Technical Documentation: audacity/audacity
ℹ️ Provenance: Multi-Tier Ingestion · CodeWiki Reference
Overview (README.md)
Audacity
Audacity is an easy-to-use, multi-track audio editor and recorder for Windows, macOS, GNU/Linux and other operating systems. More info can be found on https://www.audacityteam.org
This repository is currently undergoing major structural change.
We're currently working on Audacity 4, which means an entirely new UI and also refactorings aplenty. As such, the master branch is currently not particularly friendly to new contributors. It is still possible to submit patches to Audacity 3.x; make sure you branch off audacity3 if you choose to do so. Build instructions for 3.x can be found here; build instructions for Audacity 4 can be found here.
You can stay updated with our efforts on YouTube, Discord and our blog.
License
Audacity is open source software licensed GPLv3. Most code files are GPLv2-or-later, with the notable exceptions being /au3/lib-src (which contains third party libraries), as well as VST3-related code. Documentation is licensed CC-by 3.0 unless otherwise noted. Details can be found in the license file.
Effect View Architecture (`docs/effect-view-architecture.md`)
/* Detailed source-code truncated for AI context efficiency. */Portaudio Reported Playback Capture Latency (`docs/portaudio-reported-playback-capture-latency.md`)
Imagine there were an audio cable plugged from your output back to your input. You press play, and at some stage, the project audio will reach back the input and become recorded. Latency compensation simply consists of discarding all incoming audio until the project audio arrives. Then users could play their project, record themselves on the fly and the incoming audio would be in sync without further manual adjustment required.
Audacity already has a latency-compensation setting for the user to adjust manually. Automatically adjusting this setting optimally would be trivial as soon as we knew the round-trip latency of the audio device.
PortAudio reports these latencies, but experiments showed that they weren't reliable. Here I make an experiment that I believe to provide exact estimates of round-trip latency.
Description of experiment
- Connect the selected output device to the selected input device with an audio cable.
- Before recording starts, create a wav file.
- In the audio IO callback (on the audio driver thread)
- overwrite the output buffer with one second of silence, then with a sine wave
- write the samples of the input buffer to the wav file.
- When recording stops, close the wav file.
- Open the file in Audacity and observe the latency.
Results
| OS | Host | requested buffer size (ms) (*) | PA-reported output buffer size (ms) (**) | PA-reported input buffer size (ms) | Measured round-trip (ms) | error (reported - actual) (ms) |
|---|---|---|---|---|---|---|
| Win | WASAPI | 250 | 270 | 260 | 75 | 455 |
| Win | WASAPI | 100 | 120 | 110 | 75 | 155 |
| Win | WASAPI | 20 | 40 | 30 | 75 | -5 |
| Win | DirectSound | 250 | 250 | 62.5 | 420 | -107.5 |
| Win | DirectSound | 100 | 100 | 25 | 115 | 10 |
| Win | DirectSound | 20 | didn't work | didn't work | didn't work | didn't work |
| Win | ASIO | 500 | 50.5 | 47.9 | 100 | -1.6 |
| Win | ASIO | 100 | 50.5 | 47.9 | 100 | -1.6 |
| Win | ASIO | 20 | 26.5 | 24.7 | 54 | -2.8 |
(*) applies to both input and output, e.g. 250 means 250 for the input + 250 for the output, in total 500.
(**) for one same requested size, may differ wildly depending on whether only playback is active or playback and recording. Here we write the values obtained when both are active.
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- Canonical Reference: https://codewiki.google/github.com/audacity/audacity