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RF Compression & Decompression Guide
Previous Page RF Capture Guide
Sub-Page Speed & Decoding Testing
Next Page RF Decoding Guide
FM RF data is just like audio data just electrical signal values digitised into bits of information, but this is just more information in a different waveform pattern than normal sound waves, as such we can use lossless audio codecs like FLAC to compress captured FM RF data down A LOT or lossy ones to just break the files in fun ways too!
Now Down-Sampling is a method of cutting file size down with controlled loss of data more accurately removing background noise or dead space for example a 10-bit ADC only has 10-bits of information but can only be saved in a 16-bit format FLAC will remove those 6 extra bits of essentially 0 value information, alongside filtering cut off removing anything else recorded outside of the real-world signal range for said particular audio or video format, simmer to how a XLR microphone will have a cut off for 75hz to remove buzzing sounds from lights
This process can be done for low bandwidth formats like HiFi FM signals without much care as only 5msps or less is needed but for video formats its more a stricter practice and the bandwidth also changes between PAL/NTSC systems on a per format basis.
Down sampling is most practical with 40msps 8-bit and 40msps 16-bit or higher rate captures from the Modified CX Cards or DomesDayDuplicator while redundant samples for initial capture is never a bad thing, for stable media it can be a waste of space.
The biggest gain here is in the limited size optical discs for archival the commands can be easily tweaked on an as-needed basis for example VHS NTSC has been resampled down to 325MB/Minute without issue, and has become common practice.
However and for unstable media it's advised to keep just a normal FLAC compressed RF without re-sampling it.
Down-sampling is ideal for maximum space saving of stable media, however this is not lossless this is lossy so is not recommended unless you absolutely have to save space as this can yield upto a 10:1 ratio compared to raw captures, these values below should be taken with a grain of salt as re-sampling can very per media format and per TV system such as NTSC/PAL.
(NTSC VHS re-sampling has been fully tested)
16msps 8-bit with FLAC compression for VHS NTSC
Runtime | File Size | Storage Medium Note |
---|---|---|
45min | 15 GB | |
60min | 19.44GB | |
75min | 24.30GB | 25GB M-Disk/GlassMasterDisc |
90min | 29.16GB | S-VHS/VHS-C tape max |
120min | 38.88GB | |
150min | 48.6GB | 50GB M-Disk / 50GB GlassMasterDisc Max |
3-hour | 58.32GB | VHS SP Max |
240min | 77.76GB | |
5-hour | 97.2GB | 100GB M-Disk Max |
6-hour | 116.64GB | 128GB Sony Quad Layer BDXL Max / VHS LP Max |
RF Data at 40msps in 10-bit packed
--> ld-lds-converter
--> 16-bit singed unpacked data
--> Down Sampling via SoX or GNURadio with lowpass filtering
--> 16msps 8-bit unsinged FLAC output
The benefits of this are purely for extra space savings, but can also speed up decoding and remove interference outside of the normal signal range, however for bad tapes however only standard FLAC compression is advised as frames of information can be damaged or lost entirely.
ld-compress DdD-capture.lds
Will just compress your capture to a lossless 40msps 16-bit FLAC file in .ldf
.
Copy the scripts below into a text document, save and then add .sh extension to the end of the file, you can use these in any directory witch the script file is put inside of, you can name the scripts however you like but the below example is clear enough.
./40msps-DdD-16msps-8bit.sh DdD-capture.lds
Will make DdD-capture_NTSC_16msps_8-bit.flac
The scripts below automatically append the sample rate/bit-depth/TV System accordingly.
#!/bin/bash
echo "Conversion of 10-bit 40msps .lds to 8-bit 16msps NTSC .flac has started"
ld-lds-converter -i $1 | sox -r 40000000 -b 16 -c 1 -e signed -t raw - -b 8 -r 16000000 -e unsigned -c 1 -t raw - sinc -n 2500 0-7650000 | flac -8 --sample-rate=16000 --sign=unsigned --channels=1 --endian=little --bps=8 - -o $1_NTSC_16msps_8-bit.flac
#!/bin/bash
echo "Conversion of 10-bit 40msps .lds to 8-bit 18msps PAL .flac has started"
ld-lds-converter -i $1 | sox -r 40000000 -b 16 -c 1 -e signed -t raw - -b 8 -r 18000000 -e unsigned -c 1 -t raw - sinc -n 2500 0-8670000 | flac -8 --sample-rate=18000 --sign=unsigned --channels=1 --endian=little --bps=8 - -o $1_PAL_18msps_8-bit.flac
#!/bin/bash
echo "Conversion of 10-bit 40msps .lds to 8-bit 24msps .flac has started"
ld-lds-converter -i $1 | sox -r 40000000 -b 16 -c 1 -e signed -t raw - -b 8 -r 24000000 -e unsigned -c 1 -t raw - sinc -n 2500 0-9400000 | flac -8 --sample-rate=20000 --sign=unsigned --channels=1 --endian=little --bps=8 - -o $1_24msps_8-bit.flac
#!/bin/bash
echo "Conversion of 10-bit 40msps .lds to 8-bit 5msps .flac has started"
ld-lds-converter -i $1 | sox -r 40000000 -b 16 -c 1 -e signed -t raw - -b 8 -r 5000000 -e unsigned -c 1 -t raw - sinc -n 2500 0-3050000 | flac -8 --sample-rate=5000 --sign=unsigned --channels=1 --endian=little --bps=8 - -o $1_VHS_HiFi_5msps_8-bit.flac
Video8/Hi8 hifi is in the same modulated signal as video just on different carriers so you use 1 file for both video/hifi decoding.
There will be WSL2 passthrough scripts for windows made soon.
Download ld-tools for windows rename the folder to ld-tools-suite-windows
and place it in the C:/
directory or your boot drives main directory for scripts to work.
Windows scripts for CXADC Captures can be found here!
You will need tools installed as PATH
which is easy to do with chocolatey.
PowerShell as an administrator
Install Choco (chocolatey package manager)
Set-ExecutionPolicy Bypass -Scope Process -Force; [System.Net.ServicePointManager]::SecurityProtocol = [System.Net.ServicePointManager]::SecurityProtocol -bor 3072; iex ((New-Object System.Net.WebClient).DownloadString('https://community.chocolatey.org/install.ps1'))
Then you can install these tools system-wide (PATH) without any hassle.
Install FFmpeg
choco install ffmpeg
Install FLAC
choco install flac
Install SoX
choco install sox.portable
Manual .exe downloads SoX / FLAC / FFmpeg
You can capture in 16-bit uncompressed or 10-bit packed
Since this is uncompressed data it is easy to handle
ffmpeg INPUT.s16 -i -f s16le -ar 40k -ac 1 -acodec flac -compression_level 11 -f ogg OUTPUT.flac
This outputs a unpacked file in .flac
(Not Actually "Compressed")
2.8GB/Min 10-bit packed to 625MB/Min 16-bit FLAC
C:\ld-tools-suite-windows\ld-lds-converter.exe -u -i INPUT.lds | ffmpeg -f s16le -ar 40k -ac 1 -i - -acodec flac -compression_level 11 -f ogg OUTPUT.ldf
You can then compress this file to a FLAC compressed file.
ffmpeg -i INPUT.flac -f s16le -ar 40k -ac 1 -acodec flac -compression_level 11 -f ogg OUTPUT.flac
Rename the .flac to your media format name using standard designators & standard naming guide
Sub-Page Speed & Decoding Testing
Previous Page RF Capture Guide
Next Page RF Decoding Guide
- FAQ - Frequently Asked Questions
- Diagram Breakdowns
- Visual-Comparisons
- VCR Reports / RF Tap Examples
- Download & Contribute Data
- Speed Testing
- Visual VBI Data Guide
- Closed Captioning
- Teletext
- WSS Wide - Screen Signalling
- VITC Timecode
- VITS Signals
- XDS Data (PBS)
- Video ID IEC 61880
- Vapoursynth TBC Median Stacking Guide
- Ruxpin-Decode & TV Teddy Tapes
- Tony's GNU Radio For Dummies Guide
- Tony's GNU Radio Scripts
- DomesDay Duplicator Utilities
- ld-decode Utilities