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GEAudioBuffer.cpp Example File

demos/mobile/quickhit/ga_src/GEAudioBuffer.cpp
 /****************************************************************************
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 ** Copyright (C) 2011 Nokia Corporation and/or its subsidiary(-ies).
 ** All rights reserved.
 ** Contact: Nokia Corporation (qt-info@nokia.com)
 **
 ** This file is part of the QtDeclarative module of the Qt Toolkit.
 **
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 **
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 ** 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.
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 **     the names of its contributors may be used to endorse or promote
 **     products derived from this software without specific prior written
 **     permission.
 **
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 #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;
 };
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