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• How to write unit tests for your plugin – Introducing the Piwik Platform

  17 novembre 2014, par Thomas Steur — Development

  This is the next post of our blog series where we introduce the capabilities of the Piwik platform (our previous post was How to verify user permissions). This time you’ll learn how to write unit tests in Piwik. For this tutorial you will need to have basic knowledge of PHP, PHPUnit and the Piwik platform.

  When is a test a unit test ?

  There are many different opinions on this and it can be sometimes hard to decide. At Piwik we consider a test as a unit test if only a single method or class is being tested and if a test does not have a dependency to the filesystem, web, config, database or to any other plugin.

  If a test is slow it can be an indicator that it is not a unit test. “Slow” is of course a bit vague. We will cover how to write other type of tests, such as integration tests, in one of our next blog posts.

  Getting started

  In this post, we assume that you have already installed Piwik 2.9.0 or later via git, set up your development environment and created a plugin. If not, visit the Piwik Developer Zone where you’ll find the tutorial Setting up Piwik and other Guides that help you to develop a plugin.

  Let’s create a unit test

  We start by using the Piwik Console to create a new unit test :

  ./console generate:test --testtype unit

  The command will ask you to enter the name of the plugin the created test should belong to. I will use the plugin name “Insights”. Next it will ask you for the name of the test. Here you usually enter the name of the class you want to test. I will use “Widgets” in this example. There should now be a file plugins/Insights/tests/Unit/WidgetsTest.php which contains already an example to get you started easily :

  /**
   * @group Insights
   * @group WidgetsTest
   * @group Plugins
   */
  class WidgetsTest extends \PHPUnit_Framework_TestCase
  {
   
      public function testSimpleAddition()
      {
          $this->assertEquals(2, 1+1);
      }
   
  }
  Télécharger

  We don’t want to cover how you should write your unit test. This is totally up to you. If you have no experience in writing unit tests yet, we recommend to read articles on the topic, or a book, or to watch videos or anything else that will help you learn best.

  Running a test

  To run a test we will use the command tests:run which allows you to execute a test suite, a specific file or a group of tests.

  To verify whether the created test works we will run it as follows :

  ./console tests:run WidgetsTest

  This will run all tests having the group WidgetsTest. As other tests can use the same group you might want to pass the path to your test file instead :

  ./console tests:run plugins/Insights/tests/Unit/Widgets.php

  If you want to run all tests within your plugin pass the name of your plugin as an argument :

  ./console tests:run insights

  Of course you can also define multiple arguments :

  ./console tests:run insights WidgetsTest

  This will execute all tests within the insights plugin having the group WidgetsTest. If you only want to run unit tests within your plugin you can do the following :

  ./console tests:run insights unit

  Advanced features

  Isn’t it easy to create a unit test ? We never even created a file ! You can accomplish even more if you want : You can generate other type of tests, you can run tests on Amazon’s AWS and more. Unfortunately, not everything is documented yet so we recommend to discover more features by executing the commands ./console list tests and ./console help tests:run.

  If you have any feedback regarding our APIs or our guides in the Developer Zone feel free to send it to us.

• How to configure ffmpeg on ubuntu to convert *.3gp to pcm *.wav ? [migrated]

  31 juillet 2012, par Monica Sol

  I'm using linux Ubuntu ver 10.04.
  I need to convert file *.3gp to PCM *.wav. I'm using for that ffmpeg program.

  When it's installed from repository by using aptitude install ffmpeg it's installing some basic version of it and I cannot convert what I need.

  I've read some stuff on the Internet and I've made what there was written.
  I've installed the latest yasm ver.1.1.0 and the newest x264 - 0.125.2208. After that I got ffmpeg using git from http://ffmpeg.org/download.html (git clone git ://source.ffmpeg.org/ffmpeg.git ffmpeg).

  I`ve tried to configure ffmpeg by myself using :

  ./configure --enable-gpl --enable-version3 --enable-postproc 
  
  --enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame 
  
  --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb

  than : time make && make install.

  Till this time everything was ok. After conversion (ffmpeg -i audiotest.3gp -f s16le -ar 8000 -acodec pcm_s16le audio.wav) I wanted to check information about this PCM *.wav file (ffmpeg -i audio.wav) and I`ve got this error :

  ~# ffmpeg -i audio.wav
  
  
  
  ffmpeg version N-42619-g6b7849e Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Jul 21 2012 00:50:52 with gcc 4.4.3
  
    configuration: --enable-gpl --enable-version3 --enable-postproc --enable-nonfree --enable-swscale --enable-pthreads --enable-libmp3lame --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb
  
  
  
    libavutil      51. 65.100 / 51. 65.100
  
    libavcodec     54. 41.100 / 54. 41.100
  
    libavformat    54. 17.100 / 54. 17.100
  
    libavdevice    54.  1.100 / 54.  1.100
  
    libavfilter     3.  2.100 /  3.  2.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0. 15.100 /  0. 15.100
  
    libpostproc    52.  0.100 / 52.  0.100
  
  [aac @ 0x943d4e0] Format aac detected only with low score of 1, misdetection possible!
  
  [aac @ 0x9443740] channel element 0.0 is not allocated
  
      Last message repeated 2 times
  
  [aac @ 0x9443740] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Number of bands (16) exceeds limit (4).
  
  [aac @ 0x9443740] Number of bands (7) exceeds limit (2).
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] SSR not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
  
  [aac @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
  
  [aac @ 0x9443740] channel element 2.0 is not allocated
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] channel element 0.0 is not allocated
  
  [aac @ 0x9443740] Number of bands (31) exceeds limit (1).
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Number of bands (16) exceeds limit (2).
  
  [aac @ 0x9443740] channel element 0.7 is not allocated
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Number of scalefactor bands in group (62) exceeds limit (41).
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] channel element 0.2 is not allocated
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] channel element 0.15 is not allocated
  
  [aac @ 0x9443740] Pulse data corrupt or invalid.
  
  [aac @ 0x9443740] Number of scalefactor bands in group (48) exceeds limit (41).
  
  [aac @ 0x9443740] channel element 2.0 is not allocated
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Number of bands (16) exceeds limit (4).
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Reserved bit set.
  
      Last message repeated 1 times
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] channel element 2.0 is not allocated
  
  [aac @ 0x9443740] Number of bands (31) exceeds limit (4).
  
  [aac @ 0x9443740] Pulse data corrupt or invalid.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] SSR not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
  
  [aac @ 0x9443740] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] channel element 0.0 is not allocated
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] channel element 0.3 is not allocated
  
  [aac @ 0x9443740] Pulse data corrupt or invalid.
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Number of bands (35) exceeds limit (16).
  
  [aac @ 0x9443740] Number of scalefactor bands in group (63) exceeds limit (41).
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] channel element 0.0 is not allocated
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] channel element 0.0 is not allocated
  
  [aac @ 0x9443740] Number of bands (38) exceeds limit (10).
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] channel element 0.2 is not allocated
  
  [aac @ 0x9443740] channel element 0.7 is not allocated
  
  [aac @ 0x9443740] Reserved bit set.
  
      Last message repeated 2 times
  
  [aac @ 0x9443740] channel element 0.2 is not allocated
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Reserved bit set.
  
      Last message repeated 1 times
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] decode_band_types: Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
      Last message repeated 1 times
  
  [aac @ 0x9443740] Reserved bit set.
  
      Last message repeated 1 times
  
  [aac @ 0x9443740] Number of bands (4) exceeds limit (1).
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Reserved bit set.
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x9443740] Number of bands (31) exceeds limit (8).
  
  [aac @ 0x9443740] Invalid Predictor Reset Group.
  
  [aac @ 0x9443740] Number of bands (31) exceeds limit (2).
  
  [aac @ 0x9443740] Number of bands (28) exceeds limit (1).
  
  [aac @ 0x9443740] channel element 0.0 is not allocated
  
  [aac @ 0x9443740] Input buffer exhausted before END element found
  
  [aac @ 0x9443740] Number of bands (16) exceeds limit (2).
  
  [aac @ 0x9443740] Error decoding AAC frame header.
  
  [aac @ 0x943d4e0] decoding for stream 0 failed
  
  [aac @ 0x943d4e0] Could not find codec parameters for stream 0 (Audio: aac, 4.0, s16, 383 kb/s): unspecified sample rate
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [aac @ 0x943d4e0] Estimating duration from bitrate, this may be inaccurate
  
  audio.wav: could not find codec parameters

  Can anyone help me with this ? What I'm doing wrong ? I'm linux newbie, but I really need to get this thing works.

• Most generic ffmpeg command to truncate video [duplicate]

  18 octobre 2014, par David542

  This question already has an answer here :

  • ffmpeg command to extract first two minutes of a video
    
    1 answer
    

  I have several thousand videos that I would like to truncate the video to the first two minutes. Here is the command I’m using so far :

  ffmpeg -i %s -ss 0 -t 120 %s' % (input_path, output_path)

  However, this runs into many errors on valid files. Is there an ’ultra-generic’ command I can use to truncate the file to the first two minutes, and it will work with all videos ?