Sara Fecadu KATIE: Hello. Welcome back. So, I keep forgetting

to do this and I apologize. But the big announcement right now is that the swag is ready. But do

not go get swag now because we're about to have a really awesome talk by Sara Fecadu.

I asked Sara for a fun fact and her fun fact was that she makes  bakes a mean cookie which

unfortunately we can't all indulge in. So, as a follow up question, I said what prompted

you write this talk about an audio API. And she said, well, I had spent a year building

a checkout form and I just couldn't stand to look at it or think about it anymore and

I had to do something different. Which I think is something that literally all you have us

can probably identify really strongly with. So, anyways, Sara is gonna come up and talk

to us about the audio API. So, give it up for Sara.

[ Applause ] SARA: Hello. See if I can get my computer

started here. Okay. Welcome to my talk. Meet the packets. If not everyone has realized,

it's a play off meet the parents. I spent a lot of time working on that.

[ Laughter ] Let's see here. One second. Gonna progress?

No. Okay. We're gonna do it without the clicker. So, this will be interesting. As Katie said,

my name  oh. My whole slide deck isn't progressing. Okay. One second. There we go. Okay. Thank

you for coming to talk. As Katie said, my name is Sara Fecadu. I am from Seattle, Washington.

And I don't have a ton of hobbies besides making cookies and listening to a lot of podcasts.

And by day I'm a software developer at Nordstrom. And Nordstrom is a clothing retailer founded

in 1901. While people don't usually associate 100 year old companies with tech, we have

a thriving tech org working on innovative ways to get you what you need and feel your

best. And a year ago I was hired on to do a rewrite of Nordstrom.com's redux. And as

of last May, we have been taking 100% of customer orders. Now, why am I talking about audio

streaming? Katie may have taken my joke here, but the answer is: Form fields. Our checkout

UI has 22 form fields. And they come in different groupings for different reasons. But many

of my waking moments over the past year have been spent thinking about these form fields.

And I just wanted to do anything else. So, I was sitting on my couch one night reading

a book on packet analysis, like one does, and watching a YouTube video. And I thought

to myself, how does that work? Like, on the packet level, how does audio video streaming

work? So, to answer the larger question, I started small with: What is audio streaming?

And audio streaming is the act of sending audio files over the network. And this talk

will be about on demand audio streaming. Now, the major difference between on demand streaming

and live streaming, is with on demand streaming we need all of the packets to get across the

wire. Whereas with live streaming, you may be more interested in keeping them up with

the event and a certain amount of packet loss is acceptable. Over the past few months, I

learned that audio streaming, even when limited to on demand, is as wide a subject as it is

deep. I have picked three topics that exemplify what audio streaming is. Why it's hard and

how to get started yourself. And we will talk about audio streaming protocols, TCP congestion

control and client players. Audio streaming protocols give us a stand how to encode, segment

and ship your code to the client. TCP congestion control handles congestion on the TCP layer

of the stack. And it is relevant with on demand audio streaming because we're shipping larger

audio files and we need every single packet to make its way to the client to play audio.

A client player is any network connected device with a play and pause button. So, this could

be your phone, your TV, your laptop, et cetera. And client players not only allow us to play

our audio, but when paired with modern audio streaming protocols, they hold a lot of decision

making power. Well, audio streaming protocols are the heart of audio streaming. And today

we'll talk about adaptive bitrate streaming it &s it benefits and how to convert your

own audio files to work with two popular audio streaming protocols. Before we get started,

I wanted to go over some terms that will come up. A codec encodes data and uses compression

techniques to get the highest quality for the smallest footprint. Encoding and trans

coding is converting it from one type to another. Trans coding can convert from digital to digital.

And then move from analog to other digital files. Bitrate is how many bits it takes to

encode a second of audio. And this number usually refers to the quality of the audio

file. When I think of playing music on the Internet, I think of an HTML5 audio tag with

a source attribute set to the path of my audio file. And this is a perfectly reasonable way

to do it. You can request and receive a single file containing an entire song. And it would

be referred to as progressive streaming and the major benefit here is you only have one

file to deal with. But let's say, for instance, you have a user and they have a slow network

connection and they can't download your one file. They're stuck. So, adaptive bitrate

streaming aims to solve this problem by encoding your audio in multiple bitrates and allowing

the client player to decide which quality is best for the user to listen to your audio

uninterrupted. This allows more users to access your audio. But it does add a layer of operational

complexity because now you've got a lot more work on moving parts. The audio streaming

protocols we'll talk about not only average adaptive bitrate streaming, but also use HTTP

web servers. They do this by encoding the file, segmenting they will, placing them on

a web server and then once requested, partial audio files are sent to the client one at

a time. Here is the secret to our modern audio streaming protocols is it's more of a series

of downloads than it really is a stream. But we'll refer to it as streaming anyway. The

two most popular audio streaming protocols today are HTTP lye streaming, or HLS, and

dynamic adaptive streaming over HTTP, MPEG DASH. It was created by Apple to support streaming

to mobile devices and it is default on all Mac OS and Apple devices. And MPEG DASH was

a direct alternative to HLS. It was created by the forum who want to make MPEG DASH the

international streaming. Let's look at them side by side. HLS takes the MPC, AAC, AC 3,

or EC 3, encodes them into fragmented MP4 files. Those segmented files are in a play

list. If you have multiple bitrate streams, each stream will be in a media play list and

all of your media play lists will be in a master play list. With MPEG DASH, it is agnostic,

in theory you can convert any into MPEG DASH. It will be fragmented into a fragmented MP4

file. That will be displayed in an XML manifest file called a media presentation description.

Okay. We've talked about what files will be used and what they'll be segmented into, but

how do you get it there? You've got this audio file. What tools allow you to convert the

audio file? Well, you've got options. But most of these options are paid options. Except

for FFmpeg. Which is an open source demand line tool that among other things allows you

to convert audio files to be HLS or MPEG DASH. However, I founded learning curve for FFmpeg

to be pretty steep. And a lot of the documentation for HLS and MPEG DASH were for video streams.

Instead I used Amazon elastic trans coder. It's an AWS offering that converts files of

one type to another. In our case, we're taking an audio file and converting it to be used

with HLS and MPEG DASH. It's pretty much plug and play. You tell Amazon elastic trans coder

what type of files you have and what type of files you want and it outputs the stream

for you. And even though it's easy to use, it's not a free service. So, if you were going

to be converting a lot of files, it may be worth your time to learn more about an open

source alternative like MPEG DASH. My workflow when working with Amazon Elastic Transcoder

was to upload to an AWS object store. I told Amazon Elastic Transcoder where my audio file

was and what settings I needed it to convert my audio files to. And Amazon Elastic Transcoder

output my streams into that same S3 bucket. And I downloaded them for us to explore. This

is the basic set of files you would get with an HLS stream. And it kind of looks like a

lot. But we're going to break it down into four groups. In the top left, the master play

list. In our case, we have two bitrate streams represented and they will be linked out from

the master play list. And then in the top right you'll see those media play lists which

have each bitrate stream. And those will contain all of our links to our transport stream files

which are the fragmented audio files represented in both the bottom left and the bottom right.

On the bottom right we have our 64K bitrate stream segmented audio files. And in the bottom,

oh. Did I get that backwards? I'm not really good at right and left. But in the bottom

section you'll have your fragmented audio files. We'll take a closer look at those so

you can see really what's in it. This is the entirety of the HLS master play list. It contains

information about the specific bitrate streams and links out to those media play lists that

represent the streams themselves. Let's look at the 64K bitrate stream media playlist.

It has even more information about the stream including caching information, the target

duration of each segmented audio file, and most importantly, links out to our transport

streams. This is what one of those fragmented audio times looks like. And there's something

a little interesting going on here. If you'll notice, it's color coded and I kept trying

to figure out why. But then I realized a transport stream has the file extension .ts. And something

else has the file extension .ts, TypeScript. Ignore the colors. It's just a binary coded

file. Now our MPEG DASH audio stream has fewer files and looks more manageable. But it's

similar. We have our media presentation description, which is an XML manifest file which contains

all of our information about the stream. Then below we have our two segmented audio files.

All of the segments are encapsulated in a single file, but within them there are segments.

That's why there are fewer files in the MPEG DASH audio stream than in the other audio

stream. Look at the description. See a lot of stuff here. But there are three important

elements. All bitrate streams are represented in a representation tag. And then all bitrate

streams are enclosed in an adaptation set. Within the representation tag, we do have

our URL to our audio files. And taking a look at one of those audio files we'll see if looks

fairly similar to the segmented audio file we saw with HLS. Minus the color coding because

it's a .MP4 versus .TS. visual studio is not confused in this case.

Earlier we talked about progressive streaming which is streaming an entire audio file in

one two. We used an audio element and a source attribute with the path of our audio file.

With MPEG DASH and HLS, it's very similar. But instead of having the path to our audio

file, we have the path to the master play list for HLS or media presentation description

for MPEG DASH. We're going to take a hard left here and we're gonna talk about the second

topic in my talk. Which is TCP congestion control. And TCP is a transport layer protocol

and it has mechanisms in both its sender and receiver which are defined by the operating

systems of each to react to and hopefully avoid congestion when sending packets over

the wire. And they are called TCP congestion control. And today we talk about packet loss

congestion control and why it isn't so great. And more specific, the congestion window and

duplicate acknowledgment in packet loss based congestion control. Before we get started,

somewhere terms, bandwidth is the rate at which data can be sent. And throughput is

the rate at which data can be received. The congestion window is a TCP variable that defines

the amount of data that can be sent before the acknowledgment is received by the sender.

Let's say you have a user who has requested your audio file from the server. Your audio

packets travel down the network stack, across the physical layer, up the data link layer

in the network layer and arrives at the transport layer and unfortunately there's congestion

right before we reached our destination. Now, traffic congestion and network congestion

have very similar beginnings. Either too many cars or too many packets have entered the

roadway and there's nowhere for them to go. With traffic, you have to wait it out. Luckily

for us, TCP congestion control allows them to flow over the wire, even during congestion.

And before we get to the specifics of these TCP congestion control algorithms, let's talk

about the TCP happy path. We're going to start with a single packet sent from the sender

to the receiver flowing through the receiver's buffer. And being acknowledged by the receiver

and having an acknowledgment packet sent back to the requester. We talked about the congestion

window, the amount of data before a sender receives an acknowledgment. Another way of

thinking about the congestion window is as a sending rate. As the sender receives acknowledgments,

the congestion window grows. And as the receiver's buffers fill and they drop all excess packets,

the sender responds by shrinking the congestion window. A second way of thinking about the

congestion window is as a bucket. And as packet loss occurs, the bucket shrinks. And as acknowledgments

are received by the sender, the bucket gross. There's a slight oversight in the bucket explanation

in that the receiver has no way of telling the sender that it is dropping packets due

to congestion. But one option the sender does have is to send a duplicate acknowledgment.

And a duplicate is if they're trying to send out of order packets. They send one, two and

three. For the purposes of our example, the receiver's not going to process them right

away. So, that when we send packet four, it's full and it has nowhere to go. So, packet

four dropped due to congestion. And they move on to process packet one, send an acknowledgment,

send for packet two and for three. However, when it looks at packet five, it says I can't

process you because this would be an out of order packet. drops packet five and sends

back for three. The sender is tipped off that it needs to sends packets four and five again.