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.
Player now stops 10 secs after stop command and 10 mins after pause. At
that time the outputs have probably cut the connection themselves, but
that might be ok (needs testing).
outputs_playback_start() had the problem that was not consistently invoked: If
for instance local audio playback was running and a Airplay device was then
activated, the raop's playback_start would never be invoked (and vice versa,
of course).
Instead, the player now writes the presentation timestamp every time to the
output, so it doesn't need to keep track of it from the start.
* Untie Airtunes stuff further from player and non-Airplay outputs
* Change raop.c to use rtp_common.c (step 1)
* Change heartbeat of player to 100 ticks/sec, since we have untied from
Airtunes 352 samples per packet (which equals 126 ticks/sec at 44100)
Still a lot to be done in the player, since the rtptime's in it don't
are probably broken.
Some remotes don't respond as expected to the test. Retune will give connection
refused, because the test is made too quickly, before the service is running.
Even if we delay the test it won't work because Retune crashes.
Since the false mdns advertisements are only seen on Airplay, we only do the
test there.
The unconfigurable resync period of 10 seconds was not frequent
enough to keep my own ALSA device in sync with the AirPlay stream.
Now the period is configurable. The default is still at 10
seconds, to prevent any change in behavior unless opted in by
the user.
Currently the adjustment causes a tiny "click" distortion in the
ALSA output, so it is better to make the check as infrequent as
possible, while still being frequent enough to stay in sync
over lengthy sessions of playback.
Added source_sample_rate, target_sample_rate to alsa_session.
This is a first step toward rendering ALSA at a different
sampling rate than the AirPlay stream, so that (a) we will
be able to dynamically adjust the ALSA sampling rate for an
improved sync algorithm, and (b) later, a more generalized
resampling algorithm can accommodate very different hardware
sampling rates like 22050 Hz or 48000 Hz.
Reworked alsa_session_free() so that it can be used to
tear down a partially initialized alsa_session if an
error occurs in the middle of alsa_session_make().
This simplifies the error handling logic in alsa_session_make().
This refactoring will be helpful later when resampling is added,
because more data structures will be dynamically allocated
during initialization.
Signed-off-by: Don Cross <cosinekitty@gmail.com>
AirPlay 2 devices like Sonos One and AirPort Express with firmware 7.8
require auth-setup before ANNOUNCE, otherwise they will return 403.
Also fixed a problem where metadata did not get sent when toggling
a speaker on/off if we were playing an endless stream.
