Here, you'll find some items for Windows operating systems that I've found useful, that aren't readily available elsewhere (often old websites have disappeared). It's not intended to be comprehensive. I also intend to respect the license terms included with the distributions and I'm sure you'll do the same. I'm also mentioning some not-so-rare software that's very useful.
I've noticed that my soundcard on my work PC sounds rather hollow or gruff when playing back certain audio files, particularly those recorded at sampling rates below 32 kHz, and especially voice at 11 kHz or 8 kHz sampling rate.
I attribute this to the use of a very crude method of playback where the Digital-to-Analogue Convertor (DAC) simply slows down its clocking rate.
So, for 8 kHz sampling rate, the sample period is 0.125 ms. If your soundcard's DAC is optimised for CD and similar audio, it's likely to have a bandwidth of around 19.5 to 20.5 kHz (similar to most CD mastering). Assuming 20 kHz, the rise-time and fall-time will typically be about 0.0175 ms. If no filtering is applied during 8 kHz playback, the rise/fall-time will be about one-seventh of the new sample period. It's obvious this can cause quite a jagged staircase effect, with the jaggedness producing frequencies well above the 4 kHz maximum frequency that could be encoded at 8 kHz sample rate.
If you fail to post-filter (either by limiting the analogue bandwidth or digitally post-filtering) then the jagged staircase effect will cause aliasing - the creation of frequencies that couldn't have been represented in the original sound.
For example, at a 16 kHz sampling rate, no frequencies above half the sampling rate, 8 kHz can be properly represented. 8 kHz is the Nyquist limit. If you try to sample a 10 kHz sinusoid at 16 kHz sampling rate, you'd actually get aliasing to produce a 6 kHz sinusoid (musically, the note changes from a high D# to a F# almost one octave lower). Aliasing can thus lead to dissonance in music, and for this reason, one should low-pass filter the input appropriately before sampling at low rate (or sample at a high rate then downsample in software, applying an appropriate digital low-pass pre-filter in software). Likewise, when increasing the sampling rate, one should low-pass after upsampling to remove any effects where the interpolation (if any) hasn't been sufficiently bandwidth limited.
This aliasing principle is exactly the principle used in computer graphics when changing the pixel dimensions of an image (stretching or shrinking) that causes 'jaggies' or staircase effects on edges. There, spatial frequencies that couldn't have been in the original image, appear in the larger resized image, or frequencies that can't be represented in the smaller image become aliased to different frequencies in the process of resizing.
Obtaining a smooth, ideally-filtered playback from a digital-to-analogue convertor is one of the reasons why oversampling (e.g. resampling to 88.2 kHz) is used in many CD players. This allows precise digital filtering to be used without any risk of colouring the audible signal by using analogue filtering after a 44.1 kHz DAC.
Anyhow, resampling output plugins for WinAMP aren't that easy to come by. I was told of a couple that output via DirectSound (which my PC at work can't support), including Peter Pawlowski's out_ds.exe (95 kB self-extracting archive) which is still on his 'dead' website.
Shawn Riley kindly sent me a version of Peter's plugin that is compatible with WaveOut and Microsoft Sound Mapper drivers. I'm hosting it on this website, for the time being at least. I've added my own ReadMe file to it, providing my favoured settings, and compressed it as a 86 kB Cabinet file using PowerArchiver 2001. (This can be opened in almost any Windows environment, and is often associated with tools like Winzip and PowerArchiver if you have them installed) and should be extracted to your WinAmp plugins folder, such as C:\Program Files\Winamp\Plugins
I tend to use this plug-in sparingly, as I prefer to keep MP3-Splice as my output plugin when I'm playing mostly 44 kHz sampled files anyway.
Resampling and far more stuff related to high quality digital audio is integrated in Peter's Foobar 2000 audio player, which I now use in preference to WinAmp. It includes support for all high-quality-capable formats and some others and is superb if the music is the main thing.
PowerArchiver 2001 is a powerful archiver that behaves very much like the popular shareware WinZip (Classic, Wizard, shell extensions etc) but offers the ability to natively create CAB archives (cabinet) which are often its most compressed format (albeit multimegabyte folders take a long time to compress and need plenty of free disk space for temporary files during compression).
Like tar.gzip archives, CAB works very well on multiple files with common elements (like a whole folder of Excel or Word files with similar headers and footers). It natively supports ZIP, CAB, LHA, BH, JAR, ARC, RAR, ACE, TAR, TGZ and more. CAB can be opened but not written by WinZip (versions I've used) so this is a step up.
Another feature I enjoy is the use of Folder View, where a Windows Explorer-like tree structure allows you to around folder structures within the archive, rather than having to read a "Path" field in the window.
PowerArchiver also works in conjunction with SFX Maker to create more fully-featured self-extracting EXEcutable files out of ZIP archives, including spanned sets.
This version occasionally falls over none-too-gracefully if you run out of disk space for its temporary files while making a CAB (it's OK for nearly everything if you have a few gigs free), but its a very useful tool. Also the Move feature behaves like Add/Replace for some formats.
It has gone Shareware from 2002, but the freeware version is available from a few places, like FreeWare Pro or Yankee Web Works. Should it get hard to come-by, I may include it on this site, as permitted by the product license. At 2,137,581 bytes it's about a 6 minute download at 48 kbps.
InfoZIP is also a good, lightweight freeware command-line ZIP program that supports numerous operating systems and allows high levels of encryption if necessary.
This is an upgrade only necessary for WinAmp Lite install or WinAmp preceding version 2.8. It's already included as part of the package in Standard and Full versions of WinAmp 2.8 and later). Copy in_vorbis.dll to the plugins folder for WinAmp, usually C:\Program Files\Winamp\Plugins, close & reopen WinAmp and use Ctrl-P to access the preferences (including input plugin config and file format association)
Musepack is a superb format, particularly for audiophile quality compression, but not as widely supported as MP3. It uses sub-band encoding and performs superbly for transients and avoids pre-echo (which LAME --alt-preset standard had to work much harder to overcome in MP3 files, which use MDCT frequency-domain encoding) and it's highly tuned for optimum perceptual transparency using variable bitrate. You can download plugins for Winamp 2.xx and Winamp 3.xx from Case's page. Some good MPC examples are binaural recordings which make you feel like you're there.
A note on configuring the MPC plugin - if you enable ReplayGain loudness control, set the headroom slider to 'K14' (14 dB of peak headroom) which is equivalent to the 89 dB playback volume used in MP3gain and many other ReplayGain implementations. If you chose the original, rather quiet, 83 dB playback, the K setting is 'K20' (the default), which will make your MPCs sound very quiet if played among MP3s, OGG Vorbis and the like in your collection.
The main aim of MP3-splice is to fix the gap introduced between tracks on Live and Mix albums when tracks are individually encoded to MP3. This isn't trivial, and most attempts have problems and don't preserve intentional silence at the ends of tracks. MP3-Splice sounds perfect on a number of rips, whether CBR, ABR or VBR. Opera-lovers love the plugin too, and it doesn't care if you play an MP3, and MPC a WAV, and APE, an OGG Vorbis file in any order, it just does the job seamlessly. If your aim is musical quality, this is the business. It might crash certain visualisations (but I only care about the audio so don't use them) and will start preparing the next track 5-seconds from the end of each track it plays (so it doesn't have to close and reopen the WAVE device - a further cause of gappy playback) which stops your spectrum analyzer display, for example. Also listed on the WinAMP Classic output plugins page where you can read about a dozen rave reviews from users. I can't recommend it highly enough! The final perfection would be to allow the output to be fed to DiskWriter to create PCM WAVs for burning perfect live, mix and opera CDs from MP3s. Until then LAME --nogap encoding and decoding might work, or use of a different format which preserves the exact track length (like Ogg Vorbis or MPC).
P.S. I've tried Output_Stacker v2.xx which is supposed to allow plugins to cascade, but I can't make it work without crashing WinAMP 2.8. Ideally, I'd use MP3Splice followed by the out_wave_ssrc sample-rate convertor to both remove gaps and to resample to 44 kHz. I could also then use DiskWriter to create well-spliced PCM WAVs.
P.P.S. I'm rather reluctant to use the otherwise excellent MAD input plugin as it often misses the last few samples of a file (at least I've heard it does if your MP3 doesn't have an ID3v1 tag at the end). Undithered 16-bit output on WinAmp 2.8 sounds wonderful given the quality of my soundcard and the noise of my computer fan.
Unlike the in_wma.dll plugin bundled with newer WinAmp releases, this 91 kB plugin doesn't bypass the output plugins, allowing DSP effects and writing to disk. The latter enabled me to transcode a bulky 38 megabyte audio file of an 80-minute speech, encoded in "default 64kbps" via a mono WAV PCM file and into a 6 MB Ogg Vorbis file that's still comfortable to listen to for extended periods, especially when resampled to 44 kHz using a good sampling rate convertor output plugin
I'm a big fan of the Replay Gain standard for psychoacoustically-aware loudness standardization. I apply MP3gain's Album/Audiophile adjustment to almost all my MP3 files. This ensures I don't have to reach for the volume knob very often between tracks while preserving intended relative loudness differences within each album, preserving quality and avoiding clipping. Although "Radio/Track" setting is intended to equalize individual tracks for radio or shuffle-mode playing, the Album setting still greatly reduces annoyance when shuffling tracks in jukebox mode and is an excellent choice for me.
I also apply ReplayGain tags when encoding in other formats (e.g. using WinVorbis for analyzing Ogg Vorbis files). For these other formats, where the ReplayGain info is stored within a header or tag, the input plugin on WinAMP will support your choice of None/Radio/Album/clipping-prevention at time of playback (instead of choosing at the time you calculate the gain). The new Foobar audio player will support ReplayGain at time of playback if you append Monkey's Audio style APE tags (instead of ID3) to your MP3 files.
Unlike Cool Edit 2000, this version is an unlimited trial, but it still restricts you to two function set. This is a superb program, and you can try out all the tools it provides. The Help file is an education in itself! Find it here or here.
Cool Edit 96 supports the same file format filters as later versions, and these should be placed in the Cool Edit program folder.
Lossless encoding with APE.flt which is included in Monkey's Audio distribution, though in case you deleted the Tools folder it's below along with Ogg Vorbis filter cool_ogg.flt and the libmmd.dll file required in the Cool Edit folder to make the Ogg Vorbis filter work. Ogg Vorbis export compression options are pretty restricted, but it can import any Vorbis file.
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