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Subject: Coreaudio-api Digest, Vol 2, Issue 332
Date: Sun, 30 Oct 2005 12:04:59 -0800 (PST)
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Today's Topics:
1. Re: Channels and frames (john smith)
2. Output AU buffer size for RenderCallback (Tim Dorcey)
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Message: 1
Date: Sun, 30 Oct 2005 08:03:05 +0100
From: "john smith" <email@hidden>
Subject: Re: Channels and frames
To: email@hidden
Cc: email@hidden
Message-ID: <email@hidden>
Content-Type: text/plain; format=flowed
Stefan,
Thanks for your help. This AU manager definitely sounds like what I need.
However, I'm unable to locate it.
I assume it is something that was installed with Logic? (FWIW, I'm running
LA7).
Thanks,
Michael Olsen
>From: Stefan Gretscher <email@hidden>
>To: john smith <email@hidden>
>CC: coreaudio-api Audio <email@hidden>
>Subject: Re: Channels and frames
>Date: Fri, 28 Oct 2005 15:31:02 +0200
>
>Am 28.10.2005 um 12:34 schrieb john smith:
>>I must have misunderstood something, because when using the following
>>code, the plug-in still shows up as mono->mono and mono->stereo in
Logic.
>Logic caches the I/O capabilities of your AU - increase the version
number
>of your AU or rescan the AU in the AU Manager to have Logic update its
>cache.
>Alternativley, use a version number of 0 during debug, this will cause
>Logic to always rescan your AU when it launches.
>
>Let me know if this doesn't fix your problem.
>
>Best,
>Stefan
>
>------------------------------------
>Stefan Gretscher
>
>plug-in development & 3rd party plug-in support
>Emagic GmbH, Hamburg, Germany
>a subsidiary of Apple Computer, Inc.
>
>email: email@hidden
>phone: (+49)-4101-495-586
>(Central European Timezone)
>
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Message: 2
Date: Sun, 30 Oct 2005 11:03:42 -0800
From: "Tim Dorcey" <email@hidden>
Subject: Output AU buffer size for RenderCallback
To: <email@hidden>
Message-ID: <002901c5dd84$a5f3d850$email@hidden>
Content-Type: text/plain; charset=iso-8859-1
I found mention in archives here that the value of "inNumberFrames" passed
to a RenderCallback for default Output Unit is determined by the attached
device's "I/O proc frame count." Is this a property that clients can (or
should) change?
On my system, talking to "Built-in Audio," it seems to want about 12 msec
per render callback. This is fine, and anything in the range < 100 msec
would be fine for my app. Do I need to concern myself with getting
something outside that range, and how do I change it if so?
Also, I based my code on a "Quick and Dirty" example that was posted here,
which was nothing other than:
FindNextComponent() kAudioUnitSubType_DefaultOutput
OpenAComponent()
AudioUnitSetProperty() kAudioUnitProperty_StreamFormat
AudioUnitSetProperty() kAudioUnitProperty_SetRenderCallback
AudioUnitInitialize()
AudioUnitStart()
...
AudioUnitStop()
AudioUnitUninitialize()
CloseComponent()
Then, my render callback simply fills the AudioBufferList presented to it.
Is that all there is to it?
Also, is it correct to presume that if I Open/Start multiple instances of
the Output AU, the audio streams will be mixed at final output? And, each
could have a different stream format (e.g., sampling rate)?
Thanks,
Tim
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End of Coreaudio-api Digest, Vol 2, Issue 332
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