mirror of
https://github.com/Ardour/ardour.git
synced 2026-01-24 06:07:29 +01:00
NOOP, whitespace change only
This commit is contained in:
Robin Gareus
parent
dfeb68aeb4
commit
b3154e8ed6
3 changed files with 90 additions and 92 deletions
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@ -60,7 +60,7 @@ intptr_t Session::vst_callback (
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}
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if (effect && effect->user) {
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plug = (VSTPlugin *) (effect->user);
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plug = (VSTPlugin *) (effect->user);
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session = &plug->session();
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#ifdef COMPILER_MSVC
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SHOW_CALLBACK ("am callback 0x%x, opcode = %d, plugin = \"%s\" ", (int) pthread_self().p, opcode, plug->name());
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@ -155,7 +155,7 @@ intptr_t Session::vst_callback (
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_timeInfo.samplePos = now;
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_timeInfo.sampleRate = session->frame_rate();
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const TempoMetric& tm (session->tempo_map().metric_at (now));
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if (value & (kVstTempoValid)) {
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@ -174,22 +174,21 @@ intptr_t Session::vst_callback (
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try {
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session->tempo_map().bbt_time_rt (now, bbt);
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/* PPQ = pulse per quarter
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VST's "pulse" is our "division".
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8 divisions per bar, 1 division = quarter, so 8 quarters per bar, ppq = 1
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8 divisions per bar, 1 division = eighth, so 4 quarters per bar, ppq = 2
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4 divisions per bar, 1 division = quarter, so 4 quarters per bar, ppq = 1
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4 divisions per bar, 1 division = half, so 8 quarters per bar, ppq = 0.5
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4 divisions per bar, 1 division = fifth, so (4 * 5/4) quarters per bar, ppq = 5/4
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general: divs_per_bar / (note_type / 4.0)
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*/
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/* PPQ = pulse per quarter
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* VST's "pulse" is our "division".
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*
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* 8 divisions per bar, 1 division = quarter, so 8 quarters per bar, ppq = 1
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* 8 divisions per bar, 1 division = eighth, so 4 quarters per bar, ppq = 2
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* 4 divisions per bar, 1 division = quarter, so 4 quarters per bar, ppq = 1
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* 4 divisions per bar, 1 division = half, so 8 quarters per bar, ppq = 0.5
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* 4 divisions per bar, 1 division = fifth, so (4 * 5/4) quarters per bar, ppq = 5/4
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*
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* general: divs_per_bar / (note_type / 4.0)
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*/
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double ppq_scaling = tm.meter().note_divisor() / 4.0;
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/* Note that this assumes constant meter/tempo throughout the session. Stupid VST
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*/
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/* Note that this assumes constant meter/tempo throughout the session. Stupid VST */
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double ppqBar = double(bbt.bars - 1) * tm.meter().divisions_per_bar();
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double ppqBeat = double(bbt.beats - 1);
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double ppqTick = double(bbt.ticks) / Timecode::BBT_Time::ticks_per_beat;
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@ -197,17 +196,17 @@ intptr_t Session::vst_callback (
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ppqBar *= ppq_scaling;
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ppqBeat *= ppq_scaling;
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ppqTick *= ppq_scaling;
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if (value & (kVstPpqPosValid)) {
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_timeInfo.ppqPos = ppqBar + ppqBeat + ppqTick;
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_timeInfo.flags |= (kVstPpqPosValid);
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}
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if (value & (kVstBarsValid)) {
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_timeInfo.barStartPos = ppqBar;
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_timeInfo.flags |= (kVstBarsValid);
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}
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} catch (...) {
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/* relax */
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}
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@ -215,13 +214,13 @@ intptr_t Session::vst_callback (
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if (value & (kVstSmpteValid)) {
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Timecode::Time t;
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session->timecode_time (now, t);
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_timeInfo.smpteOffset = (t.hours * t.rate * 60.0 * 60.0) +
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(t.minutes * t.rate * 60.0) +
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(t.seconds * t.rate) +
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(t.frames) +
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_timeInfo.smpteOffset = (t.hours * t.rate * 60.0 * 60.0) +
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(t.minutes * t.rate * 60.0) +
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(t.seconds * t.rate) +
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(t.frames) +
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(t.subframes);
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_timeInfo.smpteOffset *= 80.0; /* VST spec is 1/80th frames */
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@ -257,7 +256,7 @@ intptr_t Session::vst_callback (
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_timeInfo.samplePos = 0;
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_timeInfo.sampleRate = AudioEngine::instance()->sample_rate();
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}
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return (intptr_t) &_timeInfo;
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case audioMasterProcessEvents:
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@ -286,14 +285,14 @@ intptr_t Session::vst_callback (
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case audioMasterGetParameterQuantization:
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SHOW_CALLBACK ("amc: audioMasterGetParameterQuantization\n");
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// returns the integer value for +1.0 representation,
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// or 1 if full single float precision is maintained
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// in automation. parameter index in <value> (-1: all, any)
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// returns the integer value for +1.0 representation,
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// or 1 if full single float precision is maintained
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// in automation. parameter index in <value> (-1: all, any)
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return 0;
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case audioMasterIOChanged:
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SHOW_CALLBACK ("amc: audioMasterIOChanged\n");
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// numInputs and/or numOutputs has changed
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// numInputs and/or numOutputs has changed
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return 0;
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case audioMasterNeedIdle:
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@ -333,16 +332,16 @@ intptr_t Session::vst_callback (
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case audioMasterGetPreviousPlug:
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SHOW_CALLBACK ("amc: audioMasterGetPreviousPlug\n");
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// input pin in <value> (-1: first to come), returns cEffect*
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// input pin in <value> (-1: first to come), returns cEffect*
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return 0;
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case audioMasterGetNextPlug:
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SHOW_CALLBACK ("amc: audioMasterGetNextPlug\n");
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// output pin in <value> (-1: first to come), returns cEffect*
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// output pin in <value> (-1: first to come), returns cEffect*
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case audioMasterWillReplaceOrAccumulate:
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SHOW_CALLBACK ("amc: audioMasterWillReplaceOrAccumulate\n");
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// returns: 0: not supported, 1: replace, 2: accumulate
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// returns: 0: not supported, 1: replace, 2: accumulate
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return 0;
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case audioMasterGetCurrentProcessLevel:
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@ -364,10 +363,10 @@ intptr_t Session::vst_callback (
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case audioMasterOfflineStart:
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SHOW_CALLBACK ("amc: audioMasterOfflineStart\n");
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return 0;
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case audioMasterOfflineRead:
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SHOW_CALLBACK ("amc: audioMasterOfflineRead\n");
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// ptr points to offline structure, see below. return 0: error, 1 ok
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// ptr points to offline structure, see below. return 0: error, 1 ok
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return 0;
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case audioMasterOfflineWrite:
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@ -378,7 +377,7 @@ intptr_t Session::vst_callback (
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case audioMasterOfflineGetCurrentPass:
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SHOW_CALLBACK ("amc: audioMasterOfflineGetCurrentPass\n");
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return 0;
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case audioMasterOfflineGetCurrentMetaPass:
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SHOW_CALLBACK ("amc: audioMasterOfflineGetCurrentMetaPass\n");
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return 0;
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@ -46,12 +46,12 @@ VSTPlugin::VSTPlugin (AudioEngine& engine, Session& session, VSTHandle* handle)
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, _state (0)
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, _plugin (0)
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{
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}
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VSTPlugin::~VSTPlugin ()
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{
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}
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void
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@ -78,7 +78,7 @@ VSTPlugin::activate ()
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_plugin->dispatcher (_plugin, effMainsChanged, 0, 1, NULL, 0.0f);
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}
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int
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int
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VSTPlugin::set_block_size (pframes_t nframes)
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{
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deactivate ();
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@ -93,13 +93,13 @@ VSTPlugin::default_value (uint32_t)
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return 0;
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}
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float
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float
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VSTPlugin::get_parameter (uint32_t which) const
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{
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return _plugin->getParameter (_plugin, which);
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}
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void
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void
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VSTPlugin::set_parameter (uint32_t which, float newval)
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{
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float oldval = get_parameter (which);
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@ -109,7 +109,7 @@ VSTPlugin::set_parameter (uint32_t which, float newval)
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}
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_plugin->setParameter (_plugin, which, newval);
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float curval = get_parameter (which);
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if (!PBD::floateq (curval, oldval, 1)) {
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@ -215,8 +215,8 @@ VSTPlugin::set_state (const XMLNode& node, int version)
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for (n = child->children ().begin (); n != child->children ().end (); ++n) {
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if ((*n)->is_content ()) {
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/* XXX: this may be dubious for the same reasons that we delay
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execution of load_preset.
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*/
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execution of load_preset.
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*/
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ret = set_chunk ((*n)->content().c_str(), false);
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}
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}
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@ -301,7 +301,7 @@ VSTPlugin::get_parameter_descriptor (uint32_t which, ParameterDescriptor& desc)
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memset (label, 0, sizeof (label));
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_plugin->dispatcher (_plugin, effGetParamName, which, 0, label, 0);
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desc.label = label;
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desc.integer_step = false;
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desc.lower = 0.0f;
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@ -335,7 +335,7 @@ VSTPlugin::load_preset (PresetRecord r)
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return s;
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}
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bool
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bool
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VSTPlugin::load_plugin_preset (PresetRecord r)
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{
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/* This is a plugin-provided preset.
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@ -348,7 +348,7 @@ VSTPlugin::load_plugin_preset (PresetRecord r)
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#ifndef NDEBUG
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int const p = sscanf (r.uri.c_str(), "VST:%d:%d", &id, &index);
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assert (p == 2);
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#else
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#else
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sscanf (r.uri.c_str(), "VST:%d:%d", &id, &index);
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#endif
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@ -372,13 +372,12 @@ VSTPlugin::load_plugin_preset (PresetRecord r)
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}
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#else
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_state->want_program = index;
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#endif
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return true;
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}
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bool
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bool
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VSTPlugin::load_user_preset (PresetRecord r)
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{
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/* This is a user preset; we load it, and this code also knows about the
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@ -424,7 +423,7 @@ VSTPlugin::load_user_preset (PresetRecord r)
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return false;
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} else {
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for (XMLNodeList::const_iterator j = (*i)->children().begin(); j != (*i)->children().end(); ++j) {
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if ((*j)->name() == X_("Parameter")) {
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XMLProperty* index = (*j)->property (X_("index"));
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@ -442,7 +441,7 @@ VSTPlugin::load_user_preset (PresetRecord r)
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return false;
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}
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string
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string
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VSTPlugin::do_save_preset (string name)
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{
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boost::shared_ptr<XMLTree> t (presets_tree ());
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@ -488,7 +487,7 @@ VSTPlugin::do_save_preset (string name)
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return uri;
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}
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void
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void
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VSTPlugin::do_remove_preset (string name)
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{
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boost::shared_ptr<XMLTree> t (presets_tree ());
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@ -504,7 +503,7 @@ VSTPlugin::do_remove_preset (string name)
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t->write (f);
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}
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string
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string
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VSTPlugin::describe_parameter (Evoral::Parameter param)
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{
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char name[64];
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@ -521,7 +520,7 @@ VSTPlugin::describe_parameter (Evoral::Parameter param)
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return name;
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}
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framecnt_t
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framecnt_t
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VSTPlugin::signal_latency () const
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{
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if (_user_latency) {
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@ -531,7 +530,7 @@ VSTPlugin::signal_latency () const
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return *((int32_t *) (((char *) &_plugin->flags) + 12)); /* initialDelay */
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}
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set<Evoral::Parameter>
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set<Evoral::Parameter>
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VSTPlugin::automatable () const
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{
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set<Evoral::Parameter> ret;
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@ -545,8 +544,8 @@ VSTPlugin::automatable () const
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int
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VSTPlugin::connect_and_run (BufferSet& bufs,
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ChanMapping in_map, ChanMapping out_map,
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pframes_t nframes, framecnt_t offset)
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ChanMapping in_map, ChanMapping out_map,
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pframes_t nframes, framecnt_t offset)
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{
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Plugin::connect_and_run (bufs, in_map, out_map, nframes, offset);
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@ -557,14 +556,14 @@ VSTPlugin::connect_and_run (BufferSet& bufs,
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BufferSet& silent_bufs = _session.get_silent_buffers(bufs_count);
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BufferSet& scratch_bufs = _session.get_scratch_buffers(bufs_count);
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/* VC++ doesn't support the C99 extension that allows
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/* VC++ doesn't support the C99 extension that allows
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typeName foo[variableDefiningSize];
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Use alloca instead of dynamic array (rather than std::vector which
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allocs on the heap) because this is realtime code.
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*/
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float** ins = (float**)alloca(_plugin->numInputs*sizeof(float*));
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float** outs = (float**)alloca(_plugin->numInputs*sizeof(float*));
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@ -601,48 +600,48 @@ VSTPlugin::connect_and_run (BufferSet& bufs,
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return 0;
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}
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string
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string
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VSTPlugin::unique_id () const
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{
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char buf[32];
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snprintf (buf, sizeof (buf), "%d", _plugin->uniqueID);
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return string (buf);
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}
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const char *
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const char *
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VSTPlugin::name () const
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{
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return _handle->name;
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}
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const char *
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const char *
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VSTPlugin::maker () const
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{
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return _info->creator.c_str();
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}
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const char *
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const char *
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VSTPlugin::label () const
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{
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return _handle->name;
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}
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uint32_t
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uint32_t
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VSTPlugin::parameter_count () const
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{
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return _plugin->numParams;
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}
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bool
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bool
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VSTPlugin::has_editor () const
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{
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return _plugin->flags & effFlagsHasEditor;
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}
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void
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void
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VSTPlugin::print_parameter (uint32_t param, char *buf, uint32_t /*len*/) const
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{
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char *first_nonws;
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@ -1,10 +1,10 @@
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/*
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* VST instrument support
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*
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* Derived from code that was marked:
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* Derived from code that was marked:
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* Copyright (C) Kjetil S. Matheussen 2004 (k.s.matheussen@notam02.no)
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* Alsa-seq midi-code made by looking at the jack-rack source made by Bob Ham.
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*
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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@ -41,31 +41,31 @@ create_sequencer (const char* client_name, bool isinput)
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{
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snd_seq_t * seq;
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int err;
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if ((err = snd_seq_open (&seq, "default", SND_SEQ_OPEN_DUPLEX, 0)) != 0) {
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fst_error ("Could not open ALSA sequencer, aborting\n\n%s\n\n"
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"Make sure you have configure ALSA properly and that\n"
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"/proc/asound/seq/clients exists and contains relevant\n"
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"devices (%s).",
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snd_strerror (err));
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"Make sure you have configure ALSA properly and that\n"
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"/proc/asound/seq/clients exists and contains relevant\n"
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"devices (%s).",
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snd_strerror (err));
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return NULL;
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}
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snd_seq_set_client_name (seq, client_name);
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if ((err = snd_seq_create_simple_port (seq, isinput? "Input" : "Output",
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(isinput? SND_SEQ_PORT_CAP_WRITE: SND_SEQ_PORT_CAP_READ)| SND_SEQ_PORT_CAP_DUPLEX |
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SND_SEQ_PORT_CAP_SUBS_READ|SND_SEQ_PORT_CAP_SUBS_WRITE,
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SND_SEQ_PORT_TYPE_APPLICATION|SND_SEQ_PORT_TYPE_SPECIFIC)) != 0) {
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(isinput? SND_SEQ_PORT_CAP_WRITE: SND_SEQ_PORT_CAP_READ)| SND_SEQ_PORT_CAP_DUPLEX |
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SND_SEQ_PORT_CAP_SUBS_READ|SND_SEQ_PORT_CAP_SUBS_WRITE,
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SND_SEQ_PORT_TYPE_APPLICATION|SND_SEQ_PORT_TYPE_SPECIFIC)) != 0) {
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fst_error ("Could not create ALSA port: %s", snd_strerror (err));
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snd_seq_close(seq);
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return NULL;
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}
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return seq;
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}
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static void
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static void
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queue_midi (JackVST *jvst, int val1, int val2, int val3)
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{
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||||
VstMidiEvent *pevent;
|
||||
|
|
@ -77,11 +77,11 @@ queue_midi (JackVST *jvst, int val1, int val2, int val3)
|
|||
fst_error ("event queue has no write space");
|
||||
return;
|
||||
}
|
||||
|
||||
|
||||
pevent = (VstMidiEvent *) vec[0].buf;
|
||||
|
||||
// printf("note: %d\n",note);
|
||||
|
||||
|
||||
pevent->type = kVstMidiType;
|
||||
pevent->byteSize = 24;
|
||||
pevent->deltaFrames = 0;
|
||||
|
|
@ -96,7 +96,7 @@ queue_midi (JackVST *jvst, int val1, int val2, int val3)
|
|||
pevent->midiData[1] = val2;
|
||||
pevent->midiData[2] = val3;
|
||||
pevent->midiData[3] = 0;
|
||||
|
||||
|
||||
//printf("Sending: %x %x %x\n",val1,val2,val3);
|
||||
|
||||
jack_ringbuffer_write_advance (jvst->event_queue, sizeof (VstMidiEvent));
|
||||
|
|
@ -113,8 +113,8 @@ void *midireceiver(void *arg)
|
|||
|
||||
// Try to set fifo priority...
|
||||
// this works, if we are root or newe sched-cap manegment is used...
|
||||
pthread_setschedparam( pthread_self(), SCHED_FIFO, &scp );
|
||||
|
||||
pthread_setschedparam( pthread_self(), SCHED_FIFO, &scp );
|
||||
|
||||
while (1) {
|
||||
|
||||
snd_seq_event_input (jvst->seq, &event);
|
||||
|
|
@ -158,28 +158,28 @@ void *midireceiver(void *arg)
|
|||
break;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
return NULL;
|
||||
}
|
||||
|
||||
void stop_midireceiver (JackVST *jvst)
|
||||
{
|
||||
int err;
|
||||
int err;
|
||||
snd_seq_event_t event;
|
||||
snd_seq_t *seq2 = create_sequencer ("jfstquit", true);
|
||||
|
||||
|
||||
jvst->midiquit = 1;
|
||||
|
||||
|
||||
snd_seq_connect_to (seq2, 0, snd_seq_client_id (jvst->seq),0);
|
||||
snd_seq_ev_clear (&event);
|
||||
snd_seq_ev_set_direct (&event);
|
||||
snd_seq_ev_set_subs (&event);
|
||||
snd_seq_ev_set_source (&event, 0);
|
||||
snd_seq_ev_set_controller (&event,1,0x80,50);
|
||||
|
||||
|
||||
if ((err = snd_seq_event_output (seq2, &event)) < 0) {
|
||||
fst_error ("cannot send stop event to midi thread: %s\n",
|
||||
snd_strerror (err));
|
||||
snd_strerror (err));
|
||||
}
|
||||
|
||||
snd_seq_drain_output (seq2);
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue