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9dfb188d8a
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1ff7c3b76a |
5 changed files with 109 additions and 81 deletions
10
README.md
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10
README.md
Normal file
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@ -0,0 +1,10 @@
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LV2 plugins:
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- Linux Studio Plugins `lv2-lsp-plugins`
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C requirements (headers):
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- lilv `liblilv-0-devel`
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- sndfile `libsndfile-devel`
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Python requirements:
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- lilv `python3-lilv` (not on pypi)
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- soundfile `python311-SoundFile`
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68
main.py
68
main.py
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@ -1,11 +1,11 @@
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import lilv, numpy, wave
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import lilv, numpy, time
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import soundfile
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from ctypes import c_float, c_int
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CHUNK_SIZE = 1024
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frames, sample_rate = soundfile.read('a.flac')
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num_frames = len(frames)
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print(f"{sample_rate}Hz | {int(num_frames/sample_rate/60)}:{int(num_frames/sample_rate%60)}")
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print(frames.dtype)
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frames = frames.astype(numpy.float32)
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world = lilv.World()
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@ -14,57 +14,63 @@ world.load_all()
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plugins = world.get_all_plugins()
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plugin = plugins.get_by_uri("http://lsp-plug.in/plugins/lv2/comp_delay_x2_stereo")
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if plugin is None:
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print("No plugin comp_delay_x2_stereo")
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exit()
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print(plugin.get_name())
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print("Using:", plugin.get_name())
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for i in range(plugin.get_num_ports()):
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port = plugin.get_port(i)
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print(f"Port {i}: {port.get_name()} = {port.get_range()[0]}")
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#print(f"Port {i}: {port.get_name()} = {port.get_range()[0]}")
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instance = lilv.Instance(plugin, sample_rate)
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output_left = numpy.zeros(num_frames, frames.dtype)
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output_right = numpy.zeros(num_frames, frames.dtype)
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instance.connect_port(4, numpy.array([1], dtype=numpy.int32)) # while 1 is default it needs to be set as int32 for whatever reason
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input_left_chunk = numpy.zeros(512, frames.dtype)
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input_right_chunk = numpy.zeros(512, frames.dtype)
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output_left_chunk = numpy.zeros(512, frames.dtype)
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output_right_chunk = numpy.zeros(512, frames.dtype)
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knob_mode = numpy.array([2], dtype=numpy.float32) # by the way, this can't be put into connect_port, even if just the number is a variable
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instance.connect_port(16, knob_mode)
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knob_time = numpy.array([10.0], dtype=numpy.float32) # but this can?????? it can be any of the three
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instance.connect_port(22, knob_time)
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output = numpy.zeros((num_frames, 2), frames.dtype)
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input_left_chunk = numpy.zeros(CHUNK_SIZE, frames.dtype)
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input_right_chunk = numpy.zeros(CHUNK_SIZE, frames.dtype)
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output_left_chunk = numpy.zeros(CHUNK_SIZE, frames.dtype)
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output_right_chunk = numpy.zeros(CHUNK_SIZE, frames.dtype)
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instance.connect_port(0, input_left_chunk)
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instance.connect_port(1, input_right_chunk)
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instance.connect_port(2, output_left_chunk)
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instance.connect_port(3, output_right_chunk)
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mode_left = numpy.array([2], dtype=numpy.int32)
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instance.connect_port(5, mode_left)
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time_left = numpy.array([0.0], dtype=numpy.float32)
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instance.connect_port(11, time_left)
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mode_right = numpy.array([2], dtype=numpy.int32)
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instance.connect_port(16, mode_right)
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time_right = numpy.array([500.0], dtype=numpy.float32)
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instance.connect_port(22, time_right)
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instance.activate()
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print(f"frames: {num_frames}\n")
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for start in range(0, num_frames, 512):
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end = min(start + 512, num_frames)
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print(f"\nFrames: {num_frames}")
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print(f"Chunk size: {CHUNK_SIZE}")
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print(f"So {int(numpy.ceil(num_frames/CHUNK_SIZE))} iterations\n")
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for start in range(0, num_frames, CHUNK_SIZE):
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end = min(start + CHUNK_SIZE, num_frames)
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chunk_size = end - start
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print("\033[F", start, chunk_size, end)
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print(start, chunk_size, end, " \033[F")
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input_left_chunk[:chunk_size] = frames[start:end, 0]
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input_right_chunk[:chunk_size] = frames[start:end, 1]
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output_left_chunk.fill(0)
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output_right_chunk.fill(0)
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instance.run(chunk_size)
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output_left[start:end] = output_left_chunk[:chunk_size]
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output_right[start:end] = output_right_chunk[:chunk_size]
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output[start:end, 0] = output_left_chunk[:chunk_size]
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output[start:end, 1] = output_right_chunk[:chunk_size]
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print("done processing ")
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instance.deactivate()
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merged = numpy.column_stack((output_left, output_right))
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soundfile.write('aoutt.flac', merged, sample_rate)
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if numpy.allclose(frames, output, rtol=0.5, atol=0.5):
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print("Warning: output is identical")
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soundfile.write('aoutt.flac', output, sample_rate)
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@ -1,9 +1,11 @@
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#include <audiofile.h>
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SNDFILE *audiofile_read(const char *path, SF_INFO *sfinfo) {
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printf("Input file: \"%s\"\n", path);
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SNDFILE *sndfile = sf_open(path, SFM_READ, sfinfo);
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if (sndfile == NULL) {
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fprintf(stderr, "Failed to read audio file: %s\n", sf_strerror(NULL));
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fprintf(stderr, "Failed to open audio file: %s\n", sf_strerror(NULL));
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return NULL;
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}
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@ -16,12 +18,14 @@ SNDFILE *audiofile_read(const char *path, SF_INFO *sfinfo) {
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int seconds = (double)sfinfo->frames / sfinfo->samplerate;
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int minutes = seconds / 60;
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printf("Duration: %d:%d\n", minutes, seconds % 60);
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printf("Sample rate: %dHz\n\n", sfinfo->samplerate);
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printf("Sample rate: %dHz\n", sfinfo->samplerate);
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return sndfile;
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}
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SNDFILE *audiofile_open_for_write(const char *path, SF_INFO *og_info) {
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printf("Output file: \"%s\"\n", path);
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SF_INFO sfinfo = {0};
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sfinfo.samplerate = og_info->samplerate;
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sfinfo.format = og_info->format;
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24
src/easy.c
24
src/easy.c
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@ -12,22 +12,20 @@ const LilvPlugin* easy_load_plugin(LilvWorld* world, const char* uri) {
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printf("Loaded plugin \"%s\"\n", lilv_node_as_string(plugin_name));
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lilv_node_free(plugin_name);
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uint32_t n_ports = lilv_plugin_get_num_ports(plugin);
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for (uint32_t i = 0; i < n_ports; i++) {
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uint32_t nPorts = lilv_plugin_get_num_ports(plugin);
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for (uint32_t i = 0; i < nPorts; i++) {
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const LilvPort* port = lilv_plugin_get_port_by_index(plugin, i);
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// Get port properties
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const char* name = lilv_node_as_string(lilv_port_get_name(plugin, port));
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LilvNode* def = NULL;
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lilv_port_get_range(plugin, port, &def, NULL, NULL);
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printf("Port %d: %s = %f\n", i, name, lilv_node_as_float(def));
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lilv_node_free(def);
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}
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LilvNode *nameNode = lilv_port_get_name(plugin, port);
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const char* name = lilv_node_as_string(nameNode);
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lilv_node_free(nameNode);
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LilvNodes *features = lilv_plugin_get_required_features(plugin);
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LILV_FOREACH(nodes, i, features) {
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LilvNode *n = lilv_nodes_get(features, i);
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printf("Feat: %s\n", lilv_node_as_string(n));
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LilvNode* defNode = NULL;
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lilv_port_get_range(plugin, port, &defNode, NULL, NULL);
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float defaultValue = lilv_node_as_float(defNode);
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lilv_node_free(defNode);
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printf("Port %d: %s = %f\n", i, name, defaultValue);
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}
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}
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80
src/main.c
80
src/main.c
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@ -1,5 +1,8 @@
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#include <stdio.h>
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#include <stdlib.h>
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#include <sndfile.h>
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#include <string.h>
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#include <time.h>
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#include "easy.h"
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#include "audiofile.h"
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@ -13,6 +16,22 @@ http://calf.sourceforge.net/plugins/BassEnhancer
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*/
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int main(int argc, char *argv[]) {
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if (argc < 2) {
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printf("Usage: %s <input> [output]\n", argv[0]);
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return 2;
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}
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char *input_filename = argv[1];
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char *output_filename;
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if (argc < 3) {
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output_filename = malloc((strlen(input_filename) + 4) * sizeof(char));
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strcpy(output_filename, "out-");
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strcat(output_filename, input_filename);
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} else {
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output_filename = argv[2];
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}
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LilvWorld *world = lilv_world_new();
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lilv_world_load_all(world);
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SF_INFO sfinfo = {0};
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SNDFILE *sndfile;
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if ((sndfile = audiofile_read("a.flac", &sfinfo)) == NULL) {
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if ((sndfile = audiofile_read(input_filename, &sfinfo)) == NULL) {
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return 1;
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}
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SNDFILE *out_sndfile;
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if ((out_sndfile = audiofile_open_for_write("aout.flac", &sfinfo)) == NULL) {
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if ((out_sndfile = audiofile_open_for_write(output_filename, &sfinfo)) == NULL) {
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return 1;
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}
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@ -41,6 +60,17 @@ int main(int argc, char *argv[]) {
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LilvInstance *instance = lilv_plugin_instantiate(plugin, sfinfo.samplerate, NULL);
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float knob_enabled = 1;
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lilv_instance_connect_port(instance, 4, &knob_enabled);
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float knob_mode = 2;
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lilv_instance_connect_port(instance, 16, &knob_mode);
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float knob_time = 50.0f; // ms
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lilv_instance_connect_port(instance, 22, &knob_time);
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//
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float input_left[BUFFER_SIZE];
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float input_right[BUFFER_SIZE];
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float output_left[BUFFER_SIZE];
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lilv_instance_connect_port(instance, 2, output_left); // Output L
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lilv_instance_connect_port(instance, 3, output_right); // Output R
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float sc[BUFFER_SIZE] = {0};
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lilv_instance_connect_port(instance, 4, &sc); // Output R
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lilv_instance_connect_port(instance, 5, &sc); // Output R
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/*float bypass = 0.0f;
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lilv_instance_connect_port(instance, 6, &bypass);
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float aa = 0.0f;
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lilv_instance_connect_port(instance, 8, &aa);*/
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/*int mode_r = 2;
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float time_r = 999.0f; // ms
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lilv_instance_connect_port(instance, 5, &mode_r); // Mode Right
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lilv_instance_connect_port(instance, 11, &time_r); // Time Right
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float time_dr =1.0f; // ms
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lilv_instance_connect_port(instance, 13, &time_dr); // Time Right
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float aa = 0.0f;
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lilv_instance_connect_port(instance, 4, &aa);*/
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lilv_instance_activate(instance);
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// start processing
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lilv_instance_activate(instance);
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printf("Now processing\n");
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clock_t start = clock();
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float buffer[BUFFER_SIZE * 2];
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float out_buffer[BUFFER_SIZE * 2];
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float input_buffer[BUFFER_SIZE * 2];
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float output_buffer[BUFFER_SIZE * 2];
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int frames_read;
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while ((frames_read = sf_readf_float(sndfile, buffer, BUFFER_SIZE))) {
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while ((frames_read = sf_readf_float(sndfile, input_buffer, BUFFER_SIZE))) {
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// a frame is 2 samples and a sample is a number basically and it go left right left right (or how many channels)
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for (sf_count_t i = 0; i < frames_read; i++) {
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input_left[i] = buffer[i * 2];
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input_right[i] = buffer[i * 2 + 1];
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input_left[i] = input_buffer[i * 2];
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input_right[i] = input_buffer[i * 2 + 1];
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}
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lilv_instance_run(instance, frames_read);
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for (sf_count_t i = 0; i < frames_read; i++) {
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out_buffer[i * 2] = output_left[i];
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out_buffer[i * 2 + 1] = output_right[i];
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output_buffer[i * 2] = output_left[i];
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output_buffer[i * 2 + 1] = output_right[i];
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}
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sf_writef_float(out_sndfile, out_buffer, frames_read);
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sf_writef_float(out_sndfile, output_buffer, frames_read);
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}
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printf("Done processing\n");
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clock_t end = clock();
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printf("Done processing. Took %ldns\n", (long)((double)(end - start) / CLOCKS_PER_SEC * 1000000));
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// end processing
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