Deactivate the stream in Cancel(). This fixes stuttering after a
manual song change by refilling the whole ring buffer before
reactivating the stream.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1354
Clear not only MPD's ring buffer, but also libpipewire's buffers, to
avoid playing some audio from the previous song after a manual song
change.
Fixes part 1 of https://github.com/MusicPlayerDaemon/MPD/issues/1354
Removing the LockHasClients(); this code was copied from the "httpd"
output plugin, but unlike "httpd", the SnapCast output plugin does not
feed silence while paused, so we need to implement a delay to avoid
busy-looping the CPU.
As a side effect, this eliminates the suttering after resuming
playback, because the timer now gets reset even if there is a client.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1394
Since PipeWire 0.3.53, there is no control name anymore, therefore the
name check doesn't work anymore, breaking volume change events.
This obsoletes the crash bug fix in commit 2ee57f9b0d
If the PipeWire output has not yet been enabled and no thread_loop has
been created yet, a nullptr dereference in SetVolume() was possible
because nullptr was passed to pw_thread_loop_lock().
Even if the "wave_encoder" option is disabled (and no other encoder
plugins are enabled), forcefully enable the Wave encoder (if Snapcast
is enabled).
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1500
Previous versions of MPD would call SetVolume on enabled outputs before
they are ready, causing all of MPD to crash. Checking the really_enabled
flag prevents this, though it also prevents setting volume before the
player starts.
Before (with the PipeWire output):
[i] ~$ mpc clear
volume: 81% repeat: off random: off single: off consume: off
[i] ~$ systemctl --user restart mpd.service
[i] ~$ mpc volume 100
MPD error: Connection closed by the server
[i] ~ 1 $
After:
[i] ~$ # mpd is freshly started w/o anything in the queue
[i] ~$ mpc
volume:100% repeat: off random: off single: off consume: off
[i] ~$ mpc volume 80
MPD error: problems setting volume
[i] ~ 1 $ mpc
volume:100% repeat: off random: off single: off consume: off
[i] ~$
Previous versions of MPD would, on parameter change, set the PipeWire
volume before clearing the restore_volume flag, causing the call to
short circuit and do nothing. Instead, clear the flag before the call.
Wasapi output plugin won't start playing after being paused
The cause is that the scope guard in the WASAPI work thread
(WasapiOutputPlugin.cxx, function WasapiOutputThread::Work(), in the
while (true) loop) is set up too 'late' in the execution. There is one
condition ("if (data_in_frames >= buffer_size_in_frames)") when it is
hit, the loop will continue without executing the scope guard. This
scope guard is responsible for emptying the buffer again, and if the
buffer is not emptied, the above mentioned condition will stay true.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1451
SonarLint reports the latter to be better:
std::scoped_lock basically provides the same feature as std::lock_guard,
but is more generic: It can lock several mutexes at the same time, with a
deadlock prevention mechanism (see {rule:cpp:S5524}). The equivalent code
to perform simultaneous locking with std::lock_guard is significantly more
complex. Therefore, it is simpler to use std::scoped_lock all the time,
even when locking only one mutex (there will be no performance impact).
Signed-off-by: Rosen Penev <rosenp@gmail.com>
This uncomments the code which had been present already in the first
Snapcast commit (copied from the "httpd" output plugin), but I
commented it because I did not know whether I needed to send silence
samples to all Snapcast clients.
As a side effect, this fixes playback when no Snapcast client is
connected; this was broken because Pause() always returned a positive
value when there were no clients.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1310
The ChainFilter class is extremely complicated code, and will grow to
be even more complicated when the Filter interface gets extended.
Let's just remove it; we can easily chain many TwoFilters instead.
If draining was not requested, generate silence instead if there is no
data in the ring buffer.
The problem is that pw_stream_flush() appears to disable the stream
permanently, even though there is no state_changed callback - the
stream state remains at PW_STREAM_STATE_STREAMING, but the stream is
defunct. I have no idea why and I havn't found any documentation
about it.
Closes https://github.com/MusicPlayerDaemon/MPD/issues/1219
Explanation
This adds support for DOP using the PcmExport function if the macro
ENABLE_DSD is defined. If enabled within the config-file using "dop",
the boolean dop_setting will be true. If DSD input is encountered and
the setting is on, it is checked whether the oss-device supports the
required samplerate. If that is the case, dop_active is set to true
and conversion of the input is prevented. If the sample rate is not
supported, conversion to S32 is requested. When playing back, the
PcmExport is used to pack the incoming stream into PCM. Reasoning
This is required for OSs without the required driver support for
native DSD playback that also have no ALSA. Mainly *BSD users are the
target audience for this functionality, as ALSA here is only a proxy
without full functionality. Requirements
DAC that supports the DOP standard
Building with OSS, DSD and S32-Format
Supported Formats / Required PCM Formats
DSF, DFF and WavPack-DSD will work.
DSD64, 1 Channel -> S24:176.4kHz (untested, lack of time / missing samples)
DSD64, 2 Channel -> S24:352.8kHz
DSD64, 4 Channel -> S24:705.6kHz (untested, lmissing equipment)
DSD128, 1 Channel -> S24:352.8kHz (untested, lack of time / missing samples)
DSD128, 2 Channel -> S24:705.6kHz
DSD256, 1 Channel -> S24:705.6kHz (untested, lack of time / missing samples)
Changes
inclusion of required files
adding new domain for logging
adding dop_satisfied private function
adding required member variables for storing dop state and for dop-packing
adding dop boolean parameter to many functions that are required to act a little differently when dop is active
Testing
This has been tested to work with a Sabaj Da2 on FreeBSD, where the
red status indicator LED clearly shows that DSD playback is taking
place, instead of purple for "hi-res" which is seen when converting.
Issues
I have not tested this with S24 and right now AFMT_S32_NE is
required. If not defined, ENABLE_DSD will be undef'ed. This will be
addressed in a bit, however no DAC which supports DOP but not 32Bit is
known to me. Also, AFMT_S32_NE is not defined when building on FreeBSD
which is why this is just blatantly defined in the file at the moment.
Additionally, the new dop-option is not added into any documentation
whatsoever.
We need this even when AFMT_S24_PACKED is not available, for the
correct channel order in multi-channel files. Internally, MPD uses
FLAC channel order, but OSS uses the same channel order as ALSA.
