Serpentinista

iPhone Development

2011-06-27T13:16:00.000-07:00

Adding an overlay to an image and saving the composite.

#import 
....
#pragma mark -
- (void) imagePickerController:(UIImagePickerController *)picker didFinishPickingMediaWithInfo:(NSDictionary *)info 
{   
    imageView.image = [info objectForKey:@"UIImagePickerControllerOriginalImage"];
    OverlayView *overlay = [[OverlayView alloc] initWithFrame:CGRectMake(0, 0, SCREEN_WIDTH, SCREEN_HEIGTH)];
    overlay.alpha = 0.6;  // Customize the opacity of the top image.
    [imageView addSubview:overlay];
    
    CGContextRef context = UIGraphicsGetCurrentContext();
    //UIImage *imageCopy = UIGraphicsGetImageFromCurrentImageContext();
    UIGraphicsEndImageContext();
    // Save image
    UIImageWriteToSavedPhotosAlbum(imageView.image, self,                                    @selector(image:didFinishSavingWithError:contextInfo:), context);
    [picker dismissModalViewControllerAnimated:YES];
}

Horizontal scroll

Multithread Portaudio

2011-04-05T13:00:00.001-07:00

test

The Art of cyWar

2011-04-03T14:57:00.000-07:00

http://www.nytimes.com/2010/02/23/books/23watchers.html

operations like Able Danger, a data-mining program at the Pentagon that became briefly notorious because of the erroneous claim by a few military officials who worked on it that it had been able to identify Mohammed Atta, the 9/11 hijacker, as a possible threat before the attacks.

“The T.I.A. tools crashed. They were simply incapable of processing so much information in real time. Like balloons affixed to a fire hydrant, they burst.”
--
The Watchers tells the story of these programs and the bureaucratic conflicts that evolved as the Intelligence Community tried to deal with the terrorist threat.
--
http://www.time.com/time/nation/article/0,8599,1973131,00.html
--

Reading materials will be identified throughout the course and made available via
hard copy or online reference.
Required reading includes The Watchers: The Rise of America’s Surveillance State,
by Shane Harris, ISBN 978-1-59420-245-2, Penguin Press, 2010.
Recommend reference text (not required): Introduction to Computer Security, Matt
Bishop, ISBN 0-321-24744-2, Addison-Wesley, 2005.
Recommend reference text (not required): Computer Networks, Andrew S.
Tanenbaum, ISBN 0130661023, Prentice Hall, 4th Edition, 2002.

Qt and Sound generation using libraries from audres.org

2011-03-16T11:55:00.000-07:00

Qt supports sound in various ways and Nokia/Trolltech is currently working options that make sound processing easier to work with. One of the methods to do playback is to use the Phonon module. This module operation using the gstreamer libraries, but does not allow you to control individual channels.

Computer Programming

2011-01-01T15:02:00.000-08:00

OpenCL

Audio Programming on Mac OS and Linux

2010-11-16T04:41:00.000-08:00

Portaudio And Libsndfile WAV playback with 
channel selection
/** @file pa_lsf_lessblocking.c
    @ingroup test_src
    @brief Read WAV file and playback on one channel if
         stereo file. Implemented using the blocking API
         (Pa_ReadStream(), Pa_WriteStream() ), but still
         allowing you to stop the stream if using threads.
    @author Sal Aguinaga http://serpentinista.blogspot.com
    @author Phil Burk  http://www.softsynth.com
    @author Ross Bencina rossb@audiomulch.com
*/
#include 
#include 
#include "portaudio.h"
#include 
/* #define SAMPLE_RATE  (17932) // Test failure to open with this value. */
#define SAMPLE_RATE  (44100)
#define FRAMES_PER_BUFFER (1024)
#define LATENCY_MSEC (2000) /* new */
#define NUM_SECONDS     (5)
#define NUM_CHANNELS    (2)
/* #define DITHER_FLAG     (paDitherOff)  */
#define DITHER_FLAG     (0) /**/
/* Select sample format. */
#if 1
#define PA_SAMPLE_TYPE  paFloat32
typedef float SAMPLE;
#define SAMPLE_SILENCE  (0.0f)
#define PRINTF_S_FORMAT "%.8f"
#elif 1
#define PA_SAMPLE_TYPE  paInt16
typedef short SAMPLE;
#define SAMPLE_SILENCE  (0)
#define PRINTF_S_FORMAT "%d"
#elif 0
#define PA_SAMPLE_TYPE  paInt8
typedef char SAMPLE;
#define SAMPLE_SILENCE  (0)
#define PRINTF_S_FORMAT "%d"
#else
#define PA_SAMPLE_TYPE  paUInt8
typedef unsigned char SAMPLE;
#define SAMPLE_SILENCE  (128)
#define PRINTF_S_FORMAT "%d"
#endif
/*******************************************************************/
//typedef struct  {
// float outputBuffer;
// int left_phase;
// int right_phase;
//} paTestData
/*******************************************************************/
int main(int argc, char *argv[]);
/*******************************************************************/
int main(int argc, char *argv[])
{
 printf("libsnd portaudio playback\n"); fflush(stdout);
 if (argc != 2) {
  fprintf(stderr, "Expecting wav file as argument\n");
  return 1;
 }
 
    PaStreamParameters /*inputParameters,*/ outputParameters;
    PaStream *stream;
    PaError err;
    SAMPLE *recordedSamples;
    int i;
    int totalFrames;
    int numSamples;
    int numBytes;
    SAMPLE max, average, val;
 float *buffer;
    
    // Open sound file
 SF_INFO sndInfo;
 SNDFILE *sndFile = sf_open(argv[1], SFM_READ, &sndInfo);
 if (sndFile == NULL) {
  fprintf(stderr, "Error reading source file '%s': %s\n", argv[1], sf_strerror(sndFile));
  return 1;
 }
    
 // Check format - 16bit PCM
 if (sndInfo.format != (SF_FORMAT_WAV | SF_FORMAT_PCM_16)) {
  fprintf(stderr, "Input should be 16bit Wav\n");
  sf_close(sndFile);
  return 1;
 }
 // Check channels 
 if (sndInfo.channels > 2) {
  fprintf(stderr, "Wrong number of channels\n");
  sf_close(sndFile);
  return 1;
 }
 
 if (sndInfo.channels == 1) {
  // Allocate memory
  buffer = malloc(sndInfo.frames * sizeof(float));
  //float *buffer = calloc(sndInfo.frames * sizeof(float));
  if (buffer == NULL) {
   fprintf(stderr, "Could not allocate memory for file\n");
   sf_close(sndFile);
   return 1;
  }
  
  
  // Load data
  long numFrames = sf_readf_float(sndFile, buffer, sndInfo.frames);
  
  // Check correct number of samples loaded
  if (numFrames != sndInfo.frames) {
   fprintf(stderr, "Did not read enough frames for source\n");
   sf_close(sndFile);
   free(buffer);
   return 1;
  }
  
  // Output Info
  printf(" Read %ld frames from %s,\n "
      "Channel(s):%d,\n "
      "Sample rate: %d,\n "
      "Length: %fs\n",
      numFrames, argv[1], sndInfo.channels, sndInfo.samplerate, (float)numFrames/sndInfo.samplerate);
  
  // Setup for Output
  totalFrames = (int)numFrames; /* */
  numSamples = totalFrames * sndInfo.channels;
  
  numBytes = numSamples * sizeof(SAMPLE);
  recordedSamples = (SAMPLE *) malloc( numBytes );
  if( recordedSamples == NULL )
  {
   printf("Could not allocate record array.\n");
   exit(1);
  }
  for( i=0; i 
  
  sf_close(sndFile);
  free(buffer);
 } else {  // 2 channels
     buffer = malloc(sndInfo.frames * 2 * sizeof(float));  // * 1 to put it on one channel.
  
  if (buffer == NULL) {
            fprintf(stderr, "Could not allocate memory for file\n");
            sf_close(sndFile);
            return 1;
  }
  
  long numFrames = sf_readf_float(sndFile, buffer, sndInfo.frames);
  
  if (numFrames != sndInfo.frames) {
            fprintf(stderr, "Did not read enough frames for source\n");
            sf_close(sndFile);
            free(buffer);
            return 1;
  }
  
  printf(" Read %ld frames from %s,\n "
      "Channel(s):%d,\n "
      "Sample rate: %d,\n "
      "Length: %fs\n",
      numFrames, argv[1], sndInfo.channels, sndInfo.samplerate, (float)numFrames/sndInfo.samplerate);
  
  // Setup for Output
  totalFrames = (int)numFrames; /* */
  numSamples = totalFrames * sndInfo.channels;
//  numSamples = totalFrames / 2; // changed it to 1 channel 
  /*************/
//  double duration = (double)numFrames/sndInfo.samplerate;
//  double *singleChanOutput = (double *) malloc (numFrames * sizeof(doube));
//  if (singleChanOutput == NULL) {
//   fprintf(stderr, "Could not allocate buffer for output\n");
//  }
  /*************/
//  numBytes = numSamples * sizeof(SAMPLE);
  recordedSamples = (SAMPLE *) malloc( numSamples * sizeof(SAMPLE));
  if( recordedSamples == NULL )
  {
   printf("Could not allocate record array.\n");   
   exit(1);
  }
  
  printf("numFrames = %d, numSamples =%d\n", (int)numFrames, numSamples);
  for( i=0; i2; i++ ) {
   recordedSamples[i*2] = buffer[i*2];
   recordedSamples[i*2+1] = 0;
  }
        sf_close(sndFile);
        free(buffer);  
  printf("totalFrames = %d\n"
      "numSamples = %d\n"
      ,totalFrames, numSamples);
 }
  
    // totalFrames = NUM_SECONDS * SAMPLE_RATE; /* Record for a few seconds. */
 //  numSamples = totalFrames * NUM_CHANNELS;
 
    err = Pa_Initialize();
    if( err != paNoError ) goto error;
 
    /* Measure maximum peak amplitude. */
    max = 0;
    average = 0;
    for( i=0; i
    {
        val = recordedSamples[i];
        if( val < 0 ) val = -val; /* ABS */
        if( val > max )
        {
            max = val;
        }
        average += val;
    }
    average = average / numSamples;
    printf("Sample max amplitude = "PRINTF_S_FORMAT"\n", max );
    printf("Sample average = "PRINTF_S_FORMAT"\n", average );
/*  Was as below. Better choose at compile time because this
    keeps generating compiler-warnings:
    if( PA_SAMPLE_TYPE == paFloat32 )
    {
        printf("sample max amplitude = %f\n", max );
        printf("sample average = %f\n", average );
    }
    else
    {
        printf("sample max amplitude = %d\n", max );
        printf("sample average = %d\n", average );
    }
*/
/*####################################################################
 * Playback recorded data
 ********************************************************************/
    outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
    outputParameters.channelCount = sndInfo.channels;
    outputParameters.sampleFormat =  PA_SAMPLE_TYPE;
    outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
 outputParameters.hostApiSpecificStreamInfo = NULL;
    printf("Begin playback.\n"); fflush(stdout);
    err = Pa_OpenStream(
              &stream,
              NULL, /* no input */
              &outputParameters,
              sndInfo.samplerate,
              FRAMES_PER_BUFFER,
              paClipOff,      /* we won't output out of range samples so don't bother clipping them */
              NULL, /* no callback, use blocking API */
              NULL ); /* no callback, so no callback userData */
    if( err != paNoError ) goto error;
    if( stream )
    {
        err = Pa_StartStream( stream );
        if( err != paNoError ) goto error;
        printf("Waiting for playback to finish.\n"); fflush(stdout);
  for (i=0; i
   err = Pa_WriteStream( stream, &recordedSamples[i*sndInfo.channels], min(FRAMES_PER_BUFFER,totalFrames-i) );
   if( err != paNoError ) {
    goto error;
    break;
   }
   
  }
        err = Pa_CloseStream( stream );
        if( err != paNoError ) goto error;
        printf("Done.\n"); fflush(stdout);
    }
    free( recordedSamples );
 
    Pa_Terminate();
    return 0;
error:
    Pa_Terminate();
    fprintf( stderr, "An error occured while using the portaudio stream\n" );
    fprintf( stderr, "Error number: %d\n", err );
    fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
    return -1;
}

What the Phonon: How to select individual audio channels (left, right or both)

2010-09-21T13:36:00.000-07:00

I've just searched the Trolltech site for: Does Phonon allows you to control Left or Right audio channels and there is no mention of this, so before I go building a custom WAV playing device using the example: qmusicplayer .... Does it? How?

The blog post by Garet: http://labs.qt.nokia.com/2010/05/10/low-level-audio-processing-with-qtmultimedia/ titled:
Low-level audio processing with QtMultimedia
mentions the QAudioFormat::setChannels() ... but is this for generating audio 'data' and not for selecting the physical audio channel of a given audio device?

I am using this in Mac OS, and the 'capabilities' example only returns: "Internal Speakers" for what is called Available audio devices. If I try this in Linux, will it give me more options? But as I write this I realize that these capabilities still don't give me the granularity of choosing output channels (left or right, for 2-channel audio device), right or am I in the dark about some new features?

Would Motorola departure burst Chicago’s tech bubble?

2010-02-24T20:07:00.000-08:00

Regarding:
http://www.chicagotribune.com/business/ct-biz-0224-motorola-california-20100223,0,5853421.story?page=1
From the Chicago Tribune, By Wailin Wong, Tribune reporter
8:55 a.m. CST, February 24, 2010

My Letter to Wailin Wong, on Feb 24, 2010:

Hello Wailin,

What Sanjay Jha says about Motorola's employees is infuriatingly disrespectful to the people
that have help made Motorola what it was and is. When he refers to the "talent" I would
have liked to see this explored in more detail. Who is he specifically referring to, the
management and the folks that surround him or is he referring to the legions of exceedingly
talented engineers that made it --the type of company it has been over the years and decades?

It sounds to me like an easy copout to blame the problems on the talent. The bureaucracy
and the bad decisions at the top (where real and consequential decisions are made) should
certainly bare some of the responsibility.

If the mobile phone division moves out this area, so will the footprint of many of the supporting
companies that help and support the various divisions. This means a incalculable number of
potential job losses across various sectors.

It's true that the Chicagoland area has struggled over the years to maintain a strong presence
in the technology area, and so have others, pharma being the first that comes to mind. It's
my opinion that either the state of Illinois or the greater Chicagoland community (especially those in positions of power in influence) are to blame for the lack of vision in creating not only a self-sustaining high-tech industry, but from helping it grow.

The reason Motorola has strong ties to California is not because of its offices, but it's because is where key partners and suppliers are located, Qualcomm being one of them. The angle
the section about Sanjay and California angers me, because it can't be as simple as
capriciously wanting to move the division to California, just so he's closer to his family.
The article doesn't explore the perceived benefits in full, talent is just one piece, and
talent is lured by the seductive salaries, quality of life, and other factors.

The discussion on what would it takes to help Chicago follow the lead of the West and
West coasts, or even that of Austin-Dallas Texas tech corridor misses the mark. This is
no different that what many other states, such as Michigan, are trying to do. Those
interviewed and quoted in that section should look at the bigger picture and then
propose and help advance a vision of strong a high-tech heartland complemented by
the finance sectors and its world class universities' graduates. Building this is no
easy task, but let's not kid ourselves a little bit of government money and a little
bit of venture money here and there --without a plan for sustainability will be a bit
in a Geopbyte!

C++, Qt Multithreading

2010-02-07T13:55:00.000-08:00

Exploring multithreading using Qt's QThread to run the UI on the main thread and then a high priority audio function in a second thread.

This example consists of widget that builds the UI and offers running the audio within the main thread, and an audio-thread cpp file that takes care of the audio on a separate thread. The startButton calls the audio function with in the same thread, and the newstartButton calls the new audio thread:

AudKeyWidget::AudKeyWidget(QWidget *parent)
    : QWidget(parent)
{
 createUI();
 
 connect(startButton, SIGNAL(clicked()), this, 
            SLOT(test_tone()));
 connect(newstartButton, SIGNAL(clicked()), this,
            SLOT(conti_tone()));
 connect(quitButton , SIGNAL(clicked()), qApp, 
            SLOT(quit()));

 setWindowTitle(tr("AudKey"));
 resize(550, 400);
}

The conti_tone() function is defined as:

void AudKeyWidget::conti_tone()
{
 printf("Audio slot\n");

 thread = new cAudioThread(this);
 thread->start();
 
}

A key event is defined to stop the thread, but it's not working gracefully. The objective is to hit the Escape key and stop the thread. Below is the key event and function:

void AudKeyWidget::keyPressEvent(QKeyEvent *event)
{
    switch (event->key()) {
        case Qt::Key_Escape:
            stopAudio();
            break;
        default:
            QWidget::keyPressEvent(event);
    }
}
void AudKeyWidget::stopAudio()
{
    printf("Stop Audio\n");
/* 
    if (thread->isRunning()) {
        printf("New Thread Running\n");
        //thread->stop();
        //thread->wait();
    }
*/ 
}

In the above code, I'm trying to find a method to gracefully stop the thread if isRunning().
Hello!

Got audio working off a second thread and I am able to 'stop' it while it isRunning(),
but am I halting it gracefully, and what do I have to do to quit the app while thread is running?
A pushbutton is used to start the audio thread, then it is pressed again to Stop it.

connect(newstartButton, SIGNAL(clicked()), this, SLOT(startAudioThread()));
connect(quitButton , SIGNAL(clicked()), qApp, SLOT(quit())); // This doesn't work cleanly

The slot: I use t_stop bool flag to check if the button was pressed to stop the the thread.
If the thread is started, I change the txt of the button to read "Stop Thread"

void AudKeyWidget::startAudioThread()
{
 if (t_stop) { 
  printf("Thread is running\n");
  stopAudio();
 }
 else {
  thread = new cAudioThread(this);
  thread->start();
  t_stop = !t_stop;
  if(t_stop)
  newstartButton->setText("Stop Thread");
 }
}

If we are stopping the thread, then I handle this using function stopAudio().

void AudKeyWidget::stopAudio()
{ printf("Stop Audio\n");
 if (t_stop && thread->isRunning()) {
  thread->terminate(); 
  delete thread;
  t_stop = !t_stop;
 }
}

But using terminate() offers the proper way to halt the thread? & reset things?
now my Continuous audio thread (caudiothread), some of this code was taken from the
mandelbrot example.

cAudioThread::cAudioThread(QObject *parent)
    : QThread(parent)
{
    printf("-- cAudioThread(QObject *parent) : QThead(parent) ...\n");
    restart = false;
    abort = false;

}

cAudioThread::~cAudioThread()
{
    mutex.lock();
    abort = true;
    condition.wakeOne();
    mutex.unlock();

    wait();  // Does is make sense to use this if we will almost 
  // always halt the thread rather wait for it to end?

}
void cAudioThread::run()
{
 printf("-- Independent thread, run() function started\n"); 
 forever { 
  // ++++++++++++++++++ continuous audio (tone) ++++++++++++++++++ //
  blah, blah, blah
  // --------------------------- end continuous audio
  mutex.lock();
  if (!restart)
   condition.wait(&mutex);
  restart = false;
  mutex.unlock();
 }// ends forever

}

Finally the caudiothread.h, I would be interested in how to quit while running and if stopAudio offers a clean approach to halt the thread.

class cAudioThread : public QThread
{
    Q_OBJECT

public:
    cAudioThread(QObject *parent = 0);
    ~cAudioThread();

    void playAudio( int );

protected:
    void run();

private:

    QMutex mutex;
    QWaitCondition condition;
    bool restart;
    bool abort;

};

Mac OSX Audio and C++

2009-10-06T05:29:00.000-07:00

I'm trying to program a way to extract the audio devices installed on my system, but I'm running into problems. I want to write a simple function using Xcode and in C++.

#include coreaudio/audiohardware.h

#include

#include '<'math.h'>'

int main (int argc, const char *argv[])

{

OSStatus status;

UInt32 sz;

AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices,&sz,NULL);

int nDevices = sz/sizeof(AudioDeviceID);

AudioDeviceID *devids = new AudioDeviceID[nDevices];

AudioHardwareGetProperty(kAudioHardwarePropertyDevices,&sz,devids);

printf("Device Count: %d\n", nDevices);

for(int i=0;i

{

char *s;

// get buffer list

UInt32 outputChannelCount=1;

status= AudioDeviceGetPropertyInfo(devids[i],0,false,
kAudioDevicePropertyStreamConfiguration,&sz,NULL);

AudioBufferList *bufferList=(AudioBufferList *)malloc(sz);

status= AudioDeviceGetProperty(devids[i],0,false,
kAudioDevicePropertyStreamConfiguration, &sz,&bufferList);

if (!status) {

for(int j=0;jmNumberBuffers;++j)

outputChannelCount += bufferList->mBuffers[j].mNumberChannels;

}

free(bufferList);

printf("Output Chan Count: %d\n", outputChannelCount);

// skip devices without any output channels

if(outputChannelCount==0)

// output some device info

sz=sizeof(CFStringRef);

AudioDeviceGetProperty(

devids[i],0,false,

kAudioDevicePropertyDeviceUID,

&sz,&s

);

printf("DeviceUID: [%s]\n",s);

//free(s);

AudioDeviceGetProperty(

devids[i],0,false,

kAudioObjectPropertyName,

&sz,&s

);

printf(" Name: [%s]\n",s);

//free(s);

//printf(" OutputChannels: %d",outputChannelCount);

printf(" i: %d\n",i);

}

return 0;

}

C++ Debug Methods

2009-08-28T14:18:00.000-07:00

I've been writing C++ code for under a year and I've gotten to a point where including lines for debugging is needed, plus I feel I need to make it available for others that are working on my code. I had started using simple things such as simple print statements or using the Qt qDebug function, but I recently looked my friend Dr. Steve Neely's, from Boystown, method and this is what he is using in his C source:

e = getenv("ARSC_DEBUG");
if (e){
debug = atoi(e);
}
nd = _ar_alsa_num_dev();
if (debug) {
fprintf(stderr, "ARSC: debug=%d, nd=%ld\n", debug, nd);
}

In this entry various methods will be investigated and presented.

Open Source Schematic Capture and Layout

2009-07-20T12:47:00.000-07:00

Issues with Footprints

The footprint for the Texas Instruments part: TPA6120A2 calls for a thermal pad and the bottom half of the pins to be connected to ground. A problem has risen with footprint, DRC errors highlight the pins in GEDAs pcb tool.

I have configured the symbol and called for the rest of the pins to be connected to ground, but an error flag persists. What is the best way to troubleshoot this issue?

The symbol has a property called: GND=6,7,8,9,10,11,12,13,15,21

My slotdefs are set: slotdef=1:1,2,3,4,5 and slotdef=2:20,19,18,17,16

So why does pcb highlight these pins when I connect them to GND?

Below are the symbol and layout footprints.

Post Layout Checks

Documentation

A very good source for documentation on pcb can be found in the following link:

http://geda.seul.org/wiki/geda:pcb_tips?do=export_raw%22

Figures:

MacBook Pro - Xubuntu

2008-12-18T11:26:00.000-08:00

Running Xubuntu on a MacBook Pro

Audio Issues

In general, I'm still having problems with the control of the devices that are part of the Intel HDA sound card. I can control the main speakers, but the headphone has been tricky and I am nowhere touching the SPDIF interface. The new issue is firewire audio devices. I downloaded and installed ffado and jackd and they don't work and have not been able to find a solution.

Sound on Apple MacPro and MacBook Pro

2008-10-27T08:23:00.000-07:00

MacBook Pro

After installing the alsa drivers, for sound development, the audio stopped working.
sudo lspci | grep Audio
00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)

aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC885 Analog [ALC885 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC885 Digital [ALC885 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0

-------------------------------------
chavo@tesla:~$ cat /etc/modprobe.d/alsa-base | grep model
chavo@tesla:~$ ls /etc/modprobe.d/al*
/etc/modprobe.d/aliases /etc/modprobe.d/alsa-base
chavo@tesla:~$ head -n 1 /proc/asound/card0/codec*
Codec: Realtek ALC889A
chavo@tesla:~$ zless /usr/share/doc
doc/ doc-base/
chavo@tesla:~$ zless /usr/share/doc/alsa-base/driver/A
ALSA-Configuration.txt.gz Audigy-mixer.txt.gz Audiophile-Usb.txt.gz
chavo@tesla:~$ zless /usr/share/doc/alsa-base/driver/ALSA-Configuration.txt.gz
chavo@tesla:~$ zless /usr/share/doc/alsa-base/driver/ALSA-Configuration.txt.gz
chavo@tesla:~$ zless /usr/share/doc/alsa-base/driver/ALSA-Configuration.txt.gz
chavo@tesla:~$ cat /etc/modprobe.d/alsa-base > /home/chavo/Documents/alsa_base_after_alsa_upgrade

added the option model to: chavo@tesla:~$ sudo vim /etc/modprobe.d/alsa-base
Reference: http://ubuntuforums.org/showthread.php?t=806620

The model=mbp3 did not work, but using options snd-hda-intel model=imac24, worked ok for the main speakers.

Mac Pro (mbp3) asoundrc

On the Mac Pro (Fedora 9 installation) Pulse Audio was installed, but still exhibited problems. The original .asoundrc ( in the home directory ) configured as:

pcm.pulse{
type pulse
}

ctl.pulse {
type pulse
}

pcm.phononpulse {
type plug
slave.pcm {
type pulse
}
hint {
show on
description "PulseAudio"
}
}

I created a the "sound" file under /etc/modprob.d/ but for the Mac Pro laptop, the same did not work. I changed it to "alsa-base" and that got my inputs (microphone and line-in) to work.

NFS Server Setup on Mac Pro Tower

2008-08-29T14:28:00.000-07:00

FEDORA 9 - NFS Server Setup
===========================

NFS Server
==========
/etc/exports
/usr/local 76.223.xx.xx(ro)
/home 76.223.xx.xx(rw,insecure,sync)

/etc/hosts.allow
#
# hosts.allow This file contains access rules which are used to
# allow or deny connections to network services that
# either use the tcp_wrappers library or that have been
# started through a tcp_wrappers-enabled xinetd.
#
# See 'man 5 hosts_options' and 'man 5 hosts_access'
# for information on rule syntax.
# See 'man tcpd' for information on tcp_wrappers
#
mountd: mac_os_client0, linux_client0

/etc/hosts.deny
currently empty

Next, I typed the following:
/usr/sbin/exportfs -ra

Next, verify /proc/fs/nfs/exports
# Version 1.1
# Path Client(Flags) # IPs
It comes up empty

Next, I type the following:
/sbin/chkconfig --list nfs
nfs 0:off 1:off 2:off 3:off 4:off 5:on 6:off
rpm -qa | grep nfs
nfs-utils-1.1.2-2.fc9.x86_64
system-config-nfs-1.3.40-1.fc9.noarch
nfs-utils-lib-1.1.1-5.fc9.x86_64
nfs-utils-lib-1.1.1-3.fc9.x86_64
nfs-utils-1.1.2-6.fc9.x86_64

/sbin/chkconfig portmap on
error reading information on service portmap: No such file or directory

/sbin/service nfs restart
Shutting down NFS mountd: [FAILED]
Shutting down NFS daemon: [FAILED]
Shutting down NFS quotas: [FAILED]
Shutting down NFS services: exportfs: could not open /var/lib/nfs/etab for locking
exportfs: can't lock /var/lib/nfs/etab for writing
[ OK ]
rm: cannot remove `/var/lock/subsys/nfs': Permission denied
Starting NFS services: exportfs: could not open /var/lib/nfs/etab for locking
exportfs: can't lock /var/lib/nfs/etab for writing
[ OK ]
Starting NFS quotas: [ OK ]
Starting NFS daemon: [FAILED]

but if you use
#sudo /sbin/service nfs restart, then they all pass.

Verify export mounts:
$ /usr/sbin/showmount -e
Export list for hostname.domain.xxx:
/home mac_os_client_ip, linux_client_ip
/usr/local mac_os_client_ip, linux_client_ip5

Firewall Configuration
From: http://blog.mottershead.me.uk
Run from the command line:
#system-config-firewall launches a gui (enter your root pwd)

Add to Trusted Services:

111 tcp sunrpc
111 udp sunrpc

Next, click on the Add again and User Defined box and enter each of the
following ports:

889 tcp
889 udp
890 tcp
890 udp

Next click on Apply and

Client: Mac OS
==============

When I type:
sudo mount_nfs server_ip:/home/ local_dir/

sudo mount -t nfs server_ip:/home/usr/Music /local_dir/

I keep getting error havind to do with RPC.

#rpcinfo -p nn.nn.nn.nn
rpcinfo: can't contact portmapper: RPC: Remote system error - Operation timed out

rpcinfo command also takes a long time to acknowledge or return the error.

Verifying the daemons running, I get the following
ps -aux | grep -E "nfsiod|nfsd|mountd|automount"
root 125 0.0 0.0 29308 512 ?? Ss 7:44AM 0:00.00 nfsiod -n 4
root 207 0.0 0.4 30336 3792 ?? Ss 7:44AM 0:02.53 /usr/sbin/automount -f -m /Network -
root 215 0.0 0.1 29420 1020 ?? Ss 7:44AM 0:00.29 /usr/sbin/automount -f -m /automount

I have followed the directions to setup an automount using netinfo on the Mac side,
but that did not return favorable results.

Useful links:
http://linux.derkeiler.com/Mailing-Lists/Fedora/2007-09/msg03343.html

/sysconfig/nfs
http://blog.mottershead.me.uk/blog/_archives/2008/7/17/3796637.html

http://physics.ramapo.edu/downloads/DarwinAndNFS.pdf

Useful links:
http://linux.derkeiler.com/Mailing-Lists/Fedora/2007-09/msg03343.html

/sysconfig/nfs
http://blog.mottershead.me.uk/blog/_archives/2008/7/17/3796637.html

Mandriva/Fedora on Mac Pro Quad-Core Intel Xenon - Part One

2008-08-07T09:34:00.000-07:00

Hardware Overview:
Mac Pro - MacPro3,1
Quad-Core Intel Xenon
2.8GHz
2 Processors
2GB of Memory

1. Install rEFIT
2. Using Boot Camp - Repartition the harddrive
- 32GB (actual 31.6GB)
3. Downloaded and burned to DVD mandriva-linux-free-2008-spring-dvd-x86_64.iso
4. Restart and when the system comes up check that that the partition tables are synced and no overlap problems are found with the partitions.
5. Shutdown

6. Hold the "option" key and turn on the Mac Pro, this will show you two harddrives and one CD, select the CD.

7. You will now boot into linux.

8. go through the entire installation, this section had a tricky part .. the deletion of 32G fat32 partition and writing over it.

I ended up doing everything by the book, I dedicated 30 gigs to "/" for the main installation and only 2gigs for swap. Then after going through the entire thing, the kernel did not load and it didn't boot. I ended up installing fedora9 and noticed that wha I had was two logical volumes, which I didn't understand. I'm not versed in the arts of logical volumes, so I deleted the logical volumes, deleted everything to end up back with a 32G partition. I again, dedicated 30 gig to "/" and the rest to the swap partition.

Update:

Problems with the installation of Mandriva were encountered. The problem had to do with kernel unable to mount/boot from the correct HD partitions. I had abandon this and instead installed fedora 9.

Fedora 9 on Mac Pro (Quad core)

So far the only problem is that I don't have audio working!!
Audio Update:
1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available"
2. aplay does not produce sound, period!
3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack.

I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found:

sudo cat /var/log/messages | grep ALSA
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped
Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped

Aug 15, 2008 Update:

================================================

Thanks to Stan from the alsa-project forum email list:
Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues
================================================
I think your problem is a misidentification of your sound chip. The
model being used for your 889A appears to be incompatible, thus all
the errors above. It appears that the macpro ALC885 is being selected
before it gets to the choice for the ALC889A. Perhaps a developer can
look into this.

!!HDA-Intel Codec information

!!---------------------------

Codec: Realtek ALC889A

!!Aplay/Arecord output

!!------------

APLAY

**** List of PLAYBACK Hardware Devices ****

card 0: Intel [HDA Intel], device 0: ALC885 Analog [ALC885 Analog]

Subdevices: 1/1

Subdevice #0: subdevice #0

One shows ALC889A and the other shows ALC885. In the driver code

there is a comment to the effect that the 889A is best handled as an

ALC888, yet the macpro is listed under the ALC885.

I am not a developer so can't go beyond this, but it seems that there

is a problem. There should be a macpro under the ALC888 as well as

the 885.

Here are the model options for the two candidates

ALC882/885

3stack-dig 3-jack with SPDIF I/O

6stack-dig 6-jack digital with SPDIF I/O

arima Arima W820Di1

targa Targa T8, MSI-1049 T8

asus-a7j ASUS A7J

asus-a7m ASUS A7M

macpro MacPro support

mbp3 Macbook Pro rev3

imac24 iMac 24'' with jack detection

w2jc ASUS W2JC

auto auto-config reading BIOS (default)

ALC883/888

3stack-dig 3-jack with SPDIF I/O

6stack-dig 6-jack digital with SPDIF I/O

3stack-6ch 3-jack 6-channel

3stack-6ch-dig 3-jack 6-channel with SPDIF I/O

6stack-dig-demo 6-jack digital for Intel demo board

acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)

acer-aspire Acer Aspire 9810

medion Medion Laptops

medion-md2 Medion MD2

targa-dig Targa/MSI

targa-2ch-dig Targs/MSI with 2-channel

laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)

lenovo-101e Lenovo 101E

lenovo-nb0763 Lenovo NB0763

lenovo-ms7195-dig Lenovo MS7195

haier-w66 Haier W66

3stack-hp HP machines with 3stack (Lucknow, Samba boards)

6stack-dell Dell machines with 6stack (Inspiron 530)

mitac Mitac 8252D

clevo-m720 Clevo M720 laptop series

fujitsu-pi2515 Fujitsu AMILO Pi2515

auto auto-config reading BIOS (default)

From the code, here is the comment indicating it should be treated as

an ALC888:

if (board_config <>= ALC882_MODEL_LAST) {

/* Pick up systems that don't supply PCI SSID */

switch (codec->subsystem_id) {

case 0x106b0c00: /* Mac Pro */

board_config = ALC885_MACPRO;

break;

case 0x106b1000: /* iMac 24 */

board_config = ALC885_IMAC24;

break;

case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */

case 0x106b2c00: /* Macbook Pro rev3 */

case 0x106b3600: /* Macbook 3.1 */

board_config = ALC885_MBP3;

break;

default:

/* ALC889A is handled better as ALC888-compatible */

if (codec->revision_id == 0x100103) {

alc_free(codec);

return patch_alc883(codec);

}

printk(KERN_INFO "hda_codec: Unknown model for ALC882, "

"trying auto-probe from BIOS...\n");

board_config = ALC882_AUTO;

}

------------------------------

Flashplayer on a 64bit installation of Fedora 9

Thanks to Hacktux: http://hacktux.com/fedora/9/flash

Contact me if you have any issues or questions.

IO Synchronization on Audio Devices Running on Various Linux Platforms

2008-07-15T11:22:00.000-07:00

Introduction

Linux Audio

IO Synchronization

What commercial audio devices offer the best audio IO synchronization for use in auditory research?

  On the mailing list: Research in Auditory Perception
  - Christian Borss (Inst. of Acoustic Communications of Ruhr University at Bochum (DE)
   RME multiface
  - Ivo Merks (Windows System
   RME multiface
  - Oliver Crouzet
   Edirol FA-101 and FA-66

  On www.ffado.org ( linux project working on firewire drivers )
  - Motu is hostile to linux comments

  On Ardour (digital audio workstation for linux and Mac OSX)
  - Paul Davis
   Basically there are no drivers and motu is hostile to linux
Recommended manufacturers (General no synch specific) : RME,M-Audio, Echo

  - Ardour FAQ Basic Audio I/O
What audio interfaces should I use with Ardour?

"You can use any interface supported on your platform (Linux or OS X).

For PCI /PCIe/etc. cards to install inside a computer, we recommend:
more than 26 channels
RME MADI
more than 12 channels
RME HDSP
2-12 channels
M-Audio Delta 1010
2-4 channels
M-Audio Delta 44, 66 or 2496

For portable devices to use with laptops and other portable computing
devices, we recommend:

PCMCIA/CardBus/ExpressCard
All RME Hammerfall IO Boxes (Multiface, Digiface, RPM) work with either
the Cardbus or ExpressCard interface, but please check with your Cardbus
chipset driver first. There's also the RME MADIface, a mobile MADI
interface for the ExpressCard slot (up to 64 channels).

Firewire devices
Please visit FFADO to see what Firewire devices are currently supported
on Linux.

USB devices
You'll be using a nasty little USB thing with support for just a couple
of channels. Check the ALSA soundcard matrix to verify that specific
devices are supported. Note that almost all USB1 audio interface are
supported and that almost no USB2 audio interfaces are. You can thank
the brain dead morons in the audio technology industry for this situation.

Please note that MOTU have consistently been aggressively anti-Linux
when asked about the information needed to support their devices.
Those MOTU devices that actually work on Linux do so only because of
pain-staking reverse engineering, and most do not work. We recommend
that you avoid this company's products, not because of their engineering
quality, but because of their stubborn refusal to help us create drivers
for their product. This stands in marked contrast to the cooperation
offered by RME, M-Audio and many other companies."

  On Portaudio
  - Portaudio (PA) is an equivalent of ARSC as it is a 'C' language API and
   library implemented as a layer on top of the Linux's ALSA host audio
   services. This means that ALSA supported hardware is the only option
   we have, since we have no plans to write our own drivers. Some drivers
   for firewire are available via ffado.org
  - I'll look at PA's sync tests and run this on layla3g and will check
   with Steve to interpret those results vs ARSC tstlat

  So there seems to be some evidence that sync io on linux is good using
  the RME cards! The next best are Echo(GINA3G/LAYLA3G) and the
  Roland/Edirol cards

Linux | Alsa | Echo Layla3G and Indigo IO

Layla3g, Echo, PCI
Running on Fedora core 9, the installation went fairly easy, all I had to do after installing the OS, install the firmware.

The next thing was uninstall pulse audio. Alsamixer shows 5 pcm playback devices, and as many as 255 mixer capture devices and that does not make sense.

ECHO Indigo IO

On Gentoo is a different story: Using Sabayon, a Gentoo distro, I got the Indigo card to be recognized, but it does not work and I am looking for solutions to this problem online (14-jan-2010).

ECHO Indigo IO working (to a first approximation) on Thinkpad X41

* Some initial issues with the installation: ECHO Indigo IO pcmcia card

I'm trying to get a PCMCIA card to work on Thinkpad X41 and it can't find the firmware. According to www.webalice.it/g_pochini/ead/ I should install the hotplug scripts, if I download them from

ftp.kernel.org/pub/linux/utils/kernel/hotplug/

there are hotplug-2004XXXX
hotplug-base
hotplug-core
hotplug-ng

etc

Is this recommended for SL5.1 ?

The errors I get look like this ( from dmesg )
PCI: Enabling device 0000:05:00.0 (0000 -> 0002)
ACPI: PCI Interrupt 0000:05:00.0[A] -> GSI 16 (level, low) -> IRQ 177
get_firmware(): Firmware not available (-2)
ACPI: PCI interrupt for device 0000:05:00.0 disabled
Echoaudio Indigo IO: probe of 0000:05:00.0 failed with error -2

Linux - Scientific Linux User

2008-07-08T19:57:00.000-07:00

DISPLAYS
DUAL SCREEN
Trying to drive a 1440x900 widescreen external LCD, these are some of the problems: I end up with the clone view and 3/4 of the external LCD are effectively configured, but pages use the entire LCD. The figure below shows only what is effectively active, the greyed-out section on my screen looks active.

AUDIO SETUP

One of the requirements for audio development in my particular case is to start install arsc. This requires tha my installation of Scientific Linux (SL) to have the kernel source code under /usr/src/

HARDWARE

* Thinkpad X41

* ECHO Indigo IO pcmcia card

I'm trying to get a PCMCIA card to work on Thinkpad X41 and it can't find the firmware. According to www.webalice.it/g_pochini/ead/ I should install the hotplug scripts, if I download them from

ftp.kernel.org/pub/linux/utils/kernel/hotplug/

there are hotplug-2004XXXX
hotplug-base
hotplug-core
hotplug-ng

etc

Is this recommended for SL5.1 ?

The errors I get look like this ( from dmesg )
PCI: Enabling device 0000:05:00.0 (0000 -> 0002)
ACPI: PCI Interrupt 0000:05:00.0[A] -> GSI 16 (level, low) -> IRQ 177
get_firmware(): Firmware not available (-2)
ACPI: PCI interrupt for device 0000:05:00.0 disabled
Echoaudio Indigo IO: probe of 0000:05:00.0 failed with error -2

Sound Card Fix
Looking again to the following links, respectively, I managed to fix the problem without fully installing the firmware for the OS.
1. http://ubuntuforums.org/showthread.php?p=1398393
2. https://help.ubuntu.com/community/EchoMia

The second link was very useful, because it walks you through the steps and allows one to "make clean" in case an attempt has been made to compile the alsa components.

The only step I did by-pass is the one reference by g_puchini stating that we provide a link in /lib/firmware to /usr/lib/hotplug/firmware and vice versa. While I did follow the steps to download the hotplug "scripts" I still did not do Puchini's step because I ended up with an installation that did not have the "hotplug/firmware" folders under "/usr/lib/" so I didn't know what aught to have been in the hotplug/firmware folders. Next, I restarted the Thinkpad and voila! No errors on dmesg were reported, and three devices or cards show up in the file: /proc/asound/cards, thanks to the users on Scientific-Linux-Users forum and to the authors provided in the links.

Comment or correct if you see something worth mentioning.

REVISION CONTROL

Subversion Setup

Creating repositories: Following the instructions from blog, I ended up with an error in step 8. When I type: "svnsync initialize file:///var/www/svn/repos http://localhost/repos --username=svnmirror" below is the error msg:

svnsync: 'pre-revprop-change' hook failed with error output:

Lastly, step 3 also talks about setting: "NameVirtualHost *:80" but I already have the following, 'Listen 8080'
should I append to the
"/etc/httpd/conf/httpd.conf" the line you mentioned: "NameVirtualHost *:80" ??

Apache httpd v 2.2.9 problems with fresh installations

2008-07-07T14:02:00.000-07:00

Using the source code to install apache2 there are problems with the configure option "--without-berkeley-db" and none of the sites really discuss a work-around. I had to downgrade to 2.2.3 although it might work with 2.2.8 as well, but I didn't try it. My istallation warned me that it could deal with this option. If any of you know a work around please let me know.

There is more, while it didn't give me the same warning, when I got to starting the httpd it still complaints: "... Cannot load mod_dav_svn.so into server: /usr/lib/libsvn_fs_base-1.so.0: undefined symbol: db_create"

where /usr/lib/libsvn_fs_base-1.so.0 is in fact a link to /usr/lib/libsvn_fs_base-1.so.0.0.0

I welcome your info.

April 08 Mac OS update: G4 Powerbook does not boot

2008-04-05T05:43:00.000-07:00

I press Command+option+shift+delete and all I get is a folder with question mark. When I press "C" to boot from the CD, nothing happens even though I have a linux bootable CD installed.

Pressing Shift after power-ON, nothing happens, it just sits on the first splash screen for a while, then kicks into "Staring Mac OS X" [reference: http://www.fif3.com/howto/archives/001983.html]

Apple discussions and forums show that you can the System CD to boot from it and repair the installation. What to do when you don't have the stupid System CD? There was an option for BootCD, but it does not support Mac OSX 10.4.x, so that does not work for me. I'm downloading a version of OpenSuSe LiveCD to see if I can get access to the HD.

Mean while from http://sonicchicken.net/blog/wordpress/20050926/boot-options-for-powerbook-g4/

I got the PBG4 to boot to the command line using Command-S. But after checking the HD, and it comes up ok, that's all I can think of doing. I have also pressed the T to connect it to my Mac Mini and use it to backup my Users folder.

What I want is to be able to boot from CD, any input anyone might have will be very useful.

References:
http://discussions.apple.com/thread.jspa?messageID=6547575�
http://docs.info.apple.com/article.html?path=Mac/10.4/en/cdb_strtup.html
http://docs.info.apple.com/article.html?artnum=106214

Failing to fully test features in Matlab

2007-10-30T16:24:00.001-07:00

Matlab release 14 SP3 included the data acquisition toolbox and when programs written in Matlab were compiled, using the Compiler, these programs were capable of being deployed to other machines. Except, this did not work out-of-the-box. I wrote a program about a year ago and compiled it. When the program run on the development PC, the stand-alone program works fine, but when deployed to other PCs, the program fails to talk with the hardware.

A year later I need to demo the program, but after checking on the Matlab user forums no links are found relating to this problem, but a developer claims that the problem was fixed in later releases. That's fine, but I need to run this program now and I don't have a current license to recompile the file be on my way. The current release would allow me to recompile the file and solve the issue. Contacting the support line has revealed little knowledge about a workaround to this issue on their side, and after a couple of days a solution was found.

One must register the winsound dll file. First, after running the executable, a directory is created: app_name_mcr/ in this directory go to: \app_name_mcr\app_name_mcr\toolbox\daq\daq\private

Here you will find a file: mwwinsound.dll

in windows XP (check with Mathworks for other OS's) you register the dll by typing: regsrv32 mwwinsound.dll, then run your application again.

Prestin Non-linear Capacitance Plotter: jplotNLC

2007-10-25T19:07:00.000-07:00

jplotNLC by S. Aguiñaga
This basic java app plots the NLC ascii files extracted from jClamp32 (SciSoft). Save files with asc file extension. Then select each one with jplotNLC and display prestin's NLC.

link: http://Electrophysiology.googlegroups.com/web/jplotNLC.jar

Serpentinista

iPhone Development

Multithread Portaudio

The Art of cyWar

Qt and Sound generation using libraries from audres.org

Computer Programming

Audio Programming on Mac OS and Linux

Portaudio And Libsndfile WAV playback with channel selection

What the Phonon: How to select individual audio channels (left, right or both)

Would Motorola departure burst Chicago’s tech bubble?

C++, Qt Multithreading

Mac OSX Audio and C++

C++ Debug Methods

Open Source Schematic Capture and Layout

MacBook Pro - Xubuntu

Sound on Apple MacPro and MacBook Pro

NFS Server Setup on Mac Pro Tower

Mandriva/Fedora on Mac Pro Quad-Core Intel Xenon - Part One

IO Synchronization on Audio Devices Running on Various Linux Platforms

Linux - Scientific Linux User

Apache httpd v 2.2.9 problems with fresh installations

April 08 Mac OS update: G4 Powerbook does not boot

Failing to fully test features in Matlab

Prestin Non-linear Capacitance Plotter: jplotNLC

Portaudio And Libsndfile WAV playback with
channel selection