My goal is to convert some mpeg4 files in my hard disk into mpd files that will alllow me to use it in mpeg dash streaming .i read about gpac's MP4Box capability to create mpd files and i followed the instructions of the following link to successfully compile gpac for ubuntu like in the instructions in this two links
http://gpac.wp.mines-telecom.fr/2011/04/20/compiling-gpac-on-ubuntu/
http://gpac.wp.mines-telecom.fr/2012/02/01/dash-support/
But when i try to execute any command such as
MP4Box -dash 10000 -frag 1000 -rap myFile.mp4
I get the following error
Option -dash unknown. Please check usage
I wonder is there any commands or instructions that i must execute when building gpac to add the dash and if is there any other methods to get my own MPD File not those provided by itec.
Thanks in advance !!!
Try to follow the compile instruction carefully & make sure to fetch the latest version from SVN.
MP4Box should work with your commands.
looks like you are using a outdated version of MP4Box. try downloading and compiling the latest one from here (for me the same command works): http://gpac.wp.mines-telecom.fr/downloads/gpac-nightly-builds/
Related
I want to extract audio features from a large set of files using pyaudioanalysis, using the following command line (as suggested on the github project's page):
python3 audioAnalysis.py featureExtractionDir -i data/ -mw 1.0 -ms 1.0 -sw 0.050 -ss 0.050
This seems to indeed run the feature extraction - it takes time and return to prompt without error - but does not write any csv files.
Any hint about why this does not work would be greatly appreciated. Thank you.
Found the answer:
the files to be analyzed were actually ADPCM-compressed files, which apparently does not work. Therefore, this led to the following error (in the background, it was NOT returned, and only discovered while trying to run the command in a system call in R):
ValueError: Unknown wave file format: DVI_ADPCM. Supported formats: PCM, IEEE_FLOAT
Once files were uncompressed, the pyAudioAnalysis call worked as expected.
I am using pocketsphinx for offline speech recognition. I use lmtool to get language model and dictionary.But the language model has extension .lm but pocketsphinx requires .lm.bin file. So, how can I convert this?
You just need to:
1. Download http://sourceforge.net/projects/cmusphinx/files/sphinxbase/0.8/sphinxbase-0.8-win32.zip
Unpackage sphinxbase-0.8-win32.zip. The folder will be PATH\
In my case thats C:\Users\carope9\Desktop\
Move lm file to PATH\sphinxbase-0.8-win32\bin\Release
Open CMD and write cd PATH\sphinxbase-0.8-win32\bin\Release
Write sphinx_lm_convert -i YOUR_LM_FILE -o YOUR_LM.BIN_FILE
example: sphinx_lm_convert -i es_ES.lm -o es_ES.lm.bin
Your new lm.bin file will be into PATH\sphinxbase-0.8-win32\bin\Release
If you don't use Windows need to download source files from http://sourceforge.net/projects/cmusphinx/files/sphinxbase/0.8/sphinxbase-0.8.tar.gz but I don't know how to install it I'm reading https://sourceforge.net/p/cmusphinx/discussion/help/thread/c67930c0/?limit=25
P/D: According to some people this doesn't work, it worked for me and I don't know how to correct their error. Hope it helps you.
I have started working with nw.js, creating a test app to play audio files from. I am working on OS X 10.11.
The code for the player is:
<audio id="audio_player" controls>
<source src="path/to/file/song.mp3" type="audio/mp3">
</audio>
The player and controls appear, but the file will not play. I have tried using a relative path, and an absolute path, trying both
"/path/to/file/song.mp3"
and
"file:///path/to/file/song.mp3"
schemes.
I have verified the path is valid in all cases.
This page:
http://docs.nwjs.io/en/latest/For%20Developers/Enable%20Proprietary%20Codecs/
tells me that mp3 should be supported (for v0.22.1+, though I haven't found a way to tell the version of my mp3, though is was made recently so I assume is the the latest codec.)
Just to try more stuff, I followed the instructions on these pages:
http://docs.nwjs.io/en/latest/For%20Developers/Enable%20Proprietary%20Codecs/
(It appears mp3 gained support since this doc came out.)
https://github.com/nwjs/nw.js/wiki/Using-MP3-%26-MP4-%28H.264%29-using-the--video--%26--audio--tags.
I downloaded the ffmpeg libs from here:
https://github.com/iteufel/nwjs-ffmpeg-prebuilt/releases
and placed copies here:
find . -name libffmpeg.dylib
./dist/nw.js-examples/osx64/nw.js-examples.app/Contents/Versions/67.0.3396.87/libffmpeg.dylib
./node_modules/nw/nwjs/nwjs.app/Contents/Versions/54.0.2840.99/nwjs Framework.framework/libffmpeg.dylib
./node_modules/nw-builder/cache/0.31.2-sdk/osx64/nwjs.app/Contents/Versions/67.0.3396.87/libffmpeg.dylib
cp ~/Downloads/libffmpeg.dylib ./dist/nw.js-examples/osx64/nw.js-examples.app/Contents/Versions/67.0.3396.87/libffmpeg.dylib
cp ~/Downloads/libffmpeg.dylib "./node_modules/nw/nwjs/nwjs.app/Contents/Versions/54.0.2840.99/nwjs Framework.framework/libffmpeg.dylib"
cp ~/Downloads/libffmpeg.dylib "./node_modules/nw-builder/cache/0.31.2-sdk/osx64/nwjs.app/Contents/Versions/67.0.3396.87/libffmpeg.dylib"
Still to no avail running either
npm run dev
or
npm run prod
and opening the packaged app.
I can play the file fine with the same code from a web browser.
I don't know what else to try, help would be much appreciated, thanks.
I'm new to Gromacs (and protein-analysis coding in general, but have some experience with python-based code). I'm trying to convert a .ene file received from pyDock to a more readable format (it currently opens
When I try to use different commands that the gromacs guide says accept .ene files (gmx eneconv and gmx dump), for example
gmx eneconv -f project.ene -o converted.edr
and
gmx dump -e project.ene -om read.mdp
I get the error
File 'project.ene' cannot be used by GROMACS because it does not have a recognizable extension.
The following extensions are possible for this option:
.edr
I have updated my OS and re-installed gromacs. My installation is working according to the 'Getting Started' page.
I am also open to suggestions for other programs to open and read the .ene file type.
Thanks!
"ene" is short for "energy" which contains binary data for energy, temperature, volume, etc, and is only used in older version of Gromacs. In new version of Gromacs this is replaced by "edr" file which contains portable binary data. See "gmx dump -h", you will find the following lines:
Options to specify output files:
-o [<.edr>] (fixed.edr)
Energy file
If you insist on using ene file, you should install an older version of Gromacs.
I am trying to build ffmpeg on Android. There are many tutorials. Some are very old.
So I want to try one that can use newer version of ffmpeg and Android NDK.
After long time searching, I find one, guardianproject / android-ffmpeg
The project was updated several months ago.
NDK r8 is used. ffmpeg is put from online, so a latest version.
After I follow all the instruction, I am confused which result I should use, and how to use it.
The README mentions testing, like:
# embedding metadata into a matroska video /data/local/ffmpeg -y -i test.mp4 \
-attach attach.txt -metadata:s:2 mimetype=text/plain \
-acodec copy -vcodec copy testattach.mkv
First, I fail to find the path: /data/local
Second, this is a command. How will I use it in Android?
Totall confused.
Any light?
I presume that you have successfully built the ffmpeg executable and understand the difference between static and shared libs in the build. If not you should read up on that before trying to exec it on the CLI in android.
I think that the author uses ffmpeg on the cli using either system.exec or processbuilder techniques to run and executable located on the phone in ./data/local/$yourSubDir....
see here for discussion on ffmpeg on cli in android and note that alot of people prefer to use the full JNI route employing java interfaces to wrap calls to ffmpeg.main(). IMO the JNI route is more robust for apps you intend to distribute. But it works on the cli.
the tests that confused you are just expressions for using ffmpeg that would need to be called in android using the approach you prefer ( system.exec or processbuilder ). You can get familiar with the cli testing by simply running ffmpeg in windows/ linux in your dev environment. Get a shell there and play with samples mentioned in the ffmpeg faqs.
you can look at the halfninja project on git for more examples on JNI approach.