I spent last weekend hacking at RIT’s BrickHack 4. Normally, I love to spend my hackathons building multiplayer games, but this time, I wanted to try working on an idea that I’d had in the back of my TODO list for a while. This idea came to me after I was listening to some nightcore (a remixed track of a song that speeds it up, increases its pitch, and puts more emphasis on a strong beat). Specifically, I was listening to this one on YouTube:
If you skip to the 0:36 mark, you can hear an interesting audio effect if you’re wearing headphones. The beat jumps between the left and right ear (this also happens at a few other points in this video). This was such an interesting effect that I wondered if it was possible to algorithmically create this effect for other songs. My end goal for this project was to be able take some dubstep and use stereo effects to switch the beat between the left and right ear and create an immersive listening experience. During the hackathon, I worked with my friend Andrew Searns to try and create a web application to do this (Andrew also showed me other interesting songs where the artist played around with stereo effects).
While it wasn’t as successful as we expected, we had a lot of fun coming up with an architecture to do this and I personally learned quite a bit of CSS Flexbox while designing the application interface. In this techsploration, I’ll take you through my thought process and show you how to generalize the code we wrote to build an audio processing framework in the browser.
Okay, let’s figure out what we need to make this work. Ideally, we want users to simply paste a YouTube link into our application, which will handle streaming the audio, spatializing it, and then playing it.
Obviously, the first thing we’ll need is some way to do that audio streaming. I’ll discuss how we set that up in this section, and we’ll talk about the audio spatialization part later.
Because we wanted to develop and iterate quickly, we used
(also because I happen to like node.js very much). During the hackathon, we found two well documented ways to do audio streaming from YouTube.
EDIT: I’ve revisited this project and the youtube-audio-stream package is a bit funky sometimes and isn’t as actively maintained as ytdl-core. I’ve rewritten the project to use ytdl-core, but for legacy reasons I will not change this blog post. Worry not, this change is almost a drop-in replacement and only affects around 5 lines of code.
To do this, the client sends the YouTube video ID to our server, which uses youtube-audio-stream to fetch it from YouTube and then pipe it back to the client.
We decided to offload the audio processing to the client to reduce the load on our server. This turned out to be the most pragmatic solution for our purpose and was simple to code and put together.
Now you might be looking at this and asking why we didn’t cut out the middleman and have the web browser directly request the audio stream from YouTube. Cross-Origin Resource Sharing (CORS) prevents us from making a request from the client directly to YouTube. Additionally, we wanted our server to provide a communication channel between all the clients (for reasons we will discuss in a later section).
Below is a minimal working example of the architecture described above:
This might look like a lot of code to digest, but it’s actually quite simple and most of it is just boilerplate. Let’s run through it.
audio-processor-server.js is a pretty standard boilerplate express.js server with two routes defined: one for serving the HTML page to the client, and one for serving the audio stream to the client. The server will also statically serve files in the/client folder, which is where we will store audio-processor.js.
audio-processor.html is a plain HTML page with nothing but an input field for a YouTube video ID and a submit button.
audio-processor.js is the client side script loaded into the HTML page to send a request to the server for an audio stream. We’re using an XHR request to request the audio stream so that we can specify a return type of arraybuffer. Using the Web Audio API’s AudioContext object, we can decode the arraybuffer into an AudioBuffer object containing the decoded PCM audio data. This buffer is passed into the processAudio() method (not defined in the example stub above) which you can substitute that with any function you want.
Note that the code in audio-processor.js is still relevant even if you don’t want to load audio from YouTube. You can apply it to statically loaded audio files or any other audio sources you may want to use. I recommend browsing through the Web Audio API documentation to explore all the cool things you can do with an audio buffer.
Suppose however, you want to perform the audio processing part on the server. This has several advantages to doing the processing on the client: server side caching, architecture specific processing libraries, or simply convenience. The disadvantage however, is that many concurrent requests will put a lot of load on your server.
We also experimented with doing the processing server side during the hackathon using an architecture like this:
We chose not to do this to reduce server load, but I’ll give sample code for this architecture as well if you would like to replicate this for your own projects.
audio-processor-server.js is again a simple express.js boilerplate server, but with a few extra pieces this time. In this version, we save the audio stream to the file system as a .flv file. This is not necessary if you want to just pipe the audio stream into your custom audio processor, but we found that this allowed for more flexibility and control. Like before, this server has two routes defined: one for serving the HTML page and one for serving the audio streams. Again, the server will also statically serve files in the /client folder.
audio-processor.html is the same plain HTML page with an input field for the YouTube video ID and a submit button.
audio-processor.js is a client side script loaded into the HTML page that simply requests an audio file using an HTMLAudioElement and plays it.
If you want even more granularity, you can actually combine the two methods described above to do processing on the server and additional post-processing on the client. We didn’t try this but you can probably do a lot of cool things by combining backend specific libraries with the power of the Web Audio API.
If you were only interested in building an in-browser audio processing architecture, then you can stop reading here or skip to the end. The rest of the techsploration will be about how we connected this audio streaming architecture to the Resonance Audio API for our hack.
Once we got past the challenge of figuring out the best way to manipulate an audio stream, we needed to figure out how to move the audio in stereo to create the effect we wanted. We used the Google Resonance Audio API to do this. Given an audio source and a vector position, this API allowed us to position the audio source around the listener in 3D space. You can explore this effect in their demos (best experienced with headphones).
Our plan was to generate a list of vector values based on the audio data and use the Resonance Audio API to move the sound source around while the audio was playing. This was pretty trivial to do using setInterval() and synchronizing the list of vector positions using the current frame in the audio. We could calculate exactly which vector position to position the audio source at since the sampling rate of the audio buffer was known.
The hard part (and the coolest part) was figuring out how to generate the list of vector positions properly so that the stereo panning was in sync with the beat. We didn’t want to just oscillate between left and right and we didn’t want the movement pattern to be repetitive and boring. Andrew and I experimented with various movement patterns and developed a scheme to encode them.
We named the movements patterns as Transforms and decided on 5 distinct ways in which we could move the audio around the listener: Jumps (which move the audio source to a random location around the listener), Flips (which move the audio source to the direct opposite side of the listener), Rotates (which slowly rotate the audio source around the listener), Delays (which hold the audio source in place), and Resets (which reset the audio source to the origin). Using the naive beat detection algorithm described before, we transitioned from one Transform to another on every beat.
To generate sequences of Transforms, we used a Markov chain, initially seeding every transition with equal probability. Here’s where we actually utilized the server as a relay point between the connected clients. We attempted to make the sequence generation somewhat intelligent by adding a button to the interface allowing the user to vote up or vote down the current audio spatialization pattern. The server would then take this into account by increasing or decreasing the probability of the sequences used in the song the user was listening to.
This was a fantastic project to design and play with. There are lot of interesting things to explore in the Web Audio API. I don’t know much about audio encoding but I imagine the resources there would be a lot more useful if I had a little more background knowledge about audio codecs and PCM data.