| Commit message (Collapse) | Author | Age | Files | Lines |
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Only whitespace changes in here
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The same logic is applied to the sample rate adjustments in module-rtp-recv,
module-loopback and module-combine:
- Each time an adjustment is made, the new rate can differ at most 2‰ from the
old rate. Such a step is equal to 3.5 cents (a cent is 1/100th of a
semitone) and as 5 cents is generally considered the smallest observable
difference in pitch, this results in inaudible adjustments.
- The sample rate of the stream can only differ from the rate of the
corresponding sink by 25%. As these adjustments are meant to account for
very small clock drifts, any large deviation from the base rate suggests
something is seriously wrong.
- If the calculated rate is within 20Hz of the base rate, set it to the base
rate. This saves CPU because no resampling is necessary.
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The arguments sink_dont_move and source_dont_move have been added to toggle
module automatic unloading when the sink or source were no longer
available, rather than just moving them to the next available
sink/source (via rescue streams).
Reviewed and tweaked by Colin Guthrie.
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not know anything about
All seeks/flushes that depend on the playback buffer read pointer cannot
be accounted for properly in the client since it does not know the
actual read pointer. Due to that the clients do not account for it at
all. We need do the same on the server side. And we did, but a little
bit too extreme. While we properly have not applied the changes to the
"request" counter we still do have to apply it to the "missing" counter.
This patch fixes that.
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I checked the source code, and latency_msec is missing from the list
of valid module arguments. Attaching a patch to add it.
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hooks can access it
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