Samplesize 8 or 16? - c++

Question

I try to record sound from my pc audio device. I can hear the recorded sound only when I set samplesize to 8 but when I set the samplesize to 16 , the recorded sound is only a whistle. why?
here is the code
void audioprocess::startRecording()
{
outputFile.setFileName("C:/Users/Rem/Documents/Qtproject/remaudio.wav");
outputFile.open( QIODevice::WriteOnly );
QAudioFormat format;
// set up the format you want, eg.
format.setCodec("audio/pcm");
format.setSampleRate(8000);
format.setChannelCount(1);
format.setSampleSize(8);
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::UnSignedInt);
audioInput = new QAudioInput(format);
QAudioDeviceInfo info = QAudioDeviceInfo::defaultInputDevice();
if (!info.isFormatSupported(format)) {
qWarning()<<"default format not supported try to use nearest";
format = info.nearestFormat(format);
}
QTimer::singleShot(120000, this, SLOT(stopRecording()));
audioInput->start(&outputFile);
// Records audio for 120000ms
}

Answer 1

The whistle is probably because every second sample is zero. That means you get a periodic signal at exactly half the sample frequency.
Now what does your code actually do? You haven't shown us the definition of outputFile, but the code snippet is almost literally taken from Qt documentation. And it defines outputFile as a QFile. The file there is called test.raw, and for good reason. Raw files lack a header. Thus it's impossible to determine their sample size.

Answer 2

Finally a question that's in my area of expertise. When it comes to audio development, there are a lot of factors to include into how to program an application. One of those specifically is hardware capabilities. My guess is that your audio device is being sampled too quickly and the higher end bits are just random data.
I would try to increase the sample size, the actual number of frames of audio that are being sampled before being processed, to allow the system to actually buffer the audio and process it while the audio device is receiving new audio input.
Another problem may be the Endianness of your processor. This would usually be an issue if you are on a PowerPC. Or developing for an embedded system that runs on a big Endian processor.

Answer 3

I'm going to take a wild guess here that you are leaving the sample type as unsigned int and only setting the bit depth to 16. Conventionally, 8 bit waveforms are unsigned while 16 bits are signed. Try patching in the following:
format.setSampleSize(16);
format.setSampleType(QAudioFormat::SignedInt);