Make plugin pipe audio from the outside

pw_plugin/Makefile

@@ -1,5 +1,6 @@
-.PHONY: build
+CXXFLAGS=-O3 -march=native $(CXXFLAGS)
 
+.PHONY: build
 build: main.out
 main.out: main.o
 	$(CXX) $(LDFLAGS) $^ -o $@

pw_plugin/links.sh

@@ -1,4 +1,12 @@
 #!/usr/bin/env sh
 
-pw-link audio-filter:output alsa_output.usb-Logitech_Logitech_USB_Headset-00.analog-stereo:playback_FR
-pw-link alsa_input.pci-0000_00_1b.0.analog-stereo:capture_FL audio-filter:input
+pw-link -d mpv:output_FL alsa_output.usb-Logitech_Logitech_USB_Headset-00.analog-stereo:playback_FL
+pw-link -d mpv:output_FR alsa_output.usb-Logitech_Logitech_USB_Headset-00.analog-stereo:playback_FR
+
+
+# Setup the 3 ports
+pw-link mpv:output_FR audio-filter:musinput
+pw-link audio-filter:output alsa_output.usb-Logitech_Logitech_USB_Headset-00.analog-stereo:playback_FR 
+pw-link alsa_input.pci-0000_00_1b.0.analog-stereo:capture_FL audio-filter:micinput
+
+#pw-link  alsa_input.usb-Logitech_Logitech_USB_Headset-00.mono-fallback:capture_MONO audio-filter:input

pw_plugin/main.cc

@@ -1,47 +1,80 @@
 #include <random>
 #include <cstdio>
 #include <cmath>
+#include <cstring>
 #include <iostream>
 #include "pipewire/pipewire.h"
 #include "spa/param/audio/format-utils.h"
 #include "spa/param/latency-utils.h"
 
+constexpr int samplerate = 48000;
+
 struct local_data {
 	pw_main_loop* loop;
 	pw_filter* filter;
 
-	void* iport;
-	void* oport;
+	void* mic_port; // Input data from mic
+	void* listen_port; // Input music port
+	void* hp_port; // Output data to headphone
 
-	std::random_device gen;
-	std::normal_distribution<float> nd{0, 0.3};
+	// Hopefully we never have >100ms roundtrip delay
+	float record_buffer[samplerate * 10];
+	float mic_buffer[samplerate * 10];
 
-	uint32_t buffer_index = 0;
-	float record_buffer[48000 * 10][2]; // Forçage (to R.), retour (mic L)
+	int32_t head_index = 0; // The head is where the NEXT sample is going to be saved
+
+	/*double xcor_buffer[samplerate / 10] = {0};
+	uint32_t sample_count = 0;*/
+	uint32_t call_count = 0;
 };
 
 void on_process(local_data* ld, spa_io_position* position) {
-	float* in; float* out;
-	uint32_t n_samples = position->clock.duration;
+	int32_t n_samples = (int32_t)position->clock.duration;
 
-	in = (float*)pw_filter_get_dsp_buffer(ld->iport, n_samples);
-	out = (float*)pw_filter_get_dsp_buffer(ld->oport, n_samples);
+	float* mic_in = (float*)pw_filter_get_dsp_buffer(ld->mic_port, n_samples);
+	float* listen_in = (float*)pw_filter_get_dsp_buffer(ld->listen_port, n_samples);
+	float* hp_out = (float*)pw_filter_get_dsp_buffer(ld->hp_port, n_samples);
 
-	if(in == NULL || out == NULL) {return;}
-	
-	for(uint32_t i = 0; i < n_samples; ++i) {
-		auto val = ld->nd(ld->gen);
-		if(ld->buffer_index < 48000 * 10) {
-			ld->record_buffer[ld->buffer_index][0] = val;
-			ld->record_buffer[ld->buffer_index][1] = *in++;
-		}
+	if(mic_in == NULL || listen_in == NULL || hp_out == NULL) {return;}
 
-		*out++ = val;
+	// Simply echo the outgoing data
+	int copy_frames = std::min(samplerate*10 - ld->head_index, n_samples);
 
-		ld->buffer_index+=1;
-	}
+	std::memcpy(hp_out, listen_in, n_samples * sizeof(float));
 
-	if(ld->buffer_index >= 48000 * 10) {
+	std::memcpy(&ld->record_buffer[ld->head_index], listen_in, copy_frames*sizeof(float));
+	std::memcpy(&ld->mic_buffer[ld->head_index], mic_in, copy_frames*sizeof(float));
+
+	ld->head_index += copy_frames;
+	/*
+	// Now, copy data to the circular buffer
+	int32_t fst_cpy = std::min(n_samples, (int)(samplerate/10) - ld->head_index);
+	std::memcpy(&ld->record_buffer[ld->head_index], listen_in, fst_cpy * sizeof(float));
+	ld->head_index += fst_cpy;
+
+	int32_t snd_cpy = std::max(0, n_samples - fst_cpy);
+	std::memcpy(&ld->record_buffer[0], &listen_in[fst_cpy], snd_cpy * sizeof(float));
+	if(snd_cpy != 0) { ld->head_index = snd_cpy; }
+	*/
+
+	// To save on computation time, we randomly sample only one of the mic_in
+	// inputs.
+
+	/*
+	float input = mic_in[n_samples - 1];
+	int32_t head_pos = ld->head_index - 1;
+	if(head_pos == -1) {head_pos = samplerate/10 - 1;}
+
+	// Update the xcorr
+	for(int i = 0; i < samplerate / 10; ++i) {
+		ld->xcor_buffer[i] = (1 - 1.0/512)*ld->xcor_buffer[i]
+			+ 1.0/512 * input * ld->record_buffer[head_pos]; 
+		head_pos--;
+		if(head_pos == -1) {head_pos = samplerate/10 - 1;}
+	}*/
+
+	ld->call_count++;
+	if(ld->head_index == samplerate*10) {
 		pw_main_loop_quit(ld->loop);
 	}
 }
@@ -51,25 +84,24 @@ const struct pw_filter_events filter_events {
 	.process = (void(*)(void*, spa_io_position*))on_process,
 };
 
-float compute_RMS_power(float* buffer, int stride, int n_samps) {
+float compute_RMS_power(float* buffer, int n_samps) {
 	double acc = 0;
 	for(int i = 0; i < n_samps; ++i) {
-		acc += buffer[stride*i] * buffer[stride*i];
+		acc += buffer[i] * buffer[i];
 	}
 	return (float)std::sqrt(acc/n_samps);
 }
 
 // Delta positif -> retard du micro par rapport au HP -> physique.
-float avg_correlation(float buffer[][2], int delta, int n_samps) {
+float avg_correlation(float bufferHP[], float bufferMIC[], int delta, int n_samps) {
 	double acc = 0;
 	for(int i = delta; i < n_samps; ++i) {
-		acc += buffer[i - delta][0] * buffer[i][1];
+		acc += bufferHP[i - delta] * bufferMIC[i];
 	}
 	return acc / (n_samps - delta);
 }
 
 int main(int argc, char** argv) {
-	// We do NOT seed the MT engine -> the time series will be predictable!
 	local_data local{nullptr,nullptr,};
 
 	pw_init(&argc, &argv);
@@ -91,17 +123,17 @@ int main(int argc, char** argv) {
 		&filter_events,
 		&local);
 
-	local.iport = pw_filter_add_port(local.filter,
+	local.mic_port = pw_filter_add_port(local.filter,
 		PW_DIRECTION_INPUT,
 		PW_FILTER_PORT_FLAG_MAP_BUFFERS,
 		sizeof(void*),
 		pw_properties_new(
 			PW_KEY_FORMAT_DSP, "32 bit float mono audio",
-			PW_KEY_PORT_NAME, "input",
+			PW_KEY_PORT_NAME, "micinput",
 			NULL),
 		NULL, 0);
 
-	local.oport = pw_filter_add_port(local.filter,
+	local.hp_port = pw_filter_add_port(local.filter,
 		PW_DIRECTION_OUTPUT,
 		PW_FILTER_PORT_FLAG_MAP_BUFFERS,
 		sizeof(void*),
@@ -111,6 +143,16 @@ int main(int argc, char** argv) {
 			NULL),
 		NULL, 0);
 
+	local.listen_port = pw_filter_add_port(local.filter,
+		PW_DIRECTION_INPUT,
+		PW_FILTER_PORT_FLAG_MAP_BUFFERS,
+		sizeof(void*),
+		pw_properties_new(
+			PW_KEY_FORMAT_DSP, "32 bit float mono audio",
+			PW_KEY_PORT_NAME, "musinput",
+			NULL),
+		NULL, 0);
+
 	const spa_pod* params[2];
 	uint8_t buffer[1024];
 	spa_pod_builder b = SPA_POD_BUILDER_INIT(buffer, sizeof(buffer));
@@ -138,21 +180,16 @@ int main(int argc, char** argv) {
 	pw_main_loop_destroy(local.loop);
 	pw_deinit();
 
-	float power_input = compute_RMS_power(&local.record_buffer[0][0], 2, 48000*10);
-	float power_mic = compute_RMS_power(&local.record_buffer[0][1], 2, 48000*10);
-
-	std::printf("RMS Powers : Input %f, Mic. %f\n",
-		power_input,
-		power_mic);
+	std::printf("Call count : %u\n", local.call_count);
 
 	std::printf("XCor = [\n");
 	// Look at the first 200ms...
-	for(int delta = 0; delta < 48000 / 5; ++delta) {
-		float correct_crosscor = avg_correlation(&local.record_buffer[48000], delta, 48000*9);
-		correct_crosscor /= power_input * power_input;
+	for(int delta = 0; delta < 48000/5; ++delta) {
+		float correct_crosscor = avg_correlation(&local.record_buffer[48000], &local.mic_buffer[48000], delta, 48000*9);
 		// Normalizing -> if h(n) = del_0(n) then correct_crosscor = del_0.
 		std::printf("%e,", correct_crosscor);
 	}
 	std::printf("]\n");
+
 	return 0;
 }

pw_plugin/td_xcorr.cc

@@ -0,0 +1,158 @@
+#include <random>
+#include <cstdio>
+#include <cmath>
+#include <iostream>
+#include "pipewire/pipewire.h"
+#include "spa/param/audio/format-utils.h"
+#include "spa/param/latency-utils.h"
+
+struct local_data {
+	pw_main_loop* loop;
+	pw_filter* filter;
+
+	void* iport;
+	void* oport;
+
+	std::random_device gen;
+	std::normal_distribution<float> nd{0, 0.3};
+
+	uint32_t buffer_index = 0;
+	float record_buffer[48000 * 10][2]; // Forçage (to R.), retour (mic L)
+};
+
+void on_process(local_data* ld, spa_io_position* position) {
+	float* in; float* out;
+	uint32_t n_samples = position->clock.duration;
+
+	in = (float*)pw_filter_get_dsp_buffer(ld->iport, n_samples);
+	out = (float*)pw_filter_get_dsp_buffer(ld->oport, n_samples);
+
+	if(in == NULL || out == NULL) {return;}
+	
+	for(uint32_t i = 0; i < n_samples; ++i) {
+		auto val = ld->nd(ld->gen);
+		if(ld->buffer_index < 48000 * 10) {
+			ld->record_buffer[ld->buffer_index][0] = val;
+			ld->record_buffer[ld->buffer_index][1] = *in++;
+		}
+
+		*out++ = val;
+
+		ld->buffer_index+=1;
+	}
+
+	if(ld->buffer_index >= 48000 * 10) {
+		pw_main_loop_quit(ld->loop);
+	}
+}
+
+const struct pw_filter_events filter_events {
+	PW_VERSION_FILTER_EVENTS,
+	.process = (void(*)(void*, spa_io_position*))on_process,
+};
+
+float compute_RMS_power(float* buffer, int stride, int n_samps) {
+	double acc = 0;
+	for(int i = 0; i < n_samps; ++i) {
+		acc += buffer[stride*i] * buffer[stride*i];
+	}
+	return (float)std::sqrt(acc/n_samps);
+}
+
+// Delta positif -> retard du micro par rapport au HP -> physique.
+float avg_correlation(float buffer[][2], int delta, int n_samps) {
+	double acc = 0;
+	for(int i = delta; i < n_samps; ++i) {
+		acc += buffer[i - delta][0] * buffer[i][1];
+	}
+	return acc / (n_samps - delta);
+}
+
+int main(int argc, char** argv) {
+	// We do NOT seed the MT engine -> the time series will be predictable!
+	local_data local{nullptr,nullptr,};
+
+	pw_init(&argc, &argv);
+
+	local.loop = pw_main_loop_new(NULL);
+	if(local.loop == NULL) {
+		std::cerr << "Could not create loop!\n";
+		return 1;
+	}
+
+	local.filter = pw_filter_new_simple(
+		pw_main_loop_get_loop(local.loop),
+		"audio-filter",
+		pw_properties_new(
+			PW_KEY_MEDIA_TYPE, "Audio",
+			PW_KEY_MEDIA_CATEGORY, "Filter",
+			PW_KEY_MEDIA_ROLE, "DSP",
+			NULL),
+		&filter_events,
+		&local);
+
+	local.iport = pw_filter_add_port(local.filter,
+		PW_DIRECTION_INPUT,
+		PW_FILTER_PORT_FLAG_MAP_BUFFERS,
+		sizeof(void*),
+		pw_properties_new(
+			PW_KEY_FORMAT_DSP, "32 bit float mono audio",
+			PW_KEY_PORT_NAME, "input",
+			NULL),
+		NULL, 0);
+
+	local.oport = pw_filter_add_port(local.filter,
+		PW_DIRECTION_OUTPUT,
+		PW_FILTER_PORT_FLAG_MAP_BUFFERS,
+		sizeof(void*),
+		pw_properties_new(
+			PW_KEY_FORMAT_DSP, "32 bit float mono audio",
+			PW_KEY_PORT_NAME, "output",
+			NULL),
+		NULL, 0);
+
+	const spa_pod* params[2];
+	uint8_t buffer[1024];
+	spa_pod_builder b = SPA_POD_BUILDER_INIT(buffer, sizeof(buffer));
+
+	auto latinfo = SPA_PROCESS_LATENCY_INFO_INIT(.ns = 10 * SPA_NSEC_PER_MSEC);
+	auto formatinfo = SPA_AUDIO_INFO_RAW_INIT(.format = SPA_AUDIO_FORMAT_DSP_F32,
+		.rate = 48000,
+		.channels = 1);
+
+	params[0] = spa_process_latency_build(&b, SPA_PARAM_ProcessLatency, &latinfo);
+	params[1] = spa_format_audio_raw_build(&b, SPA_PARAM_EnumFormat, &formatinfo);
+
+	if(pw_filter_connect(local.filter,
+		PW_FILTER_FLAG_RT_PROCESS,
+		params, 1) < 0) {
+		std::fprintf(stderr, "Cannot connect\n");
+		return -1;
+	}
+
+	std::printf("Waiting for connection\n");
+	pw_main_loop_run(local.loop);
+	std::printf("Successfully recorded 10s\n");
+
+	pw_filter_destroy(local.filter);
+	pw_main_loop_destroy(local.loop);
+	pw_deinit();
+
+	float power_input = compute_RMS_power(&local.record_buffer[0][0], 2, 48000*10);
+	float power_mic = compute_RMS_power(&local.record_buffer[0][1], 2, 48000*10);
+
+	std::printf("RMS Powers : Input %f, Mic. %f\n",
+		power_input,
+		power_mic);
+
+	std::printf("XCor = [\n");
+	// Look at the first 200ms...
+	for(int delta = 0; delta < 48000 / 5; ++delta) {
+		float correct_crosscor = avg_correlation(&local.record_buffer[48000], delta, 48000*9);
+		correct_crosscor /= power_input * power_input;
+		// Normalizing -> if h(n) = del_0(n) then correct_crosscor = del_0.
+		std::printf("%e,", correct_crosscor);
+	}
+	std::printf("]\n");
+	return 0;
+}