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diff --git a/examples/mobile/quickhit/ga_src/GEAudioBuffer.cpp b/examples/mobile/quickhit/ga_src/GEAudioBuffer.cpp
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+/****************************************************************************
+**
+** Copyright (C) 2012 Nokia Corporation and/or its subsidiary(-ies).
+** Contact: http://www.qt-project.org/
+**
+** This file is part of the demonstration applications of the Qt Toolkit.
+**
+** $QT_BEGIN_LICENSE:LGPL$
+** GNU Lesser General Public License Usage
+** 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, Nokia gives you certain additional
+** rights. These rights are described in the Nokia 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.
+**
+** Other Usage
+** Alternatively, this file may be used in accordance with the terms and
+** conditions contained in a signed written agreement between you and Nokia.
+**
+**
+**
+**
+**
+**
+** $QT_END_LICENSE$
+**
+****************************************************************************/
+
+#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;
+};