Renamed 'WebRTC' -> 'WebRTC Audio' so users have a better chance of knowing what this indicator means

minor client updates

Fix the 'timestretch' fixup so that seeking is done after stretching

The previous version may have the beginning of the audio off by a bit,
since it was seeking in the audio file to find the start before the
stretch was applied.

Increase the audio bitrate in the video generation

Matches the corresponding change in the audio generation.

Recording updates merge

There is no i variable here, since there aren't multiple inputs to loop
over.

The concat filter requires at least 2 segments, it doesn't support
the degenerate case.

Vorbis needs more bitrate for input with higher sample rate/stereo

Use the correct timestamp for file start seek

Rewrite audio processing to use ffmpeg

There's some fairly major changes here, including:

* All audio is resampled to 48kHz stereo on input, allowing files with
  non-standard or varying rates to be mixed
* The audio processing is now done in a single pass; ffmpeg reads all
  inputs, concatenates them, and outputs one file.

added record icon to the recording notification

Properly resized upload presentation window, removed unnecessary text