Skip to main content

Test::Harness, or a lesson about wheels not to be reinvented

Let’s say you have your unit tests in place, using something not particularly esoteric as Test::More. Good.

Now you want something to give some color to your output, so green is Good, red is Bad and you have a quicker feedback.

Time ago I wrote a quick shell script to run all the unit tests, interpret the output (in TAP format), print on screen some color output and stop the tests if something fails.

The core was:
function check_test_result() {
red='\e[0;31m'
green='\e[0;32m'
end='\033[0m'

result=`perl $1`;

echo "$result"

perl -e '{ my $input = join(" ", @ARGV); if ($input =~ /not ok/m) { exit 0; } exit 1; }' $result

if [ $? -eq 0 ]
then
echo -e "$red Not all tests passed. FAILURE$end"
exit -1
else
echo -e "$green All tests passed. SUCCESS$end"
fi
}


This works as soon as the unit tests fail (printing ‘not ok…’), but doesn’t quite work if something else is wrong (and you see the typical “your test died just after…” at the end).

Rather than improving this script, I was looking for a low hanging fruit, and eventually the easiest way I've found is to use Test::Harness prove, which BTW has color output by default.

So instead of the above (which needs also a few more lines to get the list of .t files), I can just use:

$ prove -v t/*.t


I mentioned Test::Harness prove in this post too, where I was using the JUnit module to convert from TAP to JUnit format, and get some nice code coverage report on Hudson.

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.

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...

Extracting Opus from a pcap file into an audible wav

From time to time I need to verify that the audio inside a trace is as expected. Not much in terms of quality, but more often content and duration. A few years ago I wrote a small program to transform a pcap into a wav file - the codec in use was SILK. These days I'm dealing with Opus , and I have to say things are greatly simplified, in particular if you consider opus-tools , a set of utilities to handle opus files and traces. One of those tools, opusrtp , can do live captures and write the interpreted payload into a .opus file. Still, what I needed was to achieve the same result but from a pcap already existing, i.e. "offline". So I come up with a small - quite shamlessly copy&pasted - patch to opusrtc, which is now in this fork . Once you have a pcap with an RTP stream with opus (say in input.pcap ) you can retrieve the .opus equivalent (in rtpdump.opus ) with: ./opusrtp --extract input.pcap Then you can generate an audible wav file with: ./opusd...