view data/fileio/CodedAudioFileReader.cpp @ 1061:c1e43c8d2527 tonioni

Thread-local debug was causing crash on exit with Qt 5.4.x. But we introduced that because QDebug itself was crashing when used from multiple threads. Replace with simpler fstream version
author Chris Cannam
date Tue, 31 Mar 2015 10:36:52 +0100
parents e603b44510c3
children 4d9816ba0ebe 1517d4c60e88
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/* -*- 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/Resampler.h"

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

CodedAudioFileReader::CodedAudioFileReader(CacheMode cacheMode,
                                           sv_samplerate_t targetRate,
                                           bool normalised) :
    m_cacheMode(cacheMode),
    m_initialised(false),
    m_serialiser(0),
    m_fileRate(0),
    m_cacheFileWritePtr(0),
    m_cacheFileReader(0),
    m_cacheWriteBuffer(0),
    m_cacheWriteBufferIndex(0),
    m_cacheWriteBufferSize(16384),
    m_resampler(0),
    m_resampleBuffer(0),
    m_fileFrameCount(0),
    m_normalised(normalised),
    m_max(0.f),
    m_gain(1.f)
{
    SVDEBUG << "CodedAudioFileReader::CodedAudioFileReader: rate " << targetRate << ", normalised = " << normalised << endl;

    m_frameCount = 0;
    m_sampleRate = targetRate;
}

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

    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 != "") {
        if (!QFile(m_cacheFileName).remove()) {
            cerr << "WARNING: CodedAudioFileReader::~CodedAudioFileReader: Failed to delete cache file \"" << m_cacheFileName << "\"" << endl;
        }
    }

    delete m_resampler;
    delete[] m_resampleBuffer;
}

void
CodedAudioFileReader::startSerialised(QString id)
{
    SVDEBUG << "CodedAudioFileReader::startSerialised(" << id << ")" << endl;

    delete m_serialiser;
    m_serialiser = new Serialiser(id);
}

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

    delete m_serialiser;
    m_serialiser = 0;
}

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

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

    if (m_fileRate == 0) {
        cerr << "CodedAudioFileReader::initialiseDecodeCache: ERROR: File sample rate unknown (bug in subclass implementation?)" << endl;
        throw FileOperationFailed("(coded file)", "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;
        m_resampler = new Resampler(Resampler::FastestTolerable,
                                    m_channelCount,
                                    m_cacheWriteBufferSize);
        double ratio = m_sampleRate / m_fileRate;
        m_resampleBuffer = new float
            [lrint(ceil(double(m_cacheWriteBufferSize) * m_channelCount * ratio + 1))];
    }

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

    if (m_cacheMode == CacheInTemporaryFile) {

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

            SF_INFO fileInfo;
            int fileRate = int(round(m_sampleRate));
            if (m_sampleRate != sv_samplerate_t(fileRate)) {
                cerr << "CodedAudioFileReader: WARNING: Non-integer sample rate "
                     << m_sampleRate << " presented for writing, rounding to " << fileRate
                     << endl;
            }
            fileInfo.samplerate = fileRate;
            fileInfo.channels = m_channelCount;
            
            // 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.
            fileInfo.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
    
            m_cacheFileWritePtr = sf_open(m_cacheFileName.toLocal8Bit(),
                                          SFM_WRITE, &fileInfo);

            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()) {
                    cerr << "ERROR: CodedAudioFileReader::initialiseDecodeCache: Failed to construct WAV file reader for temporary file: " << m_cacheFileReader->getError() << endl;
                    delete m_cacheFileReader;
                    m_cacheFileReader = 0;
                    m_cacheMode = CacheInMemory;
                    sf_close(m_cacheFileWritePtr);
                }

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

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

    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) {
        
        for (int c = 0; c < m_channelCount; ++c) {

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

            if (m_cacheWriteBufferIndex ==
                m_cacheWriteBufferSize * m_channelCount) {

                pushBuffer(m_cacheWriteBuffer, m_cacheWriteBufferSize, false);
                m_cacheWriteBufferIndex = 0;
            }

            if (m_cacheWriteBufferIndex % 10240 == 0 &&
                m_cacheFileReader) {
                m_cacheFileReader->updateFrameCount();
            }
        }
    }
}

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) {
        
        for (int c = 0; c < m_channelCount; ++c) {

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

            if (m_cacheWriteBufferIndex ==
                m_cacheWriteBufferSize * m_channelCount) {

                pushBuffer(m_cacheWriteBuffer, m_cacheWriteBufferSize, false);
                m_cacheWriteBufferIndex = 0;
            }

            if (m_cacheWriteBufferIndex % 10240 == 0 &&
                m_cacheFileReader) {
                m_cacheFileReader->updateFrameCount();
            }
        }
    }
}

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

    if (!m_initialised) return;

    for (float sample: samples) {
        
        m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;

        if (m_cacheWriteBufferIndex ==
            m_cacheWriteBufferSize * m_channelCount) {

            pushBuffer(m_cacheWriteBuffer, m_cacheWriteBufferSize, false);
            m_cacheWriteBufferIndex = 0;
        }

        if (m_cacheWriteBufferIndex % 10240 == 0 &&
            m_cacheFileReader) {
            m_cacheFileReader->updateFrameCount();
        }
    }
}

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

    Profiler profiler("CodedAudioFileReader::finishDecodeCache", true);

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

    pushBuffer(m_cacheWriteBuffer,
               m_cacheWriteBufferIndex / m_channelCount,
               true);

    delete[] m_cacheWriteBuffer;
    m_cacheWriteBuffer = 0;

    delete[] m_resampleBuffer;
    m_resampleBuffer = 0;

    delete m_resampler;
    m_resampler = 0;

    if (m_cacheMode == CacheInTemporaryFile) {
        sf_close(m_cacheFileWritePtr);
        m_cacheFileWritePtr = 0;
        if (m_cacheFileReader) m_cacheFileReader->updateFrameCount();
    }
}

void
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);
    }
}

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

    if (m_normalised) {
        for (sv_frame_t i = 0; i < count; ++i) {
            float v = fabsf(buffer[i]);
            if (v > m_max) {
                m_max = v;
                m_gain = 1.f / m_max;
            }
        }
    } else {
        for (sv_frame_t i = 0; i < count; ++i) {
            if (buffer[i] >  clip) buffer[i] =  clip;
        }
        for (sv_frame_t i = 0; i < count; ++i) {
            if (buffer[i] < -clip) buffer[i] = -clip;
        }
    }

    m_frameCount += sz;

    switch (m_cacheMode) {

    case CacheInTemporaryFile:
        if (sf_writef_float(m_cacheFileWritePtr, buffer, sz) < sz) {
            sf_close(m_cacheFileWritePtr);
            m_cacheFileWritePtr = 0;
            throw InsufficientDiscSpace(TempDirectory::getInstance()->getPath());
        }
        break;

    case CacheInMemory:
        m_dataLock.lockForWrite();
        for (sv_frame_t s = 0; s < count; ++s) {
            m_data.push_back(buffer[s]);
        }
        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
            (buffer,
             m_resampleBuffer,
             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 << "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
            (padding,
             m_resampleBuffer,
             padFrames,
             ratio,
             true);

        if (m_frameCount + out > sv_frame_t(double(m_fileFrameCount) * ratio)) {
            out = sv_frame_t(double(m_fileFrameCount) * ratio) - m_frameCount;
        }

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

SampleBlock
CodedAudioFileReader::getInterleavedFrames(sv_frame_t start, sv_frame_t count) const
{
    // 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 SampleBlock();
    }

    SampleBlock frames;
    
    switch (m_cacheMode) {

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

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

        sv_frame_t idx = start * m_channelCount;
        sv_frame_t i = 0;
        sv_frame_t n = count * m_channelCount;

        frames.resize(n);

        m_dataLock.lockForRead();
        while (i < n && in_range_for(m_data, idx)) {
            frames[i++] = m_data[idx++];
        }
        m_dataLock.unlock();

        frames.resize(i);
    }
    }

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

    return frames;
}