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• Updating app uses FFMpeg version used in older version of app

  29 juin 2020, par Android Developer

  I was using FFMpeg old version in my app and now i updated my app to new FFMpeg version.But when i update my old app version to new app version it still seems to use old version of FFMpeg until i uninstall old app version and then install new app version.I tried programmatically delete cache data in my new version of app using below code but with that also I face problem in some devices which still uses old FFMpeg version-

  


    private void clearData()
    {
        try {
            PackageInfo pInfo = getPackageManager().getPackageInfo(getPackageName(), 0);
            int mCurrentVersion = pInfo.versionCode;
            SharedPreferences mSharedPreferences = getSharedPreferences("xyz",  Context.MODE_PRIVATE);
            SharedPreferences.Editor mEditor = mSharedPreferences.edit();
            mEditor.apply();
            int last_version = mSharedPreferences.getInt("last_version", -1);
            if(last_version != mCurrentVersion)
            {
                deleteCache(this);
            }
            mEditor.putInt("last_version", mCurrentVersion);
            mEditor.commit();
        } catch (Exception e) {
            FirebaseCrashlytics.getInstance().recordException(e);
        }
    }
    private void deleteCache(Context context) {
        try {
            File dir = context.getCacheDir();
            deleteCacheDir(dir);
        } catch (Exception e) { FirebaseCrashlytics.getInstance().recordException(e);
        }
    }
    private boolean deleteCacheDir(File dir) {
        if (dir != null && dir.isDirectory()) {
            String[] children = dir.list();
            for (int i = 0; i < children.length; i++) {
                boolean success = deleteCacheDir(new File(dir, children[i]));
                if (!success) {
                    return false;
                }
            }
            return dir.delete();
        } else if(dir!= null && dir.isFile()) {
            return dir.delete();
        } else {
            return false;
        }
    }


  


• how to upload a video to google driver use paperclip or carriwave

  14 janvier 2016, par bách trần nguyên

  i want to upload video to google driver.
  code models
  video model

  class Video < ActiveRecord::Base
  
    has_attached_file :video,
  
     :storage => :google_drive,
  
     :google_drive_credentials => {:client_id => AppConfig.gg_drive.client_id,
  
                                :client_secret => AppConfig.gg_drive.client_secret,
  
                                :refresh_token => AppConfig.gg_drive.refresh_token,
  
                                :scope => AppConfig.gg_drive.scope,
  
                                :access_token => Token.cache_access_token_google_drive
  
                                },
  
    :styles => {
  
      :medium => {
  
        :geometry => "640x480",
  
        :format => 'mp4'
  
      },
  
      :thumb => { :geometry => "160x120", :format => 'jpeg', :time => 10} 
  
    },# hello 123
  
    :processors => [:transcoder],
  
    :google_drive_options => {
  
      :path => proc { |style| "#{style}_#{id}_#{image.original_filename}" },
  
      :public_folder_id => '0B0VNyOkzIwUZZFFGeVhycFk0dnc'
  
    }
  
  end

  in Gemfile

  gem 'google-api-client'
  
  gem 'paperclip'
  
  gem 'paperclip-googledrive'
  
  gem 'paperclip-av-transcoder'
  
  gem "paperclip-ffmpeg"

  in controller

  def create
  
      if params[:videos]
  
        params[:videos].each { |video| Video.create(video: video) }
  
      end
  
  end

  when i run , this display error

  [AV] Running command : if command -v avprobe 2>/dev/null ; then echo "true" ; else echo "false" ; fi
  [AV] Running command : if command -v ffmpeg 2>/dev/null ; then echo "true" ; else echo "false" ; fi
  Av::UnableToDetect in AlbumsController#create
  Unable to detect any supported library

  pls. how to fix this errors

• AppRTC : Google’s WebRTC test app and its parameters

  23 juillet 2014, par silvia

  If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

  When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

  We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

  Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

  Here are my favourite parameters :
  

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

  For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

  This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

  AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

  Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

  UPDATE 12 May 2014

  AppRTC now also supports the following bitrate controls :

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

  Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

  The post AppRTC : Google’s WebRTC test app and its parameters first appeared on ginger’s thoughts.