Wednesday, July 6, 2022

Control Android from Mac

 Use scrcpy (https://github.com/Genymobile/scrcpy) to control Android from a Mac:

Installation on Mac:

    brew install scrcpy

Then,

    scrcpy

If there is error related to android.permission.MANAGE_APP_OPS_MODES, enable "disable permission monitoring" in developer options.

Saturday, November 21, 2020

Create subtitle for your YouTube video on MacOS

If your language is supported by automatic captioning by YouTube, you are OK. But if not, then you can create your own. 1) Basically, follow this "Creating translated subtitles with AI" article Follow the steps in this article: https://cloud.google.com/community/tutorials/speech2srt However, some parts of the code needs to be updated to be compatible with current Speech API: 1-1) enums is deprecated #encoding = enums.RecognitionConfig.AudioEncoding.LINEAR16 encoding = speech_v1.RecognitionConfig.AudioEncoding.LINEAR16 #https://github.com/googleapis/python-speech/blob/master/UPGRADING.md#enums-and-types 1-2) long_running_recognize #operation = client.long_running_recognize(config, audio) operation = client.long_running_recognize(request={"config":config, "audio":audio}) #https://stackoverflow.com/questions/64287266/google-speech-to-text-jupyterlab-notebook-script-run-locally-using-google-cloud 1-3) nanos is deprecated #start_ms = int(w.start_time.nanos / 1000000) start_ms = int(w.start_time.microseconds / 1000000) #https://github.com/googleapis/python-speech/issues/71 #end_ms = int(w.end_time.nanos / 1000000) end_ms = int(w.end_time.microseconds / 1000000) #https://github.com/googleapis/python-speech/issues/71 2) Install ffmpeg for converting movie to audio file 3) Then you can execute the creation of SRT file to be uploaded to YouTube: 3-1) Download your video from YouTube Studio 3-2) Convert video to audio Uncompressed WAV is best for speech to text ffmpeg -i video.mp4 -ac 1 -ar 44100 audio.wav 3-3) Upload to Google bucket ../../../google-cloud-sdk/bin/gsutil cp audio.wav gs://$BUCKET_IN/ 3-4) Do speech to text python3 speech2srt.py --storage_uri gs://$BUCKET_IN/audio.wav --language_code id-ID --sample_rate_hertz 44100 --out_file audio 3-5) Upload audio.srt file to your YouTube video

Saturday, June 2, 2018

Turn PDF to text on Mac

To turn PDF to text on Mac, first install imagemagick using homebrew. Then install tesseract using homebrew. Download and put "eng.traineddata" into tessdata directory for English text.

Then run the following command:

convert -density 300 [input.pdf] -depth 8 -strip -background white -alpha off [output.tiff]
tesseract [input.tiff] [output] -c load_system_dawg=false -c language_model_penalty_non_freq_dict_word=0.1 -c language_model_penalty_non_dict_word=0.15 -c matcher_bad_match_pad=0.20


Tuesday, September 6, 2016


Gitinspector helps you analyze git activities from your repository.
Here is one of suggested options to use (using Terminal on Mac):

python gitinspector.py -Tlrw --format=htmlembedded --file-types="cs,js,asax,ascx,asmx,aspx,cshtml,fs" -x "/lib" >netisware2.html

Saturday, January 9, 2016

Rsync revisited (for Mac OS 10.10.5 Yosemite)

Get latest rsync from https://rsync.samba.org. Download also its patches, which now are in separate file. Patch.
patch -p1
patch -p1
patch -p1
Do configure and sudo make; make install
By default it will be installed to /usr/local/bin

Then execute rsync as follows:
sudo /usr/local/bin/rsync -aHxvuPESWANXN /src /dst --progress --dry-run

a equals -rlptgoD
N preserve createtimes
H preserve hardlinks
A preserve ACLs
X preserve xattr
x don't cross file system
v verbose
u skip files that are newer on the receiver

E preserve the file's executability
S handle sparse files efficiently
W copy files whole (without delta-xfer algorithm)

P equals --partial --progress, i.e., shows progress, and don't delete partially synced files when command is interrupted (to be resumed later)

Reference
http://www.lbackup.org/developer/rsync_hfs
https://discussions.apple.com/thread/6436787?start=0&tstart=0

Friday, July 10, 2015

As follow up to the entry about initial setting of Yamaha RTX on Mac, here is how to run tftp on Mac and do a firmware update:

First, start tftp server:
sudo launchctl load -w /System/Library/LaunchDaemons/tftp.plist

Then run these commands
tftp xxx.xxx.xxx.xxx (the IP address the router)
tftp> mode binary

tftp> put rtx1100.bin exec

Then stop tftp
sudo launchctl unload -w /System/Library/LaunchDaemons/tftp.plist

Thursday, June 11, 2015

How to build OpenVCS (Telepresence) on CentOS64

Use this OpenVCS:
http://code.google.com/p/openvcs/ (redirected to https://code.google.com/p/telepresence/?redir=1, version "last updated December 2014”). This version is said to be tested on CentOS64)

Build/install on VirtualBox

Get CentOS image
http://virtualboxes.org/images/centos/
(use CentOS 6.6 minimal x86_64, which is current version in December 2014)

Build and install according to https://code.google.com/p/telepresence/wiki/Support_BuildingSourceCode

OpenSSL is not built, because current version is valid

libogg, libvorbis and libtheora are not built, since they are optional.

libfaac make failed, and ffmpeg cannot be installed without this, so:
Remove line 126 from mpeg4ip.h then build (http://stackoverflow.com/questions/4320226/installing-faac-on-linux-getting-errors)

OpenAL Soft is not built, because it is optional (for 3D audio)

openOffice is optional and not built yet

When using WebRTC from Chrome connection cannot be made due to "WS handshaking not done yet” error, so modify tsip_transport_layer.c in doubango as follows (see https://code.google.com/p/telepresence/issues/detail?id=22)

Line 367:
                case event_accepted:
                case event_connected:
                        {
                                tsip_transport_stream_peer_t* peer;
                                // find peer
                                if((peer = tsip_transport_find_stream_peer_by_local_fd(transport, e->local_fd))){
                                        // If peer already exists.. do nothing :0
                                        return  0;
                                }
                                else {
                                        TSK_DEBUG_INFO("WebSocket Peer accepted/connected with fd = %d", e->local_fd);
                                        return tsip_transport_add_stream_peer(transport, e->local_fd, transport->type, tsk_true);
                                }
                        }

Doubango needs SSL to be enabled during built, otherwise, WebRTC will be connected but ended in "not acceptable”, and telepresence log will show "Remote party requesting DTLS but it is not enabled” error (see https://code.google.com/p/doubango/wiki/Building_Source_v2_0)
OpenSSL is pre-installed on CentOS but Doubango needs header files, so install openssl-devel first:
sudo yum install openssl-devel 

Then configure doubango with SSL:
./autogen.sh && ./configure --with-ssl --with-srtp --with-speexdsp --with-ffmpeg
and then build doubango.

SSL certificate (self-signed is OK) needs to be created:
https://code.google.com/p/telepresence/wiki/Configuration_Security
So, set SSL certificate:
ssh-keygen -b 1024 -f root -t dsa
Then point setting at /usr/local/sbin/telepresence.cfg to locations of pem files created above.

Telepresence make will fail with undefined reference at libtinySAK, so edit Makefile, change this LDFLAGS line:
LDFLAGS= -L/opt/openoffice4/sdk/lib
to:
LDFLAGS= -L/opt/openoffice4/sdk/lib -WI,-rpath,/usr/lib -ldl
(see http://sourceforge.net/p/cppunit/bugs/201/)

Remove openssl-devel to prevent problems with dual installation of openssl:
sudo yum remove openssl-devel

Make sure iptables is disabled to allow connections:
sudo /etc/init.d/iptables stop
(To turn off firewall on boot, do this:
chkconfig iptables off)

Edit transport configuration entry at /usr/local/sbin/telepresence.cfg, better add ip address instead of leaving the asterisks because it seems to get 0.0.0.0 as IP address, and uncomment all transport configuration entries (see https://code.google.com/p/telepresence/wiki/Support_Testing_the_system)

Start telepresence (if —config is not specified, it does not seem to use the cfg file provided by “make samples”):
telepresence —config=/usr/local/sbin/telepresence.cfg
Then log data will be shown.

Test using WebRTC. https://code.google.com/p/telepresence/wiki/Support_Testing_the_system). Access http://conf-call.org, go to settings, set the IP address of telepresence, then call.