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.
Kernel 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?
This should be a very elementary question. I have a URL like http://SERVERNAME/file.wmv. When I enter it in "Open Location" in gnome-mpLayer it connects to the server and plays the stream. But when I run
mplayer "URL"
in the terminal I get a crazy endless loop of
Playing URL. Resolving SERVERNAME for AF_INET6... Couldn't resolve name for AF_INET6: SERVERNAME Resolving SERVERNAME for AF_INET... Connecting to server SERVERNAME[xxx.xx.xxx.xx]: 80... Cache size set to 320 KBytes
I think my usage of mplayer in the terminal is correct, since I can watch other URL's.It's only this specific one that doesn't work (I am not authorized to write the URL because they want it to be private.So my question is: Does anyone know why I get this loop? Or is it possible to see how mplayer is called by gnome-mplayer and what output messages it generates?I use gnome-mplayer 0.9.9.2 and mplayer 1.0rc4-4.4.5 on Ubuntu 10.10.
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?
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?
The 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?
This 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.
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 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.
I 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.
Mencoder 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:
When I convert files with mencoder I get the output with incorrect audio bitrate. Seems that mencoder ignores the bitrate I pass to it. Here's my script:
Result file should have 128kbps audio bitrate, but here are the results: Input file: 02_seminar.avi, 42Mb video: 432x320 00:05:10 25fps DivX5 1Mbps audio: 48KHz 00:05:10 Stereo 138Kbps mp3
howto to transcode multiple mp3 files to another bitrate? I have a lot of 320bits mp3 which should be converted to 192bits before I can play them on my car stereo.
I 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'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?
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.
I am looking for an application to detect mp3 duplicates using the mp3 spectrograph (sound wave matching) or something similar because simply I have many mp3 files that are the same but with different bitrate,ID3 tags,filesize
ive had ubuntu 8.1 for awile now on my laptop and windows on desktop. ive downloaded lives media maker or movie maker what ever u widh to call it. after opening lives it says i need to download mplayer. i got to the mplayer homepage thur the downloads given here, and dont know what to do. never messed with binary or codes and dont want to realy mess things up. so looking for some advice here. how do i get mplayer on ubuntu?
or any conversion I suppose. My question is simple: How do I use this command: mplayer -ao pcm *.m4a -ao pcm:file="*.mp3" to convert whole directories and place the converted tracks into another directory. The way it's listed above it will convert all songs with a .mp4 extension into a single file with a .mp3 extension. Which of course, is not what I want, but I know of no other way to use this command.
has anyone else noticed that any program to do with FLV (flash video) files does not work on F15?i can't get mplayer or vlc to play them (totem does it seems) and mencoder and ffmpeg segfault without any useful error message when trying to convert themany ideas? i'm assuming some problem with the rpmfusion codecs or packages?
I went to install VLC and mplayer in 64-bit openSUSE 11.3 and neither of them were in the repositories. I have both the oss and non-oss repositories enabled but still can't find them. I am running KDE 4.4.4. Are these nonexistant for this version or am I just missing something?
I'm having a problem running Mplayer in Ubuntu 9.1. I've installed and uninstalled several times. Mplayer refuses to open from the applications menu.n I attempt to open from terminal it comes up with the following: mplayer: error while loading shared libraries: /usr/lib/i686/cmov/libavcodec.so.52: file too short I've also tried other players such as Gnome Mplayer to play the video and it plays really fast with no audio or video. This has really been the only problem i've run into since switching from windows to linux two weeks ago.
I've installed both mplayer and mplayer-gui packages, but I cannot find and launch mplayer with graphical interface. I want the classic mplayer's gui and seems that isn't there after installed. Doesn't appear ind applications on launcher and when I'm trying to run from terminal the mplayer-gui command I get the error:command not found. So, where is mplayer-gui and how to get it back. This problem doesn't exist in any previous ubuntu version.
When I try to open a video-file in mplayer, there are no sound. (have tried flv and mp4 files) But it plays mp3-files fine.Other media programs like vlc have no problem playing the same videoes with sound.I have checked if mute was on and if the volume was turnd up high, but that wasn't the problem.
Linux 2.6.21.5, GNU (Slackware 12.0) MPlayer 1.0rc2-4.1.2
Although mplayer is not a part of the Slack distros, I think the issue is more in connection with the system setup and thefore I place this post in this forum. What is it with mplay- er? I've installed and reinstalled slackware 12.0 many times. And each time I have built mplayer from the sources and it has run fine. In the one before the last slack 12 installation, with subsequent building and installation of MPlayer, some- thing went wrong. And the same can be said of the present installation. Every source mplayer reproduces, it does it fine, with the exception of CD-ROM. No matter the CD-ROM I insert (I have them of the highest quality, bought out of the shelf) or the CD-ROM unit I use. The result is always the same: the sound is faulty, as if the machine had not enough speed. And I repeat. MPlayer played CD-ROMs fine in the past and under the same conditions. Or almost the same conditions, because something must obviously have changed.
Some additional data:
(a) Window$MediaPlayer plays them fine on the same machine.
(b) I invoke it from the physical console.
(c) KsCD under xfce plays them fine.
(d) I am able to load an o.s. from both CD-ROM units on the machine. All the more so, should audio be "correcty" read off the CD.
(e) Now I remember the last time I compiled mplayer with GUI support.
