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//
// SampleSourceAudiofile.c - MrsWatson
// Created by Nik Reiman on 1/22/12.
// Copyright (c) 2012 Teragon Audio. All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
#if USE_AUDIOFILE
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include "audio/AudioSettings.h"
#include "io/SampleSourceAudiofile.h"
#include "io/SampleSourcePcm.h"
#include "logging/EventLogger.h"
static boolByte _openSampleSourceAudiofile(void *sampleSourcePtr, const SampleSourceOpenAs openAs)
{
SampleSource sampleSource = (SampleSource)sampleSourcePtr;
SampleSourceAudiofileData extraData = sampleSource->extraData;
if (openAs == SAMPLE_SOURCE_OPEN_READ) {
extraData->fileHandle = afOpenFile(sampleSource->sourceName->data, "r", NULL);
if (extraData->fileHandle != NULL) {
setNumChannels((const unsigned int)afGetVirtualChannels(extraData->fileHandle, AF_DEFAULT_TRACK));
setSampleRate((float)afGetRate(extraData->fileHandle, AF_DEFAULT_TRACK));
}
} else if (openAs == SAMPLE_SOURCE_OPEN_WRITE) {
int byteOrder = AF_BYTEORDER_LITTLEENDIAN;
int outfileFormat;
switch (sampleSource->sampleSourceType) {
case SAMPLE_SOURCE_TYPE_AIFF:
// AIFF is the only file format we support which is big-endian. That is,
// even on big-endian platforms (which are untested), raw PCM should still
// write little-endian data.
byteOrder = AF_BYTEORDER_BIGENDIAN;
outfileFormat = AF_FILE_AIFF;
break;
case SAMPLE_SOURCE_TYPE_WAVE:
outfileFormat = AF_FILE_WAVE;
break;
case SAMPLE_SOURCE_TYPE_FLAC:
outfileFormat = AF_FILE_FLAC;
break;
default:
logInternalError("Unsupported audiofile type %d", sampleSource->sampleSourceType);
return false;
}
AFfilesetup outfileSetup = afNewFileSetup();
afInitFileFormat(outfileSetup, outfileFormat);
afInitByteOrder(outfileSetup, AF_DEFAULT_TRACK, byteOrder);
afInitChannels(outfileSetup, AF_DEFAULT_TRACK, getNumChannels());
afInitRate(outfileSetup, AF_DEFAULT_TRACK, getSampleRate());
afInitSampleFormat(outfileSetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, DEFAULT_BITRATE);
extraData->fileHandle = afOpenFile(sampleSource->sourceName->data, "w", outfileSetup);
} else {
logInternalError("Invalid type for openAs in audiofile source");
return false;
}
if (extraData->fileHandle == NULL) {
logError("File '%s' could not be opened for %s",
sampleSource->sourceName->data,
openAs == SAMPLE_SOURCE_OPEN_READ ? "reading" : "writing");
return false;
}
sampleSource->openedAs = openAs;
return true;
}
boolByte _readBlockFromAudiofile(void *sampleSourcePtr, SampleBuffer sampleBuffer)
{
SampleSource sampleSource = (SampleSource)sampleSourcePtr;
SampleSourceAudiofileData extraData = (SampleSourceAudiofileData)(sampleSource->extraData);
const size_t bufferByteSize = sizeof(short) * getNumChannels() * getBlocksize();
AFframecount numFramesRead = 0;
if (extraData->pcmBuffer == NULL) {
extraData->pcmBuffer = (short *)malloc(bufferByteSize);
}
memset(extraData->pcmBuffer, 0, bufferByteSize);
numFramesRead = afReadFrames(extraData->fileHandle, AF_DEFAULT_TRACK,
extraData->pcmBuffer, (int)getBlocksize());
sampleBufferCopyPcmSamples(sampleBuffer, extraData->pcmBuffer);
// Set the blocksize of the sample buffer to be the number of frames read
sampleBuffer->blocksize = (unsigned long)numFramesRead;
sampleSource->numSamplesProcessed += numFramesRead;
if (numFramesRead == 0) {
logDebug("End of audio file reached");
return false;
} else if (numFramesRead < 0) {
logError("Error reading audio file");
return false;
} else {
return true;
}
}
boolByte _writeBlockToAudiofile(void *sampleSourcePtr, const SampleBuffer sampleBuffer)
{
SampleSource sampleSource = (SampleSource)sampleSourcePtr;
SampleSourceAudiofileData extraData = (SampleSourceAudiofileData)(sampleSource->extraData);
const AFframecount numSamplesToWrite = sampleBuffer->blocksize;
const size_t bufferByteSize = sizeof(short) * getNumChannels() * getBlocksize();
AFframecount numFramesWritten = 0;
if (extraData->pcmBuffer == NULL) {
extraData->pcmBuffer = (short *)malloc(bufferByteSize);
}
memset(extraData->pcmBuffer, 0, bufferByteSize);
// TODO: flip endian argument is probably wrong for some file formats (namely AIFF)!!
sampleBufferGetPcmSamples(sampleBuffer, extraData->pcmBuffer, false);
numFramesWritten = afWriteFrames(extraData->fileHandle, AF_DEFAULT_TRACK,
extraData->pcmBuffer, (int)getBlocksize());
sampleSource->numSamplesProcessed += getBlocksize() * getNumChannels();
if (numFramesWritten == -1) {
logWarn("audiofile encountered an error when writing to file");
return false;
} else if (numFramesWritten == numSamplesToWrite) {
return true;
} else {
logWarn("Short write occurred while writing samples");
return false;
}
}
void _closeSampleSourceAudiofile(void *sampleSourcePtr)
{
SampleSource sampleSource = (SampleSource)sampleSourcePtr;
SampleSourceAudiofileData extraData = (SampleSourceAudiofileData)sampleSource->extraData;
if (extraData->fileHandle != NULL) {
afCloseFile(extraData->fileHandle);
}
}
void _freeSampleSourceDataAudiofile(void *sampleSourceDataPtr)
{
SampleSourceAudiofileData extraData = (SampleSourceAudiofileData)sampleSourceDataPtr;
if (extraData->pcmBuffer != NULL) {
free(extraData->pcmBuffer);
}
free(extraData);
}
SampleSource _newSampleSourceAudiofile(const CharString sampleSourceName,
const SampleSourceType sampleSourceType)
{
SampleSource sampleSource = (SampleSource)malloc(sizeof(SampleSourceMembers));
SampleSourceAudiofileData extraData = (SampleSourceAudiofileData)malloc(sizeof(SampleSourceAudiofileDataMembers));
sampleSource->sampleSourceType = sampleSourceType;
sampleSource->openedAs = SAMPLE_SOURCE_OPEN_NOT_OPENED;
sampleSource->sourceName = newCharString();
charStringCopy(sampleSource->sourceName, sampleSourceName);
sampleSource->numSamplesProcessed = 0;
sampleSource->openSampleSource = _openSampleSourceAudiofile;
sampleSource->readSampleBlock = _readBlockFromAudiofile;
sampleSource->writeSampleBlock = _writeBlockToAudiofile;
sampleSource->closeSampleSource = _closeSampleSourceAudiofile;
sampleSource->freeSampleSourceData = _freeSampleSourceDataAudiofile;
extraData->fileHandle = NULL;
extraData->pcmBuffer = NULL;
sampleSource->extraData = extraData;
return sampleSource;
}
#endif
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