First buffer lost when trying to write to stream

[RyanG2016]

This article was an amazing read and really help me be able to create a recording app that I am using to stream audio to a server over websockets.
I am running into an issue that is really stumping me though. I need to find a way to delay the AudioQueue callback from running until the websocket is open and we get a response from the server. It usually happens within one buffer cycle (~500ms) but since I can't send until the socket is ready, the first buffer never gets to the server and seems to get dropped. If I write to a file it's all there with no issue.
I am requesting the websocket connection as soon as the user clicks record so I can't do that any better than it is.
Anyways, if you have any suggestions it would be appreciated. I should note that I'm not using objc but swift 5

[pmatsinopoulos]

Hi!
Maybe the solution is to create and open the Websocket much in advance? Can the websocket exist independently? For example, when your application starts? Why not? You might want to use it for other reasons too, not only for sending audio/recording data.

[RyanG2016]

Hi,
That is a possibility but I would prefer to not have the web socket open all of the time. Just when in use.
The app is used like a wallet talkie device and used for short 3-15 second recordings. We do get other data from the server but is transmitted through rest APIs and not over the websocket.

[pmatsinopoulos]

So, when the user clicks "Record", maybe the button label should be changed to "Getting Ready..." (or something similar) with a disabled status while the code will be setting up the websocket connection. When the websocket connection is ready, usually there is a callback something like connected() that the underlying websocket library calls. The connected() implementation on your side should change the button label status to "Recording..." (or to "Stop Recording") to indicate that recording has just started and is under way. This is when you can fire the streaming the server?

[RyanG2016]

Ideally that would be the way to do it but I want my users to be able to start and stop recording without waiting.
So essentially I want recording to start immediately and filling the buffers but not send them until the websocket state is connected which we would set when we get the server confirmation.
I have the buffer size set manually to 16000k as that is the format we need to send to the server so the audio buffer size is about 500ms. Realistically the socket will be open and ready in less than 1000ms so I need to be able to "queue" the buffer queue.