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This fixes all current code smells, bugs and issues reported by
SonarCloud static analysis. Other issues are almost exclusively false
positives. This makes much code clearer, and some minor benefits in
performance or bug evasion may be gained.
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Note: While ClientAudioStream has had a volume property, it is only now
used in the mixer.
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Across the entire audio system, audio now works in 0-1 terms instead of
0-100 as before. Therefore, volume is now a double instead of an int.
The master volume of the AudioServer changes smoothly through a
FadingProperty, preventing clicks. Finally, volume computations are done
with logarithmic scaling, which is more natural for the human ear.
Note that this could be 4-5 different commits, but as they change each
other's code all the time, it makes no sense to split them up.
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async_enqueue() is a wrapper over the async_enqueue_buffer() call
to AudioServer. This allows users to asyncronously enqueue audio
samples, when the program requires non-blocking audio streaming.
This also makes ClientConnection use east-const everywhere.
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We don't know what is a good time to wait after an audio buffer fails to
be processed by AudioServer. However, it seems like decreasing the wait
time to 10ms after such a failure should improve latency and has not
caused issues in my testing. After all, 10ms is quite some time in audio
sample magnitudes.
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This commit addresses two issues:
1. If you play a 96 KHz Wave file, the slider position is incorrect,
because it is assumed all files are 44.1 KHz.
2. For high-bitrate files, there are audio dropouts due to not
buffering enough audio data.
Issue 1 is addressed by scaling the number of played samples by the
ratio between the source and destination sample rates.
Issue 2 is addressed by buffering a certain number of milliseconds
worth of audio data (instead of a fixed number of bytes).
This makes the the buffer size independent of the source sample rate.
Some of the code is redesigned to be simpler. The code that did the
book-keeping of which buffers need to be loaded and which have been
already played has been removed. Instead, we enqueue a new buffer based
on a low watermark of samples remaining in the audio server queue.
Other small fixes include:
1. Disable the stop button when playback is finished.
2. Remove hard-coded instances of 44100.
3. Update the GUI every 50 ms (was 100), which improves visualizations.
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When using `aplay` to play audio files with a sample rate of 96000,
there were occasional one-second gaps in playback. This is
because the Audio::ClientConnection sleeps for a full second when
the audio buffer is full.
One second is too long to sleep, especially for high-bitrate files.
Changing the sleep to a more reasonable value like 100 ms ensures
we attempt to enqueue again before the audio buffer runs empty.
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This is no longer used by any of our IPC pairs.
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This changes client methods so that they return the IPC response's
return value directly - instead of the response struct - for IPC
methods which only have a single return value.
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This updates all existing code to use the auto-generated client
methods instead of post_message/send_sync.
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Instead of having a single overloaded handle method each method gets
its own unique method name now.
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SPDX License Identifiers are a more compact / standardized
way of representing file license information.
See: https://spdx.dev/resources/use/#identifiers
This was done with the `ambr` search and replace tool.
ambr --no-parent-ignore --key-from-file --rep-from-file key.txt rep.txt *
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The client ID is not useful to normal clients anymore, so stop telling
everyone what their ID is.
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This was the last remaining user of shbufs! :^)
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