Free 44100 khz software for linux

Simple Multitrack contains a set of programs that allows the simultaneous recording of an audio track while listening to a monitor mix of other tracks.

This software uses the OSS audio drivers found in the Linux kernel or the
CoreAudio API on Mac OS X. It has been tested on

Linux 2.4.20 on a Pentium III 1.4GHz with a SBLive soundcard, and
Linux 2.4.20 on a Pentium I 120MHz laptop with a Crystal soundcard.
Mac OS 10.3.3 on a PowerBook G4.

Both Linux machines are running Slackware, so please tell me about your successes or problems on other distros.

Please read the BUGS section of this document. If you have bug reports, bug fixes, questions, comments, criticism, improvements, or documentation corrections please e-mail me.

OVERVIEW:

./build
cd bin;
source shellsetup;

This will compile the programs and put them in your path.
It will also make your shell prompt shorter, so you have room.
It will also define a little function that is explained below.

to record initial track:
srp < /dev/zero > my_file

to listen to a track:
mix 2 1 1 1 my_file | srp > /dev/null

to listen to one track while recording a new track:
mix 2 1 1 1 file_1 | srp > file_2

to listen to two tracks while recording a new track:
mix 2 1 .5 .5 file_1 .5 .5 file_2 | srp > file_3

Simple Multitrack is basically two programs. They were designed to be used together, but they might be useful on their own. I will describe each program separately before discussing them together.

mix is a command line program. It mixes one or more monophonic audio files into a single output stream. The output stream can have any number of channels: mono, stereo, quad, whatever you like. The output stream is written to standard out.

The input files are specified as command line arguments, as are the gain settings for each input. Invocation goes like this (in stereo mode):

mix nocs mgain l_gain_1 r_gain_1 file_1 l_gain_2 r_gain_2 file_2 ...

where nocs is the number of output channels, and mgain is the master gain.

Command line arguments to mix after the nocs and mgain arguments are the channels. For N output channels, you will have N gain arguments and then the name of the file. Therefore, the arguments after nocs and mgain must appear in (nocs+1)-tuples.

mix will continue to pump out an endless stream of silence after the end of the input files is reached. This behavior is different from most UNIX command line programs, which exit at the end of their input data, which closes their stdout.

mix can be exited with the keyboard interrupt, ctrl-c. The input files must be 16-bit 44100 samples/sec monophonic raw signed word files. The output stream is in 16-bit 44100 samples/sec stereo raw signed word format. (Thank God, its in word format!) The program sox can be used to convert from most audio formats to and from most other audio formats.

EXAMPLES:

mix 2 1 0 1 my_file
This pans my_file completely to the right.

mix 2 1 1 1 my_file
This centers my_file.

mix 2 .5 1 0 flute 0 1 viola
This puts the flute completely to the left and viola completely to the right.
The master_gain is set to .5 to reduce the overall level of the mix by 3dB.

mix 2 .5 1 .3 flute .2 .9 viola
This is similar but gentler.

mix 2 .5 2 .6 flute .2 .9 viola
This is the same but the flute is louder.

mix 2 1 2 .6 flute .2 .9 viola .3 .3 violin
The violin is added in the center.

Note: If you get clipping errors when using mix, you can lower the mgain factor instead of adjusting all of the individual channel gains, although that works too.

srp is a command line program. It enables simultaneous recording and playback of audio using a sound card. It will only work if your sound card and its driver support full-duplex operation correctly.

srp reads a stream of stereo 16-bit 44100 samples/sec raw signed word data on standard in and plays that stream out on the soundcard. Meanwhile, it reads from the left channel of the soundcard and writes a mono 16-bit 44100 samp/sec stream on standard out.

Virtual Drum Machine is a simple drum machine.
It works for little endian/linux kind of machines. You may let it work on others machines, but you probably will get troubles with it.
You definitely need oss (or maybe alsa) for sound output, and a posix-like operating system. To let it work on a big endian machine should be painful.
You write a rhythm, then you compile it, then you are able to play it to your sound card or save it to a file.
The Virtual Drum Machine is made of
- the Rhythm Compiler,
- the runtime library.
The Virtual Drum Machine is in the public domain. Who needs a license? money makers? Protection against robbery? let me laugh... Read any text of law, you will see where the robbers reside.
A simple file would look like :
void main_rhythm(void)
{
tempo = 120;
- a
. b
. b
.
- a
.b
- a
.
. b
. b
- a c
. b
}
Install:
Do a "./configure" in the drums directory, then "make", then "make install", it should be alright. You can listen to some examples in the examples/ directory.
Who yo use it?
Write a rhythm. Compile it with "rc". Run the produced program. You are done.
See the examples/ directory to get the point.
When you run an example, try "-h" to get the available options.
It should be self-explanatory.
The rhythm compiler has several options. By running "rc --help", all should be clear.
Technical Details:
The compiler will parse the input file line by line.
If a line starts with "*" or "." (not counting leading white spaces), the whole line is seen as a rhythm line, and is transformed into C code. If not, it is passed as is to the C file.
Beware! You MUST NOT start any C code line by "*" or "."!
You can create as much functions as you want, write any C code you want. But remember that a line starting by "*" or "." is seen as a rhythm line and is translated by "rc" into C code.
You must provide a "void main_rhythm(void)" function, that will be called by the library. It is the starting point of your rhythm. It can be "void main_rhythm(int argc, char *argv[])" too, with common meaning for those parameters (non-C coders will have trouble with the Virtual Drum Machine).
You can change the tempo (ex. "tempo=100;") or the volume (ex "vol=0.4;") at any time. Each sample comes with its own volume and panning (ex. "a.vol = 0.1;" "a.pan=-0.8;"). Volumes range from 0 to what you want. 1 is for the normal volume. Panning ranges from -1 (left) to 1 (right). 0 is center. All values are double. You can use "volume" instead of "vol", and "panning" instead of "pan". There is no global panning, if you want all left, set all samples to left.
To run in stereo mode, dont forget "-s" when running the generated program. It is mono by default.
You absolutely need to compile and run the examples, and read them to get the point out of it!
The "rc.conf" file contains configuration informations. You specify the sample by "sample" followed by its name (the one you will use in your rhythm files), then the file that will be played. The name of the sample must start by a letter, followed by letters and/or numbers (it must be a valid C identifier, without "_" though). The configuration file contains the install directory, used by "rc" to compile your rhythms. Take a look at the one that is provided to see how to use it.
The sound files are simple wav files. They all should be of the same rate, which can be specified to the generated program, using the "-f" option (44100 is the default). (The library only handles very basic wav files, if yours dont work, you probably will have to modify the library for the program to handle it.)
When you add a sample, you must modify "rc.conf" for the changes to appear. The samples are hard-linked to the produced program, so if you change "sample a /some/dir/file1.wav" by "sample a /one/other/dir/file2.wav" in the configuration file, the previously generated programs will still use "/some/dir/file1.wav". You will have to compile them again to take the changes into account.
Enhancements:
- The code has been modified to let gcc 4 compile it.

The FAAC project includes the AAC encoder FAAC and decoder FAAD2.
FAAC supports several MPEG-4 object types (LC, LTP, HE AAC, Main, PS) and file formats (raw AAC, MP4, ADTS AAC), multichannel and gapless en/decoding as well as MP4 metadata tags.
The codecs are compatible with standard-compliant audio applications using one or more of these profiles.
General FAAC compiling instructions:
1. Make sure you have autoconf, automake and libtool installed. For MP4 support, you must have libmp4v2 (included in faad2) installed.
2. cd to FAAC source dir
3. Run:
./bootstrap
./configure
make
make install
Usage:
faac [options]
Options:
-a X Set average bitrate to approximately X kbps per channel (i.e. using -a 64 averages at 128 kbps/stereo).
-c < bandwidth > Set the bandwidth in Hz (default value depends on sample rate)
-q < quality > Set quantizer quality (default: 100, averages at approx. 128 kbps VBR for a normal stereo input file at 16 bit and 44.1 kHz sample rate).
--tns Enable TNS coding.
--notns Disable TNS coding.
-n Disable mid/side coding.
-m X AAC MPEG version, X can be 2 or 4 (default: MPEG-2, so for the sake of interoperability with non-standard compliant players like QuickTime 6 you should set it to "4").
-o X AAC object type, X can be LC, MAIN or LTP (default: LC, for the same reason as with the MPEG version dont use Main or LTP).
-r RAW AAC output file (i.e. without ADTS headers).
-P Raw PCM input mode.
-R Raw PCM input sample rate in Hz (default: 44100 Hz).
-B Raw PCM input bit depth (default: 16 bits, also possible 8 bits).
-C Raw PCM input channels (default: 2).
- < stdin > If you simply use a hyphen/minus sign instead of an input file name, FAAC can encode directly from stdin, thus enabling piping within other applications like foobar2000 or mp4live.
Note: VBR output bitrate depends on -q AND -c, so you should only vary the default setting -q 100 -c 16000 if you know what youre doing and/or want to experiment with other cutoff frequencies at a given quality setting.
The ABR setting with -a is an approximate average bitrate that does not use a bit reservoir, i.e -a 64 and -q 100 at 44.1 kHz will result in exactly the same output file.
The following list should give some orientation for useful -q and -c settings, based on FAAC v1.17. The resulting VBR bitrates are referring to an average sounding stereo file with 16bit, 44.1 kHz, i.e. ct_reference.wav in this case. Multiplexing these AAC files to MP4 with e.g. mp4creator will result in a ~3 kbps lower bitrate because of the stripped ADTS headers:
-q 130 -c 22000 -m 4 (~218 kbps)
-q 120 -c 20000 -m 4 (~194 kbps)
-q 110 -c 18000 -m 4 (~158 kbps)
-q 100 -c 16000 -m 4 (~129 kbps)
-q 90 -c 14000 -m 4 (~103 kbps)
-q 80 -c 12000 -m 4 (~79 kbps)
-q 70 -c 10000 -m 4 (~62 kbps)
The added -m 4 switch does not change the bitrate or sound of course, but is recommended for most AAC/MP4 players that use an updated FAAD2-based plugin from this year (Winamp 2.x, foobar2000 etc.) or cant decode MPEG-2 AAC LC files like QuickTime 6. Philips Expanium users should not use this switch, because their CD portable does not know MPEG-4 AAC files.
Enhancements:
- Small bug fixes since last version

1. Ultra-low latency (typically from 3 to 9 ms)
2. Full audio bandwidth (44.1 kHz and 48 kHz)
3. Stereo support
4. Packet loss concealment
5. Constant bit-rates from 32 kbps to 128 kbps and above
6. A fixed-point version of the encoder and decoder
7. The CELT codec is meant to close the gap between Vorbis and Speex for applications where both high quality audio and low delay are desired.

Enhancements:

• Has just been released, with many quality improvements, including better stereo coupling, better handling of transients, and better handling of highly tonal signals.
• Packet loss robustness has been improved through the optional use of independent (intra) frames.
• Supports a larger dynamic range, suitable for encoding 24-bit audio (float version only).
• There is also a very early VBR implementation.
  

dsprec is a program that reads samples from the /dev/dsp device and outputs them to standard output.It allows you to specify the sample rate, word size, and number of channels.

dsprec was written for FreeBSD. It also works with Linux. It might work with other UNIX systems too. If you make it work on another OS, please tell me so I can mention it here.

This is a typical usage:

dsprec rate 44100 bits 16 channels 2 > outfile.raw

Sample rate, word size, and number of channels are untouched if they are not specified. You should always specify them.

Installation

cd /usr/local/src
fetch http://tomclegg.net/software/dsprec-0.2.tar.gz
tar xzf dsprec-0.2.tar.gz
cd dsprec-0.2
make
make install

You need sound support (device pcm) in your kernel in order to run dsprec.

ivcall is a small utility which may be used to make automated telephone calls with your isdn4linux supported ISDN card. Outgoing calls are supported as well as incoming calls.
The audio data recieved from the peer is written to STDOUT, audio data read from STDIN is send to the peer. The audio data is in raw 8 bit uLaw 8 KHz format, without any headers.
Installation:
./configure
make
make install
Enhancements:
- cleanups
- add softfax support using spandsp.

fdmf is portable perl/C software for finding pairs of music files in a collection that are likely to contain the same music. It works on the music itself, not on the filename, tags, or headers. The project uses an audio fingerprint, or perceptual hash to recognize the duplicate files. It is currently under heavy development, so it might be buggy, broken, or otherwise bad. But it works for me. Please email me and tell me if it works for you. Bug reports are appreciated.

It would help the development of this software if I had a larger set of files to test it on. If you feel like lending music files to this effort, it would be greatly appreciated. The optimal form would be a CD or DVD of MP3 or OGG files which could be sent via postal mail. The media will be mailed back to you after it has been used, along with some extra disks of music for your testing purposes. Please email me if you are interested in lending files.

USAGE:

fdmf < music_dir >
vector_pairs

USAGE EXAMPLE:

./fdmf /home/bob/music/
./vector_pairs

vector_pairs will print on stdout a list of pairs of files that may contain the same music.

NOTE: The first time you run this program it will do a lot of processing. It takes 10 minutes to process a music_dir containing 140 mp3 files on my Apple G4. It will attempt to cache the result so that subsequent runs are faster. You can interrupt it and your progress will not be lost.

ANOTHER NOTE: On my current development box, an IBM T20, mplayer is the fastest mp3 decoder. mpg123 is also quite fast, much faster than mpg321. Some Linux distros come with mpg123 being a symbolic link to mpg321 since mpg123 is not under GPL. Bottom line: you can use any program for decoding if it can take the name of a music file as a command line arg and write the raw decoded 44100/stereo/16-bit stream on stdout. You might want to take some measurements on your system to see which decoder is fastest.

mp3info software shows details of an mp3-file on the console.

Screendump

folkert@belle:~/Personal/src/mp3info$ mp3info -v -f /data4/mp3/7zuma7 - deep inside - 10 - heroin chic.mp3
mpeg3library version 1.5.4
Number of audio streams: 1
Stream 0:
2 channels and consists of 11455240 samples.
sample rate: 44100, duration: 00:04:19
audio format: MPEG
File indicates that it has no video streams.
id3 v1 tag info
title: Heroin Chic
artist: 7Zuma7
album: Deep Inside...
year: 2000
comment:
track: 10
genre: Metal

Integration in Mutt

Add the following lines to ~/.mailcap:
audio/mp3; /usr/local/bin/mp3info -v -f %s ; copiousoutput
audio/mpeg; /usr/local/bin/mp3info -v -f %s ; copiousoutput
audio/x-mp3; /usr/local/bin/mp3info -v -f %s ; copiousoutput

and add the following lines to ~/.muttrc:

auto_view audio/mp3
auto_view audio/mpeg
auto_view audio/x-mp3