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.
(1) to be multithreaded,
(2) to work on multichannel audio and
(3) to work on 48k and 44.1k audio.
Allow multiple impulse response (IR) files with a new setting: "convolution_ir_files"
When convolution starts, Shairport Sync will look for an IR file with a sample rate
matching the input (44.1k or 48k) and channel count.
If one can't be found, it will look for a single-channel IR file with the same rate.
It will always choose the first match in the file list supplied "convolution_ir_files".
Allow multithreading -- use the "convolution_thread_pool_size" to set the number of threads to use.
Deprecate "convolution" -- use "convolution_enabled" instead.
Deprecate "convolution_max_length" -- use "convolution_max_length_in_seconds" instead.
Deprecate "convolution_ir_file" -- use "convolution_ir_files" instead.
Update corresponding D-Bus methods and properties.
Update the loudness code to work on 48k and well as 44.1k audio and with multichannel audio.
Deprecate "loudness" -- use "loudness_enabled" instead.
Update corresponding D-Bus methods and properties.
Fix a deprecated FFmpeg warning.
Update HiFi-LoFi FFT convolver to latest available.
- 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.
Specifically, firstly, the disable_standby_mode_silence_threshold, which is the
amount of audio in the output device's hardware buffer. It should normally
be close to the value givein in the audio_backend_buffer_desired_length_in_seconds
setting. If it drops to this value, silence is added to the buffer to prevent
the output device from becoming idle.
Secondly, the disable_standby_mode_silence_scan_interval is the time between checks
of the output device's hardware buffer.
The code for generating frames of possibly-dithered silence has been simplified.
Improve support for big-endian CPUs by adding support for explicit -endian formats, i.e. S16_LE, S16_BE, S24_LE, S24_BE, S32_LE, S32_BE.
For the "disable_standby_mode" setting, change "while_active" to "auto".
lululombard
Mike Brady
aaronk6
dasl-
Mike Brady
Guillaume Revaillot
Pieter De GendtandPieter De Gendt
Mike Brady
David Crook