good for now

.gitignore

@@ -1,2 +1,4 @@
 build/
 .vscode/
+
+*.flac

Makefile

@@ -2,7 +2,7 @@
 CC ?= gcc
 
 CFLAGS = -Wall -Wextra -std=gnu17 -I include $(shell pkg-config --cflags lilv-0)
-LDFLAGS = -llilv-0
+LDFLAGS = -llilv-0 -lsndfile
 
 # Directory for build outputs
 BUILD_DIR := build

include/audiofile.h

@@ -0,0 +1,9 @@
+#ifndef AUDIOFILE_H
+#define AUDIOFILE_H
+
+#include <sndfile.h>
+
+SNDFILE *audiofile_read(const char *path, SF_INFO *sfinfo);
+SNDFILE *audiofile_open_for_write(const char *path, SF_INFO *og_info);
+
+#endif

include/easy.h

@@ -2,6 +2,7 @@
 #define EASY_H
 
 #include <lilv/lilv.h>
+
 const LilvPlugin* easy_load_plugin(LilvWorld* world, const char* uri);
 
 #endif

main.py

@@ -0,0 +1,70 @@
+import lilv, numpy, wave
+import soundfile
+from ctypes import c_float, c_int
+
+frames, sample_rate = soundfile.read('a.flac')
+num_frames = len(frames)
+print(f"{sample_rate}Hz | {int(num_frames/sample_rate/60)}:{int(num_frames/sample_rate%60)}")
+print(frames.dtype)
+frames = frames.astype(numpy.float32)
+
+world = lilv.World()
+world.load_all()
+
+plugins = world.get_all_plugins()
+
+plugin = plugins.get_by_uri("http://lsp-plug.in/plugins/lv2/comp_delay_x2_stereo")
+
+print(plugin.get_name())
+
+for i in range(plugin.get_num_ports()):
+    port = plugin.get_port(i)
+    print(f"Port {i}: {port.get_name()} = {port.get_range()[0]}")
+
+instance = lilv.Instance(plugin, sample_rate)
+
+output_left = numpy.zeros(num_frames, frames.dtype)
+output_right = numpy.zeros(num_frames, frames.dtype)
+
+input_left_chunk = numpy.zeros(512, frames.dtype)
+input_right_chunk = numpy.zeros(512, frames.dtype)
+output_left_chunk = numpy.zeros(512, frames.dtype)
+output_right_chunk = numpy.zeros(512, frames.dtype)
+
+instance.connect_port(0, input_left_chunk)
+instance.connect_port(1, input_right_chunk)
+instance.connect_port(2, output_left_chunk)
+instance.connect_port(3, output_right_chunk)
+
+mode_left = numpy.array([2], dtype=numpy.int32)
+instance.connect_port(5, mode_left)
+
+time_left = numpy.array([0.0], dtype=numpy.float32)
+instance.connect_port(11, time_left)
+
+mode_right = numpy.array([2], dtype=numpy.int32)
+instance.connect_port(16, mode_right)
+
+time_right = numpy.array([500.0], dtype=numpy.float32)
+instance.connect_port(22, time_right)
+
+instance.activate()
+
+print(f"frames: {num_frames}\n")
+for start in range(0, num_frames, 512):
+    end = min(start + 512, num_frames)
+    chunk_size = end - start
+    print("\033[F", start, chunk_size, end)
+
+    input_left_chunk[:chunk_size] = frames[start:end, 0]
+    input_right_chunk[:chunk_size] = frames[start:end, 1]
+
+    instance.run(chunk_size)
+
+    output_left[start:end] = output_left_chunk[:chunk_size]
+    output_right[start:end] = output_right_chunk[:chunk_size]
+
+instance.deactivate()
+
+merged = numpy.column_stack((output_left, output_right))
+soundfile.write('aoutt.flac', merged, sample_rate)

src/audiofile.c

@@ -0,0 +1,37 @@
+#include <audiofile.h>
+
+SNDFILE *audiofile_read(const char *path, SF_INFO *sfinfo) {
+    SNDFILE *sndfile = sf_open(path, SFM_READ, sfinfo);
+    if (sndfile == NULL) {
+        fprintf(stderr, "Failed to read audio file: %s\n", sf_strerror(NULL));
+        return NULL;
+    }
+
+    if (sfinfo->channels != 2) {
+        fprintf(stderr, "File must have 2 channels\n");
+        sf_close(sndfile);
+        return NULL;
+    }
+
+    int seconds = (double)sfinfo->frames / sfinfo->samplerate;
+    int minutes = seconds / 60;
+    printf("Duration: %d:%d\n", minutes, seconds % 60);
+    printf("Sample rate: %dHz\n\n", sfinfo->samplerate);
+
+    return sndfile;
+}
+
+SNDFILE *audiofile_open_for_write(const char *path, SF_INFO *og_info) {
+    SF_INFO	sfinfo = {0};
+    sfinfo.samplerate = og_info->samplerate;
+    sfinfo.format = og_info->format;
+    sfinfo.channels = og_info->channels;
+
+    SNDFILE *sndfile = sf_open(path, SFM_WRITE, &sfinfo);
+    if (sndfile == NULL) {
+        fprintf(stderr, "Failed to open audio file: %s\n", sf_strerror(NULL));
+        return NULL;
+    }
+
+    return sndfile;
+}

src/easy.c

@@ -1,4 +1,4 @@
-#include <easy.h>
+#include "easy.h"
 
 const LilvPlugin* easy_load_plugin(LilvWorld* world, const char* uri) {
     const LilvPlugins* plugin_list = lilv_world_get_all_plugins(world);
@@ -11,6 +11,24 @@ const LilvPlugin* easy_load_plugin(LilvWorld* world, const char* uri) {
         LilvNode *plugin_name = lilv_plugin_get_name(plugin);
         printf("Loaded plugin \"%s\"\n", lilv_node_as_string(plugin_name));
         lilv_node_free(plugin_name);
+
+        uint32_t n_ports = lilv_plugin_get_num_ports(plugin);
+        for (uint32_t i = 0; i < n_ports; i++) {
+            const LilvPort* port = lilv_plugin_get_port_by_index(plugin, i);
+            // Get port properties
+            const char* name = lilv_node_as_string(lilv_port_get_name(plugin, port));
+
+            LilvNode* def = NULL;
+            lilv_port_get_range(plugin, port, &def, NULL, NULL);
+            printf("Port %d: %s = %f\n", i, name, lilv_node_as_float(def));
+            lilv_node_free(def);
+        }
+
+        LilvNodes *features = lilv_plugin_get_required_features(plugin);
+        LILV_FOREACH(nodes, i, features) {
+            LilvNode *n = lilv_nodes_get(features, i);
+            printf("Feat: %s\n", lilv_node_as_string(n));
+        }
     }
 
     return plugin;

src/main.c

@@ -1,6 +1,10 @@
-#include <lilv/lilv.h>
-#include <easy.h>
 #include <stdio.h>
+#include <sndfile.h>
+
+#include "easy.h"
+#include "audiofile.h"
+
+#define BUFFER_SIZE 256 // this is per channel
 
 /*
 http://lsp-plug.in/plugins/lv2/comp_delay_x2_stereo
@@ -8,13 +12,24 @@ http://lsp-plug.in/plugins/lv2/para_equalizer_x32_lr
 http://calf.sourceforge.net/plugins/BassEnhancer
 */
 
-const LilvPlugin* easy_load_plugin(LilvWorld* world, const char* uri);
-
-int main() {
+int main(int argc, char *argv[]) {
     LilvWorld *world = lilv_world_new();
     lilv_world_load_all(world);
 
-    //
+    // load audio file
+
+    SF_INFO sfinfo = {0};
+    SNDFILE *sndfile;
+    if ((sndfile = audiofile_read("a.flac", &sfinfo)) == NULL) {
+        return 1;
+    }
+
+    SNDFILE *out_sndfile;
+    if ((out_sndfile = audiofile_open_for_write("aout.flac", &sfinfo)) == NULL) {
+        return 1;
+    }
+    
+    // load lilv plugin
 
     const LilvPlugin* plugin = easy_load_plugin(world, "http://lsp-plug.in/plugins/lv2/comp_delay_x2_stereo");
     if (plugin == NULL) {
@@ -22,8 +37,77 @@ int main() {
         return 1;
     }
 
-    //
+    // prepare lilv instance
 
-    LilvInstance* instance = lilv_plugin_instantiate(plugin, 48000.0, NULL);
+    LilvInstance *instance = lilv_plugin_instantiate(plugin, sfinfo.samplerate, NULL);
+
+    float input_left[BUFFER_SIZE];
+    float input_right[BUFFER_SIZE];
+    float output_left[BUFFER_SIZE];
+    float output_right[BUFFER_SIZE];
+
+    lilv_instance_connect_port(instance, 0, input_left); // Input L
+    lilv_instance_connect_port(instance, 1, input_right); // Input R
+    lilv_instance_connect_port(instance, 2, output_left); // Output L
+    lilv_instance_connect_port(instance, 3, output_right); // Output R
+
+    float sc[BUFFER_SIZE] = {0};
+    lilv_instance_connect_port(instance, 4, &sc); // Output R
+    lilv_instance_connect_port(instance, 5, &sc); // Output R
+
+    /*float bypass = 0.0f;
+    lilv_instance_connect_port(instance, 6, &bypass);
+
+    float aa = 0.0f;
+    lilv_instance_connect_port(instance, 8, &aa);*/
+
+    /*int mode_r = 2;
+    float time_r = 999.0f; // ms
+
+    lilv_instance_connect_port(instance, 5, &mode_r); // Mode Right
+    lilv_instance_connect_port(instance, 11, &time_r); // Time Right
+
+    float time_dr =1.0f; // ms
+    lilv_instance_connect_port(instance, 13, &time_dr); // Time Right
+
+    float aa = 0.0f;
+    lilv_instance_connect_port(instance, 4, &aa);*/
+    
+    lilv_instance_activate(instance);
+
+    // start processing
+
+    printf("Now processing\n");
+
+    float buffer[BUFFER_SIZE * 2];
+    float out_buffer[BUFFER_SIZE * 2];
+    int frames_read;
+    while ((frames_read = sf_readf_float(sndfile, buffer, BUFFER_SIZE))) {
+        for (sf_count_t i = 0; i < frames_read; i++) {
+            input_left[i] = buffer[i * 2];
+            input_right[i] = buffer[i * 2 + 1];
+        }
+
+        lilv_instance_run(instance, frames_read);
+
+        for (sf_count_t i = 0; i < frames_read; i++) {
+            out_buffer[i * 2] = output_left[i];
+            out_buffer[i * 2 + 1] = output_right[i];
+        }
+
+        sf_writef_float(out_sndfile, out_buffer, frames_read);
+    }
+
+    printf("Done processing\n");
+
+    // end processing
+
+    lilv_instance_deactivate(instance);
+    lilv_instance_free(instance);
+
+    lilv_world_free(world);
+
+    sf_close(sndfile);
+    sf_close(out_sndfile);
 }