Audio distortion in AES16e

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   I am using AES16e with custom application, which plays 8 channel audio using xaudio2. I have set the buffer size using LMixer to 256(default). I am getting glitches like double audio(repeat) randomly. Increasing the buffer size increases the glitches.
I am sending time code for synchronization with other device through one of the channel in the audio. Reducing the buffer size results in audio drop and breaks the sync.

I am using i7 3770 processor with 4 GB ram. My CPU load is normally around 70-80%.
Firmware : 13.2 and Lynx mixer version : 2.0.0. Build 22.

Any help regarding would be great.

--
Vicky

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Vicky Members Profile Send Private Message Find Members Posts Add to Buddy List Caracal Joined: March/25/2015 Location: India Points: 3	Topic: Audio distortion in AES16e Posted: March/25/2015 at 6:00am
	I am using AES16e with custom application, which plays 8 channel audio using xaudio2. I have set the buffer size using LMixer to 256(default). I am getting glitches like double audio(repeat) randomly. Increasing the buffer size increases the glitches. I am sending time code for synchronization with other device through one of the channel in the audio. Reducing the buffer size results in audio drop and breaks the sync. I am using i7 3770 processor with 4 GB ram. My CPU load is normally around 70-80%. Firmware : 13.2 and Lynx mixer version : 2.0.0. Build 22. Any help regarding would be great. -- Vicky

David A Hoatson Members Profile Send Private Message Find Members Posts Add to Buddy List Admin Group Forum Administrator Joined: October/01/2003 Location: Idaho Points: 4993	Posted: March/25/2015 at 7:03am
	You have a lot of things going on that are really beyond the level of support we can provide on this forum. Custom application which we know nothing about. Custom time code over digital audio to external device which you didn't describe. You didn't mention what operating system you are using. I will try and help as best I am able to. The LMixer.exe application is really only to be used with the older PCI cards, and not with the AES16e. You should be using the new LynxMixer.exe with the AES16e. The buffer size in LMixer.exe does nothing for the AES16e with the WaveRT (WDM) driver. That is only used with the ASIO driver. Since you are using XAudio2, I assume you are using WASAPI, but you didn't say. You say you are using a custom application, have you tried playing back audio with LynxDemo.exe? You can always generate your time code to a WAV file then just play that back with LynxDemo.exe. It should be exactly the same functionality. How are you synchronizing the clocking on your external device? Is the AES16e set to Internal? If so, what clock source is your external device using? Edited by David A Hoatson - March/25/2015 at 7:06am
	Thank you, David A. Hoatson Lynx Studio Technology, Inc. Co-founder, Chief Software Engineer

Vicky Members Profile Send Private Message Find Members Posts Add to Buddy List Caracal Joined: March/25/2015 Location: India Points: 3	Posted: March/26/2015 at 1:31am
	Here is the scene. OS win7. The custom application play av at 24 fps. Audio is 7.1 channel 48khz. The external device is an automation device which does some special effects. It is driven by a time code in a specific format which we send on 8th channel of the audio(5.1 main audio, 7 channel silence, time code 8th channel). We have working hardware av decoder board which produces output in the same way on its AES output and is working perfectly with our automation device. We are now trying to do similar thing on software using Lynx AES16 on which we are facing the issue. The external device does not depends on external clock. It just plays the 5.1 audio and does some automation based on timecode. My first observation was the timecode was breaking with buffer size of 256 and lower. There was no audio drop when i changed it to 512 and above. But as per your saying that LMixer.exe should not be use with AES16e, I am still wondering how changing the buffer size in that change the behavior. l

David A Hoatson Members Profile Send Private Message Find Members Posts Add to Buddy List Admin Group Forum Administrator Joined: October/01/2003 Location: Idaho Points: 4993	Posted: March/26/2015 at 9:07am
	There is no "buffer size" setting with the AES16e when using WASAPI (is that what you are using? you still didn't say...). The buffer size is determined by your application when it opens the device. Unless the external device is using a sample rate converter, it most certainly would depend on an external clock otherwise it wouldn't work properly (and would dropout audio just as you described). In any digital audio system, there can only be one master clock and all other devices must either slave to it, or use a sample rate converter. Did you try LynxDemo.exe with a pre-rendered audio file?
	Thank you, David A. Hoatson Lynx Studio Technology, Inc. Co-founder, Chief Software Engineer