annotate data/fileio/CodedAudioFileReader.cpp @ 1008:d9e0e59a1581

When using an aggregate model to pass data to a transform, zero-pad the shorter input to the duration of the longer rather than truncating the longer. (This is better behaviour for e.g. MATCH, and in any case the code was previously truncating incorrectly and ending up with garbage data at the end.)

/* -*- 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,
                                           int 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);
        float ratio = float(m_sampleRate) / float(m_fileRate);
        m_resampleBuffer = new float
            [lrintf(ceilf(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;
            fileInfo.samplerate = m_sampleRate;
            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, int nframes)
{
    QMutexLocker locker(&m_cacheMutex);

    if (!m_initialised) return;

    for (int 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, int nframes)
{
    QMutexLocker locker(&m_cacheMutex);

    if (!m_initialised) return;

    for (int 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 (int i = 0; i < (int)samples.size(); ++i) {

        float sample = samples[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::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, int sz, bool final)
{
    m_fileFrameCount += sz;

    float ratio = 1.f;
    if (m_resampler && m_fileRate != 0) {
        ratio = float(m_sampleRate) / float(m_fileRate);
    }
        
    if (ratio != 1.f) {
        pushBufferResampling(buffer, sz, ratio, final);
    } else {
        pushBufferNonResampling(buffer, sz);
    }
}

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

    if (m_normalised) {
        for (int 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 (int i = 0; i < count; ++i) {
            if (buffer[i] >  clip) buffer[i] =  clip;
        }
        for (int 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) < (int)sz) {
            sf_close(m_cacheFileWritePtr);
            m_cacheFileWritePtr = 0;
            throw InsufficientDiscSpace(TempDirectory::getInstance()->getPath());
        }
        break;

    case CacheInMemory:
        m_dataLock.lockForWrite();
        for (int s = 0; s < count; ++s) {
            m_data.push_back(buffer[s]);
        }
	MUNLOCK_SAMPLEBLOCK(m_data);
        m_dataLock.unlock();
        break;
    }
}

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

    if (sz > 0) {

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

        pushBufferNonResampling(m_resampleBuffer, out);
    }

    if (final) {

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

        int padSamples = padFrames * m_channelCount;

        SVDEBUG << "frameCount = " << m_frameCount << ", equivFileFrames = " << m_frameCount / ratio << ", m_fileFrameCount = " << m_fileFrameCount << ", padFrames= " << padFrames << ", padSamples = " << padSamples << endl;

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

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

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

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

void
CodedAudioFileReader::getInterleavedFrames(int start, int count,
                                           SampleBlock &frames) 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;
    }

    switch (m_cacheMode) {

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

    case CacheInMemory:
    {
        frames.clear();
        if (!isOK()) return;
        if (count == 0) return;
        frames.reserve(count * m_channelCount);

        int idx = start * m_channelCount;
        int i = 0;

        m_dataLock.lockForRead();
        while (i < count * m_channelCount && idx < (int)m_data.size()) {
            frames.push_back(m_data[idx]);
            ++idx;
        }
        m_dataLock.unlock();
    }
    }

    if (m_normalised) {
        for (int i = 0; i < (int)(count * m_channelCount); ++i) {
            frames[i] *= m_gain;
        }
    }
}