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• Announcing Piwik Community Meetup in Munich : Register now !

  5 mai 2014, par Piwik Core Team

  We’re excited to announce our second Piwik community Meetup ! This will be a unique opportunity to connect with other Piwik users, and meet the core team behind Piwik.

  Updated : presentation slides

  The Making Of The Analytics Platform Of The Future

  Piwik in Enterprise

  Piwik Analytics Platform

  Guest lecture : TV to Web Analytics

  Guest lecture : Oxid analytics with Piwik

  When and where is the meetup ? (update, see below)

  Location : Munich, Germany
  Date : Tuesday July 29th 2014
  Time : 5PM
  Language : German/English
  Cost : Free !

  Register now !

  Who can join ?

  All Piwik community members (users, translators, contributors) are warmly invited to join the meetup. Almost all of the core team will be present, we’re looking forward to meeting you !

  What to expect for the Piwik meetup ?

  The meetup will consist of three speakers giving quick 15-20 minute presentations followed by Q&A.

  • Discover some of the upcoming features
  • Learn tricks to make the most out of Piwik
  • Networking and socialising… and an after party in a local bar to continue the discussion !

  Who can join the meetup ?

  This meetup is open to all Piwik users and members of the community.

  • If you are using Piwik to improve your websites and apps, or generally curious about digital analytics and marketing
  • If you are interested in the platform, integrating your app with Piwik or building plugins, come meet with other developers and creators of Piwik

  Timing

  • Doors open at 17:00
  • Starts at 17:30
  • Ends at 20:00 – 20:30

  Schedule

  • 17:30 – Welcome speech (Peter Boehlke (german))
  • 17:40 – Piwik for governments & corporations – Piwik PRO case study (Maciej Zawadziński english)
  • 18:05 – Break (25 minutes) – Coffee, Tea, pastries and cold buffet (free)
  • 18:30 – Overview of the platform Piwik, custom data tracking, publishing on the new Marketplace (Thomas Steur (german))
  • 18:55 – Break (10 minutes)
  • 19:05 – Piwik users present interesting real world use cases (german)
    – TV-to-Web analytics (Jasper Sasse)
    – Piwik from a SEO’s perspective (Thomas Zeithaml)
    – Using the Piwik Framework to analyze Shop-Data (Joachim Barthel)
  • 19:30 – Break (10 minutes)
  • 19:40 – Next big features, milestones, future roadmap (Matthieu Aubry english)
  • 20:05 – Break (5 minutes)
  • 20:10 – Summary & end of the conference (german)
  • After party at a nearby bar or restaurant (open end)

  Call for Papers

  We would like to hear about how you use Piwik ! If you’d like to present your interesting use case on the conference (speaking time 5 to 7 minutes), please contact us at hello@piwik.org !

  Meetup location

  Munich Workstyle
  Landwehrstraße 61
  80336 München
  Location / Directions

  Parking space is limited : We recommend to use public transport !
  Stations nearby :
  
   S-Bahn : Hauptbahnhof, Stachus (both 700m)
  
   U-Bahn : Stachus (700m), Theresienwiese (400m)

  Beverage pricing
  
   Mineral water : EUR 3,10 (0,75l)
  
   Softdrinks / juices : EUR 2,10
  
   Beer : EUR 2,80

  

  A special thank you to our sponsor Mayflower GmbH !
  
  
  

  Register now

  Seats are limited, please register today to secure your seat :
  Register now !

• Improving Google Cloud Speech-to-Text accuracy

  6 juillet 2020, par lr_optim

  I'm working on a project where I need to perform these steps :

  


  

  1. Record a voice call (.webm -file)

  


  2. Split the webm -file into chunks with ffmpeg and convert the file into wav

  


  3. Transcribe the chunks using SpeechRecognition -library and Google Cloud API

  


  


  I've faced problems with the transcription accuracy and wondering if there is something I could do to improve it. At the time I'm splitting the original file into 30s chunks. I thought there might be one problem, that I might be missing words because of splitting so I've tried also with longer chunks under 60s but didn't notice any improve in accuracy.
Reading trough the speechRecognition docs I decided to set r.energy_threshold = 4000, I also tried to set the energy_treshold dynamically like this :

  


  with sr.AudioFile(name) as source:
    r.dynamic_energy_threshold = True
    r.adjust_for_ambient_noise(source, duration = 1)
    audio = r.record(source)


  


  I've also tested en-US and en-GB to see if there's some difference but there isn't as much as I'd want. The program is supposed to work with english language spoken by nordic people. If someone has experience about choosing a right language model for people speaking with accent, please let me know.

  


  This is the ffmpeg command is use to split the webm file into chunks : command = ['ffmpeg', '-i', filename, '-f', 'segment', '-segment_time', '30', parts_dir + outputname + '%09d.wav']

  


  Is there somethig I could do better ? I'm wondering if the quality is not good enough an Google is having hard time because of that ?

  


  The main problem is I'm getting bad results (lots of wrong words) from Google and wondering if there is something I could do about it.

  


• Convert an RTSP/RTMP-Livestream with G.711 audio into RTMP/RTSP with aac-audio

  31 août 2018, par Alex Fuhr

  im new at this forum and my english skills are not the best !

  I have a website where i publish the videostreams of the cameras to show what happens inside during the nesting-time live ! An guy with high IT-skills has build me a little Server for Restream it (Datarhei-Restreamer) But this guy has still no time and worse response-times...

  To my Problem : The Restreamer dont support the "G.711" Audio-Codec from the cameras and the Livestream are still without audio at the website. So, i need to convert the Livestreams (RTSP and RTMP- in H.264) so that the audio changes to "aac" or something other supported. But i have no plan how to do this. I tried it with FFMPEG but i dont find the correct commands to get the my result. There is something with an Streaming-server to send the new created stream to - i dont get it into my head to do this (i need just a stream that are viewable with VLC player and then as input for my restreamer-server, jsut the same like ca

  I want to change the source-stream into the correct codec (audio from G.711 to AAC, the rest like source) and then, put this "new" stream into my Restreamer-Server and it will work fine ! (Tested with XSplitbroadcaster, but dont runs on Raspberry, only 1 instance runable but 2 livestreams needs to be encoded at same time) And this programm has annoying bugs (endless and not removeable error-messages, but running stream)

  I have a new second raspberry that are planned as "live-encoder" for the restreamer-raspberry were the "new" streams are are going in (rtmp/rtsp-input on a graphical ui) I try it still with FFMPEG but still no result...

  Sorry about this long text with all the language-issues but im really frustrated with it because i have purchased 2 new cameras with total 450 euros just to get the livestream with sound now :(