8kHz sample rate option broken for stream sounds in CS5

I've made several test files where all I do is add a sound to a keyframe, set it to Stream sound, add a sufficient number of frames to hear the sound (about 300), then change publish settings to override as MP3 with "use 8kHz sample rate" option.
Testing movie yields a SWF that has NO SOUND at all.
Changing the sound to Event sound and republishing PLAYS THE SOUND AS EXPECTED.
Changing it back to Stream sound and turning off the "use 8kHz sample rate" option again yields a SWF that has NO SOUND at all.
It seems to make no difference what sound I import and use.
I'm on a Mac Octocore.
This Publish Setting appears to be broken in Flash CS5.

The only setting that I could find in compressor that lets your change the bitrate to 44.1 is when you create a new dolby digital setting and then under the inspectors audio tab/Target System button, change the button to Generic AC-3. When done, you can change the Sample Rate to 44.1.
Hope this helps?

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I have to agree with you. There are several forum discussions on bit rate being as high as 256 kbps but I don't see how it could be more than 96 kbps based on the poor sound quality I'm hearing. I'm comparing it to an internet radio station that is 128 kbps and sounds much better.
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I would like to implement a lock-in amplifier in labview (version 8.5). I understand that there is a developer kit for the LIA available, and have checked it out; however, it is incompatible with my equipment.
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Here it says the minimum rate is 50kHz. Sorry, but ther is no option to change that.
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Sample rate too high for swf or PowerPoint export

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I have a few presentations that need to be exported to swf or PowerPoint. Each of the slides use audio Voice Over. The problem is that when I go to export I'm told that because the audio uses a sample rate of 48khz it can't be exported.
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AAC.
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OK, thanks for clearing that up.
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Message Edited by el_loco on 29-Sep-2008 06:49 PM
640K Should be enough for everybody
El_Loco Nokia Video Blog

Sample rate/quicktime pro audio/video

Hi all:
Have a few questions related to Quicktime pro on mac related to sample rate of audio and video...any help understanding much appreciated...
I am a music composer using logic pro and recently started working with filmakers who send me quicktime files of the video to score to...
regarding sample rate of audio as played back in quicktime...I'm a bit confused....
When playing back either an MP3,wave,Aiff at 44.1k or 48k
seems that quicktime plays at the right speed and pitch of audio...Does Quicktime use it's own audio processor to convert sample rates in real time at playback?
When I play a video track with audio recorded at 44.1k or even an MP3...and playback on my soundcard set at 48k...it does play at correct pitch...even though theory of audio tells me that the sample rate is very inportant especially when playing back...sample rate needs to be the same on both the soundcard audio clock and the software...
Is this part of the magic of quicktime or am I missing something.
Even a bit more confused when I got a dialog file from a filmaker attached to his quicktime file that he thougnt was in 44.1k...was actually 48k...so when taken out of quicktime and put in another enviornment for editing that became really important and avident that the samplerate audio rules outside of quicktime applied...
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All you may ever want to know about QT, audio and OS X:
http://developer.apple.com/documentation/MusicAudio/Reference/CoreAudio/index.ht ml
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http://www.adobe.com/cfusion/webforums/forum/messageview.cfm?catid=578&threadid=1199097
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I find this behavior for a DAW very strange. If I have a multitrack open of one bit rate and my clock set to another, how does that work? How does Audition 6 play and record audio at differnt sample rates than the interface clock? When I open a multitrack that's 96 kHz and my clock is set to 48, what sample rate am I getting? The audio file being saved says 96 kHz but is it?
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loneraver1 wrote:
If true, this is indeed strange. I have used just about every DAW under the sun for the last 10 years based on what my employer purchased and I have never seen a DAW allow you to record sample rates where the project didn't match the incoming clock.
There should be an option that locks the audio interface internal clock to the opened project. It makes it annoying any time I have to switch between recording in 96 kHz to 48 kHz. That's annoying on it's own right, but what's really annoying is that it's an extra step that I have to train our archival technicians who don't have a techical background in audio.
I can definitely say it's true for my set-up!
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FWIW, if I want to record something at 44.1 I always ensure the card and AA settings are 44.1; but for "playback" (or video editing which I do quite a lot of and which requires 48 - my video editor software will not "play" unless the audio sampling rate of the card is 48 - ) 48 is my "default" setting.
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Hmm... there are plenty of possible reasons for this ...here are couple of them (not mentioned in your list of suggestions given by Creative support)
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Bob Lyons wrote:
Might as well save 5 or 6 screen sets with your custom template. That's going to save you a lot of time setting up new projects.
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"audio source rate and capture sample rates did not match & footage may be out of synch"<<</div>
William,
The message itself seems to be a bug that started with FCP 5 - most users that receive this message have reported that there is NO actual problem.
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Hello,
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Hi,
If you are in the HTML5 canvas document, this is expected. Only "Event" sync option is available since this is the only one supported by Sound.js.
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If you are buying them it makes sense to opt for 48.

8kHz sample rate option broken for stream sounds in CS5

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