Free lamer software for linux

This is a unified diff (you probably need GNU patch to apply it) to lame 3.58. Please note that this diff is now obsolete as recent beta versions of lame include a version of this code. Just type "make mp3rtp". The code was broken, but the CVS code is reported to work (and interoperate with playRTPMPEG) as of Feb 21 2000.

Please note that the output stream will only have the correct speed if the input is live recorded stream from your sound card or lame encodes exactly in real-time on your CPU!

RTP is the Realtime Transport Protocol as defined in RFC 1889. It is the transport mechanism of your choice to multicast an mp3 stream.

tooLAME is an optimized Mpeg Audio 1/2 Layer 2 encoder.

It is based heavily on:

- the ISO dist10 code
- improvement to algorithms as part of the LAME project

Installation:

1. edit Makefile
at least change the architecture type (ARCH) to suit your machine.
2. make

Usage:

./toolame [options] < input > < output >

Input File
tooLAME parses AIFF and WAV files for file info
raw PCM is assumed if no header is found
for stdin use a -

Output File
file is automatically renamed from *.* to *.mp2
for stdout use a -

Input Options
-s [int]
if inputting raw PCM sound, you must specify the sample rate
default sample rate is 44.1khz.

-a
downmix from stereo to mono
if the incoming file is stereo, combine the audio into
a single channel

-x
force byte-swapping of the input. (current endian detection is dodgy,
so if toolame produces only noise, use -x )

-g
swap the LR channels of a stereo file

Output Options
-m [char]
the encoding mode (default j)
s stereo
d dual channel
j joint stereo
m mono

-p [int]
which psy model to use (default 1)
Different models for the psychoacoustics
Models: -1 to 4

-b [int]
the total bitrate
For 48/44.1/32kHz default = 192
For 24/22.05/16kHz default = 96

-v [int]
Switch on VBR mode.
The higher the number the better the quality.
Useful range -10 to 10.
See README.VBR for details.

Operation
-f
fast mode turns off calculation of the psychoacoustic model.
Instead a set of default values are assumed

-q [int]
quick mode calculates the psy model every num frames.

Misc
-d emp
de-emphasis (default n)
-c
mark as copyright
-o
mark as original
-e
add error protection
-r
force padding bits off
-D
add DAB extensions
-t [int]
talkativity setting. 0 = no message. 3 = too much information

FAME project is a set of tools for encoding multimedia content.

It currently consists in the following tools:

libfame

libfame is a video encoding library.

It can currently encode MPEG-1 and MPEG-4 rectangular video, as well as MPEG-4 video with arbitrary shape.

Compliance

Provide bitstreams compliant to the MPEG-1, MPEG-2 and MPEG-4 video standards.

Speed

Provide a fast implementation of the techniques used in MPEG standards.

Flexibility

Allow the user to choose between different options for speed, compression ratio and quality.

Portability Support many different platforms and architectures.

fame

fame is a multimedia encoder, which captures video from a video4linux device, and optionnaly sound, for MPEG encoding.

It is based on libfame and lame and thus supports the same output formats as these two libraries.

TCMixer is a compact X11 audio mixer.

Can be controlled from keyboard. Supports mixer callbacks. Very fast Xlib based UI routines, optimized mixer code.

Version 2.0 adds complete keyboard control over the mixer. Read the ASCII diagram below:

.-----..---.
| TAB || Q | .---.
`-----`--- | ^ |
`---
.---..---..---..---..---..---..---. .---..---..---.
| Z || X || C || || || || M | | < || v || > |
`---`---`---`---`---`---`--- `---`---`---

Tab, Right arrow: next channel
Mod-Tab, Left arrow: previous channel
Up Arrow volume up
Down Arrow volume down
Q quit
M mute
Z,X,C Line, Mic, CD record source

The keys are grabbed as scancodes, so I think this should work on all international/whatnot keyboards. zxc are in the lower left corner and the m key is 3 keys after, so its like "zxc...m" on the bottom row. "Q" should be right next to the Tab key. Scancodes returned by the keys located there should be the same regardless of keyboard layout. If not, let me know.

Current channel is marked with a green LED on the slider bar. Slider slide, changing volume. There is a row of buttons on the top. If a button is yellow, it means the channel is on and not muted. If its off, the channel is muted. Push it again to go back to the last known volume setting (remembered before pushing the button). If a channel was muted and you moved the slider, without pushing the button, it will become unmuted... If the button is red, you will not see a mixer slider for that channel, and it means your soundcard doesnt support that particular channel (most likely going to be true for Bass / Treble channels on elcheapo cards and also on some PCI cards (AudioPCI for example)).

Most modern sound card controls are here, and this should be enough for basic volume adjustment when you play your cd or mp3s or whatever. Missing controls will be deleted from the face of the mixer, but it will not be resized, so if your card is so lame that half the controls are "red" on the mixer, then you should find a new card.

Choose recording source by using radio buttons under the sliders by clicking on them or pressing z, x or c keys to select Line, Mic, or CD as recording source. Again, if that doesnt work, complain to your sound card manufacturer.

Oh yah, you quit by clicking anywhere in the words "TCMixer", or pressing the "q" button on keyboard.

Since that lameness called DEVFS has been included in the kernel, people might have shit like /dev/bus/pci/slot0/card0/controller0/sound/mixer for their mixer device, so you can use the -m option with tcmixer to specify your own mixer device:

% ./mixer -m /dev/dev/fs/is/lame/sound/mixer

It works for me, with ALSA. It should work with OSS, but I never bothered to test it. Some OSS drivers are known to only have volume steps from say 0 to 64 instead of 0 to 100 like ALSA. Up to this day I dont know which card does this. I have received some reports that this could be your standard SB16 if used with kernel sound drivers under kernel 2.2. If your card does this, you can email me, and tell me the card name, oss version, shit from /dev/sndstat, etc etc - if you want that fixed you will need to tell me as much information as you can possibly find about your sound shit, because the only people who reported this to me so far didnt even bother to tell me what soundcard they have. This 0 to 64 stepping will break the mixer, and since fuckheads who wrote OSS did not provide an API to check the min/max ranges for a particular channel I cannot test for this situation at run-time. Sorry. Use ALSA, it will fake your lame 0 to 64 card into a smooth 0 to 100 curve.

AudConvert is an application that is designed to take any audio format and convert it to any other audio format.
The idea for AudConvert came from my need to turn my Ogg Vorbis collection into MP3s for portable devices.
Yes, this process sometimes will result in lower quality, but sometimes it must be done.
Main features:
- Input any directory of files, get out the same directory structure (or flat directory) of newly encoded files.
- Multi-threaded: Encode up to 8 files simultaneously.
This is the first release of this software and it needs a lot of testing.
Supported Inputs:
- Ogg Vorbis (oggdec)
- MP3 (mpg123)
- FLAC (flac)
Supported Outputs:
- Ogg Vorbis (oggenc)
- MP3 (lame)

BladeEnc is a freeware MP3 encoder. It is based on the same ISO compression routines as mpegEnc, so you can expect roughly the same, or better, quality . The main difference is the appearance and speed.
BladeEnc doesnt have a nice, user-friendly interface like mpegEnc, but it is more than three times faster, and it works with several popular front-end graphical user interfaces.
BladeEncs output quality is one of those rare subjects that completely divides the world in two parts. Either you love it or you hate it, there never seems to be an opinion inbetween. Different audiophiles and mp3 experts tends to come to completely different conclusions depending on their methods and testsamples.
The reason for this is of course that BladeEnc is a very different mp3 encoder (compared to Fraunhofer, LAME etc) with a very unique approach to mp3 encoding.
In order to compress sound to an mp3 file, you need to make certain sacrifices in quality. Taking into account how we percieve sound, the mp3 encoder tries to remove the details that it believes us to be least likely to notice. How much that needs to be removed depends on the bitrate and the encoder often has the choice of doing different kinds of sacrifices.
It can remove low volume tones that are "shadowed" by high volume tones of similar frequencies, remove the high frequency part of the sound spectrum, cut down the stereo effect (so called joint stereo) and simply decrease the samplerate. What approach is the best depends on a lot of things, like the style of music and the selected bitrate.
Main features:
- Sourcecode available under the LGPL-license!
- Stereo or Mono output. Can downmix to Mono on the fly.
- Supports the following bitrates: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kBit/s. However, for bitrates lower than 128 kBit we seriously recommend you to use another encoder.
- Flags like Private, Original and Copyright can be set.
- Input samples can be in either 32, 44.1 or 48 kHz.
- Both 8 and 16-bit samples are supported.
- Working CRC checksum generation (since 0.80). The ISO reference code had broken CRC calculations, which has been inherited into every ISO based encoder that havent added a fix for it.
- Can be plugged directly into many popular 3rd party products, giving them integrated mp3 encoding abilities.
- Encodes chunks of data from memory to memory, no need to use files or pipes.
- Can be compiled for nearly any operating system still in use.
- Commandline based, makes it easy to include BladeEnc into BAT files and shell scripts.
- Only mp3 encoder that supports gapless encoding.
- Reads standard uncompressed WAV- and AIFF-files as well as well as RAW PCM-data.
- Batch encoding. Can encode any number of samples in a row.
- Wildcards supported. You can for example encode all WAV-files in a directory by typing *.WAV".
- Input samples can be automatically deleted after encoding.
- Large selection of graphical frontends available.
- Task priority can be set from the commandline and is by default set to LOWEST so that you still can use your computer effectively while encoding (Windows & OS/2 only).
- Full support for pipes and redirection (stdin and stdout).
- Textbased configuration file where you can change default settings.

h4ckeramba is a system monitoring SuperKaramba theme.

I made this superkaramba theme to make my desktop look a bit more hacker style- yeah, it might be a little lame but I like its simplicity. What do you think?

BTW you can choose between a bigger and a smaller clock style, I recommend the bigger one, its the default. Dont forget to set your hda and hdb filesystem to the right mounted partition.