I am wanting to reduce around 9.9gigs of music to 8gigs to fit on my phone. I have done some snooping and noticed a program called Lame. I also noticed a code:
for x in [ `ls -1 *.mp3` ]; do lame --preset 32 $x new32-${x}; done
I was wondering if someone would be able to explain the code and how I go about converting the bitrate to 96?
Ok, so I find myself ripping audio CDs frequently, which I then lame to mp3's to put on my media player. I usually define the --ta and --tl (artist and album) ID3 tags and batch encode each album, but don't bother with the track tags as I'd have to do each one seperately.
So, I'm working on a script to do all this for me, extracting info from 'pwd' etc. to fill in the blanks for --ta, --tl and --tt (track name). All is working well, except that I can't get sed to pass on the "" character to lame to escape spaces.
Here's what I've got so far: (trouble spot is bolded - no need to pay attention to the rest of it)
Code:
All this does is pass a 'space' on to lame, which it takes as an invalid argument.
i'm trying to rebuild mplayer and facing the same error - lame-devel package can't be found as when i try to use rpmbuild. I'm not sure why and how to fix it. Here are steps and information:
[Code]...
I have a rack of four 1TB drives all partitioned identically with three primary partitions. On each drive
- the first partition is only 64MB;
- the second is a large 900GB partition and
- the last holds all the remaining space
mdadm has been used to set up
/dev/md0 - RAID1, comprised of /dev/sda1 and /dev/sdb1
/dev/md1 - RAID5, comprised of /dev/sda2, /dev/sdb2, /dev/sdc2, /dev/sdd2
/dev/md2 - RAID5, comprised of /dev/sda3, /dev/sdb3, /dev/sdc3, /dev/sdd3
OK, so it was a silly mistake to make - but I am now need to increase the size of /dev/md0. My thinking is to reduce the size of md1 so that I can grow md0.
On md1 I have two logical volumes. I've successfully reduced the size of the volume so that I can reduce the size of md1. Now I'm at the nervous stage; I can find little written on the topic of shrinking RAID5 arrays - and even if I do this I'm unsure if I can move partitions around to regain the space I so desire.
so i have f12 installed on my hd with lvm using the whole extent of the HD , i want to reduce it so i can dual boot it with a windows system, i managed to reduce the logical volume to free some space, but i cant seem to reduce the physical volume, is this possible and how ?
View 2 Replies View Relatedi've downloaded LAME, but I don't know exactly how to use it in order to begin ripping CDS to the mp3 format.
View 9 Replies View RelatedThis may be a minor irritation, but I'm puzzled by it nonetheless. I'm ripping my CDs to MP3 using Sound Juicer with the lamemp3enc plugin and following gstreamer pipeline.
audio/x-raw-int,rate=44100,channels=2 ! lamemp3enc name=enc target=0 quality=0 ! xingmux ! id3v2mux
All was great a couple days ago, but suddenly the files' audio properties show 48 kbps bitrate when it should be well over 200. I even changed the pipeline to use "target=1 bitrate=256" with the same result. The files are being encoded as expected and that is reflected in the file size and the Statistics view in VLC.
I have two computers running Ubuntu 10.10. One has all the latest updates, but the other has not been updated in several days. This problem is happening on the former, but not the latter. gst-inspect lamemp3enc shows both computers have the same version of the encoder.
I thought I'd put it out to the forum here before submitting a bug in launchpad.
I need to download lame package to my offline ubuntu [URL] but its involved so many dependencies to download. Is there anyway to ease the pain of downloading each dependencies separately?
View 1 Replies View RelatedIs there a difference in quality between converting an audio file (e.g. *.wav) to MP3 using LAME or FFMPEG? Or do they produce the same results?
View 8 Replies View RelatedTrying to play a *.mp4 video using mplayer and the picture is jumpy and the sound completely screwed. It say it is guessing the bitrate thus:
Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding)
FAAD: compressed input bitrate missing, assuming 128kbit/s!
AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000->176400)
Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio) decoder)
In fact the bitrate should be 139 or possibly 132. So assume that is what is wrong. Have assiduously coated myself with nourishing smegma and the like but am unable to work out how to set the bitrate in this case.
I have the same mp3 on two different distros, Fedora 14 and Ubuntu 10.10, both using gnome. The md5sum of the mp3 is identical on both distros, but on Ubuntu the bitrate is 96kbps and on Fedora it is apparently 257kbps. Discovered this using RhythmBox but also appears in Nautilus browser, under the properties tag.
View 1 Replies View RelatedKernel 2.6.21.5, Slakware 12.0 I have plenty of files like this:
Code:
$ file 23-1.mp3
23-1.mp3: Audio file with ID3 version 23.0 tag, MP3 encoding
$
All I want to know about a file like this is the bitrate is has been created with. Can this data be inside the tag? What's a cli program that lets me know that information?
When using libmp3lame with ffmpeg is there a way to use lame options?
Code:
ffmpeg -i foo -acodec libmp3lame foo.mp3
For example:- --vbr-new, --abr <bitrate>, -V (n) and -q <arg>
I think it is possible to pipe ffmpeg to lame then use the options.But is it possible to do it without a pipe.
Code:
ffmpeg -i foo -f wav - | lame --vbr-new -V 4 - foo.mp3
I am trying to find out how I can estimate the IP Layer bitrate I would get from an ethernet link? Say at 100 Mbits/s or 200 Mbits/s (powerline)? Is there a formula that I could use to calculate that?
View 2 Replies View RelatedThe subject of this post is actually a question e.g. is there a mp3 ripper that allows the bitrate to be modified. I have looked at cdparanioa, which I believe is the foundation of most of the Linux rippers, but it does not allow modification of the bitrate. I realise I could ripper in flacc format but I have an mp3 player as does my grandaughter so I need the mp3 format. Incidentally is there a player(portable) that will play flacc encoded files?
View 1 Replies View RelatedI'm new to gStreamer and need to determine the bitrate of a stream. Has anyone here worked with this before and does anyone know how to do it?
View 1 Replies View RelatedMy new Wheezy install is looking pretty good and I'd like to copy some of my CDs to my harddrive. I'd like them in mp3 format, but unlike other distros, i don't seem to be able to find Lame, which is needed. I've searched for answers, but not come up with anything that works, yet. I know this question is probably asked a lot, but can anyone help me with this one? I have contrib and non-free enabled in my /etc/apt/sources.list.
View 4 Replies View RelatedHow do I get 'lame' in my path? I just installed Asunder, and I want to rip CDs in MP3 format to my hard drive. I get an error message that reads:Quote:'lame' was not found in your path. Asunder requires it to play MP3 files. All MP3 functionality is disabled.
View 4 Replies View RelatedI have 1 router 2 pcs. We had to move one pc away from the router so I bought a wifi card. Everything went fine, the card works and I have internet but the network is incredible slow.
I am using Ubuntu Lucid Lynx 10.04
lspci | grep Wire
Code:
Code:
Then:
Code:
But there are no changes.
I've been testing with the SoundConverter software and I want to write a script for it.Most music files on my pc are *.flac. But I want to convert some albums to my mp3 player with a script. Everything works fine. I do this:Code:soundconverter -b -m audio/mpeg -s .mp3 *.flacBut the quality is 128kbps.Is there any way to change the bitrate (in the terminal ofcourse)?And if this is not possible, is there an alternative that copies the tags correct like SoundConverter?
View 5 Replies View RelatedI am a happy Ubuntu user, I run it on all of my servers.
A few days ago I setup a new slice, and now I want to use it to convert ULAW (μ-law) WAV files into MP3's.
If I would only install LAME this would give me an 'Unsupported data format: 0x0007'. Because LAME doesn't support ulaw encoded wav's only Linear PCM's.
I have read that one can use the Libsndfile and use it with LAME --fileio=libsndfile (or something like that) when 'MAKE'-ing Lame. In this case because of the additional encoding library sndfile Lame can read and encode the ulaw into mp3.
I have also read another solution is to first convert the ulaw-wav into a pcm-wav with sox, and afterwards convert the output into an mp3 with lame. This seems to be a lot of extra system resources.
I am looking for a hands on tutorial howto wget, compile, install and run LAME with Libsndfile on Ubuntu Lucid (commandline only) so I can encode incoming ulaw files into mp3's?
Where can I download a copy of gstreamer0.10-lame plug-in? I looked in Synaptic Package Manager but did not see it listed.
View 2 Replies View Relatedhow to use lame to convert wav files to mp3. When using K3B sometime later, I noticed that mp3 was now listed as an option for the formats I could rip CD tracks in. I'm pretty sure it was never in the list before, and it says not just "mp3" but "mp3 (lame)" so it is obvious that my installing lame made mp3 an option in this list.The thing is, it doesn't work. I've tried burning in mp3 now, and every time, some error prevents it. The debugging output says:
Quote:
System
-----------------------
K3b Version: 1.0.5
KDE Version: 3.5.10
[code]...
and the error message, showing the command that failed, is shown in this screenshot I attached. (It wouldn't let me copy the text to the clipboard.)Any idea why it doesn't work? If I could rip tracks as MP3 in K3B, I could rip them all at once, and it would be much faster than using the lame command on each file in a terminal window.
First I've compiled and installed lame and xvid with --prefix=/usr.
Then I've compiled and install mplayer with --prefix=/usr.
Checking for libmp3lame ... yes (in FFmpeg: yes)
Checking for Xvid ... yes
mencoder -ovc help shows xvid (it works) but no mp3lame at all. And mplayer uses ffmpeg [ffmp3float], which extremely distors tracks. I'm sure it's not alsa, because music played online sounds normal.
I have finally found a native Linux music player that's as light and as simple as Foobar(unfortunately Foobar under Wine is giving me problems so I needed to find a native player). creating custom columns for the player. And since I can't find any DeadBeef forums, I'm hoping people here are familiar with this player.
So, the two main custom columns I'm looking to create are Codec and Bitrate. Codec to show if the tracker is mp3, ogg, flac, etc. And Bitrate to show the bitrate of each track.
Now there's an option in DeadBeef to create columns, but I'm having trouble figuring out how to get the column to show the info that I want. I was hoping that the way you get custom columns to show up in Foobar would work in DeadBeef. But inputting %bitrate% and %codec% into the columns in DeadBeef does not work.
how to get the two columns that I'm wanting to display the info that I want?
Or at least that's what I figured, when watching videos from ....., etc., the sound in the left channel turns into distorted high frequency noise, anyone had the same issue? Is there a fix for this? Sound in other applications works just fine.EDIT: Oh, and I'm using the proprietary flash plugin, has worked just fine on my Ubuntu Studio on the exact same computer.
View 5 Replies View Relatedprompt$sox mix2.wav soundstrack.mp3 sox FAIL formats: can't open output file `soundstrack.mp3': SoX was compiled without MP3 encoding support so i have seen a lot of other posts about this issue but none had an answer for me. i've run sox on numerous other mac and linux machines and never had a problem with .mp3 stuff provided i have lame and libmad installed. but on my current machine (ubunut 10.10) i can't seem to get sox to do anything with .mp3 despite having lame installed, reinstalling sox, etc.
View 3 Replies View RelatedI really need to burn my 64 kbps uncompressed A-Law PCM file to Audio CD uncompressed as is and make this 8 hour file appear on CD Players. What app does it for me? It must fit on CD! Goldwave makes a-law / u-law wav files BTW. I can do this with linux or windows with any application. tell me how or where to download. I have the wav files already.
View 1 Replies View RelatedI have a large collection of music albums sorted in folders which are named like this: "Zombie Ritual - 2004 - Night Of The Zombie Party" (%{artist} - %{year} - %{album}). I want to rename them so as to be indicative of the bitrate, for example, "Zombie Ritual - 2004 - Night Of The Zombie Party" => "Zombie Ritual - 2004 - Night Of The Zombie Party (@320)". It will be hard to do this manually. I tried to use EasyTag and Kid3 to do this, but they cannot add bitrate to tags.
View 6 Replies View RelatedMencoder settings... I'm doing a simple transcoding of an avi file from FFmpeg MPEG4 codec to XVID MPEG-4 codec. The problem is that the audio bitrate is changing from 192 kbps to 32 kbps. The audio codec for both files is mp3 (MPEG-1 layer 3). (I'm transcoding the FFmpeg MPEG4 encoded avi because my blu-ray player doesn't see, let alone play it. Any XVID MPEG-4 encoded avi files the blu-ray player sees and plays fine). I ran mencoder trying the four settings below but the audio bitrate still comes out at 32 kbps:
Code:
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=192 -o camping_us.avi
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=687 -o camping_us.avi
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=9146 -o camping_us.avi
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=192000 -o camping_us.avi
How to keep the bitrate at 192 kbps?