Yes. Check out the Song Maker experiment, which lets you make and share your own songs.
Do I need to make an account?
Nope. Just open any experiment and start playing.
How were these built?
All our experiments are all built with freely accessible web technology such as Web Audio API, WebMIDI, Tone.js, and more. These tools make it easier for coders to build new interactive music experiences. You can get the open-source code to lots of these experiments here on Github.
What devices do these work on?
You can play with these experiments across devices – phones, tablets, laptops – just by opening the site on a web browser such as Chrome.
TAK stands for (T)om’s lossless (A)audio (k)ompressor. Besides, it’s a throwback to a (not very philanthropic) character from Stephen King’s “Regulators”. Early semi-public evaluation versions operated under the working title YALAC .
High compression . The strongest mode is on a par with Monkey’s Audio High and OptimFrog’s Normal; for specific files such as classical music or voice recordings, it often outperforms both. This classification is based on the evaluation of hundreds of files of various styles; it definitely does not apply to every single file.
High compression speed . I am currently not aware of any other compressor that works faster than TAK’s Turbo or Fast mode and achieves similar compression rates.
Multi-core compressor . The compressor optionally generates up to four threads in order to take advantage of multi-core cpus.
Very high decompression speed . It is at the level of FLAC and therefore significantly higher than most symmetrical compressors.
Support for every popular audio format (not yet fully implemented).
Streaming support . An info frame, which contains all the information required for decoding, is inserted into the compressed audio data every 2 seconds.
Fault tolerance . A single bit error never damages the audio data for more than a maximum of 250 ms, since it is stored in completely independent frames of a maximum of this duration. The decoder processes even extremely damaged files, optionally replacing or removing the affected data with silence.
Error detection . Every single frame is protected by a 24-bit checksum (CRC).
MD5 checksums for quick identification of audio material (e.g. for searching for duplicates).
Fast, sample-accurate access to any playback position . The file header contains a look-up table with index positions every second. Even without this table, efficient random access is possible; for this purpose, the synchronization codes of the frame headers and / or the offset values optionally recorded in the frame header, which refer to the beginning of the previous and next frame, can be used.
Metadata . A flexible and expandable structure allows the recording of non-audio data such as images or cuesheets.
Playback plugins for Winamp and Foobar are currently available.
An SDK provides other developers with decoding functions for integration into their applications. An extension to include coding functions is planned.
HyBrit Head is an hybrid amp model based on two famous British gears.
The PLS channel is actually an hybrid of the two (tweaked) channels (Normal and Treble) of the real thing. Both channels are actually processed and the mix of the two signals can be adjusted through the PLS MIX knob. The other channel, MCJ, is also a tweaked version of the real thing. So, this amp simulator doesn’t represent anything specifically but it does have the character of the British amp.
mypiano_jukebox is a mypiano_chung bass.dll based MIDI jukebox, MIDI files & folder player with a virtual acoustic piano recorded on Isabelle’s upright piano, with a smartphone . The sounds, reverb, chorus and volume are variable with the number of played notes and the sustain switch, just like a real piano.
Added a public domain Kawai, City piano and Steinway samples.
What sets this frontend for ffmpeg apart from other stereo simulators is it creates an illusion of actual stereo separation. More importantly, it produces none of the weird phasing, and/or time delay artifacts. And very little, if any of the tone discoloration, when those others aren’t meticulously set up just right. All this is accomplished by using the ffmpeg crossover audio filter to split the sound into 8 frequency bands. The split points are based on center frequencies of a typical 1/3 octave equalizer. Those 8 bands are then panned in varying degrees to left and right. The varying width of each band is set to achieve the best balance between the left and right channels.
WaveGain is an application of the ReplayGain algorithms to standard PCM wave files. Calculated gain adjustments are applied directly to the audio data, instead of just writing metadata as traditionally done for other formats like MP3, FLAC and Ogg Vorbis. The replaygain values can also be added as metadata in a custom RIFF chunk named ‘gain’. This could theoretically allow WAV files to have same lossless functionality as other formats where audio data is not altered. But since no current players are aware of this “standard”, the metadata is used only by WaveGain for the “–undo-gain” feature, which is lossy.
qaac requires Apple Application Support that is included in iTunes or QuickTime. If you are using 64 bit Windows, recent 64 bit iTunes (ver 12.1 or later) is recommended. 64 bit installer for iTunes (iTunes6464Setup.exe) includes both of 32 bit and 64 bit Apple Application Support, therefore you can run both of qaac.exe and qaac64.exe with its installation.
AAC-LC, AAC-HE, ALAC encoding are supported. m4a container (just a MP4 container with ‘M4A ‘ brand, created by Apple) is used by default, but you can also mux into ADTS.
Filenames and tags are treated with Unicode. Very long file names are supported.
Support for multichannel / surround.
Fine quality control over AAC encoding.
Support for hi-resolution / multichannel ALAC.
WAV, raw PCM, ALAC, and all LPCM formats supported by CoreAudio AudioFile interface (such as AIFF, CAF, and Sun/AU) are directly available as input. MP3 is also decodable through CoreAudio.
Cue sheet input is also available.
FLAC, Wavpack, TAK, and other LPCM formats are optionally supported when libFLAC.dll, wavpackdll.dll, tak_deco_lib.dll, and libsndfile-1.dll are installed.
Piped input is available for WAV and raw PCM.
Piped output (streaming) is supported for ADTS.
Automatically fetch tags from AIFF, FLAC, Wavpack, Tak, and ALAC. Major tags are copied to the result. Also, you can manually set tags via command line options.