1.10.3

CHANGELOG.md

@@ -1,6 +1,7 @@
 # swyh-rs Changelog
 
-- 1.10.3 (unreleased)
+- 1.10.3 (Apr 82024 dheijl)
+  - GUI: try to stop any players still streaming when the swyh-rs windows is closed. If the players don't stop streaming voluntarily exit anyway (after a 5 second time-out).
 
 - 1.10.2 (Apr 8 2024 dheijl)
   - clip player button text

README.md

@@ -23,7 +23,7 @@ A "Stream-What-You-Hear" implementation written in Rust, MIT licensed.
 
 ## Current Release
 
-The current release is 1.10.2, refer to the [Changelog](CHANGELOG.md) for more details.
+The current release is 1.10.3, refer to the [Changelog](CHANGELOG.md) for more details.
 
 ## Why this SWYH alternative ?
 
@@ -124,13 +124,14 @@ The icon was designed by @numanair, thanks!
 ### How does it work?
 
 - audio is captured from the default audio device (WasApi on Windows, Alsa on Linux, not tested on Mac), but you can choose any audio source you want. Changing the sound source needs a restart of the app to take effect.
-- On Windows you can check in the soundmixer that the audio device you're capturing is the device that is actually playing audio. On Linux you can use [pavucontrol](https://freedesktop.org/software/pulseaudio/pavucontrol/) to enable the audio monitor for the audio device you are capturing.
+- On Windows you can check in the **soundmixer** that the audio device you're capturing is the device that is actually playing audio. On Linux you can use [pavucontrol](https://freedesktop.org/software/pulseaudio/pavucontrol/) to enable the audio monitor for the audio device you are capturing.
+- you can (and probably should) use the "_RMS monitor_" feature to verify that swyh-rs is actually capturing audio.
 - a built-in audio streaming web server is started on port 5901.
 - all media renderers are discoverded using SSDP on the local network, this takes about four seconds to complete. By default the network that connects to the internet is chosen (so that on a multihomed Windows machine the most likely interface is selected). If necessary you can choose another network from the network dropdown, for instance if you use a VPN.
 - then a button is shown for every renderer found
 - if you click the button for a renderer the OpenHome or AvTransport protocol is used to let the renderer play the captured audio from the webserver
 - audio is always sent in audio/l16 PCM format, no matter the input source, using the sample rate of the source, unless you enable 24 bit LPCM (see below).
-- some renderers will stop when detecting a pause between songs or for some other unknown reason. You can use the *_Autoresume_" checkbox if you encounter this problem. But always try to disable the "_Chunked Transfer Encoding_" first to see if this fixes the problem before you enable AutoResume. Since version 1.3.2 AutoResume should work with OpenHome renderers too (tested with Bubble UPNP Server and Chromecast/Nest Audio).
+- some renderers will stop when detecting a pause between songs or for some other unknown reason. You can use the "_Autoresume_" checkbox if you encounter this problem. But always try to disable the "_Chunked Transfer Encoding_" first to see if this fixes the problem before you enable AutoResume. Since version 1.3.2 AutoResume should work with OpenHome renderers too (tested with Bubble UPNP Server and Chromecast/Nest Audio).
 - there is an "_Autoreconnect_" checkbox, if set the last used renderer will be automatically activated on program start
 - since 1.4.0 there is a dropdown that lets you choose between FLAC, LPCM or WAV format. Preferred format is FLAC, WAV or LPCM should only be used if FLAC does not work. Also, only FLAC will work with 24 bit.
 - there is (since 1.3.20) a check box "_24 bit_". It causes audio to be streamed in 24 bit LPCM format (audio/L24) with the sampling rate of the audio source. It only works reliably with the FLAC format. 24 bit works with Bubble/UPNP too with LPCM, but not with hardware streamers.