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simplify audio code, allow passing of Head object

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This commit is contained in:
Stephen Birarda 2013-01-22 17:56:12 -08:00
parent 8a7db1f63c
commit 4d914128ea
6 changed files with 97 additions and 439 deletions
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2
Source/SerialInterface.h
View file

@ -10,7 +10,7 @@ int init_port (int baud);
int read_sensors(int first_measurement, float * avg_adc_channels, int * adc_channels, int * samples_averaged, int * LED_state);
#define NUM_CHANNELS 6
#define SERIAL_PORT_NAME "/dev/tty.usbmodem1411"
#define SERIAL_PORT_NAME "/dev/tty.usbmodemfa141"
// Acceleration sensors, in screen/world coord system (X = left/right, Y = Up/Down, Z = fwd/back)
#define ACCEL_X 4

354
Source/audio.cpp
View file

@ -2,49 +2,36 @@
// audio.cpp
// interface
//
// Created by Seiji Emery on 9/2/12.
// Copyright (c) 2012 __MyCompanyName__. All rights reserved.
// Created by Stephen Birarda on 1/22/13.
// Copyright (c) 2013 Rosedale Lab. All rights reserved.
//
/**
* @file audio.cpp
* Low level audio i/o portaudio wrapper.
*
* @author Seiji Emery
*
*/
#include <cstdlib>
#include <cstdio>
#include <cstring>
#include "audio.h"
// static member definitions
// (required – will cause linker errors if left out...):
bool Audio::initialized;
Audio::AudioData *Audio::data;
PaStream *Audio::stream;
PaError Audio::err;
float Audio::AudioData::inputGain;
PaStream *Audio::stream;
Audio::AudioData *Audio::data;
/**
* Audio callback used by portaudio.
* Communicates with Audio via a shared pointer to Audio::data.
* Writes input audio channels (if they exist) into Audio::data->buffer,
multiplied by Audio::data->inputGain.
multiplied by Audio::data->inputGain.
* Then writes Audio::data->buffer into output audio channels, and clears
the portion of Audio::data->buffer that has been read from for reuse.
the portion of Audio::data->buffer that has been read from for reuse.
*
* @param[in] inputBuffer A pointer to an internal portaudio data buffer containing data read by portaudio.
* @param[out] outputBuffer A pointer to an internal portaudio data buffer to be read by the configured output device.
* @param[in] frames Number of frames that portaudio requests to be read/written.
(Valid size of input/output buffers = frames * number of channels (2) * sizeof data type (float)).
(Valid size of input/output buffers = frames * number of channels (2) * sizeof data type (float)).
* @param[in] timeInfo Portaudio time info. Currently unused.
* @param[in] statusFlags Portaudio status flags. Currently unused.
* @param[in] userData Pointer to supplied user data (in this case, a pointer to Audio::data).
Used to communicate with external code (since portaudio calls this function from another thread).
Used to communicate with external code (since portaudio calls this function from another thread).
* @return Should be of type PaStreamCallbackResult. Return paComplete to end the stream, or paContinue to continue (default).
Can be used to end the stream from within the callback.
Can be used to end the stream from within the callback.
*/
int audioCallback (const void *inputBuffer,
@ -52,83 +39,19 @@ int audioCallback (const void *inputBuffer,
unsigned long frames,
const PaStreamCallbackTimeInfo *timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData)
void *userData)
{
Audio::AudioData *data = (Audio::AudioData*)userData;
float *input = (float*)inputBuffer;
float *output = (float*)outputBuffer;
Audio::AudioData *data = (Audio::AudioData *) userData;
int16_t *input = (int16_t *) inputBuffer;
int16_t *output = (int16_t *) outputBuffer;
#if WRITE_AUDIO_INPUT_TO_OUTPUT
if (input != NULL) {// && Audio::writeAudioInputToOutput) {
// combine input into data buffer
// temp variables (frames and bufferPos need to remain untouched so they can be used in the second block of code)
unsigned int f = (unsigned int)frames,
p = data->bufferPos;
for (; p < data->bufferLength && f > 0; --f, ++p) {
#if WRITE_AUDIO_INPUT_TO_BUFFER
data->buffer[p].l +=
data->inputBuffer[p].l = (*input++) * data->inputGain;
data->buffer[p].r +=
data->inputBuffer[p].r = (*input++) * data->inputGain;
#else
data->buffer[p].l += (*input++) * data->inputGain;
data->buffer[p].r += (*input++) * data->inputGain;
#endif
}
if (f > 0) {
// handle data->buffer wraparound
for (p = 0; f > 0; --f, ++p) {
#if WRITE_AUDIO_INPUT_TO_BUFFER
data->buffer[p].l +=
data->inputBuffer[p].l = (*input++) * data->inputGain;
data->buffer[p].r +=
data->inputBuffer[p].r = (*input++) * data->inputGain;
#else
data->buffer[p].l += (*input++) * data->inputGain;
data->buffer[p].r += (*input++) * data->inputGain;
#endif
}
}
// check if we have input data
if (input != NULL) {
memcpy(data->buffer, input, data->bufferLength * 2);
}
#elif WRITE_AUDIO_INPUT_TO_BUFFER
if (input != NULL) {// && Audio::writeAudioInputToBuffer) {
unsigned int f = (unsigned int)frames,
p = data->bufferPos;
for (; p < data->bufferLength && f > 0; --f, ++p) {
data->inputBuffer[p].l = (*input++) * data->inputGain;
data->inputBuffer[p].r = (*input++) * data->inputGain;
}
if (f > 0) {
// handle data->buffer wraparound
for (p = 0; f > 0; --f, ++p) {
data->inputBuffer[p].l = (*input++) * data->inputGain;
data->inputBuffer[p].r = (*input++) * data->inputGain;
}
}
}
#endif
// Write data->buffer into outputBuffer
if (data->bufferPos + frames >= data->bufferLength) {
// wraparound: write first section (end of buffer) first
// note: buffer is just an array of a struct of floats, so it can be typecast to float*
memcpy(output, (float*)(data->buffer + data->bufferPos), // write data buffer
(data->bufferLength - data->bufferPos) * 2 * sizeof(float));
memset((float*)(data->buffer + data->bufferPos), 0, // clear data buffer
(data->bufferLength - data->bufferPos) * 2 * sizeof(float));
frames -= (data->bufferLength - data->bufferPos); // adjust frames to be written
data->bufferPos = 0; // reset position to start
}
memcpy(output, (float*)(data->buffer + data->bufferPos), // write data buffer
frames * 2 * sizeof(float));
memset((float*)(data->buffer + data->bufferPos), 0, // clear data buffer
frames * 2 * sizeof(float));
data->bufferPos += frames; // update position
memcpy(output, data->buffer, data->bufferLength * 2);
return paContinue;
}
@ -137,25 +60,32 @@ int audioCallback (const void *inputBuffer,
* Should be called at the beginning of program exection.
* @seealso Audio::terminate
* @return Returns true if successful or false if an error occurred.
Use Audio::getError() to retrieve the error code.
Use Audio::getError() to retrieve the error code.
*/
bool Audio::init()
bool Audio::init()
{
return Audio::init(NULL);
}
bool Audio::init(Head* mainHead)
{
initialized = true;
data = new AudioData();
data->linkedHead = mainHead;
err = Pa_Initialize();
if (err != paNoError) goto error;
err = Pa_OpenDefaultStream(&stream,
err = Pa_OpenDefaultStream(&stream,
1, // input channels
1, // output channels
paFloat32, // sample format
paInt16, // sample format
22050, // sample rate (hz)
512, // frames per buffer
audioCallback, // callback function
(void*)data); // user data to be passed to callback
(void *) data); // user data to be passed to callback
if (err != paNoError) goto error;
err = Pa_StartStream(stream);
@ -174,221 +104,29 @@ error:
* Close the running audio stream, and deinitialize portaudio.
* Should be called at the end of program execution.
* @return Returns true if the initialization was successful, or false if an error occured.
The error code may be retrieved by Audio::getError().
The error code may be retrieved by Audio::getError().
*/
bool Audio::terminate ()
{
if (!initialized)
if (!initialized) {
return true;
initialized = false;
// err = Pa_StopStream(stream);
// if (err != paNoError) goto error;
} else {
initialized = false;
err = Pa_CloseStream(stream);
if (err != paNoError) goto error;
delete data;
err = Pa_Terminate();
if (err != paNoError) goto error;
return true;
}
err = Pa_CloseStream(stream);
if (err != paNoError) goto error;
delete data;
err = Pa_Terminate();
if (err != paNoError) goto error;
return true;
error:
fprintf(stderr, "-- portaudio termination error --\n");
fprintf(stderr, "PortAudio error (%d): %s\n", err, Pa_GetErrorText(err));
return false;
}
/**
* Write a stereo audio stream (float*) to the audio buffer.
* Values should be clamped between -1.0f and 1.0f.
* @param[in] offset Write offset from the start of the audio buffer.
* @param[in] length Length of audio channels to be read.
* @param[in] left Left channel of the audio stream.
* @param[in] right Right channel of the audio stream.
*/
void Audio::writeAudio (unsigned int offset, unsigned int length, float const *left, float const *right) {
if (data->buffer == NULL)
return;
if (length > data->bufferLength) {
fprintf(stderr, "Audio::writeAudio length exceeded (%d). Truncating to %d.\n", length, data->bufferLength);
length = data->bufferLength;
}
unsigned int p = data->bufferPos + offset;
if (p > data->bufferLength)
p -= data->bufferLength;
for (; p < data->bufferLength && length > 0; --length, ++p) {
data->buffer[p].l = *left++;
data->buffer[p].r = *right++;
}
if (length > 0) {
p = 0;
for (; length > 0; --length, ++p) {
data->buffer[p].l = *left++;
data->buffer[p].r = *right++;
}
}
}
/**
* Write a repeated stereo sample (float) to the audio buffer.
* Values should be clamped between -1.0f and 1.0f.
* @param[in] offset Write offset from the start of the audio buffer.
* @param[in] length Length of tone.
* @param[in] left Left component.
* @param[in] right Right component.
*/
void Audio::writeTone (unsigned int offset, unsigned int length, float const left, float const right) {
if (data->buffer == NULL)
return;
if (length > data->bufferLength) {
fprintf(stderr, "Audio::writeTone length exceeded (%d). Truncating to %d.\n", length, data->bufferLength);
length = data->bufferLength;
}
unsigned int p = data->bufferPos + offset;
if (p > data->bufferLength)
p -= data->bufferLength;
for (; p < data->bufferLength && length > 0; --length, ++p) {
data->buffer[p].l = left;
data->buffer[p].r = right;
}
if (length > 0) {
p = 0;
for (; length > 0; --length, ++p) {
data->buffer[p].l = left;
data->buffer[p].r = right;
}
}
}
/**
* Write a stereo audio stream (float*) to the audio buffer.
* Audio stream is added to the existing contents of the audio buffer.
* Values should be clamped between -1.0f and 1.0f.
* @param[in] offset Write offset from the start of the audio buffer.
* @param[in] length Length of audio channels to be read.
* @param[in] left Left channel of the audio stream.
* @param[in] right Right channel of the audio stream.
*/
void Audio::addAudio (unsigned int offset, unsigned int length, float const *left, float const *right) {
if (data->buffer == NULL)
return;
if (length > data->bufferLength) {
fprintf(stderr, "Audio::addAudio length exceeded (%d). Truncating to %d.\n", length, data->bufferLength);
length = data->bufferLength;
}
unsigned int p = data->bufferPos + offset;
if (p > data->bufferLength)
p -= data->bufferLength;
for (; p < data->bufferLength && length > 0; --length, ++p) {
data->buffer[p].l += *left++;
data->buffer[p].r += *right++;
}
if (length > 0) {
p = 0;
for (; length > 0; --length, ++p) {
data->buffer[p].l += *left++;
data->buffer[p].r += *right++;
}
}
}
/**
* Write a repeated stereo sample (float) to the audio buffer.
* Sample is added to the existing contents of the audio buffer.
* Values should be clamped between -1.0f and 1.0f.
* @param[in] offset Write offset from the start of the audio buffer.
* @param[in] length Length of tone.
* @param[in] left Left component.
* @param[in] right Right component.
*/
void Audio::addTone (unsigned int offset, unsigned int length, float const left, float const right) {
if (data->buffer == NULL)
return;
if (length > data->bufferLength) {
fprintf(stderr, "Audio::writeTone length exceeded (%d). Truncating to %d.\n", length, data->bufferLength);
length = data->bufferLength;
}
unsigned int p = data->bufferPos + offset;
if (p > data->bufferLength)
p -= data->bufferLength;
for (; p < data->bufferLength && length > 0; --length, ++p) {
data->buffer[p].l += left;
data->buffer[p].r += right;
}
if (length > 0) {
p = 0;
for (; length > 0; --length, ++p) {
data->buffer[p].l += left;
data->buffer[p].r += right;
}
}
}
/**
* Clear a section of the audio buffer.
* @param[in] offset Offset from the start of the audio buffer.
* @param[in] length Length of section to clear.
*/
void Audio::clearAudio(unsigned int offset, unsigned int length) {
if (data->buffer == NULL)
return;
if (length > data->bufferLength) {
fprintf(stderr, "Audio::clearAudio length exceeded (%d). Truncating to %d.\n", length, data->bufferLength);
length = data->bufferLength;
}
unsigned int p = data->bufferPos + offset;
if (p > data->bufferLength)
p -= data->bufferLength;
if (length + p < data->bufferLength) {
memset((float*)(data->buffer + p), 0,
sizeof(float) * 2 * length);
} else {
memset((float*)(data->buffer + p), 0,
sizeof(float) * 2 * (data->bufferLength - p));
memset((float*)(data->buffer + p), 0,
sizeof(float) * 2 * (data->bufferLength + p - data->bufferLength));
}
}
/**
* Read audio input into the target buffer.
* @param[in] offset Offset from the start of the input audio buffer to read from.
* @param[in] length Length of the target buffer.
* @param[out] left Left channel of the target buffer.
* @param[out] right Right channel of the target buffer.
*/
void Audio::readAudioInput (unsigned int offset, unsigned int length, float *left, float *right) {
#if WRITE_AUDIO_INPUT_TO_BUFFER
if (data->inputBuffer == NULL)
return;
if (length + offset > data->bufferLength) {
fprintf(stderr, "Audio::readAudioInput length exceeded (%d + %d). Truncating to %d + %d.\n", offset, length, offset, data->bufferLength - offset);
length = data->bufferLength - offset;
}
unsigned int p = data->bufferPos + offset;
if (p > data->bufferLength)
p -= data->bufferLength;
for (; p < data->bufferLength && length > 0; --length, ++p) {
*left++ = data->inputBuffer[p].l;
*right++ = data->inputBuffer[p].r;
}
if (length > 0) {
p = 0;
for (; length > 0; --length, ++p) {
*left++ = data->inputBuffer[p].l;
*right++ = data->inputBuffer[p].r;
}
}
#else
return;
#endif
}

151
Source/audio.h
View file

@ -2,140 +2,60 @@
// audio.h
// interface
//
// Created by Seiji Emery on 9/2/12.
// Copyright (c) 2012 __MyCompanyName__. All rights reserved.
// Created by Stephen Birarda on 1/22/13.
// Copyright (c) 2013 Rosedale Lab. All rights reserved.
//
#ifndef interface_audio_h
#define interface_audio_h
#ifndef __interface__audio__
#define __interface__audio__
#include <iostream>
#include "portaudio.h"
#include "head.h"
// Note: main documentation in audio.cpp
/**
* If enabled, direct the audio callback to write the audio input buffer
* directly into the audio output buffer.
*/
#define WRITE_AUDIO_INPUT_TO_OUTPUT 1
/**
* If enabled, create an additional buffer to store audio input
* and direct the audio callback to write the audio input to this buffer.
*/
#define WRITE_AUDIO_INPUT_TO_BUFFER 0
// Note: I initially used static const bools within the Audio class and normal
// 'if' blocks instead of preprocessor - under the assumption that the compiler
// would optimize out the redundant code.
// However, as that apparently did not work (for some reason or another - even
// with full compiler optimization turned on), I've switched to using preprocessor
// macros instead (which is probably faster anyways (at compile-time)).
/**
* Low level audio interface.
*
* Contains static methods that write to an internal audio buffer, which
is read from by a portaudio callback.
* Responsible for initializing and terminating portaudio. Audio::init()
and Audio::terminate() should be called at the beginning and end of
program execution.
*/
class Audio {
public:
// Initializes portaudio. Should be called at the beginning of program execution.
static bool init ();
// Deinitializes portaudio. Should be called at the end of program execution.
static bool terminate ();
// Write methods: write to internal audio buffer.
static void writeAudio (unsigned int offset, unsigned int length, float const *left, float const *right);
static void addAudio (unsigned int offset, unsigned int length, float const *left, float const *right);
static void writeTone (unsigned int offset, unsigned int length, float const left, float const right);
static void addTone (unsigned int offset, unsigned int length, float const left, float const right);
static void clearAudio (unsigned int offset, unsigned int length);
// Read data from internal 'input' audio buffer to an external audio buffer.
// (*only* works if WRITE_AUDIO_INPUT_TO_BUFFER is enabled).
static void readAudioInput (unsigned int offset, unsigned int length, float *left, float *right);
/**
* Set the audio input gain. (multiplier applied to mic input)
*/
static void setInputGain (float gain) {
data->inputGain = gain;
}
/**
* Get the internal portaudio error code (paNoError if none).
* Use in conjunction with Audio::init() or Audio::terminate(), as it is not
impacted by any other methods.
*/
const PaError getError () { return err; }
// initializes audio I/O
static bool init();
static bool init(Head* mainHead);
// terminates audio I/O
static bool terminate();
private:
/**
* Set to true by Audio::init() and false by Audio::terminate().
* Used to prevent Audio::terminate() from deleting uninitialized memory.
*/
static bool initialized;
/**
* Internal audio data.
* Used to communicate with the audio callback code via a shared pointer.
*/
static struct AudioData {
/**
* Internal (stereo) audio buffer.
* Written to by Audio I/O methods and the audio callback.
* As this is a ring buffer, it should not be written to directly – thus methods
like Audio::writeAudio are provided.
*/
struct BufferFrame{
float l, r;
} *buffer, *inputBuffer;
/**
* Length of the audio buffer.
*/
const static unsigned int bufferLength = 1000;
/**
* Current position (start) within the ring buffer.
* Updated by the audio callback.
*/
unsigned int bufferPos;
/**
* Audio input gain (multiplier applied to the incoming audio stream).
* Use Audio::setInputGain() to modify this.
*/
static float inputGain;// = 1.f;
// struct for left/right data in audio buffer
struct BufferFrame {
int16_t left, right;
} *buffer;
AudioData () : bufferPos(0) {
inputGain = 1.0f;
Head* linkedHead;
// length in bytes of audio buffer
const static unsigned int bufferLength = 1024;
AudioData() {
// alloc memory for buffer
buffer = new BufferFrame[bufferLength];
memset((float*)buffer, 0, sizeof(float) * bufferLength * 2);
#if WRITE_AUDIO_INPUT_TO_BUFFER
inputBuffer = new BufferFrame[bufferLength];
memset((float*)inputBuffer, 0, sizeof(float) * bufferLength * 2);
#else
inputBuffer = NULL;
#endif
memset(buffer, 0, sizeof(int16_t) * bufferLength * 2);
}
~AudioData () {
~AudioData() {
delete[] buffer;
#if WRITE_AUDIO_INPUT_TO_BUFFER
delete[] inputBuffer;
#endif
}
}*data;
/**
* Internal audio stream handle.
*/
static PaStream *stream;
/**
* Internal error code (used only by Audio::init() and Audio::terminate()).
*/
} *data;
// protects constructor so that public init method is used
Audio();
// hold potential error returned from PortAudio functions
static PaError err;
Audio (); // prevent instantiation (private constructor)
// audio stream handle
static PaStream *stream;
// give access to AudioData class from audioCallback
friend int audioCallback (const void*, void*, unsigned long, const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags, void*);
};
@ -147,5 +67,4 @@ int audioCallback (const void *inputBuffer,
PaStreamCallbackFlags statusFlags,
void *userData);
#endif
#endif /* defined(__interface__audio__) */

2
Source/cube.cpp
View file

@ -9,7 +9,7 @@
#include "cube.h"
#define MAX_CUBES 250000
#define SMALLEST_CUBE 0.005
#define SMALLEST_CUBE 0.2
float cubes_position[MAX_CUBES*3];
float cubes_scale[MAX_CUBES];

15
Source/main.cpp
View file

@ -103,7 +103,7 @@ ParticleSystem balls(0,
);
Cloud cloud(0, // Particles
Cloud cloud(50000, // Particles
box, // Bounding Box
false // Wrap
);
@ -310,7 +310,11 @@ void init(void)
int i;
if (audio_on) {
Audio::init();
if (serial_on) {
Audio::init(&myHead);
} else {
Audio::init();
}
printf( "Audio started.\n" );
}
@ -609,11 +613,11 @@ void display(void)
glDisable(GL_DEPTH_TEST);
glDisable(GL_LIGHTING);
lattice.render(WIDTH, HEIGHT);
// lattice.render(WIDTH, HEIGHT);
//drawvec3(100, 100, 0.15, 0, 1.0, 0, myHead.getPos(), 0, 1, 0);
glPointParameterfvARB( GL_POINT_DISTANCE_ATTENUATION_ARB, pointer_attenuation_quadratic );
myFinger.render();
// myFinger.render();
if (mouse_pressed == 1)
{
@ -794,9 +798,6 @@ void key(unsigned char k, int x, int y)
float add[] = {0.001, 0.001, 0.001};
field_add(add, pos);
}
if ((k == 't') && (audio_on)) {
Audio::writeTone(0, 400, 1.0f, 0.5f);
}
if (k == '1')
{
myHead.SetNewHeadTarget((randFloat()-0.5)*20.0, (randFloat()-0.5)*20.0);

12
interface.xcodeproj/project.pbxproj
View file

@ -8,7 +8,6 @@
/* Begin PBXBuildFile section */
5325C25016AF4DBE0051A40B /* agent.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C22916AF4DBE0051A40B /* agent.cpp */; };
5325C25116AF4DBE0051A40B /* audio.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C22B16AF4DBE0051A40B /* audio.cpp */; };
5325C25216AF4DBE0051A40B /* cloud.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C22D16AF4DBE0051A40B /* cloud.cpp */; };
5325C25316AF4DBE0051A40B /* cube.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C22F16AF4DBE0051A40B /* cube.cpp */; };
5325C25416AF4DBE0051A40B /* field.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C23116AF4DBE0051A40B /* field.cpp */; };
@ -25,6 +24,7 @@
5325C25F16AF4DBE0051A40B /* SerialInterface.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C24916AF4DBE0051A40B /* SerialInterface.cpp */; };
5325C26016AF4DBE0051A40B /* texture.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C24B16AF4DBE0051A40B /* texture.cpp */; };
5325C26116AF4DBE0051A40B /* util.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C24D16AF4DBE0051A40B /* util.cpp */; };
5325C26416AF4E2C0051A40B /* audio.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5325C26216AF4E2C0051A40B /* audio.cpp */; };
532C7AF216AF298D00B1A969 /* CVBlob.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 532C792A16AF298900B1A969 /* CVBlob.framework */; };
532C7CCC16AF301E00B1A969 /* grayson-particle.png in CopyFiles */ = {isa = PBXBuildFile; fileRef = 532C7AC316AF298D00B1A969 /* grayson-particle.png */; };
532C7CCD16AF301E00B1A969 /* int-texture256-v2.png in CopyFiles */ = {isa = PBXBuildFile; fileRef = 532C7AC416AF298D00B1A969 /* int-texture256-v2.png */; };
@ -81,8 +81,6 @@
/* Begin PBXFileReference section */
5325C22916AF4DBE0051A40B /* agent.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = agent.cpp; sourceTree = "<group>"; };
5325C22A16AF4DBE0051A40B /* agent.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = agent.h; sourceTree = "<group>"; };
5325C22B16AF4DBE0051A40B /* audio.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = audio.cpp; sourceTree = "<group>"; };
5325C22C16AF4DBE0051A40B /* audio.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = audio.h; sourceTree = "<group>"; };
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5325C22F16AF4DBE0051A40B /* cube.cpp */,
@ -619,6 +617,8 @@
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5325C24F16AF4DBE0051A40B /* world.h */,
5325C26216AF4E2C0051A40B /* audio.cpp */,
5325C26316AF4E2C0051A40B /* audio.h */,
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