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Editorial Team
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Asked: 2026-06-09T10:30:01+00:00

I really need help with this issue. I’m developing an iOS application with audio

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I really need help with this issue. I’m developing an iOS application with audio units, the recorded audio needs to at 8bit / 8000 hertz sample rate using alaw format. How ever I’m getting a distorted voice coming out the speaker.

I came across this sample online:

http://www.stefanpopp.de/2011/capture-iphone-microphone/comment-page-1/

while trying to debug my app I used my audioFormat in his application and I am getting the same distorted sound. I guessing I either have incorrect settings or I need to do something else to enable this to work. Given the application in the link and the below audioFormat can anyone tell me if I’m doing something wrong or missing something ? I don’t know a lot about this stuff, thanks.

Audio Format:

AudioStreamBasicDescription audioFormat;
    audioFormat.mSampleRate         = 8000;
    audioFormat.mFormatID           = kAudioFormatALaw;
    audioFormat.mFormatFlags        = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
    audioFormat.mFramesPerPacket    = 1;
    audioFormat.mChannelsPerFrame   = 1;
    audioFormat.mBitsPerChannel     = 8;
    audioFormat.mBytesPerPacket     = 1;
    audioFormat.mBytesPerFrame      = 1;
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1 Answer

  1. Editorial Team

    Eventually got it to play correctly. I’m posting here to help out anyone else facing similar issues.

    Main issue I was facing is that there is a huge difference between the simulator and an actual device. Running the app on the device the sound quality was better but it kept skipping every second or 2, I found a setting that seemed to fix this and a setting to change the buffer size / duration. (The duration setting does not work on the simulator, some of my issues were needing it to run at a certain rate to sync with something else, this was causing distorted sounds)

    status = AudioSessionInitialize(NULL, kCFRunLoopDefaultMode, NULL, audioUnit);
UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord;
status = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(audioCategory), &audioCategory);
[self hasError:status:__FILE__:__LINE__];

Float32 preferredBufferSize = 0.005805; // in seconds
status = AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(preferredBufferSize), &preferredBufferSize);
[self hasError:status:__FILE__:__LINE__];

status = AudioSessionSetActive(true);

    The first audio session property is what stopped the skipping making it play much more smoothly. The second adjusts the buffer duration, this is in seconds how often the callbacks are fired and will give you a different buffer size. Its best effort meaning it will get as close as it can to the value you provide but it seems to have a list of available sizes and picks the closest.

    See the post I link to in my question for a very good tutorial / sample program to get started with this stuff.

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