The backend holds `_port_callback_mutex` while disconnecting ports.
In some cases disconnecting a port can drop the last reference
resulting in a port-deletion from the connection handler.
This in turn will eventually aquire the `_port_callback_mutex`
and deadlock.
This is now circumvented by using atomic operations instead of
taking a lock to set the `_port_change_flag`.
The flag is also used to trigger a latency update in some cases,
atomic is preferable to taking a lock to set this flag.
--
Full bt: https://paste.debian.net/1184056/
Short:
#1 in pthread_mutex_lock ()
#2 in ARDOUR::PortEngineSharedImpl::port_connect_add_remove_callback()
#3 in ARDOUR::BackendPort::~BackendPort()
#4 in ARDOUR::DummyPort::~DummyPort()
#6 in ARDOUR::DummyAudioPort::~DummyAudioPort()
#7 in boost::checked_delete<ARDOUR::BackendPort>(ARDOUR::BackendPort*)
#12 in boost::shared_ptr<ARDOUR::ProtoPort>::reset()
#13 in ARDOUR::Port::drop()
#14 in ARDOUR::Port::~Port()
#15 in ARDOUR::AudioPort::~AudioPort()
#17 in ARDOUR::AudioEngine::add_pending_port_deletion(ARDOUR::Port*)
#20 in boost::detail::sp_counted_base::release()
#37 in ARDOUR::PortManager::connect_callback() at libs/ardour/port_manager.cc:788
#38 in ARDOUR::DummyAudioBackend::main_process_thread() at libs/backends/dummy/dummy_audiobackend.cc:1018
The warning "samples per period does not match." never triggered.
Previously not being able to set the requested buffersize was a
fatal error.
This adds support for soundcards that only support msec.
e.g. recent HDA Intel via SOF (Sound Open Firmware)
This allow to restore original engine port-names as set
by the backend. ALSA MIDI, CoreAudio, CoreMIDI and PortAudio
drivers can provide human readable physical port names for
some devices.
When exporting long sessions with freewheeling, pulseaudio
may meanwhile suspend the corked audio device. The "FAIL_ON_SUSPEND"
option then prevents ardour to uncork it after export, and the
audio-backend is halted.
This potentially breaks various assumptions (e.g. no resampling,
fixed buffersize) when the stream is moved to a different device.
Then again it's pulseaudio, which is unsuitable for pro-audio to
begin with.
This fixes an issue with some soundcards e.g. "AxeFx III".
Device configuration fails unless set_hwpar() is performed
for the capture device before configuring the playack
device (half duplex is fine, too).
This is mainly for RME RayDAT that has a fixed buffersize of 16k:
dev_name : hw:HDSPMxc2f6c5,0
channels : 36
min_rate : 32000
max_rate : 192000
min_bufz : 16384
max_bufz : 16384
min_nper : 4
max_nper : 512
However nperiod configuration determines the effective latency
regardless.
This is similar to https://github.com/jackaudio/jack1/blob/master/drivers/alsa/alsa_driver.c#L476-L486
