Free 44100 software for linux

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.

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

Tapiir is a simple and flexible audio effects processor, inspired on the classical magnetic tape delay systems used since the early days of electro-acoustic music composition.

Tapiir project provides a graphical user interface consisting of six delay lines, or "taps", which can introduce an almost arbitrarily big or small delay to their inputs and can be feed back to each other.

A wide set of effects can be easily achieved by properly configuring and connecting the delay lines: complex echo patterns, resonances, filtering, etc. Delays, interconnections and gains can all be controlled in real time.

TAPIIR internal processing modules consist of six delay-lines, each with a mixer at its input and a gain control at its output, and a stereo output mixer. Stereo input from an external source, typically a musical instrument, is routed to all input mixers. In addition to this, the output of each delay line is also routed to the input mixers of all delay lines, including itself. Figure 1 shows the diagram of TAPIIRs internals.

This cross-feeding of audio signals throughout the system of delay-lines and mixers, allows the user to create a very large variety of stereo delay effects. Very simple echos or ping-pong effects can be achieved easily, but more complex effects such as early reflection echos, reverbs, complex rhythmic and arrhythmic patterns and even Karplus-Strong like synthesis is also possible. It is important to observe, that these more complex effects are only possible by using sample accurate processing.

This article appears in Proceedings of the COST G-6 Conference on Digital Audio Effects (DAFX-01), Limerick, Ireland, December 6-8, 2001

Sample accuracy

Conventional hardware effect processors are often rather limited in the lenght of there delay-lines. It is unusual to encounter accuracy higher than 1 msec, and even 10 msec is used frequently, and maximum delay-lenght are limited as well.

Obviously, this limitation in hardware effect processors is deliberate, both out of technical concerns or marketing. Most users are not interested in higher accuracy, and the standard user interface of hardware effects processors - buttons or at the most an alpha-dial - would make it a painful job to adjust. Also, one can imagine that lower accuracy means less computational cost, and therefore lower overall cost of the effect processing hardware.

For advanced users however, this limitation can be annoying. Of course, many of the effects obtained with very short delay times, such as reverb or filtering, are usually also implemented in the same hardware, but it can be very interesting to combine all these with longer delay-time effects; it would be necessary to use several processors connected together to do this.

The implementation of TAPIIR, however, is sample accurate. This means that extremely short delay times can be used, 0.023 msec when using a sample-rate of 44100 Hz. In addition to this fine control over delay-lengths, the sample accuracy is also implented for feedback and even cross-feeding between the various delay-lines, This is achieved by the fact that the internal processing core of TAPIIR is written in such a way, that the input and output values of the delay-lines and mixers are passed on 1 at a time, instead of buffer-by-buffer.

Filtering with delays

Obviously, the effects obtained by sample accurate processing of delay-lines go far beyond the simple echo effects. This includes the creation of FIR filters and - using feedback - IIR filters (this has been the inspiration for the name TAPIIR). In these cases, the mixer gains function as filter coefficients. This means that TAPIIR can efficiently be used for filtering, with flexible filter design. In a future version, TAPIIR could contain a pole/zero editor that automatically sets the mixer values to create the corresponding filter.

The maximum delay-length that can be achieved is only limited by the physical RAM memory of the system TAPIIR runs on. To give an example, with 32 MB of free memory, a total delay-length of more than 6 minutes can be used. While this might seem rather useless for normal effect processing, it clearly has musical applications. Several compositions have been written that make use of long delay times. Originally performed with the use of tape-delays, they could take great profit of the use of digital techniques for sound quality. The use of hard disk space with sufficient fast access would take away time limitation even more.

Delay-length control

The graphical user interface of TAPIIR allows the user to take full advantage of the delay-length accuracy, but at the same time it tries to maintain user-friendly and manageable, by offering value-sliders for larger scales as well. Delay-time can be entered in time in seconds in number of samples. Sliders control the digits of the delay-length, with an accuracy of 5 decimals. An additional feature is the use of tempo/signature. In that case, delay-length in not represented in seconds, but in beats, and the sliders control the subdivision of beats according to the signature. Obviously, in many circumstances this representation is a lot more useful, in a musical sense, than time in seconds.

libreplay library will save the output of realplay in a raw format, so you can do what you want with this uncompressed data later in time.

When you use realplay, sometimes you would like to save the sound that you ear, to be able to listen to it later in time. But realplay doesnt have a save or export button, so libreplay dose that for you.

Installation

Just do a make in the directory, should be alright. There will be the file libreplay.so which contains everything usefull.

Usage

This documentation is only valid for version 0.3 of libreplay. Avoid using 0.1, it crashes too much realplay. (Version 0.2 does not have LIBREPLAY_AUDIO support.)

You must set the variable LIBC_PATH to the place where your libc is (something like /lib/libc.so.6). If you are under bash, just do export LIBC_PATH=/lib/libc.so.6
(replace /lib/libc.so.6 by your libc of course).

Do
LD_PRELOAD=./libreplay.so realplay in the directory where libreplay is.

realplay will create /tmp/realplayer_stream_XX_speed_YY_channels_ZZ_format_AA.raw for each stream you play.

XX will be replaced by the stream number (from 0 to whatever the number of streams you download).

YY will be the speed of the stream (44100 if the stream is at 44100Hz, for instance).

ZZ will be the number of channels of the stream (1 for mono, 2 for stereo).

AA will be the size of each sample of the stream (I only saw 16 for it, meaning: 16 bits, signed, little endian).

Some streams may be empty, because realplay sometimes opens and closes the soundcard whitout outputting any sound.

The output is a raw file, use sox or you favorite conversion program to transform it into a format you like more.

For instance, using sox, you would do

sox -t raw -s -w -c 2 -r 44100 realplayer_stream_6_speed_44100_channels_1_format_16.raw out.wav copy

-t raw for saying its a raw file,
-s -w for saying its 16b signed,
-c 2 for saying there are 2 channels,
-r 44100 for saying its a 44100Hz stream.

It has been tested against realplayer 8.

By setting the environment variable LIBREPLAY_AUDIO, you can save live streams. Some streams create problems when they are saved faster than they are played; the server will send again some audio data, and the result is bad. With this option, the stream is played and saved at the same time. If you have problems with your saved streams, try this option.

You may use the scripts realplay.dump and realplay.dump_live, which include all the necessary to let realplay be used with libreplay.
realplay.dump will only save the stream.
realplay.dump_live will save it and play it.
You may need to edit these scripts, to adapt the paths, for them to fit your system. These scripts are only here to help you; if they dont work, you still will need to read the documentation (this file) and understand it.

Gregory did a little perl utility to, maybe, facilitate the usage of this little tool. realexport.tar.gz

realexport.sh something.rm is the way to use it, according to Gregory. (It may work, it may fail, not tested.) You will need sox and lame available on your system.