2. Clean up get_audio_buffer_size_and_occupancy() and replace it with get_audio_buffer_occupancy().
3. Add a new general setting: audio_decoded_buffer_desired_length_in_seconds for managing the size of the player buffer queue when the source is buffered audio. Default is 0.75 seconds so that we can step back up to that time if older buffers appear.
- 48,000 frames per second operation,
- 48k lossless stereo, 5.1 and 7.1 surround sound
- Multichannel and multi rate operation on ALSA, PipeWire, PulseAudio, FreeBSD, stdout and unix pipe output backends.
Automatic, flexible and controllable output format (rate, sample format and channel count) selection.
Full FFmpeg integration to support transcoding, resampling, and new audio formats.
Better operation on lower powered devices down to e.g. Raspberry Pi B.
Reduced Docker image sizes with a slimmed-down FFmpeg library.
Enhanced timestamp handling for better synchronization.
Improved the sync error calculation.
A new "vernier" resampling and interpolation method for low-power CPUs.
Bug fixes and minor enhancements.
Note -- there are many breaking changes from previous versions of Shairport Sync!
Clean up and improve the way lead-in sillence is used to give a more accurate sync time to the first packet of audio.
Allow very good sync to occur with 0.3 seconds of lead-in silence.
Improve the use of the buffer offset.
Format some other files.
Previously I have made spelling changes as caught by Debian's lintian
tool, which has a short list of common misspellings that it tests for.
This time I decided to run codespell against the source and fix all the
obvious problems it came up with. It also uses a list of misspellings
but it's much larger than lintian's, but also has some false positives
so I went over the list by hand.
The command I used to do check the source was:
git ls-tree -rz --name-only HEAD | xargs -0 codespell -L minimise,errorstring
Mike Brady
Lukas Rusak
Mike Brady
Mike Brady
Chris Boot
shtirlic
James Laird
Dimitri Sokolyuk
MuffinmanandJames Laird
Paul Lietar