Some of the logs were only useful for debug scenarios, so they have
been downgraded to DEBUG. Others were in INFO but supposed to be in
WARNING so those were upgraded.
Segments where readability was hampered were fixed by appropriate
format skipping directives. New code should hopefully be black
compatible. The moment they remove the -S option is probably the moment
I stop using black though.
Rework FFmpeg player and add FFmpegOpusAudio
I have extracted some of the base FFmpeg source code into its own
base class and reimplemented the PCM and the new Opus variants.
Support avconv probing
Also fix a few things
Update `__all__`
Fix the bugs
Rework probe functions and add factory function
Probing involves subprocess so it has been reworked into an async
factory function.
Add docs + a few tweaks
* Removed unnecessary read() and is_opus() functions from FFmpegAudio
* Clear self._stdout in cleanup()
* Add 20 second process communication timeout to probe functions
* Capped probe function bitrate values at 512
Change AudioPlayer to use more accurate, monotonic time.perf_counter()
Add lazy opus loading
The library now no longer loads libopus on import, only on
opus.Encoder creation or manually.
Fix review nits
Use bare raise statement when reraising the exception that occured, and
remove unused exception variables. Also remove a pointless exception
handler in discord.opus.
This is a complete redesign of the old voice code.
A list of major changes is as follows:
* The voice websocket will now automatically reconnect with
exponential back-off just like the regular Client does.
* Removal of the stream player concept.
* Audio now gracefully pauses and resumes when a disconnect is found.
* Introduce a discord.AudioSource concept to abstract streams
* Flatten previous stream player functionality with the
VoiceClient, e.g. player.stop() is now voice_client.stop()
* With the above re-coupling this means you no longer have to
store players anywhere.
* The after function now requires a single parameter, the error,
if any existed. This will typically be None.
A lot of this design is experimental.
Enable forward error correction + packet loss percent tuning in opus encoder. Additionally, use some sane defaults. This should fix people hearing robo in music when packet loss is happening.
