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• How Piwik uses Travis CI to deliver a reliable analytics platform to the community

  26 mai 2014, par Matthieu Aubry — Development, Meta

  In this post, we will explain how the Piwik project uses continuous integration to deliver a quality software platform to dozens of thousands of users worldwide. Read this post if you are interested in Piwik project, Quality Assurance or Automated testing.

  Why do we care about tests ?

  Continuous Integration brings us agility and peace of mind. From the very beginning of the Piwik project, it was clear to us that writing and maintaining automated tests was a necessity, in order to create a successful open source software platform.

  Over the years we have invested a lot of time into writing and maintaining our tests suites. This work has paid off in so many ways ! Piwik platform has fewer bugs, fewer regressions, and we are able to release new minor and major versions frequently.

  Which parts of Piwik software are automatically tested ?

  • Piwik back-end in PHP5 : we use PHPUnit to write and run our PHP tests : unit tests, integration tests, and plugin tests.
  • piwik.js Tracker : the JS tracker is included into all websites that use Piwik. For this reason, it is critical that piwik.js JavaScript tracker always works without any issue or regression. Our Javascript Tracker tests includes both unit and integration tests.
  • Piwik front-end : more recently we’ve started to write JavaScript tests for the user interface partially written in AngularJS.
  • Piwik front-end screenshots tests : after each change to Piwik, more than 150 different screenshots are automatically taken. For example, we take screenshots of each of the 8-step installation process, we take screenshots of the password reset workflow, etc. Each of these screenshot is then compared pixel by pixel, with the “expected” screenshot, and we can automatically detect whether the last code change has introduced an undesired visual change. Learn more about Piwik screenshot tests.

  How often do we run the tests ?

  The tests are executed by Travis CI after each change to the Piwik source code. On average all our tests run 20 times per day. Whenever a Piwik developer pushes some code to Github, or when a community member issues a Pull request, Travis CI automatically runs the tests. In case some of the automated tests started failing after a change, the developer that has made the change is notified by email.

  Should I use Travis CI ?

  Over the last six years, we have used various Continuous Integration servers such as Bamboo, Hudson, Jenkins… and have found that the Travis CI is the ideal continuous integration service for open source projects that are hosted on Github. Travis CI is free for open source projects and the Travis CI team is very friendly and reactive ! If you work on commercial closed source software, you may also use Travis by signing up to Travis CI Pro.

  Summary

  Tests make the Piwik analytics platform better. Writing tests make Piwik contributors better developers. We save a lot of time and effort, and we are not afraid of change !

  Here is the current status of our builds :
  Main build :
  Screenshot tests build :

  PS : If you are a developer looking for a challenge, Piwik is hiring a software developer to join our engineering team in New Zealand or Poland.

• FFmpeg (merge two audio)

  10 septembre 2017, par Ameer Alkateeb

  I have two long audio files. I want to merge cut some part from each video
  and merge both parts in one file.

  In below command the problem there is no audio in the second audio part. It contain the first part and the second is empty.

  What is the problem ?

  ffmpeg -f lavfi -i color=c=black 
  
  -ss 157.824 -t 99.818 
  
  -i "file1.mp4" 
  
  -ss 315.764 -t 50.308 
  
  -i "file2.mp4" 
  
  -s 854x480 
  
  -aspect 1.779167
  
  -r 25 
  
  -c:v libx264 
  
  -b:v 800k 
  
  -c:a aac 
  
  -strict experimental 
  
  -b:a 128k 
  
  -f mp4 
  
  -t 150.126 -async 1 
  
  -y "output.mp4"

• Stream RTP to FFMPEG using SDP

  9 avril 2021, par Johnathan Kanarek

  I get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream.
I want to forward this RTP data to ffmpeg and from there I can save it to file, or push it as RTMP stream to other media servers.
I guess that the best way would be to create SDP file that describes both the audio and video streams and send the packets through new sockets.

  



  The ffmpeg command is :

  



  ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i test.sdp -vcodec libx264 -acodec aac -y output.mp4

  



  I tried to send the packets through UDP :

  



  v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 RTP/AVP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=sendrecv
m=video 33302 RTP/AVP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=sendrecv


  



  But I always get (removed the boring parts) :

  



  Opening an input file: test.sdp.

[sdp @ 0x103dea0]
Format sdp probed with size=2048 and score=50
[sdp @ 0x103dea0] audio codec set to: (null)
[sdp @ 0x103dea0] audio samplerate set to: 44100
[sdp @ 0x103dea0] audio channels set to: 1
[sdp @ 0x103dea0] video codec set to: (null)
[udp @ 0x10402e0] end receive buffer size reported is 131072
[udp @ 0x10400c0] end receive buffer size reported is 131072
[sdp @ 0x103dea0] setting jitter buffer size to 500
[udp @ 0x1040740] bind failed: Address already in use
[AVIOContext @ 0x1046980] Statistics: 473 bytes read, 0 seeks
test.sdp: Invalid data found when processing input


  



  Note that I get it even if I don't open socket at all or send anything to this port, as if the ffmpeg itself tries to open these ports more than once.

  



  I tried also to open two (video and audio) TCP servers and define SDP with TCP :

  



  v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 TCP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=setup:active
a=connection:new
a=sendrecv
m=video 33302 TCP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=setup:active
a=connection:new
a=sendrecv


  



  However I don't see any incoming connection into my TCP servers and I get the following from ffmpeg :

  



  Opening an input file: test.sdp.

[sdp @ 0xdddea0]
Format sdp probed with size=2048 and score=50

[sdp @ 0xdddea0]
audio codec set to: (null)

[sdp @ 0xdddea0]
audio samplerate set to: 44100
[sdp @ 0xdddea0] audio channels set to: 1
[sdp @ 0xdddea0] video codec set to: (null)
[udp @ 0xde02e0] end receive buffer size reported is 131072
[udp @ 0xde00c0] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[udp @ 0xde0740] end receive buffer size reported is 131072

[udp @ 0xde0180] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[sdp @ 0xdddea0] Before avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 nb_streams:2
[libvpx @ 0xdeea80] v1.3.0
[libvpx @ 0xdeea80] --target=x86_64-linux-gcc --enable-pic --disable-install-srcs --as=nasm --enable-shared --prefix=/usr --libdir=/usr/lib64

[sdp @ 0xdddea0] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[sdp @ 0xdddea0] After avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 frames:0
Input #0, sdp, from 'test.sdp':
  Metadata:
    title           : 7199daf55e496b370e36cd1d25b1ef5b9dff6858
  Duration: N/A, bitrate: N/A
    Stream #0:0, 0, 1/90000: Audio: opus, 48000 Hz, mono, fltp
    Stream #0:1, 0, 1/90000: Video: vp8, 1 reference frame, none, 90k tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output file output.mp4.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 0xde3660] Setting default whitelist 'file,crypto'
Successfully opened the file.

detected 1 logical cores
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_fmt' to value 'fltp'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0xde3940] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_fmts' to value 'fltp'
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 0xde0220] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed

Output #0, mp4, to 'output.mp4':

  Metadata:

    title           :
7199daf55e496b370e36cd1d25b1ef5b9dff6858


    encoder         :
Lavf57.56.100


    Stream #0:0
, 0, 1/48000
: Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, mono, fltp, delay 1024, 69 kb/s


    Metadata:

      encoder         :
Lavc57.64.100 aac


Stream mapping:

  Stream #0:0 -> #0:0 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)

test.sdp: Connection timed out
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[output stream 0:0 @ 0xde3b40] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
[mp4 @ 0xe6a540] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
[mp4 @ 0xe6a540] Encoder did not produce proper pts, making some up.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
size=       1kB time=00:00:00.04 bitrate= 157.9kbits/s speed=0.00426x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3268.000000%
Input file #0 (test.sdp):
  Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
  Input stream #0:1 (video): 0 packets read (0 bytes);
  Total: 0 packets (0 bytes) demuxed
Output file #0 (output.mp4):
  Output stream #0:0 (audio): 0 frames encoded (0 samples); 2 packets muxed (25 bytes);
  Total: 2 packets (25 bytes) muxed
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0xde37a0] Statistics: 30 seeks, 25 writeouts
[aac @ 0xde2b00] Qavg: 47249.418

[AVIOContext @ 0xde6980] Statistics: 593 bytes read, 0 seeks


  



  Note to the "Connection timed out" in the log above.

  



  I guess that both my SDPs are wrong, any suggestions ?

  



  Alternatives to SDP are also most welcomed.