Moves speaker selection, volume handling and startup to outputs.c, plus adds
the ability to "resurrect" a speaker that disconnects.
The purpose of moving the code is to concentrate device handling in one place.
Also changes how we deal with speaker selection. The player will now generally
not alter a user selection, even if the device fails. The purpose of this is to
maintain selection both if the device briefly fails, and if the user switches
off the device (we stop playback) and later turns it on + starts new playback.
Instead of using OPTIONS we use SET_PARAMETER with progress metadata to avoid
disconnects from Apple TVs, Homepods and possibly also Airport Expresses.
The player will write 24 bit samples using 3 bytes, not 4, so the appropriate
sample format is SND_PCM_FORMAT_S24_3LE, not SND_PCM_FORMAT_S24_LE.
For extra protection we also use snd_pcm_bytes_to_frames() instead of BTOS(),
because that way we can be more certain that the buffer is not too short for
snd_pcm_writei().
Since it is unknown how to do real sync on Chromecast, this commit instead adds
a primitive delay to the stream, so that it is at least somewhat closer to
Airplay/local audio.
Also some cleanup of unused stuff.
Fixes bugs which were due to incorrect handling of unsigned integer wrap-around:
1. Calling packet_resend() with seqnum + len greater than UINT16_MAX => infinite loop
2. Calling rtp_packet_get() with session->seqnum - seqnum greater than pktbuf_next => wrong packet
* Fix "clicks" during playback, especially on low buffer size devices
Bug had two causes: Trying to write to the prebuf ringbuffer when it was full
and writing new audio to the device without first having drained the prebuf,
thus writing out of order.
* Use snd_pcm_drain() so alsa doesn't report underrun on playback session end
Removes SNDRV_PCM_IOCTL_SYNC_PTR errors
* Fix missing error check of the return value from snd_pcm_avail (now use snd_pcm_avail_delay)
This change is preparation to use ffmpeg's resampling capabilities to keep local
audio in sync (by up/downsampling slightly). This requires that sample rates are
not fixed for a transcode profile.
Added benefit of this is that we don't need quite as many xcode profiles.
In the output implementations playback_stop() was somewhat redundant,
since device_stop() does the same.
The timer should make sure that we always close outputs (previously
they were in some cases kept open).
The commit also includes some renaming.
