[solved] Sample data import - result is a blank audio track

OS
Fedora 35, Cinnamon spin

Audacity about screen - installed by dnf command

The Build
Program build date:
Apr 10 2022
Commit Id:
ff6d67 of Wed Dec 22 16:35:36 2021 +0100
Build type:
CMake Debug build (debug level 1), 64 bits
Compiler:
GCC 11.2.1
Installation Prefix:
/usr
Settings folder:
/home/geir/.audacity-data
Core Libraries
wxWidgets
(Cross-platform GUI library)
3.1.5
PortAudio
(Audio playback and recording)
v19
libsoxr
(Sample rate conversion)
Enabled
File Format Support
libmad
(MP3 Importing)
Enabled
libvorbis
(Ogg Vorbis Import and Export)
Enabled
libid3tag
(ID3 tag support)
Enabled
libflac
(FLAC import and export)
Enabled
libtwolame
(MP2 export)
Enabled
QuickTime
(Import via QuickTime)
Disabled
ffmpeg
(FFmpeg Import/Export)
Disabled
gstreamer
(Import via GStreamer)
Disabled
Features
Theme
(Dark Theme Extras)
Disabled
Nyquist
(Plug-in support)
Enabled
LADSPA
(Plug-in support)
Enabled
Vamp
(Plug-in support)
Enabled
Audio Units
(Plug-in support)
Disabled
VST
(Plug-in support)
Enabled
LV2
(Plug-in support)
Enabled
PortMixer
(Sound card mixer support)
Enabled
SoundTouch
(Pitch and Tempo Change support)
Enabled
SBSMS
(Extreme Pitch and Tempo Change support)
Enabled

The problem is that I’m not able to import the audio successfully. I don’t get any error messages, but the resulting audio track does not seems to include any actual data - but get a length.

Things I’ve tried out / is done

  • Tried different text encoding, including unicode and west european (but in limited number of attempts)
  • Different line endings, both Windows and Linux line endings are tested, same result (use Geany for this and last point)
  • Tried to replace all numbers to force 5 decimal places (just a random thing I did because I saw another forum thread where OP had used 5 decimals).

I don’t understand if there is something I do wrong here? I expect to see a wave curve (probably not pretty), but get nothing.
IA.txt (5.31 KB)
Screenshot from 2022-08-03 21-24-00.png

It works for me, but note that IA.txt has only 680 numbers / samples, so at a sample rate of 44100 Hz that’s only 0.015 second.

After importing the samples, try pressing “Ctrl + F” (zoom to fit project in window).

First Track000.png

Thank you very much, I didn’t expect it being that short. Found the waveform and problem solved :smiley:

:smiley: Nothing unusual there. Most folk underestimate how much data is required by digital audio.
There’s an easy way to work out the length:

If the sample rate is 44100 Hz, then there are 44100 samples per second (for stereo it’s 44100 samples per second per channel).
So to calculate the length, divide the number of samples by the sample rate. In your example that is:
680 / 44100 = 0.01542 seconds (a little over 15 milliseconds).