NAME¶

SYNOPSIS¶

butt [option] <path_to_file>

DESCRIPTION¶

butt (broadcast using this tool) is an easy to use, multi OS streaming tool. It supports ShoutCast and IceCast and runs on Linux, MacOS and Windows. The main purpose of butt is to stream live audio data from your computers Mic or Line input to an Shoutcast or Icecast server. Recording is also possible. It

is not intended to be a server by itself or automatically stream a set of audio files.

It Works with SHOUTcast and Icecast.

It runs on all three major operating systems. Mac OS X, Linux and Windows.

It supports aac+, mp3, ogg/vorbis, ogg/opus and flac for streaming.

It supports aac+, mp3, ogg/vorbis, ogg/opus, flac and wav for recording.

It is able to connect to a server after starting up automatically.

It is able to start a recording after connecting to a server automatically.

Recording can be split after a user defined amount of time.

Current song can either be updated manually or automatically by reading a file.

Configuration files can be imported and exported.

Status display shows infos about the current state (click on it).

Automatically reconnects in case the connection was interrupted.

It has a VU Meter with peak hold.

It is able to attentuate and amplify the input volume.

It has a 5-band EQ.

It can read song names from different apps in MacOS and Linux.

Display colors can be changed as desired.

QUICK START¶

In order to connect to a server, you need to add a new server in the config window. Just open the settings window and click on [ADD]. Now fill in the input fields with the server data and click on the new [ADD].

Adding Stream Infos is not necessary for connecting to a server.

CONFIGURATION¶

        [Save]: Saves your current settings to the standard configuration file or to the file that was passed to the -c option.
        [Export]: Saves your current settings to the given file.
        [Import]: Loads the selected file and applies the settings.

MAIN WINDOW¶

        The [>] symbol shines yellow if butt is connected to a server.
        The [O] symbol shines orange if the [start rec. when connected] checkbox is activated.
        The [O] symbol shines red if butt is currently recording.

Gain slider: The slider is only visible when the little [more/less] button below the [settings] button was clicked. With this slider you can attenuate and amplify the input signal between -24dB and +24dB, respectively. Double clicking the slider resets the gain to 0dB. Use this slider only to fine tune your input signal. It does not change the operating systems input volume setting. Instead, the input signal is multiplied by the given factor. Thus adding to much gain will also add lots of noise.

STREAMING¶

You can stream with 3 different codecs: mp3, ogg/vorbis and ogg/opus. In case opus is selected the samplerate is always upsampled to 48kHz. Upsampling needs lot of CPU power. You can change the upsampling algorithm in the [Advanced] settings on the [Audio] tab. Upsampling is deactivated if you select 48kHz as sample rate.

Unfortunately, it is not possible to update Stream Infos while streaming. You need to reconnect for updating the Stream Infos.

However, at least you can update the current song on the fly. You only need to type the song into the Song Name input field at the [Stream] tab and hit Enter or click [OK].

butt can also update the song automatically from a text file. The first line of the text file must be the name of the song. As soon as butt detects that the file has been changed it updates the name of the song on the server. A plugin for the amarok audio player can be found in the player_plugins/ directory. Do not hesitate to write a plugin for other audio players. Just mail them to me and I’ll add them to the butt package.

Stream infos:

        In the [main] settings window you can add stream infos. This allows you to deliver more details of your stream station. For example the genre of your music, description of your station, web address etc.

RECORDING¶

In case opus is selected the samplerate is always upsampled to 48kHz. Upsampling needs lots of CPU power. You can change the upsampling algorithm in the [Advanced] settings on the [Audio] tab or disable it by selecting 48kHz as sample rate. Not all sound interfaces support a samplerate of 48kHz, though.

To record you first need to select the destination folder and specify a file name in the [Rec] tab. butt will replace the variables %d, %m and %y with the current day, month and year. e.g. rec_(%m_%d_%y).mp3 → rec_(03_28_2008).mp3. Other possible time variables are %H (hours) %M (minutes) %S (seconds).

With the %i variable you can add an index number to your file name. This means with rec_%i.mp3 butt first tries to open rec_0.mp3. If that file already exists, butt tries rec_1.mp3 and so on…

If the start recording when connected checkbox is activated butt starts the recording immediately after being connected to a server.

To manually start the recording press the record symbol. To stop recording simply click on the record symbol again.

You can also tell butt to split your recording into separat files every n minutes. Just enter a number higher than 0 into the Split file every [..] minutes field. Let’s assume your file name is rec_(%m_%d_%y)_%i.mp3 Then the first file is expanded to rec_(03_28_2008)_0-1.mp3, the second after n minutes to rec_(03_28_2008)_0-2.mp3, the third to rec_(03_28_2008)_0-3.mp3, you got it. If the sync to full hour checkbox is activated the automatic file splitting is synchronized to the full hour. That means if the time is 8:55 and file splitting is set to 15 minutes, the second file starts at 9:00 and the third at 9:15.

AUTHOR¶

This manual page was written by Paulo Henrique de Lima Santana (phls) <phls@softwarelivre.org> for the Debian project (but may be used by others).