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Session::process() returns early with Session::_silent set to true. AudioBuffer::set_data()
was never set for (at least) the LTC output port. PortManager::cycle_end() calls
AudioBuffer::silence() which used to assume that get_buffer() must have been called. But it
was not, because that should have happened in Session::process().
So check AudioBuffer::data() and call get_buffer() if required.
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not try to use its own _solo_ignore to track Muteable::can_solo() or solo isolate state
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To export a MIDI region to a file, the code used MidiRegion::clone()
since it takes care of creating a new file-backed source with the wanted
contents. Nevertheless, it had several side-effects:
- it created and registered a new region which is confusing to users
- it only exported notes that were in the region range, but didn't
remove the region start offset from MIDI events, essentially producing
a spurious silence at the beginning of the exported file (this is not
a problem for region cloning because the newly created region is made
aware of the offset and caters for it).
Add a dedicated code path for export, that uses the new offsetting
capabilities of MidiModel::write_section_to().
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MidiModel::write_section_to() only wrote events to the given source if
those events had a time in the given range. Make it able to optionally
offset event times so that the start of the written range corresponds to
time 0 in the source.
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..since memory-size and segmentation increased.
(large sizes favor TLSF, smaller segments are in favor realloc-pool)
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post-processing is no longer just Normalization. RealtimeExport
does Encoding - faster than realtime - using the same infrastructure.
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Don't query after every cycle, some plugins inject license checks
when a host queries latency (!)
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This allows a TmpFile pointer to be either a Sync or Async (Threaded)
writer. As result we must be able to handle both RT and non RT processing.
Still, post-processing (normalization and encoding) should always
happen faster than realtime (freewheeling).
Since jack does not allow a client to change to freewheeling from within
the process-callback, the async-writer disk-thread FileFlushed is used
to initiate post-processing.
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When flushing the buffers of Delivery processors owned by a Route/Track,
inner deliveries of PortInsert processors were missed since PortInsert
is not a Delivery subclass, but rather owns a Delivery as a private
member. Expose a flush_buffers() for PortInsert and call it too.
This is correct since (external) Send is a Delivery subclass, so that
just makes the send part of inserts behave as external sends do.
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Route::no_roll(), Route::roll(), Track::no_roll(), AudioTrack::roll()
and MidiTrack::roll() all had the exact same loop for flushing buffers
of their Delivery processors. That was a lot of replicated code that had
to be kept synchronised by hand. Put that code into a protected method
Route::flush_processor_buffers_locked() which is called instead.
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* add a threaded TmpFile Writer
* update API calls to that process_export_fw() can be used as
process_function
The idea is to re-use export infrastructure from normalization:
export to a tmp-file and then encode target formats after that.
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This deprecates Evoral::midi_note_name(). we don't maintain i18n
for libevoral.
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changes do not move it.
The user may drag the marker, edit in the Location UI, or use nudge, to set the end
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Ardour takes the closest pin/port match yet still offers a plugin with
variable i/o access to all port-buffers.
the "reported" (user visible) pin-count now defaults to the actual
selected configuration (precise) and lua_dsp_configure() can
override it.
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this simplifies lua-bindings and also let's the compiler worry about
constant primitive types.
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- should be a no-op.
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rounding.
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- review / comment would be appreciated here.
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tempo dilation
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ctor.
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- for those not in the know, this series provides a way to
remove the temporal distortion introduced when using an
audio frame-based gui for music-locked objects.
In short, the gui uses an audio frame representation to move
objects. It displays the object using frame_at_beat(), quantizing
the time value to audio frames. This is fine until the user selects
that frame but expects it to be interpreted as a beat.
Thus beat_at_frame() would not produce the user-expected beat
(temporal quantization error of up to 0.5 audio samples).
This is one method of mapping audio time to music time accurately.
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- use exact beats to determine frame position.
- see comments in tempo.cc for more.
- this hasn't been done for split yet, but dragging and
trimming are supported.
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this fixes an issue with jack1 and jack_latency_recompute() since must not
send a server request from inside the server callback.
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* support all Plugin APIs (not implementation specific)
* also check for latency changes when plugins are hard en/disabled
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Required for state save/restore to know about the master
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Allows for more robust out-of-band (negative) value
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Newly added routes,removed routes etc. all correctly assign or
unassign to the group's VCA.
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