM_WRITE, &fileInfo);
#endif

            if (m_cacheFileWritePtr) {

                // Ideally we would do this now only if we were in a
                // threaded mode -- creating the reader later if we're
                // not threaded -- but we don't have access to that
                // information here

                m_cacheFileReader = new WavFileReader(m_cacheFileName);

                if (!m_cacheFileReader->isOK()) {
                    SVDEBUG << "ERROR: CodedAudioFileReader::initialiseDecodeCache: Failed to construct WAV file reader for temporary file: " << m_cacheFileReader->getError() << endl;
                    delete m_cacheFileReader;
                    m_cacheFileReader = nullptr;
                    m_cacheMode = CacheInMemory;
                    sf_close(m_cacheFileWritePtr);
                }

            } else {
                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;
                m_cacheMode = CacheInMemory;
            }

        } catch (const DirectoryCreationFailed &f) {
            SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: failed to create temporary directory! Falling back to in-memory cache" << endl;
            m_cacheMode = CacheInMemory;
        }
    }

    if (m_cacheMode == CacheInMemory) {
        m_data.clear();
    }

    if (m_trimFromEnd >= (m_cacheWriteBufferFrames * m_channelCount)) {
        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";
        m_trimFromEnd = 0;
    }

    m_initialised = true;
}

void
CodedAudioFileReader::addSamplesToDecodeCache(float **samples, sv_frame_t nframes)
{
    QMutexLocker locker(&m_cacheMutex);

    if (!m_initialised) return;

    for (sv_frame_t i = 0; i < nframes; ++i) {

        if (m_trimFromStart > 0) {
            --m_trimFromStart;
            continue;
        }
        
        for (int c = 0; c < m_channelCount; ++c) {

            float sample = samples[c][i];
            m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;

        }

        pushCacheWriteBufferMaybe(false);
    }
}

void
CodedAudioFileReader::addSamplesToDecodeCache(float *samples, sv_frame_t nframes)
{
    QMutexLocker locker(&m_cacheMutex);

    if (!m_initialised) return;

    for (sv_frame_t i = 0; i < nframes; ++i) {

        if (m_trimFromStart > 0) {
            --m_trimFromStart;
            continue;
        }
        
        for (int c = 0; c < m_channelCount; ++c) {

            float sample = samples[i * m_channelCount + c];
        
            m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;
        }

        pushCacheWriteBufferMaybe(false);
    }
}

void
CodedAudioFileReader::addSamplesToDecodeCache(const floatvec_t &samples)
{
    QMutexLocker locker(&m_cacheMutex);

    if (!m_initialised) return;

    for (float sample: samples) {

        if (m_trimFromStart > 0) {
            --m_trimFromStart;
            continue;
        }
        
        m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;

        pushCacheWriteBufferMaybe(false);
    }
}

void
CodedAudioFileReader::finishDecodeCache()
{
    QMutexLocker locker(&m_cacheMutex);

    Profiler profiler("CodedAudioFileReader::finishDecodeCache");

    if (!m_initialised) {
        SVDEBUG << "WARNING: CodedAudioFileReader::finishDecodeCache: Cache was never initialised!" << endl;
        return;
    }

    pushCacheWriteBufferMaybe(true);

    delete[] m_cacheWriteBuffer;
    m_cacheWriteBuffer = nullptr;

    delete[] m_resampleBuffer;
    m_resampleBuffer = nullptr;

    delete m_resampler;
    m_resampler = nullptr;

    if (m_cacheMode == CacheInTemporaryFile) {

        sf_close(m_cacheFileWritePtr);
        m_cacheFileWritePtr = nullptr;
        if (m_cacheFileReader) m_cacheFileReader->updateFrameCount();

    } else {
        // I know, I know, we already allocated it...
        StorageAdviser::notifyPlannedAllocation
            (StorageAdviser::MemoryAllocation,
             (m_data.size() * sizeof(float)) / 1024);
    }

    SVDEBUG << "CodedAudioFileReader: File decodes to " << m_fileFrameCount
            << " frames" << endl;
    if (m_fileFrameCount != m_frameCount) {
        SVDEBUG << "CodedAudioFileReader: Resampled to " << m_frameCount
                << " frames" << endl;
    }
    SVDEBUG << "CodedAudioFileReader: Signal abs max is " << m_max
            << ", " << m_clippedCount
            << " samples clipped, first non-zero frame is at "
            << m_firstNonzero << ", last at " << m_lastNonzero << endl;
    if (m_normalised) {
        SVDEBUG << "CodedAudioFileReader: Normalising, gain is " << m_gain << endl;
    }
}

void
CodedAudioFileReader::pushCacheWriteBufferMaybe(bool final)
{
    if (final ||
        (m_cacheWriteBufferIndex ==
         m_cacheWriteBufferFrames * m_channelCount)) {

        if (m_trimFromEnd > 0) {
        
            sv_frame_t framesToPush =
                (m_cacheWriteBufferIndex / m_channelCount) - m_trimFromEnd;

            if (framesToPush <= 0 && !final) {
                // This won't do, the buffer is full so we have to push
                // something. Should have checked for this earlier
                throw std::logic_error("Buffer full but nothing to push");
            }

            pushBuffer(m_cacheWriteBuffer, framesToPush, final);
            
            m_cacheWriteBufferIndex -= framesToPush * m_channelCount;

            for (sv_frame_t i = 0; i < m_cacheWriteBufferIndex; ++i) {
                m_cacheWriteBuffer[i] =
                    m_cacheWriteBuffer[framesToPush * m_channelCount + i];
            }

        } else {

            pushBuffer(m_cacheWriteBuffer,
                       m_cacheWriteBufferIndex / m_channelCount,
                       final);

            m_cacheWriteBufferIndex = 0;
        }

        if (m_cacheFileReader) {
            m_cacheFileReader->updateFrameCount();
        }
    }
}

sv_frame_t
CodedAudioFileReader::pushBuffer(float *buffer, sv_frame_t sz, bool final)
{
    m_fileFrameCount += sz;

    double ratio = 1.0;
    if (m_resampler && m_fileRate != 0) {
        ratio = m_sampleRate / m_fileRate;
    }
        
    if (ratio != 1.0) {
        pushBufferResampling(buffer, sz, ratio, final);
    } else {
        pushBufferNonResampling(buffer, sz);
    }

    return sz;
}

void
CodedAudioFileReader::pushBufferNonResampling(float *buffer, sv_frame_t sz)
{
    float clip = 1.0;
    sv_frame_t count = sz * m_channelCount;

    // statistics
    for (sv_frame_t j = 0; j < sz; ++j) {
        for (int c = 0; c < m_channelCount; ++c) {
            sv_frame_t i = j * m_channelCount + c;
            float v = buffer[i];
            if (!m_normalised) {
                if (v > clip) {
                    buffer[i] = clip;
                    ++m_clippedCount;
                } else if (v < -clip) {
                    buffer[i] = -clip;
                    ++m_clippedCount;
                }
            }
            v = fabsf(v);
            if (v != 0.f) {
                if (m_firstNonzero == 0) {
                    m_firstNonzero = m_frameCount;
                }
                m_lastNonzero = m_frameCount;
                if (v > m_max) {
                    m_max = v;
                }
            }
        }
        ++m_frameCount;
    }

    if (m_max > 0.f) {
        m_gain = 1.f / m_max; // used when normalising only
    }

    switch (m_cacheMode) {

    case CacheInTemporaryFile:
        if (sf_writef_float(m_cacheFileWritePtr, buffer, sz) < sz) {
            sf_close(m_cacheFileWritePtr);
            m_cacheFileWritePtr = nullptr;
            throw InsufficientDiscSpace(TempDirectory::getInstance()->getPath());
        }
        break;

    case CacheInMemory:
        m_dataLock.lock();
        try {
            m_data.insert(m_data.end(), buffer, buffer + count);
        } catch (const std::bad_alloc &e) {
            m_data.clear();
            SVCERR << "CodedAudioFileReader: Caught bad_alloc when trying to add " << count << " elements to buffer" << endl;
            m_dataLock.unlock();
            throw e;
        }
        m_dataLock.unlock();
        break;
    }
}

void
CodedAudioFileReader::pushBufferResampling(float *buffer, sv_frame_t sz,
                                           double ratio, bool final)
{
//    SVDEBUG << "pushBufferResampling: ratio = " << ratio << ", sz = " << sz << ", final = " << final << endl;

    if (sz > 0) {

        sv_frame_t out = m_resampler->resampleInterleaved
            (m_resampleBuffer,
             m_resampleBufferFrames,
             buffer,
             int(sz),
             ratio,
             false);

        pushBufferNonResampling(m_resampleBuffer, out);
    }

    if (final) {

        sv_frame_t padFrames = 1;
        if (double(m_frameCount) / ratio < double(m_fileFrameCount)) {
            padFrames = m_fileFrameCount - sv_frame_t(double(m_frameCount) / ratio) + 1;
        }

        sv_frame_t padSamples = padFrames * m_channelCount;

        SVDEBUG << "CodedAudioFileReader::pushBufferResampling: frameCount = " << m_frameCount << ", equivFileFrames = " << double(m_frameCount) / ratio << ", m_fileFrameCount = " << m_fileFrameCount << ", padFrames = " << padFrames << ", padSamples = " << padSamples << endl;

        float *padding = new float[padSamples];
        for (sv_frame_t i = 0; i < padSamples; ++i) padding[i] = 0.f;

        sv_frame_t out = m_resampler->resampleInterleaved
            (m_resampleBuffer,
             m_resampleBufferFrames,
             padding,
             int(padFrames),
             ratio,
             true);

        SVDEBUG << "CodedAudioFileReader::pushBufferResampling: resampled padFrames to " << out << " frames" << endl;

        sv_frame_t expected = sv_frame_t(round(double(m_fileFrameCount) * ratio));
        if (m_frameCount + out > expected) {
            out = expected - m_frameCount;
            SVDEBUG << "CodedAudioFileReader::pushBufferResampling: clipping that to " << out << " to avoid producing more samples than desired" << endl;
        }

        pushBufferNonResampling(m_resampleBuffer, out);
        delete[] padding;
    }
}

floatvec_t
CodedAudioFileReader::getInterleavedFrames(sv_frame_t start, sv_frame_t count) const
{
    Profiler profiler("CodedAudioFileReader::getInterleavedFrames");
    
    // Lock is only required in CacheInMemory mode (the cache file
    // reader is expected to be thread safe and manage its own
    // locking)

    if (!m_initialised) {
        SVDEBUG << "CodedAudioFileReader::getInterleavedFrames: not initialised" << endl;
        return {};
    }

    floatvec_t frames;
    
    switch (m_cacheMode) {

    case CacheInTemporaryFile:
        if (m_cacheFileReader) {
            frames = m_cacheFileReader->getInterleavedFrames(start, count);
        }
        break;

    case CacheInMemory:
    {
        if (!isOK()) return {};
        if (count == 0) return {};

        sv_frame_t ix0 = start * m_channelCount;
        sv_frame_t ix1 = ix0 + (count * m_channelCount);

        // This lock used to be a QReadWriteLock, but it appears that
        // its lock mechanism is significantly slower than QMutex so
        // it's not a good idea in cases like this where we don't
        // really have threads taking a long time to read concurrently
        m_dataLock.lock();
        sv_frame_t n = sv_frame_t(m_data.size());
        if (ix0 > n) ix0 = n;
        if (ix1 > n) ix1 = n;
        frames = floatvec_t(m_data.begin() + ix0, m_data.begin() + ix1);
        m_dataLock.unlock();
        break;
    }
    }

    if (m_normalised) {
        for (auto &f: frames) f *= m_gain;
    }

    return frames;
}