static GstFlowReturn
gst_lv2_filter_transform_data (GstLV2Filter * self,
GstMapInfo * in_map, GstMapInfo * out_map)
{
GstAudioFilterClass *audiofilter_class;
GstLV2FilterClass *lv2_class;
GstLV2FilterGroup *lv2_group;
GstLV2FilterPort *lv2_port;
guint j, nframes, samples, out_samples;
gfloat *in = NULL, *out = NULL;
nframes = in_map->size / sizeof (float);
audiofilter_class = GST_AUDIO_FILTER_GET_CLASS (self);
lv2_class = (GstLV2FilterClass *) audiofilter_class;
samples = nframes / lv2_class->in_group.ports->len;
/* multi channel inputs */
lv2_group = &lv2_class->in_group;
in = g_new0 (gfloat, nframes);
if (lv2_group->ports->len > 1)
gst_lv2_filter_deinterleave_data (lv2_group->ports->len, in,
samples, (gfloat *) in_map->data);
for (j = 0; j < lv2_group->ports->len; ++j) {
lv2_port = &g_array_index (lv2_group->ports, GstLV2FilterPort, j);
lilv_instance_connect_port (self->instance, lv2_port->index,
in + (j * samples));
}
lv2_group = &lv2_class->out_group;
out_samples = nframes / lv2_group->ports->len;
out = g_new0 (gfloat, samples * lv2_group->ports->len);
for (j = 0; j < lv2_group->ports->len; ++j) {
lv2_port = &g_array_index (lv2_group->ports, GstLV2FilterPort, j);
lilv_instance_connect_port (self->instance, lv2_port->index,
out + (j * out_samples));
}
lilv_instance_run (self->instance, samples);
if (lv2_group->ports->len > 1)
gst_lv2_filter_interleave_data (lv2_group->ports->len,
(gfloat *) out_map->data, out_samples, out);
g_free (out);
g_free (in);
return GST_FLOW_OK;
}
void
LV2Module::run (uint32 nframes)
{
if (worker)
worker->processWorkResponses();
lilv_instance_run (instance, nframes);
if (worker)
worker->endRun();
}
void EFFECT_LV2::process(void)
{
for(unsigned long m = 0; m < plugins_rep.size(); m++)
lilv_instance_run(plugins_rep[m]->me, buffer_repp->length_in_samples());
}
bool LV2Effect::ProcessStereo(int count,
WaveTrack *left,
WaveTrack *right,
sampleCount lstart,
sampleCount rstart,
sampleCount len)
{
/* Allocate buffers */
if (mBlockSize == 0)
{
mBlockSize = left->GetMaxBlockSize() * 2;
fInBuffer = new float *[mAudioInputs.GetCount()];
for (size_t i = 0; i < mAudioInputs.GetCount(); i++)
{
fInBuffer[i] = new float[mBlockSize];
}
fOutBuffer = new float *[mAudioOutputs.GetCount()];
for (size_t i = 0; i < mAudioOutputs.GetCount(); i++)
{
fOutBuffer[i] = new float[mBlockSize];
}
}
/* Instantiate the plugin */
LilvInstance *handle = lilv_plugin_instantiate(mData,
left->GetRate(),
gLV2Features);
if (!handle)
{
wxMessageBox(wxString::Format(_("Unable to load plug-in %s"), pluginName.c_str()));
return false;
}
/* Write the Note On to the MIDI event buffer and connect it */
LV2_Event_Buffer *midiBuffer = NULL;
int noteOffTime;
if (mMidiInput)
{
midiBuffer = lv2_event_buffer_new(40, 2);
LV2_Event_Iterator iter;
lv2_event_begin(&iter, midiBuffer);
uint8_t noteOn[] = { 0x90, mNoteKey, mNoteVelocity };
lv2_event_write(&iter, 0, 0, 1, 3, noteOn);
noteOffTime = mNoteLength * left->GetRate();
if (noteOffTime < len && noteOffTime < mBlockSize) {
uint8_t noteOff[] = { 0x80, mNoteKey, 64 };
lv2_event_write(&iter, noteOffTime, 0, 1, 3, noteOff);
}
lilv_instance_connect_port(handle, mMidiInput->mIndex, midiBuffer);
}
for (size_t p = 0; p < mAudioInputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mAudioInputs[p].mIndex, fInBuffer[p]);
}
for (size_t p = 0; p < mAudioOutputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mAudioOutputs[p].mIndex, fOutBuffer[p]);
}
for (size_t p = 0; p < mControlInputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mControlInputs[p].mIndex,
&mControlInputs[p].mControlBuffer);
}
for (size_t p = 0; p < mControlOutputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mControlOutputs[p].mIndex,
&mControlOutputs[p].mControlBuffer);
}
float latency = 0.0;
if (mLatencyPortIndex >= 0)
{
lilv_instance_connect_port(handle, mLatencyPortIndex, &latency);
}
lilv_instance_activate(handle);
// Actually perform the effect here
sampleCount originalLen = len;
sampleCount ls = lstart;
sampleCount rs = rstart;
sampleCount ols = ls;
sampleCount ors = rs;
bool noteOver = false;
sampleCount delayed = 0;
sampleCount delay = 0;
bool cleared = false;
while (len || delayed)
{
int block = mBlockSize;
if (len)
{
if (block > len)
{
block = len;
}
if (left && mAudioInputs.GetCount() > 0)
{
left->Get((samplePtr)fInBuffer[0], floatSample, ls, block);
}
if (right && mAudioInputs.GetCount() > 1)
{
right->Get((samplePtr)fInBuffer[1], floatSample, rs, block);
}
}
else if (delayed)
{
// At the end if we don't have enough left for a whole block
if (block > delayed)
{
block = delayed;
}
// Clear the input buffer so that we only pass zeros to the effect.
if (!cleared)
{
for (int i = 0; i < mBlockSize; i++)
{
fInBuffer[0][i] = 0.0;
}
if (right)
{
memcpy(fInBuffer[1], fOutBuffer[0], mBlockSize);
}
cleared = true;
}
}
lilv_instance_run(handle, block);
if (delayed == 0 && latency != 0)
{
delayed = delay = latency;
}
if (delay >= block)
{
delay -= block;
}
else if (delay > 0)
{
sampleCount oblock = block - delay;
if (left && mAudioOutputs.GetCount() > 0)
{
left->Set((samplePtr)(fOutBuffer[0] + delay), floatSample, ols, oblock);
}
if (right && mAudioOutputs.GetCount() > 1)
{
right->Set((samplePtr)(fOutBuffer[1] + delay), floatSample, ors, oblock);
}
ols += oblock;
ors += oblock;
delay = 0;
}
else
{
if (left && mAudioOutputs.GetCount() > 0)
{
left->Set((samplePtr)fOutBuffer[0], floatSample, ols, block);
}
if (right && mAudioOutputs.GetCount() > 1)
{
right->Set((samplePtr)fOutBuffer[1], floatSample, ors, block);
}
ols += block;
ors += block;
}
if (len)
{
len -= block;
noteOffTime -= block;
// Clear the event buffer and add the note off event if needed
if (mMidiInput)
{
lv2_event_buffer_reset(midiBuffer, 1,
(uint8_t *)midiBuffer +
sizeof(LV2_Event_Buffer));
if (!noteOver && noteOffTime < len && noteOffTime < block)
{
LV2_Event_Iterator iter;
lv2_event_begin(&iter, midiBuffer);
uint8_t noteOff[] = { 0x80, mNoteKey, 64 };
lv2_event_write(&iter, noteOffTime, 0, 1, 3, noteOff);
noteOver = true;
}
}
}
else if (delayed)
{
delayed -= block;
}
ls += block;
rs += block;
if (mAudioInputs.GetCount() > 1)
{
if (TrackGroupProgress(count, (ls-lstart)/(double)originalLen))
{
return false;
}
}
else
{
if (TrackProgress(count, (ls-lstart)/(double)originalLen))
{
return false;
}
}
}
lilv_instance_deactivate(handle);
lilv_instance_free(handle);
return true;
}
static GstFlowReturn
gst_lv2_source_fill (GstBaseSrc * base, guint64 offset,
guint length, GstBuffer * buffer)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstLV2SourceClass *klass = (GstLV2SourceClass *) GST_BASE_SRC_GET_CLASS (lv2);
GstLV2Class *lv2_class = &klass->lv2;
GstLV2Group *lv2_group;
GstLV2Port *lv2_port;
GstClockTime next_time;
gint64 next_sample, next_byte;
guint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
guint j, k, l;
gfloat *out = NULL, *cv = NULL, *mem;
gfloat val;
/* example for tagging generated data */
if (!lv2->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "lv2 wave", NULL);
eclass = GST_ELEMENT_CLASS (parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
lv2->tags_pushed = TRUE;
}
if (lv2->eos_reached) {
GST_INFO_OBJECT (lv2, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
bpf = GST_AUDIO_INFO_BPF (&lv2->info);
/* if no length was given, use our default length in samples otherwise convert
* the length in bytes to samples. */
if (length == -1)
samples = lv2->samples_per_buffer;
else
samples = length / bpf;
/* if no offset was given, use our next logical byte */
if (offset == -1)
offset = lv2->next_byte;
/* now see if we are at the byteoffset we think we are */
if (offset != lv2->next_byte) {
GST_DEBUG_OBJECT (lv2, "seek to new offset %" G_GUINT64_FORMAT, offset);
/* we have a discont in the expected sample offset, do a 'seek' */
lv2->next_sample = offset / bpf;
lv2->next_time =
gst_util_uint64_scale_int (lv2->next_sample, GST_SECOND, samplerate);
lv2->next_byte = offset;
}
/* check for eos */
if (lv2->check_seek_stop &&
(lv2->sample_stop > lv2->next_sample) &&
(lv2->sample_stop < lv2->next_sample + samples)
) {
/* calculate only partial buffer */
lv2->generate_samples_per_buffer = lv2->sample_stop - lv2->next_sample;
next_sample = lv2->sample_stop;
lv2->eos_reached = TRUE;
GST_INFO_OBJECT (lv2, "eos reached");
} else {
/* calculate full buffer */
lv2->generate_samples_per_buffer = samples;
next_sample = lv2->next_sample + (lv2->reverse ? (-samples) : samples);
}
bytes = lv2->generate_samples_per_buffer * bpf;
next_byte = lv2->next_byte + (lv2->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (lv2, "samplerate %d", samplerate);
GST_LOG_OBJECT (lv2,
"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = lv2->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!lv2->reverse) {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + lv2->next_time;
GST_BUFFER_DURATION (buffer) = next_time - lv2->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = lv2->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (lv2), GST_BUFFER_TIMESTAMP (buffer));
lv2->next_time = next_time;
lv2->next_sample = next_sample;
lv2->next_byte = next_byte;
GST_LOG_OBJECT (lv2, "generating %u samples at ts %" GST_TIME_FORMAT,
samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
/* multi channel outputs */
lv2_group = &lv2_class->out_group;
if (lv2_group->ports->len > 1) {
out = g_new0 (gfloat, samples * lv2_group->ports->len);
for (j = 0; j < lv2_group->ports->len; ++j) {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, j);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
out + (j * samples));
GST_LOG_OBJECT (lv2, "connected port %d/%d", j, lv2_group->ports->len);
}
} else {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, 0);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
(gfloat *) map.data);
GST_LOG_OBJECT (lv2, "connected port 0");
}
/* cv ports */
cv = g_new (gfloat, samples * lv2_class->num_cv_in);
for (j = k = 0; j < lv2_class->control_in_ports->len; j++) {
lv2_port = &g_array_index (lv2_class->control_in_ports, GstLV2Port, j);
if (lv2_port->type != GST_LV2_PORT_CV)
continue;
mem = cv + (k * samples);
val = lv2->lv2.ports.control.in[j];
/* FIXME: use gst_control_binding_get_value_array */
for (l = 0; l < samples; l++)
mem[l] = val;
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index, mem);
k++;
}
lilv_instance_run (lv2->lv2.instance, samples);
if (lv2_group->ports->len > 1) {
gst_lv2_source_interleave_data (lv2_group->ports->len,
(gfloat *) map.data, samples, out);
g_free (out);
}
g_free (cv);
gst_buffer_unmap (buffer, &map);
return GST_FLOW_OK;
}
void Lv2Plugin::doProcess(bool rolling, jack_position_t &pos, jack_nframes_t nframes, jack_nframes_t time) {
// pull in new control mappings
Lv2ControlMapping *freshMapping;
while(newControlMappingsQueue.pop(freshMapping)) {
freshMapping->getConnection()->addMapping(freshMapping);
}
// process MIDI inputs
Lv2MidiInput *midiInput = midiInputList.getFirst();
while(midiInput) {
midiInput->process(rolling, pos, nframes, time);
midiInput = midiInputList.getNext(midiInput);
}
// process and connect audio inputs
for(uint32_t i = 0; i < audioInputCount; i++) {
AudioConnection *connection = audioInput[i].getConnection();
if(connection) {
connection->getSource()->process(rolling, pos, nframes, time);
lilv_instance_connect_port(instance, audioInputIndex[i], connection->getAudio());
} else {
lilv_instance_connect_port(instance, audioInputIndex[i], AudioConnection::getDummyBuffer());
}
}
// update control port values from scheduled control events
Lv2ControlEvent* evt = controlBuffer.getNextEvent(rolling, pos, nframes);
while(evt) {
evt->getPort()->value = evt->getValue();
ObjectCollector::scriptCollector().recycle(evt);
evt = controlBuffer.getNextEvent(rolling, pos, nframes);
}
// process control connections
Lv2ControlConnection *connection = this->controlConnections.getFirst();
while(connection) {
connection->process(rolling, pos, nframes, time);
connection = controlConnections.getNext(connection);
}
// clear event output buffers
Lv2MidiOutput *midiOutput = midiOutputList.getFirst();
while(midiOutput) {
midiOutput->clear();
midiOutput = midiOutputList.getNext(midiOutput);
}
// set up audio output buffers
for(uint32_t i = 0; i < audioOutputCount; i++) {
lilv_instance_connect_port(instance, audioOutputIndex[i], audioOutput[i]->getAudio());
}
// run the plugin
lilv_instance_run(instance, nframes);
// fire MIDI events
fireMidiEvents(pos);
// emit worker responses
if(worker) {
worker->respond();
}
}
static double
bench(const LilvPlugin* p, uint32_t sample_count, uint32_t block_size)
{
URITable uri_table;
uri_table_init(&uri_table);
LV2_URID_Map map = { &uri_table, uri_table_map };
LV2_Feature map_feature = { LV2_URID_MAP_URI, &map };
LV2_URID_Unmap unmap = { &uri_table, uri_table_unmap };
LV2_Feature unmap_feature = { LV2_URID_UNMAP_URI, &unmap };
const LV2_Feature* features[] = { &map_feature, &unmap_feature, NULL };
float* const buf = (float*)calloc(block_size * 2, sizeof(float));
float* const in = buf;
float* const out = buf + block_size;
if (!buf) {
fprintf(stderr, "Out of memory\n");
return 0.0;
}
LV2_Atom_Sequence seq = {
{ sizeof(LV2_Atom_Sequence_Body),
uri_table_map(&uri_table, LV2_ATOM__Sequence) },
{ 0, 0 } };
const char* uri = lilv_node_as_string(lilv_plugin_get_uri(p));
LilvNodes* required = lilv_plugin_get_required_features(p);
LILV_FOREACH(nodes, i, required) {
const LilvNode* feature = lilv_nodes_get(required, i);
if (!lilv_node_equals(feature, urid_map)) {
fprintf(stderr, "<%s> requires feature <%s>, skipping\n",
uri, lilv_node_as_uri(feature));
free(buf);
uri_table_destroy(&uri_table);
return 0.0;
}
}
LilvInstance* instance = lilv_plugin_instantiate(p, 48000.0, features);
if (!instance) {
fprintf(stderr, "Failed to instantiate <%s>\n",
lilv_node_as_uri(lilv_plugin_get_uri(p)));
free(buf);
uri_table_destroy(&uri_table);
return 0.0;
}
float* controls = (float*)calloc(
lilv_plugin_get_num_ports(p), sizeof(float));
lilv_plugin_get_port_ranges_float(p, NULL, NULL, controls);
const uint32_t n_ports = lilv_plugin_get_num_ports(p);
for (uint32_t index = 0; index < n_ports; ++index) {
const LilvPort* port = lilv_plugin_get_port_by_index(p, index);
if (lilv_port_is_a(p, port, lv2_ControlPort)) {
lilv_instance_connect_port(instance, index, &controls[index]);
} else if (lilv_port_is_a(p, port, lv2_AudioPort) ||
lilv_port_is_a(p, port, lv2_CVPort)) {
if (lilv_port_is_a(p, port, lv2_InputPort)) {
lilv_instance_connect_port(instance, index, in);
} else if (lilv_port_is_a(p, port, lv2_OutputPort)) {
lilv_instance_connect_port(instance, index, out);
} else {
fprintf(stderr, "<%s> port %d neither input nor output, skipping\n",
uri, index);
lilv_instance_free(instance);
free(buf);
free(controls);
uri_table_destroy(&uri_table);
return 0.0;
}
} else if (lilv_port_is_a(p, port, atom_AtomPort)) {
lilv_instance_connect_port(instance, index, &seq);
} else {
fprintf(stderr, "<%s> port %d has unknown type, skipping\n",
uri, index);
lilv_instance_free(instance);
free(buf);
free(controls);
uri_table_destroy(&uri_table);
return 0.0;
}
}
lilv_instance_activate(instance);
struct timespec ts = bench_start();
for (uint32_t i = 0; i < (sample_count / block_size); ++i) {
lilv_instance_run(instance, block_size);
}
const double elapsed = bench_end(&ts);
lilv_instance_deactivate(instance);
lilv_instance_free(instance);
uri_table_destroy(&uri_table);
if (full_output) {
printf("%d %d ", block_size, sample_count);
}
printf("%lf %s\n", elapsed, uri);
free(buf);
free(controls);
return elapsed;
}