From 26defddd8281f81027f3ebfbdb3734de38803cc3 Mon Sep 17 00:00:00 2001
From: KubaPro010 <kuba.pro010@gmail.com>
Date: Sat, 4 Jan 2025 17:41:42 +0100
Subject: [PATCH] add MonoPass (great name)

---
 README.md              |   3 +
 src/mono_passthrough.c | 162 +++++++++++++++++++++++++++++++++++++++++
 2 files changed, 165 insertions(+)
 create mode 100644 src/mono_passthrough.c

diff --git a/README.md b/README.md
index a7239fd..77b7e06 100644
--- a/README.md
+++ b/README.md
@@ -41,6 +41,9 @@ Has a fine quality, but as it goes for 12 khz fm signals
 # StereoSCAMod
 Stereo SCA, like normal SCA but encodes L-R onto 80 khz, only demodulator of this right now is gnu radio
 
+# MonoPass
+want to keep mono for a reason but have the lpf and preemphasis, do so
+
 # How to compile?
 To compile you need `cmake` and `libpulse-dev`, if you have those then do these commands:
 ```
diff --git a/src/mono_passthrough.c b/src/mono_passthrough.c
new file mode 100644
index 0000000..fd91f46
--- /dev/null
+++ b/src/mono_passthrough.c
@@ -0,0 +1,162 @@
+#include <stdio.h>
+#include <pulse/simple.h>
+#include <pulse/error.h>
+#include <stdlib.h>
+#include <math.h>
+#include <stdint.h>
+#include <signal.h>
+#include <string.h>
+
+#include "../lib/constants.h"
+#include "../lib/filters.h"
+
+#include "options.h"
+
+#define SAMPLE_RATE 16000
+
+#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
+#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
+#define BUFFER_SIZE 512
+#define CLIPPER_THRESHOLD 0.525 // Adjust this as needed
+
+#define MONO_VOLUME 0.6f // L+R Signal
+
+#ifdef PREEMPHASIS
+#define PREEMPHASIS_TAU 0.00005  // 50 microseconds, use 0.000075 if in america
+#endif
+
+#ifdef LPF
+#define LPF_CUTOFF 15000
+#endif
+
+volatile sig_atomic_t to_run = 1;
+
+float clip(float sample) {
+    if (sample > CLIPPER_THRESHOLD) {
+        return CLIPPER_THRESHOLD;  // Clip to the upper threshold
+    } else if (sample < -CLIPPER_THRESHOLD) {
+        return -CLIPPER_THRESHOLD;  // Clip to the lower threshold
+    } else {
+        return sample;  // No clipping
+    }
+}
+
+static void stop(int signum) {
+    (void)signum;
+    printf("\nReceived stop signal. Cleaning up...\n");
+    to_run = 0;
+}
+
+int main() {
+    printf("MonoPass : Mono filter for FM made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
+    // Define formats and buffer atributes
+    pa_sample_spec audio_format = {
+        .format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64
+        .channels = 1,
+        .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
+    };
+
+    pa_buffer_attr input_buffer_atr = {
+        .maxlength = buffer_maxlength,
+	    .fragsize = buffer_tlength_fragsize
+    };
+    pa_buffer_attr output_buffer_atr = {
+        .maxlength = buffer_maxlength,
+        .tlength = buffer_tlength_fragsize,
+	    .prebuf = buffer_prebuf
+    };
+
+    printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
+
+    pa_simple *input_device = pa_simple_new(
+        NULL,
+        "MonoPass",
+        PA_STREAM_RECORD,
+        INPUT_DEVICE,
+        "Audio Input",
+        &audio_format,
+        NULL,
+        &input_buffer_atr,
+        NULL
+    );
+    if (!input_device) {
+        fprintf(stderr, "Error: cannot open input device.\n");
+        return 1;
+    }
+
+    printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
+
+    pa_simple *output_device = pa_simple_new(
+        NULL,
+        "MonoPass",
+        PA_STREAM_PLAYBACK,
+        OUTPUT_DEVICE,
+        "Audio",
+        &audio_format,
+        NULL,
+        &output_buffer_atr,
+        NULL
+    );
+    if (!output_device) {
+        fprintf(stderr, "Error: cannot open output device.\n");
+        pa_simple_free(input_device);
+        return 1;
+    }
+
+#ifdef PREEMPHASIS
+    Emphasis preemp;
+    init_emphasis(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE);
+#endif
+#ifdef LPF
+    LowPassFilter lpf;
+    init_low_pass_filter(&lpf, LPF_CUTOFF, SAMPLE_RATE);
+#endif
+
+    signal(SIGINT, stop);
+    signal(SIGTERM, stop);
+    
+    int pulse_error;
+    float input[BUFFER_SIZE]; // Input from device, interleaved stereo
+    float mpx[BUFFER_SIZE]; // MPX, this goes to the output
+    while (to_run) {
+        if (pa_simple_read(input_device, input, sizeof(input), &pulse_error) < 0) {
+            fprintf(stderr, "Error reading from input device: %s\n", pa_strerror(pulse_error));
+            to_run = 0;
+            break;
+        }
+    
+        for (int i = 0; i < BUFFER_SIZE; i++) {
+            float in = input[i];
+
+#ifdef PREEMPHASIS
+#ifdef LPF
+            float lowpassed = apply_low_pass_filter(&lpf, in);
+            float preemphasized = apply_pre_emphasis(&preemp, lowpassed);
+            float current_input = clip(preemphasized);
+#else
+            float preemphasized = apply_pre_emphasis(&preemp, in);
+            float current_input = clip(preemphasized);
+#endif
+#else
+#ifdef LPF
+            float lowpassed = apply_low_pass_filter(&lpf, in);
+            float current_input = clip(lowpassed);
+#else
+            float current_input = clip(in);
+#endif
+#endif
+
+            mpx[i] = current_input * MONO_VOLUME;
+        }
+
+        if (pa_simple_write(output_device, mpx, sizeof(mpx), &pulse_error) < 0) {
+            fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error));
+            to_run = 0;
+            break;
+        }
+    }
+    printf("Cleaning up...\n");
+    pa_simple_free(input_device);
+    pa_simple_free(output_device);
+    return 0;
+}