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gstvstaudioprocessor.cpp
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gstvstaudioprocessor.cpp
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/*
* Copyright (C) 2017 Sebastian Dröge <[email protected]>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#include <sys/types.h>
#include <sys/stat.h>
/* For g_stat () */
#include <glib/gstdio.h>
#include "plugin.h"
#include "gstvstaudioprocessor.h"
#include <gst/audio/audio.h>
#include <gst/base/base.h>
#include <common/memorystream.h>
#include <vst/hosting/module.h>
#include <vst/vstcomponent.h>
#include <vst/vsteditcontroller.h>
#include <vst/hosting/stringconvert.h>
#include <vst/hosting/parameterchanges.h>
#include <pluginterfaces/vst/ivstprocesscontext.h>
#include <pluginterfaces/vst/ivstaudioprocessor.h>
#if defined(G_OS_WIN32)
#include <windows.h>
#include <shlobj.h>
#pragma comment(lib, "Shell32")
#endif
GST_DEBUG_CATEGORY_STATIC(gst_vst_audio_processor_debug);
#define GST_CAT_DEFAULT gst_vst_audio_processor_debug
using namespace Steinberg;
static GstElementClass *parent_class = nullptr;
static GQuark audio_processor_info_quark;
static void gst_vst_audio_processor_class_init(GstVstAudioProcessorClass * klass);
static void gst_vst_audio_processor_sub_class_init(GstVstAudioProcessorClass * klass);
static void gst_vst_audio_processor_init(GstVstAudioProcessor * self, GstVstAudioProcessorClass * klass);
static GstFlowReturn gst_vst_audio_processor_sink_chain(GstPad * pad,
GstObject * parent, GstBuffer * buffer);
static gboolean gst_vst_audio_processor_sink_event(GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_vst_audio_processor_src_query(GstPad * pad,
GstObject * parent, GstQuery * query);
static void gst_vst_audio_processor_finalize(GObject * object);
static void gst_vst_audio_processor_get_property(GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_vst_audio_processor_set_property(GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_vst_audio_processor_sub_get_property(GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_vst_audio_processor_sub_set_property(GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_vst_audio_processor_change_state(GstElement *
element, GstStateChange transition);
// The different states the audio processor can be in
typedef enum {
STATE_NONE = 0,
STATE_CREATED,
STATE_INITIALIZED,
STATE_SETUP,
STATE_ACTIVE,
STATE_PROCESSING,
} State;
enum {
PROP_0 = 0,
PROP_MAX_SAMPLES_PER_CHUNK,
};
#define DEFAULT_MAX_SAMPLES_PER_CHUNK (1024)
// Communication between edit controller and component happens over this. We
// don't really use this as we don't want to use the GUI provided by the
// controller.
class GstVstAudioProcessorComponentHandler: public Vst::IComponentHandler {
public:
GstVstAudioProcessorComponentHandler(GstVstAudioProcessor * processor): processor(processor) { FUNKNOWN_CTOR }
virtual ~GstVstAudioProcessorComponentHandler() { FUNKNOWN_DTOR }
DECLARE_FUNKNOWN_METHODS
tresult PLUGIN_API beginEdit(Vst::ParamID id) override {
GST_FIXME_OBJECT(processor, "beginEdit not implemented");
return kNotImplemented;
}
tresult PLUGIN_API performEdit(Vst::ParamID id, Vst::ParamValue valueNormalized) override {
GST_FIXME_OBJECT(processor, "performEdit not implemented");
return kNotImplemented;
}
tresult PLUGIN_API endEdit(Vst::ParamID id) override {
GST_FIXME_OBJECT(processor, "endEdit not implemented");
return kNotImplemented;
}
tresult PLUGIN_API restartComponent(int32 flags) override {
// TODO: Maybe want to do something with the other ones?
GST_DEBUG_OBJECT(processor, "restartComponent(0x%08x)", flags);
if (flags & Vst::kLatencyChanged) {
gst_element_post_message(GST_ELEMENT_CAST(processor),
gst_message_new_latency(GST_OBJECT_CAST(processor)));
}
return kResultOk;
}
GstVstAudioProcessor * processor;
};
IMPLEMENT_REFCOUNT(GstVstAudioProcessorComponentHandler);
DECLARE_CLASS_IID(GstVstAudioProcessorComponentHandler, 0x8f2a46d5, 0x148a4e40, 0xab996b56, 0xc2c615cf);
tresult GstVstAudioProcessorComponentHandler::queryInterface(const char * iid, void ** obj) {
QUERY_INTERFACE(iid, obj, FUnknown::iid, FUnknown);
QUERY_INTERFACE(iid, obj, Vst::IComponentHandler::iid, Vst::IComponentHandler);
*obj = nullptr;
return kNoInterface;
}
struct _GstVstAudioProcessor {
GstElement element;
GstPad *srcpad, *sinkpad;
// Properties
gint max_samples_per_chunk;
// Protected by object lock
Vst::ParameterChanges *parameter_changes;
gdouble *parameter_values;
// State
// Protected by stream lock
GstSegment segment;
GstAudioInfo info;
GstClockTime latency;
State state;
std::shared_ptr<VST3::Hosting::Module> module;
IPtr<Vst::IComponent> component;
IPtr<Vst::IEditController> edit_controller;
IPtr<Vst::IAudioProcessor> audio_processor;
IPtr<GstVstAudioProcessorComponentHandler> component_handler;
// Temporary buffer space used for deinterleaving
gpointer in_data[2];
gpointer out_data[2];
guint data_len;
};
struct _GstVstAudioProcessorClass {
GstElementClass parent_class;
const GstVstAudioProcessorInfo *processor_info;
};
// Property definition
typedef struct {
Vst::ParamID param_id;
const gchar *name;
const gchar *nick;
const gchar *description;
GType type; // float, bool, int
gint32 max_value; // for int
gdouble default_value;
gboolean read_only;
GParamSpec *pspec;
} GstVstAudioProcessorProperty;
// Class information extracted from component
struct _GstVstAudioProcessorInfo {
const gchar *name;
GstCaps *caps;
const gchar *path;
VST3::UID class_id;
// properties[0] -> GObject property ID 1
GstVstAudioProcessorProperty *properties;
guint n_properties;
};
GType
gst_vst_audio_processor_get_type(void)
{
static volatile gsize type = 0;
if (g_once_init_enter(&type)) {
GType _type;
static const GTypeInfo info = {
sizeof (GstVstAudioProcessorClass),
nullptr,
nullptr,
(GClassInitFunc) gst_vst_audio_processor_class_init,
nullptr,
nullptr,
sizeof (GstVstAudioProcessor),
0,
(GInstanceInitFunc) gst_vst_audio_processor_init,
nullptr
};
_type = g_type_register_static(GST_TYPE_ELEMENT, "GstVstAudioProcessor",
&info, (GTypeFlags) 0);
g_once_init_leave(&type, _type);
}
return type;
}
static void
gst_vst_audio_processor_sub_class_init(GstVstAudioProcessorClass * klass)
{
auto gobject_class = G_OBJECT_CLASS(klass);
auto element_class = GST_ELEMENT_CLASS(klass);
auto audio_processor_klass = GST_VST_AUDIO_PROCESSOR_CLASS(klass);
auto processor_info = (const GstVstAudioProcessorInfo *)
g_type_get_qdata(G_TYPE_FROM_CLASS(klass), audio_processor_info_quark);
// This happens for the base class and abstract subclasses
if (!processor_info)
return;
audio_processor_klass->processor_info = processor_info;
// Add pad templates
auto templ = gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS, processor_info->caps);
gst_element_class_add_pad_template(element_class, templ);
templ = gst_pad_template_new("src", GST_PAD_SRC, GST_PAD_ALWAYS, processor_info->caps);
gst_element_class_add_pad_template(element_class, templ);
auto longname = g_strdup_printf("VST3 Audio processor - %s", processor_info->name);
gst_element_class_set_metadata(element_class,
processor_info->name,
"Audio/Filter",
longname, "Sebastian Dröge <[email protected]>");
g_free(longname);
// Register all our properties, if any
if (processor_info->n_properties > 0) {
gobject_class->set_property = gst_vst_audio_processor_sub_set_property;
gobject_class->get_property = gst_vst_audio_processor_sub_get_property;
for (auto i = 0U; i < processor_info->n_properties; i++) {
auto property = &processor_info->properties[i];
GParamSpec *param_spec;
if (property->type == G_TYPE_DOUBLE) {
param_spec = g_param_spec_double(property->name, property->nick, property->description,
-G_MAXDOUBLE, G_MAXDOUBLE,
property->default_value,
(GParamFlags) ((property->read_only ? G_PARAM_READABLE : G_PARAM_READWRITE) | GST_PARAM_CONTROLLABLE));
} else if (property->type == G_TYPE_BOOLEAN) {
param_spec = g_param_spec_boolean(property->name, property->nick, property->description,
property->default_value > 0.5,
(GParamFlags) ((property->read_only ? G_PARAM_READABLE : G_PARAM_READWRITE) | GST_PARAM_CONTROLLABLE));
} else {
param_spec = g_param_spec_int(property->name, property->nick, property->description,
0, property->max_value,
property->default_value,
(GParamFlags) ((property->read_only ? G_PARAM_READABLE : G_PARAM_READWRITE) | GST_PARAM_CONTROLLABLE));
}
property->pspec = param_spec;
g_object_class_install_property(gobject_class, i + 1, param_spec);
}
}
}
static void
gst_vst_audio_processor_class_init(GstVstAudioProcessorClass * klass)
{
auto gobject_class = G_OBJECT_CLASS(klass);
auto gstelement_class = GST_ELEMENT_CLASS(klass);
parent_class = GST_ELEMENT_CLASS(g_type_class_peek_parent(klass));
gobject_class->set_property = gst_vst_audio_processor_set_property;
gobject_class->get_property = gst_vst_audio_processor_get_property;
gobject_class->finalize = gst_vst_audio_processor_finalize;
g_object_class_install_property (gobject_class, PROP_MAX_SAMPLES_PER_CHUNK,
g_param_spec_int ("max-samples-per-chunk", "Max Samples per Chunk",
"Maximum number of samples to process per chunk", 1,
G_MAXINT, DEFAULT_MAX_SAMPLES_PER_CHUNK,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY)));
gstelement_class->change_state = gst_vst_audio_processor_change_state;
}
static void
gst_vst_audio_processor_init(GstVstAudioProcessor * self, GstVstAudioProcessorClass * klass)
{
auto sink_templ = gst_element_class_get_pad_template(GST_ELEMENT_CLASS(klass), "sink");
self->sinkpad = gst_pad_new_from_template (sink_templ, "sink");
gst_pad_set_chain_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_vst_audio_processor_sink_chain));
gst_pad_set_event_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_vst_audio_processor_sink_event));
GST_PAD_SET_PROXY_CAPS (self->sinkpad);
gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
auto src_templ = gst_element_class_get_pad_template(GST_ELEMENT_CLASS(klass), "src");
self->srcpad = gst_pad_new_from_template (src_templ, "src");
gst_pad_set_query_function (self->srcpad,
GST_DEBUG_FUNCPTR (gst_vst_audio_processor_src_query));
GST_PAD_SET_PROXY_CAPS (self->srcpad);
gst_pad_use_fixed_caps (self->srcpad);
gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
self->state = STATE_NONE;
self->module = nullptr;
self->component = nullptr;
self->audio_processor = nullptr;
self->edit_controller = nullptr;
self->component_handler = nullptr;
self->max_samples_per_chunk = DEFAULT_MAX_SAMPLES_PER_CHUNK;
// Initialize all properties as stored here with their default values
self->parameter_values = g_new0(gdouble, klass->processor_info->n_properties);
for (auto i = 0U; i < klass->processor_info->n_properties; i++)
self->parameter_values[i] = klass->processor_info->properties[i].default_value;
}
static void
gst_vst_audio_processor_finalize(GObject * object)
{
auto self = GST_VST_AUDIO_PROCESSOR(object);
if (self->parameter_changes)
delete self->parameter_changes;
g_free(self->parameter_values);
G_OBJECT_CLASS(parent_class)->finalize(object);
}
static void
gst_vst_audio_processor_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec)
{
auto self = GST_VST_AUDIO_PROCESSOR(object);
switch (property_id) {
case PROP_MAX_SAMPLES_PER_CHUNK:
g_value_set_int (value, self->max_samples_per_chunk);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_vst_audio_processor_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec)
{
auto self = GST_VST_AUDIO_PROCESSOR(object);
switch (property_id) {
case PROP_MAX_SAMPLES_PER_CHUNK:
self->max_samples_per_chunk = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_vst_audio_processor_sub_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec)
{
auto self = GST_VST_AUDIO_PROCESSOR(object);
auto klass = GST_VST_AUDIO_PROCESSOR_GET_CLASS(self);
if (property_id > klass->processor_info->n_properties + 1 || property_id == 0) {
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
return;
}
GST_OBJECT_LOCK(self);
auto property = &klass->processor_info->properties[property_id - 1];
if (property->type == G_TYPE_DOUBLE)
g_value_set_double(value, self->parameter_values[property_id - 1]);
else if (property->type == G_TYPE_BOOLEAN)
g_value_set_boolean(value, self->parameter_values[property_id - 1] > 0.5);
else
g_value_set_int(value, self->parameter_values[property_id - 1]);
GST_OBJECT_UNLOCK(self);
}
static void
gst_vst_audio_processor_sub_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec)
{
auto self = GST_VST_AUDIO_PROCESSOR(object);
auto klass = GST_VST_AUDIO_PROCESSOR_GET_CLASS(self);
if (property_id > klass->processor_info->n_properties + 1 || property_id == 0) {
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
return;
}
GST_OBJECT_LOCK(self);
auto property = &klass->processor_info->properties[property_id - 1];
// Store value in our cache
if (property->type == G_TYPE_DOUBLE) {
self->parameter_values[property_id - 1] = g_value_get_double(value);
} else if (property->type == G_TYPE_BOOLEAN) {
self->parameter_values[property_id - 1] = g_value_get_boolean(value);
} else {
self->parameter_values[property_id - 1] = g_value_get_int(value);
}
// If we have an edit controller, convert our plain values to normalized
// values and store the parameter changes and let the edit controller know
// about it
//
// We always use plain values, but controller and component use normalized
// values between 0.0 and 1.0
if (self->edit_controller) {
if (!self->parameter_changes)
self->parameter_changes = new Vst::ParameterChanges();
Steinberg::int32 idx = 0;
auto queue = self->parameter_changes->addParameterData(property->param_id, idx);
auto value = self->edit_controller->plainParamToNormalized(property->param_id,
self->parameter_values[property_id - 1]);
queue->addPoint(0, value, idx);
self->edit_controller->setParamNormalized(property->param_id, value);
}
GST_OBJECT_UNLOCK(self);
}
static gboolean
gst_vst_audio_processor_open(GstVstAudioProcessor *self)
{
auto klass = GST_VST_AUDIO_PROCESSOR_GET_CLASS(self);
std::string err;
std::string path(klass->processor_info->path);
self->state = STATE_NONE;
auto mod = VST3::Hosting::Module::create(path, err);
if (!mod) {
GST_ERROR_OBJECT(self, "Failed to load module '%s': %s", path.c_str(), err.c_str());
return FALSE;
}
auto factory = mod->getFactory();
auto component = factory.createInstance<Vst::IComponent>(klass->processor_info->class_id);
if (!component) {
GST_ERROR_OBJECT(self, "Failed to create instance for '%s'", klass->processor_info->name);
return FALSE;
}
auto res = component->initialize(gStandardPluginContext);
if (res != kResultOk) {
GST_ERROR_OBJECT(self, "Component can't be initialized: 0x%08x", res);
return FALSE;
}
// Check if this supports the IAudioProcessor interface
IPtr<Vst::IAudioProcessor> audio_processor;
Vst::IAudioProcessor *audio_processor_ptr = nullptr;
if (component->queryInterface(Vst::IAudioProcessor::iid, (void **) &audio_processor_ptr) != kResultOk
|| !audio_processor_ptr) {
GST_ERROR_OBJECT(self, "Component does not implement IAudioProcessor interface");
return FALSE;
}
audio_processor = shared(audio_processor_ptr);
// Get the controller
IPtr<Vst::IEditController> edit_controller;
Vst::IEditController *edit_controller_ptr = nullptr;
if (component->queryInterface(Vst::IEditController::iid, (void**) &edit_controller_ptr) != kResultOk
|| !edit_controller_ptr) {
FUID controller_cid;
// ask for the associated controller class ID
if (component->getControllerClassId(controller_cid) == kResultOk && controller_cid.isValid ()) {
// create its controller part created from the factory
edit_controller = factory.createInstance<Vst::IEditController>(controller_cid.toTUID());
if (edit_controller) {
// initialize the component with our context
res = edit_controller->initialize(gStandardPluginContext);
if (res != kResultOk) {
GST_ERROR_OBJECT(self, "Can't initialize edit controller: 0x%08x", res);
return FALSE;
}
}
}
} else {
edit_controller = shared(edit_controller_ptr);
}
if (!edit_controller) {
GST_ERROR_OBJECT(self, "No edit controller found");
return FALSE;
}
// activate busses, just in case
res = component->activateBus(Vst::MediaTypes::kAudio, Vst::BusDirections::kInput, 0, TRUE);
if (res != kResultOk) {
GST_ERROR_OBJECT(self, "Failed to activate input bus: 0x%08x", res);
return FALSE;
}
res = component->activateBus(Vst::MediaTypes::kAudio, Vst::BusDirections::kOutput, 0, TRUE);
if (res != kResultOk) {
GST_ERROR_OBJECT(self, "Failed to activate output bus: 0x%08x", res);
return FALSE;
}
self->component_handler = owned(new GstVstAudioProcessorComponentHandler(self));
// the host set its handler to the controller
edit_controller->setComponentHandler(self->component_handler);
// connect the 2 components
Vst::IConnectionPoint* iConnectionPointComponent = nullptr;
Vst::IConnectionPoint* iConnectionPointController = nullptr;
component->queryInterface(Vst::IConnectionPoint::iid, (void**)&iConnectionPointComponent);
edit_controller->queryInterface(Vst::IConnectionPoint::iid, (void**)&iConnectionPointController);
if (iConnectionPointComponent && iConnectionPointController) {
iConnectionPointComponent->connect(iConnectionPointController);
iConnectionPointController->connect(iConnectionPointComponent);
}
// synchronize controller to component by using setComponentState
MemoryStream stream;
if (component->getState(&stream) == kResultOk) {
stream.truncate();
edit_controller->setComponentState(&stream);
}
// synchronize our cached property values with the component and controller
self->parameter_changes = new Vst::ParameterChanges();
for (auto i = 0U; i < klass->processor_info->n_properties; i++) {
auto property = &klass->processor_info->properties[i];
if (property->read_only)
continue;
Steinberg::int32 idx = 0;
auto queue = self->parameter_changes->addParameterData(property->param_id, idx);
auto value = edit_controller->plainParamToNormalized(property->param_id,
self->parameter_values[i]);
queue->addPoint(0, value, idx);
edit_controller->setParamNormalized(property->param_id, value);
}
self->state = STATE_INITIALIZED;
self->module = mod;
self->component = component;
self->audio_processor = audio_processor;
self->edit_controller = edit_controller;
return TRUE;
}
static GstStateChangeReturn
gst_vst_audio_processor_change_state(GstElement * element,
GstStateChange transition)
{
auto self = GST_VST_AUDIO_PROCESSOR(element);
auto state_ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
gst_audio_info_init(&self->info);
gst_segment_init(&self->segment, GST_FORMAT_TIME);
if (!gst_vst_audio_processor_open(self))
state_ret = GST_STATE_CHANGE_FAILURE;
break;
default:
break;
}
state_ret = GST_ELEMENT_CLASS(parent_class)->change_state(element, transition);
if (state_ret == GST_STATE_CHANGE_FAILURE)
return state_ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (self->state >= STATE_PROCESSING)
self->audio_processor->setProcessing(false);
if (self->state >= STATE_ACTIVE)
self->component->setActive(false);
self->state = STATE_SETUP;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (self->state >= STATE_SETUP) {
self->component->terminate();
self->edit_controller->terminate();
}
self->state = STATE_NONE;
self->audio_processor = nullptr;
self->component = nullptr;
self->edit_controller = nullptr;
self->module = nullptr;
self->component_handler = nullptr;
g_free(self->in_data[0]);
self->in_data[0] = nullptr;
g_free(self->in_data[1]);
self->in_data[1] = nullptr;
g_free(self->out_data[0]);
self->out_data[0] = nullptr;
g_free(self->out_data[1]);
self->out_data[1] = nullptr;
self->data_len = 0;
break;
default:
break;
}
return state_ret;
}
static void
deinterleave_data(GstVstAudioProcessor *self, gconstpointer in_data, guint len)
{
if (self->info.finfo->format == GST_AUDIO_FORMAT_F32) {
if (self->info.channels == 1) {
memcpy (self->in_data[0], in_data, len * self->info.bpf);
} else {
const float *f_in_data = (const float *) in_data;
for (auto i = 0; i < 2; i++) {
float *f_out_data = (float *) self->in_data[i];
for (auto j = 0U; j < len; j++) {
f_out_data[j] = f_in_data[i + j * 2];
}
}
}
} else {
if (self->info.channels == 1) {
memcpy (self->in_data[0], in_data, len * self->info.bpf);
} else {
const double *f_in_data = (const double *) in_data;
for (auto i = 0; i < 2; i++) {
double *f_out_data = (double *) self->in_data[i];
for (auto j = 0U; j < len; j++) {
f_out_data[j] = f_in_data[i + j * 2];
}
}
}
}
}
static void
interleave_data(GstVstAudioProcessor *self, gpointer out_data, guint len)
{
if (self->info.finfo->format == GST_AUDIO_FORMAT_F32) {
if (self->info.channels == 1) {
memcpy (out_data, self->out_data[0], len * self->info.bpf);
} else {
float *f_out_data = (float *) out_data;
for (auto i = 0; i < 2; i++) {
const float *f_in_data = (const float *) self->out_data[i];
for (auto j = 0U; j < len; j++) {
f_out_data[i + j * 2] = f_in_data[j];
}
}
}
} else {
if (self->info.channels == 1) {
memcpy (out_data, self->out_data[0], len * self->info.bpf);
} else {
double *f_out_data = (double *) out_data;
for (auto i = 0; i < 2; i++) {
const double *f_in_data = (const double *) self->out_data[i];
for (auto j = 0U; j < len; j++) {
f_out_data[i + j * 2] = f_in_data[j];
}
}
}
}
}
static GstFlowReturn
gst_vst_audio_processor_sink_chain(GstPad * pad, GstObject * parent,
GstBuffer * in_buffer)
{
auto self = GST_VST_AUDIO_PROCESSOR(parent);
auto klass = GST_VST_AUDIO_PROCESSOR_GET_CLASS(self);
auto ret = GST_FLOW_OK;
if (self->state < STATE_SETUP) {
gst_buffer_unref(in_buffer);
GST_ERROR_OBJECT(self, "Not negotiated yet");
return GST_FLOW_NOT_NEGOTIATED;
}
if (!GST_BUFFER_PTS_IS_VALID (in_buffer)) {
GST_ERROR_OBJECT(self, "Need buffers with valid timestamps");
gst_buffer_unref(in_buffer);
return GST_FLOW_ERROR;
}
// FIXME: Can we drain somehow? We should on disconts
if (GST_BUFFER_IS_DISCONT (in_buffer)) {
GST_DEBUG_OBJECT(self, "Discontinuity, restarting component");
if (self->state >= STATE_PROCESSING)
self->audio_processor->setProcessing(false);
if (self->state >= STATE_ACTIVE)
self->component->setActive(false);
}
if (self->state < STATE_ACTIVE) {
GST_DEBUG_OBJECT(self, "Activating component");
auto res = self->component->setActive(true);
if (res != kResultOk) {
GST_ERROR_OBJECT(self, "Failed to set active: %08x", res);
gst_buffer_unref(in_buffer);
return GST_FLOW_ERROR;
}
self->state = STATE_ACTIVE;
}
if (self->state < STATE_PROCESSING) {
GST_DEBUG_OBJECT(self, "Set component to processing");
auto res = self->audio_processor->setProcessing(true);
if (res != kResultOk && res != kNotImplemented) {
GST_ERROR_OBJECT(self, "Failed to set processing: %08x", res);
gst_buffer_unref(in_buffer);
return GST_FLOW_ERROR;
}
self->state = STATE_PROCESSING;
}
// We process the input buffer in chunks of at most the configured
// max-samples-per-chunk, and while doing so keep track of our current
// timestamp, stream time and sample position
GstMapInfo in_map;
gst_buffer_map(in_buffer, &in_map, GST_MAP_READ);
auto in_data = in_map.data;
auto num_samples = in_map.size / self->info.bpf;
auto stream_time = gst_segment_to_stream_time(&self->segment, GST_FORMAT_TIME, GST_BUFFER_PTS(in_buffer));
auto sample_position = (gint64) gst_util_uint64_scale(GST_BUFFER_PTS(in_buffer), self->info.rate, GST_SECOND);
auto sample_start_position = sample_position;
do {
gst_object_sync_values(GST_OBJECT_CAST(self), stream_time +
gst_util_uint64_scale(sample_position - sample_start_position, GST_SECOND, self->info.rate));
auto chunk_size = MIN(self->data_len, num_samples);
// Fill input buffers and metadata
deinterleave_data(self, (gconstpointer) in_data, chunk_size);
Vst::AudioBusBuffers input;
input.numChannels = self->info.channels;
input.silenceFlags = GST_BUFFER_FLAG_IS_SET(in_buffer, GST_BUFFER_FLAG_GAP) ? G_MAXUINT64 : 0;
if (self->info.finfo->format == GST_AUDIO_FORMAT_F32)
input.channelBuffers32 = (Vst::Sample32 **) self->in_data;
else
input.channelBuffers64 = (Vst::Sample64 **) self->in_data;
// Fill output buffer metadata
Vst::AudioBusBuffers output;
output.numChannels = self->info.channels;
output.silenceFlags = 0;
if (self->info.finfo->format == GST_AUDIO_FORMAT_F32)
output.channelBuffers32 = (Vst::Sample32 **) self->out_data;
else
output.channelBuffers64 = (Vst::Sample64 **) self->out_data;
// Set up process context with information about the system state
Vst::ProcessContext process_context;
process_context.state = Vst::ProcessContext::kPlaying |
Vst::ProcessContext::kRecording |
Vst::ProcessContext::kSystemTimeValid;
process_context.sampleRate = self->info.rate;
process_context.projectTimeSamples = sample_position;
// FIXME: Should we pretend real-time processing here?
process_context.systemTime = gst_util_get_timestamp();
// Set up process data
Vst::ProcessData data;
data.processMode = Vst::kPrefetch;
data.symbolicSampleSize = self->info.finfo->format == GST_AUDIO_FORMAT_F32 ? Vst::kSample32 : Vst::kSample64;
data.numSamples = chunk_size;
data.numInputs = 1;
data.numOutputs = 1;
data.inputs = &input;
data.outputs = &output;
data.processContext = &process_context;
// Check if we have any pending input parameter changes
GST_OBJECT_LOCK(self);
auto parameter_changes = self->parameter_changes;
self->parameter_changes = nullptr;
GST_OBJECT_UNLOCK(self);
data.inputParameterChanges = parameter_changes;
// Allocate space for any output parameter changes
Vst::ParameterChanges out_parameter_changes;
data.outputParameterChanges = &out_parameter_changes;
// And finally do the actual processing of this chunk
auto res = self->audio_processor->process(data);
// We have to delete the pointer here, the processor does not do that
if (parameter_changes) {
delete parameter_changes;
data.inputParameterChanges = nullptr;
}
// Update out parameter changes
auto out_changes_count = out_parameter_changes.getParameterCount();
for (auto i = 0; i < out_changes_count; i++) {
auto queue = out_parameter_changes.getParameterData(i);
auto point_count = queue->getPointCount();
if (point_count > 0) {
Vst::ParamValue value;
Steinberg::int32 sample_offset = 0;
if (queue->getPoint(point_count - 1, sample_offset, value) == kResultOk) {
GstVstAudioProcessorProperty *prop = nullptr;
auto param_id = queue->getParameterId();
auto plain_value = self->edit_controller->normalizedParamToPlain(param_id, value);
guint k;
for (k = 0; k < klass->processor_info->n_properties; k++) {
if (klass->processor_info->properties[k].param_id == param_id) {
prop = &klass->processor_info->properties[k];
break;
}
}
// Cache the new value and notify anybody interested
if (prop) {
self->parameter_values[k] = plain_value;
g_object_notify_by_pspec(G_OBJECT(self), prop->pspec);
}
// And let the edit controller know about this change too
self->edit_controller->setParamNormalized(param_id, value);
}
}
}
if (res != kResultOk) {
ret = GST_FLOW_ERROR;
break;
}
// If there's output, allocate a new buffer and fill it
// FIXME: We assume that input length == output length currently
// for the timestamp calculation. This is not necessarily true:
// there could be latency involved. But none of the plugins this was
// tested with makes use of that
if (data.numSamples != (gint) chunk_size) {
GST_FIXME_OBJECT(self, "Output number of samples different than input: %d != %lu", data.numSamples, chunk_size);
}
if (data.numSamples > 0) {
auto out_buffer = gst_buffer_new_and_alloc(data.numSamples * self->info.bpf);
GstMapInfo out_map;
gst_buffer_map(out_buffer, &out_map, GST_MAP_WRITE);
auto out_data = out_map.data;
interleave_data(self, (gpointer) out_data, data.numSamples);
gst_buffer_unmap(out_buffer, &out_map);
GST_BUFFER_PTS(out_buffer) = GST_BUFFER_PTS(in_buffer) +
gst_util_uint64_scale(sample_position - sample_start_position, GST_SECOND, self->info.rate);
GST_BUFFER_DURATION(out_buffer) = gst_util_uint64_scale(chunk_size, GST_SECOND, self->info.rate);
ret = gst_pad_push(self->srcpad, out_buffer);
}
num_samples -= chunk_size;
in_data += chunk_size * self->info.bpf;
sample_position += chunk_size;
} while (ret == GST_FLOW_OK && num_samples > 0);
gst_buffer_unmap(in_buffer, &in_map);
gst_buffer_unref(in_buffer);
return ret;
}
static gboolean
gst_vst_audio_processor_sink_event(GstPad * pad, GstObject * parent,
GstEvent * event)
{
auto self = GST_VST_AUDIO_PROCESSOR(parent);
gboolean ret = FALSE;
switch (GST_EVENT_TYPE(event)) {
case GST_EVENT_CAPS:{
GstCaps *caps;
GstAudioInfo info;
gboolean changed;
gst_event_parse_caps(event, &caps);
ret = gst_audio_info_from_caps(&info, caps);
changed = ret && !gst_audio_info_is_equal(&info, &self->info);
if (ret && changed) {
GST_DEBUG_OBJECT(self, "Got caps %" GST_PTR_FORMAT, caps);
self->info = info;
// Need to shut down the component to be able to configure any new
// sample rate, channel configuration or sample format
// FIXME: Can we drain somehow?
if (self->state >= STATE_PROCESSING)
self->audio_processor->setProcessing(false);
if (self->state >= STATE_ACTIVE)
self->component->setActive(false);
self->state = STATE_SETUP;
Vst::SpeakerArrangement arrangement[1] = { info.channels == 1 ? Vst::SpeakerArr::kMono : Vst::SpeakerArr::kStereo };
auto res = self->audio_processor->setBusArrangements(arrangement, 1, arrangement, 1);
if (res != kResultOk) {
GST_ERROR_OBJECT(self, "Failed to set bus arrangments: 0x%08x", res);
self->state = STATE_INITIALIZED;
ret = FALSE;
gst_event_unref(event);
break;
}
Vst::ProcessSetup setup = {
Vst::kPrefetch,
info.finfo->format == GST_AUDIO_FORMAT_F32 ? Vst::kSample32 : Vst::kSample64,
self->max_samples_per_chunk,
(double) info.rate
};
res = self->audio_processor->setupProcessing(setup);
if (res != kResultOk) {
GST_ERROR_OBJECT(self, "Failed to setup processing: %08x", res);
self->state = STATE_INITIALIZED;
ret = FALSE;
gst_event_unref(event);
break;
}
// Reallocate our buffers for both channel
auto bps = info.bpf / info.channels;
g_free(self->in_data[0]);
self->in_data[0] = g_malloc0(bps * self->max_samples_per_chunk);
g_free(self->in_data[1]);
self->in_data[1] = info.channels == 2 ? g_malloc0(bps * self->max_samples_per_chunk) : nullptr;
g_free(self->out_data[0]);
self->out_data[0] = g_malloc0(bps * self->max_samples_per_chunk);
g_free(self->out_data[1]);
self->out_data[1] = info.channels == 2 ? g_malloc0(bps * self->max_samples_per_chunk) : nullptr;
self->data_len = self->max_samples_per_chunk;
// Update latency
auto latency_samples = self->audio_processor->getLatencySamples();
auto latency = gst_util_uint64_scale_int(latency_samples, GST_SECOND, info.rate);
if (latency != self->latency) {
self->latency = latency;