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-rw-r--r--examples/mobile/quickhit/ga_src/GEAudioBuffer.cpp392
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diff --git a/examples/mobile/quickhit/ga_src/GEAudioBuffer.cpp b/examples/mobile/quickhit/ga_src/GEAudioBuffer.cpp
deleted file mode 100644
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--- a/examples/mobile/quickhit/ga_src/GEAudioBuffer.cpp
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-/****************************************************************************
-**
-** Copyright (C) 2012 Digia Plc and/or its subsidiary(-ies).
-** Contact: http://www.qt-project.org/legal
-**
-** This file is part of the demonstration applications of the Qt Toolkit.
-**
-** $QT_BEGIN_LICENSE:LGPL$
-** Commercial License Usage
-** Licensees holding valid commercial Qt licenses may use this file in
-** accordance with the commercial license agreement provided with the
-** Software or, alternatively, in accordance with the terms contained in
-** a written agreement between you and Digia. For licensing terms and
-** conditions see http://qt.digia.com/licensing. For further information
-** use the contact form at http://qt.digia.com/contact-us.
-**
-** GNU Lesser General Public License Usage
-** Alternatively, this file may be used under the terms of the GNU Lesser
-** General Public License version 2.1 as published by the Free Software
-** Foundation and appearing in the file LICENSE.LGPL included in the
-** packaging of this file. Please review the following information to
-** ensure the GNU Lesser General Public License version 2.1 requirements
-** will be met: http://www.gnu.org/licenses/old-licenses/lgpl-2.1.html.
-**
-** In addition, as a special exception, Digia gives you certain additional
-** rights. These rights are described in the Digia Qt LGPL Exception
-** version 1.1, included in the file LGPL_EXCEPTION.txt in this package.
-**
-** GNU General Public License Usage
-** Alternatively, this file may be used under the terms of the GNU
-** General Public License version 3.0 as published by the Free Software
-** Foundation and appearing in the file LICENSE.GPL included in the
-** packaging of this file. Please review the following information to
-** ensure the GNU General Public License version 3.0 requirements will be
-** met: http://www.gnu.org/copyleft/gpl.html.
-**
-**
-** $QT_END_LICENSE$
-**
-****************************************************************************/
-
-#include <QDebug>
-#include <math.h>
-#include "GEAudioBuffer.h"
-
-using namespace GE;
-
-
-struct SWavHeader {
- char chunkID[4];
- unsigned int chunkSize;
- char format[4];
-
- unsigned char subchunk1id[4];
- unsigned int subchunk1size;
- unsigned short audioFormat;
- unsigned short nofChannels;
- unsigned int sampleRate;
- unsigned int byteRate;
-
- unsigned short blockAlign;
- unsigned short bitsPerSample;
-
- unsigned char subchunk2id[4];
- unsigned int subchunk2size;
-
-};
-
-CAudioBuffer::CAudioBuffer() {
- m_data = 0;
- m_dataLength = 0;
- m_sampleFunction = 0;
-};
-
-
-CAudioBuffer::~CAudioBuffer() {
- reallocate(0);
-}
-
-void CAudioBuffer::reallocate( int length ) {
- if (m_data) delete [] ((char*)m_data);
- m_dataLength = length;
- if (m_dataLength>0) {
- m_data = new char[ m_dataLength ];
- } else m_data = 0;
-};
-
-
-CAudioBuffer* CAudioBuffer::loadWav( QString fileName ) {
- QFile *wavFile = new QFile( fileName );
-
-
- if (wavFile->open(QIODevice::ReadOnly)) {
- SWavHeader header;
-
- wavFile->read( header.chunkID, 4 );
- if (header.chunkID[0]!='R' || header.chunkID[1]!='I' || header.chunkID[2]!='F' || header.chunkID[3]!='F') return 0; // incorrect header
-
- wavFile->read( (char*)&header.chunkSize,4 );
- wavFile->read( (char*)&header.format,4 );
-
- if (header.format[0]!='W' || header.format[1]!='A' || header.format[2]!='V' || header.format[3]!='E') return 0; // incorrect header
-
- wavFile->read( (char*)&header.subchunk1id,4 );
- if (header.subchunk1id[0]!='f' || header.subchunk1id[1]!='m' || header.subchunk1id[2]!='t' || header.subchunk1id[3]!=' ') return 0; // incorrect header
-
- wavFile->read( (char*)&header.subchunk1size,4 );
- wavFile->read( (char*)&header.audioFormat,2 );
- wavFile->read( (char*)&header.nofChannels,2 );
- wavFile->read( (char*)&header.sampleRate,4 );
- wavFile->read( (char*)&header.byteRate,4 );
- wavFile->read( (char*)&header.blockAlign,2 );
- wavFile->read( (char*)&header.bitsPerSample,2 );
-
- qDebug() << fileName << " opened";
-
- while (1) {
- if (wavFile->read( (char*)&header.subchunk2id,4 ) != 4) return 0;
- if (wavFile->read( (char*)&header.subchunk2size,4 ) != 4) return 0;
- //int deb_size = header.subchunk2size;
- //char tes[4];
- //memcpy(tes, header.subchunk2id, 4 );
- //if (header.subchunk2id[0]!='d' || header.subchunk2id[1]!='a' || header.subchunk2id[2]!='t' || header.subchunk2id[3]!='a') return 0; // incorrect header
- if (header.subchunk2id[0]=='d' && header.subchunk2id[1]=='a' && header.subchunk2id[2]=='t' && header.subchunk2id[3]=='a') break; // found the data, chunk
- // this was not the data-chunk. skip it
- if (header.subchunk2size<1) return 0; // error in file
- char *unused = new char[header.subchunk2size];
- wavFile->read( unused, header.subchunk2size );
- delete [] unused;
- }
-
-
-
- // the data follows.
- if (header.subchunk2size<1) return 0;
-
- CAudioBuffer *rval = new CAudioBuffer;
- rval->m_nofChannels = header.nofChannels;
- rval->m_bitsPerSample = header.bitsPerSample;
- rval->m_samplesPerSec = header.sampleRate;
- rval->m_signedData = 0; // where to know this?
- rval->reallocate( header.subchunk2size );
-
- wavFile->read( (char*)rval->m_data, header.subchunk2size );
-
- // choose a good sampling function.
- rval->m_sampleFunction = 0;
- if (rval->m_nofChannels==1) {
- if (rval->m_bitsPerSample == 8) rval->m_sampleFunction = sampleFunction8bitMono;
- if (rval->m_bitsPerSample == 16) rval->m_sampleFunction = sampleFunction16bitMono;
- } else {
- if (rval->m_bitsPerSample == 8) rval->m_sampleFunction = sampleFunction8bitStereo;
- if (rval->m_bitsPerSample == 16) rval->m_sampleFunction = sampleFunction16bitStereo;
- }
-
- return rval;
-
-
- } else {
- qDebug() << fileName << " NOT opened";
- return 0;
- }
-
- delete wavFile;
-};
-
-
-CAudioBuffer* CAudioBuffer::loadWav( FILE *wavFile ) {
- // read the header.
- SWavHeader header;
- fread( header.chunkID, 4, 1, wavFile );
- if (header.chunkID[0]!='R' || header.chunkID[1]!='I' || header.chunkID[2]!='F' || header.chunkID[3]!='F') return 0; // incorrect header
-
- fread( &header.chunkSize, 4, 1, wavFile );
- fread( header.format, 4, 1, wavFile );
- if (header.format[0]!='W' || header.format[1]!='A' || header.format[2]!='V' || header.format[3]!='E') return 0; // incorrect header
-
- fread( header.subchunk1id, 4, 1, wavFile );
- if (header.subchunk1id[0]!='f' || header.subchunk1id[1]!='m' || header.subchunk1id[2]!='t' || header.subchunk1id[3]!=' ') return 0; // incorrect header
-
- fread( &header.subchunk1size, 4, 1, wavFile );
- fread( &header.audioFormat, 2, 1, wavFile );
- fread( &header.nofChannels, 2, 1, wavFile );
- fread( &header.sampleRate, 4, 1, wavFile );
- fread( &header.byteRate, 4, 1, wavFile );
-
- fread( &header.blockAlign, 2, 1, wavFile );
- fread( &header.bitsPerSample, 2, 1, wavFile );
-
- fread( header.subchunk2id, 4, 1, wavFile );
- if (header.subchunk2id[0]!='d' || header.subchunk2id[1]!='a' || header.subchunk2id[2]!='t' || header.subchunk2id[3]!='a') return 0; // incorrect header
- fread( &header.subchunk2size, 4, 1, wavFile );
-
-
- // the data follows.
- if (header.subchunk2size<1) return 0;
-
- CAudioBuffer *rval = new CAudioBuffer;
- rval->m_nofChannels = header.nofChannels;
- rval->m_bitsPerSample = header.bitsPerSample;
- rval->m_samplesPerSec = header.sampleRate;
- rval->m_signedData = 0; // where to know this?
- rval->reallocate( header.subchunk2size );
-
- fread( rval->m_data, 1, header.subchunk2size, wavFile );
-
-
-
- // choose a good sampling function.
- rval->m_sampleFunction = 0;
- if (rval->m_nofChannels==1) {
- if (rval->m_bitsPerSample == 8) rval->m_sampleFunction = sampleFunction8bitMono;
- if (rval->m_bitsPerSample == 16) rval->m_sampleFunction = sampleFunction16bitMono;
- } else {
- if (rval->m_bitsPerSample == 8) rval->m_sampleFunction = sampleFunction8bitStereo;
- if (rval->m_bitsPerSample == 16) rval->m_sampleFunction = sampleFunction16bitStereo;
- }
-
- return rval;
-};
-
-
-
-AUDIO_SAMPLE_TYPE CAudioBuffer::sampleFunction8bitMono( CAudioBuffer *abuffer, int pos, int channel ) {
- return (AUDIO_SAMPLE_TYPE)(((unsigned char*)(abuffer->m_data))[pos]-128)<<8;
-};
-
-AUDIO_SAMPLE_TYPE CAudioBuffer::sampleFunction16bitMono( CAudioBuffer *abuffer, int pos, int channel ) {
- return (AUDIO_SAMPLE_TYPE)(((short*)(abuffer->m_data))[pos]);
-};
-
-AUDIO_SAMPLE_TYPE CAudioBuffer::sampleFunction8bitStereo( CAudioBuffer *abuffer, int pos, int channel ) {
- return ((AUDIO_SAMPLE_TYPE)(((char*)(abuffer->m_data))[pos*abuffer->m_nofChannels + channel])<<8);
-};
-
-
-AUDIO_SAMPLE_TYPE CAudioBuffer::sampleFunction16bitStereo( CAudioBuffer *abuffer, int pos, int channel ) {
- return (AUDIO_SAMPLE_TYPE)(((short*)(abuffer->m_data))[pos*abuffer->m_nofChannels + channel]);
-};
-
-CAudioBufferPlayInstance *CAudioBuffer::playWithMixer( CAudioMixer &mixer ) {
- CAudioBufferPlayInstance *i = (CAudioBufferPlayInstance*)mixer.addAudioSource( new CAudioBufferPlayInstance( this ));
- return i;
-};
-
-
-CAudioBufferPlayInstance::CAudioBufferPlayInstance() {
- m_fixedPos = 0;
- m_fixedInc = 0;
- m_buffer = 0;
- m_fixedLeftVolume = 4096;
- m_fixedRightVolume = 4096;
- m_destroyWhenFinished = true;
- m_finished = false;
-};
-
-CAudioBufferPlayInstance::CAudioBufferPlayInstance( CAudioBuffer *startPlaying ) {
- m_fixedPos = 0;
- m_fixedInc = 0;
- m_fixedLeftVolume = 4096;
- m_fixedRightVolume = 4096;
- m_destroyWhenFinished = true;
- m_finished = false;
- playBuffer( startPlaying, 1.0f, 1.0f );
-};
-
-void CAudioBufferPlayInstance::playBuffer( CAudioBuffer *startPlaying, float volume, float speed, int loopTimes ) {
- m_buffer = startPlaying;
- m_fixedLeftVolume = (int)(4096.0f*volume);
- m_fixedRightVolume = m_fixedLeftVolume;
- m_fixedPos = 0;
- //m_fixedInc = ( startPlaying->getSamplesPerSec() * (int)(4096.0f*speed)) / AUDIO_FREQUENCY;
- setSpeed( speed );
- m_loopTimes = loopTimes;
-
-};
-
-CAudioBufferPlayInstance::~CAudioBufferPlayInstance() {
-
-};
-
-
-void CAudioBufferPlayInstance::stop() {
- m_buffer = 0;
- m_finished = true;
-};
-
-void CAudioBufferPlayInstance::setSpeed( float speed ) {
- if (!m_buffer) return;
- m_fixedInc = (int)( ((float)m_buffer->getSamplesPerSec() * 4096.0f*speed) / (float)AUDIO_FREQUENCY );
-};
-
-void CAudioBufferPlayInstance::setLeftVolume( float vol ) {
- m_fixedLeftVolume = (int)(4096.0f*vol);
-};
-
-void CAudioBufferPlayInstance::setRightVolume( float vol ) {
- m_fixedRightVolume = (int)(4096.0f*vol);
-};
-
-bool CAudioBufferPlayInstance::canBeDestroyed() {
- if (m_finished==true &&
- m_destroyWhenFinished==true) return true; else return false;
-};
-
-
-
-// Does not do any bound-checking. Must be checked before called.
-int CAudioBufferPlayInstance::mixBlock( AUDIO_SAMPLE_TYPE *target, int samplesToMix ) {
- SAMPLE_FUNCTION_TYPE sampleFunction = m_buffer->getSampleFunction();
- if (!sampleFunction) return 0; // unsupported sampletype
- AUDIO_SAMPLE_TYPE *t_target = target+samplesToMix*2;
- int sourcepos;
- //int tempCounter = 0;
-
- if (m_buffer->getNofChannels() == 2) { // stereo
- while (target!=t_target) {
- sourcepos = m_fixedPos>>12;
- target[0] = (((((sampleFunction)( m_buffer, sourcepos, 0) * (4096-(m_fixedPos&4095)) + (sampleFunction)( m_buffer, sourcepos+1, 0) * (m_fixedPos&4095) ) >> 12) * m_fixedLeftVolume) >> 12);
- target[1] = (((((sampleFunction)( m_buffer, sourcepos, 1) * (4096-(m_fixedPos&4095)) + (sampleFunction)( m_buffer, sourcepos+1, 1) * (m_fixedPos&4095) ) >> 12) * m_fixedRightVolume) >> 12);
- m_fixedPos+=m_fixedInc;
- target+=2;
- //tempCounter++;
- };
- } else { // mono
- int temp;
- while (target!=t_target) {
- sourcepos = m_fixedPos>>12;
- temp = (((sampleFunction)( m_buffer, sourcepos, 0 ) * (4096-(m_fixedPos&4095)) + (sampleFunction)( m_buffer, sourcepos+1, 0 ) * (m_fixedPos&4095) ) >> 12);
- target[0] = ((temp*m_fixedLeftVolume)>>12);
- target[1] = ((temp*m_fixedRightVolume)>>12);
- m_fixedPos+=m_fixedInc;
- target+=2;
- //tempCounter++;
- };
-
- };
-
- return samplesToMix;
-};
-
-
-
-int CAudioBufferPlayInstance::pullAudio( AUDIO_SAMPLE_TYPE *target, int bufferLength ) {
- if (!m_buffer) return 0; // no sample associated to mix..
-
- int channelLength = ((m_buffer->getDataLength()) / (m_buffer->getNofChannels()*m_buffer->getBytesPerSample()))-2;
-
- int samplesToWrite = bufferLength/2;
- int amount;
- int totalMixed = 0;
-
-
- while (samplesToWrite>0) {
- int samplesLeft = channelLength - (m_fixedPos>>12);
- int maxMixAmount = (int)(((long long int)(samplesLeft)<<12) / m_fixedInc ); // This is how much we can mix at least
- //int maxMixAmount = (int)((float)samplesLeft / ((float)m_fixedInc/4096.0f));
- //if (maxMixAmount<1) maxMixAmount = 1; // NOTE, THIS MIGHT CAUSE PROBLEMS. NEEDS CHECKING
- if (maxMixAmount>samplesToWrite) {
- maxMixAmount=samplesToWrite;
- }
-
- if (maxMixAmount > 0) {
- amount=mixBlock(target+totalMixed * 2, maxMixAmount);
-
- if (amount == 0)
- {
- // Error!
- break;
- }
-
- totalMixed+=amount;
- } else {
- amount = 0;
- m_fixedPos = channelLength<<12;
- }
-
- // sample is ended,.. check the looping variables and see what to do.
- if ((m_fixedPos>>12)>=channelLength) {
- m_fixedPos -= (channelLength<<12);
- if (m_loopTimes>0) m_loopTimes--;
- if (m_loopTimes==0) {
- stop();
- return totalMixed;
- }
- }
-
- samplesToWrite-=amount;
- if (samplesToWrite<1) break;
- };
- return totalMixed*2;
-};