diff --git a/meson.build b/meson.build
index 1437e59a..b29506bc 100644
--- a/meson.build
+++ b/meson.build
@@ -25,6 +25,7 @@ openrtx_src = ['openrtx/src/state.c',
                'openrtx/src/queue.c',
                'openrtx/src/rtx.c',
                'openrtx/src/gps.c',
+               'openrtx/src/dsp.cpp',
                'openrtx/src/memory_profiling.cpp']
 
 openrtx_inc = ['openrtx/include',
diff --git a/openrtx/include/dsp.h b/openrtx/include/dsp.h
new file mode 100644
index 00000000..f822239b
--- /dev/null
+++ b/openrtx/include/dsp.h
@@ -0,0 +1,75 @@
+/***************************************************************************
+ *   Copyright (C) 2020 by Federico Amedeo Izzo IU2NUO,                    *
+ *                         Niccolò Izzo IU2KIN,                            *
+ *                         Silvano Seva IU2KWO,                            *
+ *                         Frederik Saraci IU2NRO                          *
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 3 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ *   This program 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 General Public License for more details.                          *
+ *                                                                         *
+ *   You should have received a copy of the GNU General Public License     *
+ *   along with this program; if not, see <http://www.gnu.org/licenses/>   *
+ ***************************************************************************/
+
+#ifndef DSP_H
+#define DSP_H
+
+#include <inttypes.h>
+#include <stdlib.h>
+
+typedef int16_t audio_sample_t;
+
+/*
+ * This header contains various DSP utilities which can be used to condition
+ * input or output signals when implementing digital modes on OpenRTX.
+ */
+
+#ifdef __cplusplus
+#include <array>
+extern "C" {
+#endif
+
+/*
+ * Compensate for the filtering applied by the PWM output over the modulated
+ * signal. The buffer will be processed in place to save memory.
+ *
+ * @param buffer: the buffer to be used as both source and destination.
+ * @param length: the length of the input buffer.
+ */
+void dsp_pwmCompensate(audio_sample_t *buffer, uint16_t length);
+
+/*
+ * Remove any DC bias from the audio buffer passed as parameter.
+ * The buffer will be processed in place to save memory.
+ *
+ * @param buffer: the buffer to be used as both source and destination.
+ * @param length: the length of the input buffer.
+ */
+void dsp_dcRemoval(audio_sample_t *buffer, uint16_t length);
+
+#ifdef __cplusplus
+}
+
+/*
+ * Applies a generic FIR filter on the audio buffer passed as parameter.
+ * The buffer will be processed in place to save memory.
+ *
+ * @param buffer: the buffer to be used as both source and destination.
+ * @param length: the length of the input buffer.
+ * @param taps: an array of coefficients which defines the transfer function.
+ */
+template<size_t order>
+void dsp_applyFIR(audio_sample_t *buffer,
+                  uint16_t length,
+                  std::array<float, order> taps);
+
+#endif // __cplusplus
+
+#endif /* DSP_H */
diff --git a/openrtx/src/dsp.cpp b/openrtx/src/dsp.cpp
new file mode 100644
index 00000000..4dc04d69
--- /dev/null
+++ b/openrtx/src/dsp.cpp
@@ -0,0 +1,70 @@
+/***************************************************************************
+ *   Copyright (C) 2020 by Federico Amedeo Izzo IU2NUO,                    *
+ *                         Niccolò Izzo IU2KIN,                            *
+ *                         Silvano Seva IU2KWO,                            *
+ *                         Frederik Saraci IU2NRO                          *
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 3 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ *   This program 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 General Public License for more details.                          *
+ *                                                                         *
+ *   You should have received a copy of the GNU General Public License     *
+ *   along with this program; if not, see <http://www.gnu.org/licenses/>   *
+ ***************************************************************************/
+
+#include <dsp.h>
+
+/*
+ * Applies a generic FIR filter on the audio buffer passed as parameter.
+ * The buffer will be processed in place to save memory.
+ */
+template<size_t order>
+void dsp_applyFIR(audio_sample_t *buffer,
+                  uint16_t length,
+                  std::array<float, order> taps)
+{
+    for(int i = length - 1; i >= 0; i--) {
+        float acc = 0.0f;
+        for(uint16_t j = 0; j < order; j++) {
+            if (i >= j)
+                acc += buffer[i - j] * taps[j];
+        }
+        buffer[i] = (audio_sample_t) acc;
+    }
+}
+
+/*
+ * Compensate for the filtering applied by the PWM output over the modulated
+ * signal. The buffer will be processed in place to save memory.
+ */
+void dsp_pwmCompensate(audio_sample_t *buffer, uint16_t length)
+{
+    // FIR filter designed by Wojciech SP5WWP
+    std::array<float, 5> taps = { 0.01f, -0.05f, 0.88, -0.05f, 0.01f };
+    dsp_applyFIR(buffer, length, taps);
+}
+
+/*
+ * Remove any DC bias from the audio buffer passed as parameter.
+ * The buffer will be processed in place to save memory.
+ */
+void dsp_dcRemoval(audio_sample_t *buffer, uint16_t length)
+{
+    // Compute the average of all the samples
+    float acc = 0.0f;
+    for (int i = 0; i < length; i++) {
+        acc += buffer[i];
+    }
+    float mean = acc / (float) length;
+
+    // Subtract it to all the samples
+    for (int i = 0; i < length; i++) {
+        buffer[i] -= mean;
+    }
+}