Ubuntu Multimedia :: Editing Flac Compression In Banshee?
Jan 29, 2011
I want to rip all my CDs to flac at the lowest compression (space is not an issue) via Banshee. I have tried a few tracks but the seem to be ripped at a higher compression to the ones I have done via sound juicer (set at compression 0)
How do I adjust the flac settings in Banshee to do this? The option to edit the settings is not available for flac? I guess there is a config file somewhere?
I have Ubuntu 11.04 and I have some music in flac-format. However, when I try to transfer that music to my ipod nano 5g using Banshee, I only have the option of converting it to pcm, not i.e. mp3.
Is it possible to set Banshee up, so it converts flac to mp3 instead of PCM?
I use mplayer to play my media files. Occasionally I want to take screenshots of media I am playing. I use the -vf screenshot option to take screenshots which generates shotxxxx.png. The issue is that all those png are not compressed and usually large. When dealing with Hi-Def media they are extremely large, each is 4-5 MB in size at the least. Is there a way I can set the option for compression of png images. If I use imagemagick the same files get compressed to like 1.5MB of png file.
some reason it seems like the lowest it goes is ~64kbps (~ implying variable bitrate).So yeah, any thing that'll let me do the compression limbo better? (How low can you go? ) A different program? Unlock bonus stage? What
I am trying to convert hundreds of BMPs into a single AVI file, without compression. Since every pixel matters in my case, I don't want any kind of compression. It's fine if the output file is extremely large.
I'd like to know if it's possible to achieve this with packages coming with Ubuntu 11.04. I have tried ffmpeg/mencoder, but they either compressed the output file, or the output file is not playable in totem. I am a new user for both tools. there is actually a way to get uncompressed avi from these tools.
"BMP to AVI Sequencer" [URL] does the job perfectly. I successfully converted 180 1080P BMPs into a 1G avi, running at 30 fps. Unfortunately I need a command line tool this time.
I finally got K9copy to work and I'm using it to extract DVDs to my hard drive. I extract a dual layer DVDs it becomes a 5GB DVD. In the settings, I have DVD size as 9GB because I assumed this would not cause any compression. When I use k9copy assistant to change the shrink factor before ripping, I can only change the shrink factor of some titles to 1.67 from 2.50. I want the shrink factor to be 1 since this seems to ensure that the output of the file is the same size as the title on the dvd. I want the copied DVD to be the same size as the original. I have no problem with hard drive space since I have 6 terrabytes of free space.
I'm head engineer for a community radio station. One of our compressors has recently died and I want to replace it with software compression on our streaming machine. I've been looking at VST/LADSPA/JOST etc and I've got nothing but confused. Hence I'm coming here looking for help.
My spec is this. We need a system to run a compressor and EQ, probably a graphic and a parametric. The compressor needs to be multiband and have proper control of the compression curve. A way of configuring it without too much command line work would be an advantage. All needs to run on an Ubuntu server with not massive overhead as it's not exactly the latest hardware. I suspect it would be possible to run WINE for windows requiring VSTs if needed.
I was hoping some of you good people might be able to point me in the direction of a suitable host and plugins to achieve the effect described.
How to install XMMS from source with the Flac plugin. It was originally based on howto's from blogs. I have tested this on Karmic Koala and it should work fine.
We will start off with XMMS. We'll take the plugin later..
I have tried everything within my power and knowledge to get xmms2 to play my flac files, but I just get no sound. I've had to resort to mplayer for now for flac. For the record, yes I have the flac plugin installed along with the all-plugins package.
I would like to know which repositories I have to add to allow k3b to rip cds into flac because even though I have installed k3b codecs, I cannot rip cds into flac.
Amarok will play flac files just fine, but when i try to index them into the library via "Update collection" they are not added.
All other threads i can find on this kind of thing also have Amarok not playing flac....but if i drag and drop them in, they play just fine, so i know its not a codec problem.
Is there something i have to enable? Like in some media players, you can set which formats are scanned for...but i cant seem to find such a feature available in any of the Amarok settings...
I have a whole bunch of wma files I want to convert to a more liked format like wv flac or shn. Shntool does not seem to want to do that soundconverter and soundkonverter both refuse too. My wma format is wma1 or wmal cannot tell if it is 1 the number or l the letter L. 1. They play in xine straight off. Xine is in your synaptic 2. To convert quickly install dBpoweramp under wine again quick and no fuss.
I have all my CDs in FLAC format for playback in my home but like most folks I have a portable.Now, converting my collection to Mp3 fits on my player but its a pain to manually convert each new album. It would be cool to script something that could be run on cron or manually to keep them synced.
As K3b is the standard ripper I suppose it must be possible rip an audio cd to flac? In the plugin menu I see that the flac decoder is installed but not the encoder. Did someone already succeeded in installing the flac encoder plugin?
FYI: With virtual folders I am able to copy from the flac directory. It works to encode with flac like this but I experience the last seconds of the song are gone...
I am looking for a command line command to convert ~2500 .flac files to .ogg files. All of the .flac files are in one folder and I would like to have the .ogg files put in a folder labled OGG - I would like to retain song information etc if possible.
Background: I'm digitizing an old record. I used Sound Recorder to make FLACs of each side of the LP, and now I need to edit them down to individual tracks. I'm using the development version of Ubuntu 10.04.
Problem: Audacity ignores these FLAC files when I attempt to import or open them.
The FLAC files play if I hover the mouse over them in GNOME.
Audacity gladly imports FLAC files I ripped from CD last year (using Sound Juicer I think).
But when I try to do the same for my recorded FLAC files, nothing happens. I tried running audacity in a terminal window, and there is no additional output when I attempt the import. This comes out when I start it:
Looking at the file properties, they're basically the same except the Juicer-generated file has values filled in for title, artist, album, year. They are all FLAC Stereo 44100 Hz. I thought it might be a file-size problem, so I made a 5-second test with Sound Recorder, and that also refuses to import.
So, why do only the ripped files work with Audacity?
I need to split big FLAC file(s) to single tracks. Is there any Ubuntu app similar to Medieval CUE Splitter? Also, I need spectrum analyzer app to determine is FLAC file real lossless or is transcoded from some lossy format. Something like this.
I am using 11.2, KDE 4.3.5 with Grip and Flac to rip and encode CDs. Two questions:- Have read flac -help and do not understand the -o option which is used in a few example threads. The man page suggests that -o should be used with subsequent parameters such as --output_name=FILENAME for example. In the example posts this general option is used on its own. What for?
The installation when first used, already had a string of options including --best on the command line. I cannot find any documentation for this. I assume it is a preset in the same way as for example --preset extreme in lame.
I have some downloaded files of radio programme from BBC which are .aac. I have never come across this type before. Googled it so now I am a bit wiser and VLC seems to play them without problem but they are not accepted by my upnp client devices which are happier with flac or mp3. I would prefer flac but what is the preferred conversion software.
I am running ubuntu 10.04 and try to get 24bit/96 Khz flacs heard with Amarok (Xine engine). Amarok after some seconds jumps to the next title -> no sound. No problems with Rhythmbox.