Skip to main content

Use the Warnings, Luke

Browsing StackOverflow on perl-related topics, I have two main considerations:
1. Answers are typically very good, concise and useful
2. Questions are submitted with code that doesn't have 'warnings' enabled

I'd say that a considerable portion of the questions submitted would not be posted, or would be less generic, if authors used 'use warnings;' in their code.

Then add a pinch of the great perl critic, and possibly only half of the questions would really be submitted.

If you're using perl, or plan to use it, I strongly recommend to:
1. Always set 'use warnings;'
2. Always submit your code to perl critic (and keep a copy of Perl Best Practices handy).
3. First create the tests, then write the code. That's the only reasonable way (unless you're working on a one-liner for a quick admin task). TDD is your friend.

See more on Perl Critic here.
Labels:

Popular posts from this blog

Troubleshooting TURN

  WebRTC applications use the ICE negotiation to discovery the best way to communicate with a remote party. I t dynamically finds a pair of candidates (IP address, port and transport, also known as “transport address”) suitable for exchanging media and data. The most important aspect of this is “dynamically”: a local and a remote transport address are found based on the network conditions at the time of establishing a session. For example, a WebRTC client that normally uses a server reflexive transport address to communicate with an SFU. when running inside the home office, may use a relay transport address over TCP when running inside an office network which limits remote UDP targets. The same configuration (defined as “iceServers” when creating an RTCPeerConnection will work in both cases, producing different outcomes.
Read more

About ICE negotiation

Disclaimer: I wrote this article on March 2022 while working with Subspace, and the original link is here:  https://subspace.com/resources/ice-negotiation . This post in my personal blog is a way to ensure it doesn't get lost. There is nothing service-specific in it, I've made only minor edits and I hope it can be a good technical reference on the topic. WebRTC is a set of protocols that allow applications, typically running on Web browsers, to exchange media (audio, video, data) with other entities. Before media can flow, however, the WebRTC entities need to discover what type of connection is possible, and among the possible connections, what’s the best to be used. This needs to happen as fast as possible, so that users can perceive the service as instantaneous as possible. WebRTC includes protocols like STUN and TURN that are designed to facilitate the establishment of connections when a direct connection is not possible. The typical case is a computer inside a home or o...
Read more

VoIP calls encoded with SILK: from RTP to WAV

SILK is a codec defined by Skype, but can be found in many VoIP clients, like CSipSimple . It comes in different flavours (sample rates and frame sizes), from narrowband (8 KHz) to wideband (24 KHz). Since Wireshark doesn't allow you to decode an RTP stream carrying SILK frames, I was curious to find a programmatic way to do it. In fact, this has also allowed to me to earn a "tumbleweed" badge in stackoverflow . You may argue that a Wireshark plugin would be the right solution, but that's probably for another day. Initially I thought it was sufficient to read the specification for RTP payload when using SILK ; the truth is that I had to reverse engineer a solution by looking at SILK SDK's test vectors. There, I discovered that a file containing SILK audio doesn't have the file header indicated in the IETF draft ("!#SILK"), but a slightly different one ("!#SILK_V3"). More importantly, each encoded frame is not preced...
Read more