Playing back a WAV file streamed gradually over a network connection in iOS

Question

I'm working with a third party API that behaves as follows:

• I have to connect to its URL and make my request, which involves POSTing request data;
• the remote server then sends back, "chunk" at a time, the corresponding WAV data (which I receive in my NSURLConnectionDataDelegate's didReceiveData callback).

By "chunk" for argument's sake, we mean some arbitrary "next portion" of the data, with no guarantee that it corresponds to any meaningful division of the audio (e.g. it may not be aligned to a specific multiple of audio frames, the number of bytes in each chunk is just some arbitrary number that can be different for each chunk, etc).

Now-- correct me if I'm wrong, I can't simply use an AVAudioPlayer because I need to POST to my URL, so I need to pull back the data "manually" via an NSURLConnection.

So... given the above, what is then the most painless way for me to play back that audio as it comes down the wire? (I appreciate that I could concatenate all the arrays of bytes and then pass the whole thing to an AVAudioPlayer at the end-- only that this will delay the start of playback as I have to wait for all the data.)

Answer 1

I will give a bird's eye view to the solution. I think that this will help you a great deal in the direction to find a concrete, coded solution.

iOS provides a zoo of audio APIs and several of them can be used to play audio. Which one of them you choose depends on your particular requirements. As you wrote already, the AVAudioPlayer class is not suitable for your case, because with this one, you need to know all the audio data in the moment you start playing audio. Obviously, this is not the case for streaming, so we have to look for an alternative.

A good tradeoff between ease of use and versatility are the Audio Queue Services, which I recommend for you. Another alternative would be Audio Units, but they are a low level C API and therefor less intuitive to use and they have many pitfalls. So stick to Audio Queues.

Audio Queues allow you to define callback functions which are called from the API when it needs more audio data for playback - similarly to the callback of your network code, which gets called when there is data available.

Now the difficulty is how to connect two callbacks, one which supplies data and one which requests data. For this, you have to use a buffer. More specifically, a queue (don't confuse this queue with the Audio Queue stuff. Audio Queue Services is the name of an API. On the other hand, the queue I'm talking about next is a container object). For clarity, I will call this one buffer-queue.

To fill data into the buffer-queue you will use the network callback function, which supplies data to you from the network. And data will be taken out of the buffer-queue by the audio callback function, which is called by the Audio Queue Services when it needs more data.

You have to find a buffer-queue implementation which supports concurrent access (aka it is thread safe), because it will be accessed from two different threads, the audio thread and the network thread. Alternatively to finding an already thread safe buffer-queue implementation, you can take care of the thread safety on your own, e.g. by executing all code dealing with the buffer-queue on a certain dispatch queue (3rd kind of queue here; yes, Apple and IT love them).

Now, what happens if either

• The audio callback is called and your buffer-queue is empty, or

• The network callback is called and your buffer-queue is already full?

In both cases, the respective callback function can't proceed normally. The audio callback function can't supply audio data if there is none available and the network callback function can't store incoming data if the buffer-queue is full.

In these cases, I would first try out blocking further execution until more data is available or respectively space is available to store data. On the network side, this will most likely work. On the audio side, this might cause problems. If it causes problems on the audio side, you have an easy solution: if you have no data, simply supply silence as data. That means that you need to supply zero-frames to the Audio Queue Services, which it will play as silence to fill the gap until more data is available from the network. This is the concept that all streaming players use when suddenly the audio stops and it tells you "buffering" next to some kind of spinning icon indicating that you have to wait and nobody knows for how long.