diff options
author | David Robillard <d@drobilla.net> | 2009-05-07 06:30:50 +0000 |
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committer | David Robillard <d@drobilla.net> | 2009-05-07 06:30:50 +0000 |
commit | 7183242b8c8d9296f94a035fb66b1eae06fd3496 (patch) | |
tree | ee48e2019f2330baee7c280090a48e43336682b3 /libs/ardour/amp.cc | |
parent | 97b5eb1580d53197dc93a805d1995b82660cdfe3 (diff) |
The great audio processing overhaul.
The vast majority of Route signal processing is now simply in the list of
processors. There are definitely regressions here, but there's also
a lot of things fixed. It's far too much work to let diverge anymore
regardless, so here it is.
The basic model is: A route has a fixed set of input channels (matching
its JACK input ports and diskstream). The first processor takes this
as input. The next processor is configured using the first processor's
output as input, and is allowed to choose whatever output it wants
given that input... and so on, and so on. Finally, the last processor's
requested output is used to set up the panner and create whatever Jack
ports are needed to output the data.
All 'special' internal processors (meter, fader, amp, insert, send) are
currently transparent: they read any input, and return the same set
of channels back (unmodified, except for amp).
User visible changes:
* LV2 Instrument support (tracks with both MIDI and audio channels)
* MIDI in/out plugin support
* Generic plugin replication (for MIDI plugins, MIDI/audio plugins)
* Movable meter point
Known Bugs:
* Things seem to get weird on loaded sessions
* Output delivery is sketchy
* 2.0 session loading was probably already broken...
but it's definitely broken now :)
Please test this and file bugs if you have any time...
git-svn-id: svn://localhost/ardour2/branches/3.0@5055 d708f5d6-7413-0410-9779-e7cbd77b26cf
Diffstat (limited to 'libs/ardour/amp.cc')
-rw-r--r-- | libs/ardour/amp.cc | 125 |
1 files changed, 119 insertions, 6 deletions
diff --git a/libs/ardour/amp.cc b/libs/ardour/amp.cc index 263ce82086..aa20f3e389 100644 --- a/libs/ardour/amp.cc +++ b/libs/ardour/amp.cc @@ -20,23 +20,123 @@ #include <cmath> #include <algorithm> #include "ardour/amp.h" -#include "ardour/buffer_set.h" #include "ardour/audio_buffer.h" +#include "ardour/buffer_set.h" +#include "ardour/configuration.h" +#include "ardour/io.h" +#include "ardour/session.h" namespace ARDOUR { +Amp::Amp(Session& s, IO& io) + : Processor(s, "Amp") + , _io(io) + , _mute(false) + , _apply_gain(true) + , _apply_gain_automation(false) + , _current_gain(1.0) + , _desired_gain(1.0) +{ +} + +bool +Amp::can_support_io_configuration (const ChanCount& in, ChanCount& out) const +{ + out = in; + return true; +} + +bool +Amp::configure_io (ChanCount in, ChanCount out) +{ + if (out != in) { // always 1:1 + return false; + } + + return Processor::configure_io (in, out); +} + +void +Amp::run_in_place (BufferSet& bufs, nframes_t start_frame, nframes_t end_frame, nframes_t nframes) +{ + gain_t* gab = _session.gain_automation_buffer(); + + if (_mute && !bufs.is_silent()) { + Amp::apply_gain (bufs, nframes, _current_mute_gain, _desired_mute_gain, false); + if (_desired_mute_gain == 0.0f) { + bufs.is_silent(true); + } + } + + if (_apply_gain) { + + if (_apply_gain_automation) { + + if (_io.phase_invert()) { + for (BufferSet::audio_iterator i = bufs.audio_begin(); i != bufs.audio_end(); ++i) { + Sample* const sp = i->data(); + for (nframes_t nx = 0; nx < nframes; ++nx) { + sp[nx] *= -gab[nx]; + } + } + } else { + for (BufferSet::audio_iterator i = bufs.audio_begin(); i != bufs.audio_end(); ++i) { + Sample* const sp = i->data(); + for (nframes_t nx = 0; nx < nframes; ++nx) { + sp[nx] *= gab[nx]; + } + } + } + + } else { /* manual (scalar) gain */ + + if (_current_gain != _desired_gain) { + + Amp::apply_gain (bufs, nframes, _current_gain, _desired_gain, _io.phase_invert()); + _current_gain = _desired_gain; + + } else if (_current_gain != 0.0f && (_io.phase_invert() || _current_gain != 1.0f)) { + + /* no need to interpolate current gain value, + but its non-unity, so apply it. if the gain + is zero, do nothing because we'll ship silence + below. + */ + + gain_t this_gain; + + if (_io.phase_invert()) { + this_gain = -_current_gain; + } else { + this_gain = _current_gain; + } + + for (BufferSet::audio_iterator i = bufs.audio_begin(); i != bufs.audio_end(); ++i) { + Sample* const sp = i->data(); + apply_gain_to_buffer(sp, nframes, this_gain); + } + + } else if (_current_gain == 0.0f) { + for (BufferSet::audio_iterator i = bufs.audio_begin(); i != bufs.audio_end(); ++i) { + i->clear(); + } + } + } + } +} /** Apply a declicked gain to the audio buffers of @a bufs */ void -Amp::run_in_place (BufferSet& bufs, nframes_t nframes, gain_t initial, gain_t target, bool invert_polarity) +Amp::apply_gain (BufferSet& bufs, nframes_t nframes, + gain_t initial, gain_t target, bool invert_polarity) { - if (nframes == 0) + if (nframes == 0) { return; + } - if (bufs.count().n_audio() == 0) + if (bufs.count().n_audio() == 0) { return; - - // assert(bufs.buffer_capacity(DataType::AUDIO) >= nframes); + } // if we don't need to declick, defer to apply_simple_gain if (initial == target) { @@ -98,5 +198,18 @@ Amp::apply_simple_gain (BufferSet& bufs, nframes_t nframes, gain_t target) { } +XMLNode& +Amp::state (bool full_state) +{ + return get_state(); +} + +XMLNode& +Amp::get_state() +{ + XMLNode* node = new XMLNode(state_node_name); + node->add_property("type", "amp"); + return *node; +} } // namespace ARDOUR |