audioconverter and unpacked source

Subject: audioconverter and unpacked source
From: Christopher Skogen <email@hidden>
Date: Thu, 30 Nov 2006 20:49:02 -0500

hi all,

i've been using the audioconverter for a while now and have had no trouble with it til now. i'm trying to convert from an unpacked integer format to 32-bit float.

here's the source asbd. i'm trying to describe a 16 bit signed integer sample, low aligned, stored in a 4 byte signed integer.

fmt.mSampleRate = 44100;

fmt.mFramesPerPacket = 1;

fmt.mChannelsPerFrame = 1;

fmt.mBitsPerChannel = 16;

fmt.mBytesPerFrame = 4;

fmt.mBytesPerPacket = 4;

fmt.mFormatID = audioFormatLinearPCM;

fmt.mFormatFlags = kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsSignedInteger;

fmt.mReserved = 0;

here's the dest asbd. i'm trying to describe a 32 bit float sample.

fmt.mSampleRate = 44100;

fmt.mFramesPerPacket = 1;

fmt.mChannelsPerFrame = 1;

fmt.mBitsPerChannel = 32;

fmt.mBytesPerFrame = 4;

fmt.mBytesPerPacket = 4;

fmt.mFormatID = audioFormatLinearPCM;

fmt.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;

fmt.mReserved = 0;

the converter is created and runs, but the problem is in the float values created. it works fine on a ppc mac, but on my intel mac i get a strange result.

my first sample in is -32768 or 0x0080FFFF. after conversion, i should get -1 or 0x000080BF. instead, on the intel mac i get 0x000080B7.

my understanding of unpacked, low-aligned leads me to believe that the FFFF of the input sample would be ignored by the audio converter. i then tried modifying all four bytes of the input and found that any change to the four input bytes was changing the output of the converter. what am i missing? or... how am i misunderstanding this?

any help would be appreciated,

christopher skogen

gizmolabs, inc.

email@hidden

office: (404) 687-0008

fax: (404) 687-1010

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