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Most audio tape-based computer data formats (record and play on standard audio tape deck) are designed to work with limited frequency response and low SNR, speed variations, and phase shifts. They typically only need telephone grade bandwidth and SNR (4 KHz bandwidth, 40-55 dB SNR). You could use a lower sample rate such as 11 or 8 KHz / 8 bit.
Do consider the playback mechanism, though. It's no use using 8/8 format, or some specialised codec format, if the playback device doesn't support it.
Personally, I'd use MP3 encoding.  About 10 years ago I had to record the training sequence and data stream of a 2400 bps modem and send it to an overseas lab. I recorded it at 16/44.1 mono and converted to 128 Kbps MP3. The lab had no difficulty playing it into their analyser. I just now took one of the original files and compressed it with LAME at quality 9 (VBR, 45-85 Kbps) for a 20:1 reduction.  Overlaying the input and output waveforms shows an almost perfect match.
General - (fb2k) / Re: Replaygain settings per folder
Last post by lvqcl -
automatically turn on track-based replay gain
Remove album gain tags from these files.
General - (fb2k) / Replaygain settings per folder
Last post by hlloyge -
Hello all!

I have one question (and proposal) for possible situation.

I have more than few compilation folders where I put different music from different albums, and then playing them back. I would like to automatically turn on track-based replay gain and set boost gain when I load that folder into foobar. For this to be achieved, I have one possible solution - manually create a file in the folder, text, for example, rgain.set, and inside that file would be information for replaygain configuration of foobar2000 player:


(for example).

This file, when found in folder when adding files to playlist, would override current RG configuration and be applied to files in that folder by default. Manual change could be possible, this would just load 'default' configuration for that folder.
Is this possible to do, and is there interest for that? I am very interested :)
This may not be applicable to your situation, but the most straightforward way to accomplish this would be to port the utility that generates the wav file from the computer data, and then simply have Android generate and play the audio on-the-fly from the original data you want to transfer. Of course, this assumes you have the source code and programming skills.

Otherwise, WavPack lossy might be a good choice because it leaves the waveform unaltered. Regular lossy codes based on human hearing models are probably not recommended.
You can try other losless codecs, like Monkey's Audio or Wavpack. But, I've seen people had compressed C64 tapes in mp3 format and loading them from portable CD player through some kind of interface, replacing cassette player.

I'm looking for a codec that's suitable to compress the audio of old computer tapes. I have an audio file that is generated, so it's flawless in terms of noise. I will be using some device (a phone most likely) to play this audio file to a retro computer instead of using it's ageing tape player.

A small mono wav-file is 1705kB, flac can compress it down to 688kB. 7z can compress the wav to 5kB! The flac-file can be compressed down to 25kB with 7z. (There are around 1kB of actual computer data in the audio file.)

I understand that this is a very specific scenario when it comes to audio encoding. Are there any other codecs that are more suited for these tasks?

If not, does anyonw know any Android player that transparently can decompress a 7z and play the wav inside it on the fly?

General - (fb2k) / Re: issues with album artist in WAV files?
Last post by masi -
This did the trick.

Opus / Re: opusgain scanner implementation status
Last post by dutch109 -
There is basic support for transferring REPLAYGAIN tags in FLAC files into Opus gains.
* If you use ffmpeg to encode your files, REPLAYGAIN tags will be copied over.
The Opus RFC explicitly discourages against this:
   To avoid confusion with multiple normalization schemes, an Opus
   comment header SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN,
   REPLAYGAIN_ALBUM_PEAK tags, unless they are only to be used in some
   context where there is guaranteed to be no such confusion.
   [EBU-R128] normalization is preferred to the earlier REPLAYGAIN
   schemes because of its clear definition and adoption by industry.
   Peak normalizations are difficult to calculate reliably for lossy
   codecs because of variation in excursion heights due to decoder
   differences.  In the authors' investigations, they were not applied
   consistently or broadly enough to merit inclusion here.

Good audio tools such as foobar will also do gain calculations for you and apply the tags.

Not sure that helps much if your player ignores the tags!
Yeah I tried foobar2000 mobile on Android, and it handles the output gain header + R128 tags properly.
However it has its own set of flaws compared to GMMP (little possible customization compared to the foobar2000 on Windows that I remember, incomplete Bluetooth support, album arts sometimes not showing up...), which prevents me from labeling it as "good audio tool".

An official opusgain tool isn't high on the list of priorities.
Why is that? I am not familiar with the Ogg container format or the Opus codec internals, but using libebur128 + the vorbisgain code as a base, it does not strike me as very complex to implement. If all that is missing is some development time, I'd be happy to help (I write C/C++ code for a living).

* loudness-scanner supports scanning opus files and adding REPLAYGAIN tags to them.
Thanks, didn't know that one. Will try and report back.
General Audio / Re: Powering Up Audio System
Last post by dc2bluelight -
Can't believe this, but today I just ran into a piece of gear with the inrush protection devices!  Now, it's not a piece of audio gear, it's an older low power FM broadcast transmitter, 200W, uses a HUGE linear power supply. Their design calls for a moderate 6A slow-blow input fuse, and the power goes through a combo IEC filter/fuse holder/voltage selector, then on to the inrush devices and a pair of MOVs.  Since the inrush devices (yes, 2 of them) toasted themselves,  (we took a surge that took out the MOVs too), I bypassed the inrush devices temporarily until parts arrive so I could get the TX on the air.  Couldn't actually get the thing all...with anything less than a 20A fuse because of the inrush! 

The product is older, and new designs of this product use switching PS, so this issue is over.  But I'm amused that I was just posting in this thread and just a day later that design lands in my lap. 
Audio Hardware / Re: Interview Time
Last post by silverprout -
What is a "legacy SS amp"? How far back are you going in time? 10 years? 20 years? More?

Modern class D amplifiers are also "damped" by their output filters, some of them can be peaky in the highs if the there is an impedance mismatch.