1. Add the ability to work back through queued buffers to find the first buffer that’s older than the given one. This is to enable the replacement of deciphered buffers with buffers that arrive later but are timed earlier. Added functions that relate to this are: get_revised_seqno() to wind back through the queued buffers and clear_buffers_from() to free buffers that are already occupied.

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.
2025-12-14 10:17:38 +00:00
parent ee56f778fd
commit 07b60e7112
10 changed files with 119 additions and 32 deletions
@@ -72,9 +72,10 @@ general =
 //		Use it, for example, to compensate for a fixed delay in the audio back end.
 //		E.g. if the output device, e.g. a soundbar, takes 100 ms to process audio, set this to -0.1 to deliver the audio
 //		to the output device 100 ms early, allowing it time to process the audio and output it perfectly in sync.
-//	audio_backend_buffer_desired_length_in_seconds = 0.2; // If set too small, buffer underflow occurs on low-powered machines.
+//	audio_backend_buffer_desired_length_in_seconds = 0.2; // This is the desired size of the buffer to be maintained in the external output system, e.g. the DAC in ALSA. If set too small, buffer underflow occurs on low-powered machines.
 //		Too long and the response time to volume changes becomes annoying.
-//		Default is 0.2 seconds in the alsa backend, 0.35 seconds in the pa backend and 1.0 seconds otherwise.
+//	audio_decoded_buffer_desired_length_in_seconds = 1.0; // Advanced feature. This is the desired size of the buffer of fully deciphered and decoded audio maintained within Shairport Sync prior to sending it to the external output system , e.g. the DAC in ALSA.
+//		Valid for AirPlay 2 Buffered Audio streams only.
 //	audio_backend_buffer_interpolation_threshold_in_seconds = 0.075; // Advanced feature. If the buffer size drops below this, stop using time-consuming interpolation like soxr to avoid dropouts due to underrun.
 //	audio_backend_silent_lead_in_time = "auto"; // This optional advanced setting, either "auto" or a positive number, sets the length of the period of silence that precedes the start of the audio.
 //		The default is "auto" -- the silent lead-in starts as soon as the player starts sending packets.