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Home/ Questions/Q 3481076
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Editorial Team
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Editorial Team
Asked: May 18, 20262026-05-18T10:24:44+00:00 2026-05-18T10:24:44+00:00

I’m decoding a video format that has an accompanying audio track in a separate

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I’m decoding a video format that has an accompanying audio track in a separate file. Per the specs, I render a frame of video every 1/75th second. And the length of the video file is the same as the length of the audio track.

I’m playing the audio with Audio Queue Services (which I chose because I figured there would be situations where I needed precise timing control — just the sort of situation I’m encountering!). It’s a big API and I haven’t progressed much past the sample code in Apple’s programming guide (though I have wrapped things up in a nicer ObjC API).

In ideal situations, things work fine with the basic playback setup. The video and audio stays synced and both end at the same time (within my own ability to tell the difference). However, if performance hiccups (or I attach the Leaks Instrument or something), they quickly get out of sync.

This is the first time I’ve ever written something of this nature: I have no prior experience with sound or video. I certainly have no experience with Audio Queue Services. So I’m not sure where to go from here.

Have you done something like this? Do you have some advice or tips or tricks to offer? Is there some fundamental piece of documentation I need to read? Any help would be greatly appreciated.

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  1. Editorial Team
    Editorial Team
    2026-05-18T10:24:44+00:00Added an answer on May 18, 2026 at 10:24 am

    First off, I’ve never actually coded anything like this so I’m shooting from the hip. Also, I’ve done a decent amount of programming with the HAL and AUHAL but never with AudioQueue so my approach might not be the best way to use AQ.

    Obviously the first thing to decide is whether to sync the audio to the video or the video to the audio. From the question it seems you’ve decided the video will be the master and the audio should sync to it.

    I would approach this by keeping track of the number of frames of video rendered, along with the frame rate. Then, when enqueuing your audio buffers, rather than passing a monotonically increasing value for the startTime adjust the buffer’s start time to match any discontinuities observed in the video. This is a bit vague because I don’t know exactly where your audio is coming from or how you are enqueuing it, but hopefully the principle is clear.

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