diff --git a/similar/arch/sdl/digi_mixer.cpp b/similar/arch/sdl/digi_mixer.cpp
index efeef34a9..8ea92c49d 100644
--- a/similar/arch/sdl/digi_mixer.cpp
+++ b/similar/arch/sdl/digi_mixer.cpp
@@ -15,6 +15,7 @@
  */
 
 #include <bitset>
+#include <span>
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
@@ -189,12 +190,12 @@ static int32_t coeffs_halfband[FILTER_LEN] =
 
 // Fixed-point FIR filtering
 // Not optimal: consider optimization with 1/4, 1/2 band filters, and symmetric kernels
-static std::unique_ptr<int16_t[]> filter_fir(int16_t *signal, int signalLen, const int32_t (&coeffs)[FILTER_LEN])
+static std::unique_ptr<int16_t[]> filter_fir(const std::span<const int16_t> signal, const std::span<const int32_t, FILTER_LEN> coeffs)
 {
-	auto output = std::make_unique<int16_t[]>(signalLen);
+	auto output = std::make_unique<int16_t[]>(signal.size());
 	// A FIR filter is just a 1D convolution
 	// Keep only signalLen samples
-	for(int nn = 0; nn < signalLen; nn++)
+	for (const auto nn : xrange(signal.size()))
 	{
 		// Determine start/stop indices for convolved chunk
 		constexpr std::size_t coeffsLen = FILTER_LEN;
@@ -202,7 +203,7 @@ static std::unique_ptr<int16_t[]> filter_fir(int16_t *signal, int signalLen, con
 		 * unsigned subtraction to underflow.
 		 */
 		const std::size_t min_idx = (nn + 1 > coeffsLen ? nn + 1 - coeffsLen : 0u);
-		const std::size_t max_idx = std::min(nn, signalLen - 1);
+		const std::size_t max_idx = std::min(nn, signal.size() - 1);
 		if (min_idx > max_idx)
 			continue;
 
@@ -220,14 +221,14 @@ static std::unique_ptr<int16_t[]> filter_fir(int16_t *signal, int signalLen, con
 	return output;
 }
 
-static std::unique_ptr<int16_t[]> upsample(uint8_t *input, const std::size_t upsampledLen, int inputLen, int factor)
+static std::unique_ptr<int16_t[]> upsample(const std::span<const uint8_t> input, const std::size_t upsampledLen, const int factor)
 {
 	/* Caution: `output` is sparsely initialized, so the value-initialization
 	 * from `make_unique` is necessary.  This site cannot be converted to
 	 * `make_unique_for_overwrite`.
 	 */
 	auto output = std::make_unique<int16_t[]>(upsampledLen);
-	for(int ii = 0; ii < inputLen; ii++)
+	for (const auto ii : xrange(input.size()))
 	{
 		// Save input sample, convert to signed, and scale
 		output[ii*factor] = 256 * (int16_t(input[ii]) - 128);
@@ -235,10 +236,9 @@ static std::unique_ptr<int16_t[]> upsample(uint8_t *input, const std::size_t ups
 	return output;
 }
 
-
-static void replicateChannel(int16_t *input, int16_t *output, int inputLen, int chFactor)
+static void replicateChannel(const std::span<const int16_t> input, int16_t *const output, int chFactor)
 {
-	for(int ii = 0; ii < inputLen; ii++)
+	for (const auto ii : xrange(input.size()))
 	{
 		// Duplicate and interleave as many channels as needed
 		for(int jj = 0; jj < chFactor; jj++)
@@ -249,13 +249,11 @@ static void replicateChannel(int16_t *input, int16_t *output, int inputLen, int
 	}
 }
 
-
-static void convert_audio(uint8_t *input, int16_t *output, int inputLen, int upFactor, int chFactor)
+static void convert_audio(const std::span<const uint8_t> input, int16_t *const output, int upFactor, int chFactor)
 {
 	// First upsample
-	int upsampledLen = inputLen*upFactor;
-
-	auto stage1 = upsample(input, upsampledLen, inputLen, upFactor);
+	const auto upsampledLen = input.size() * upFactor;
+	auto stage1 = upsample(input, upsampledLen, upFactor);
 
 	// We expect a 4x upscaling 11025 -> 44100
 	// But maybe 2x for d2x in some cases
@@ -263,10 +261,10 @@ static void convert_audio(uint8_t *input, int16_t *output, int inputLen, int upF
 
 	// Apply LPF filter to smooth out upscaled points
 	// There will be some uniform amplitude loss here, but less than -3dB
-	auto stage2 = filter_fir(stage1.get(), upsampledLen, coeffs);
+	auto stage2 = filter_fir({stage1.get(), upsampledLen}, coeffs);
 
 	// Replicate channel
-	replicateChannel(stage2.get(), output, upsampledLen, chFactor);
+	replicateChannel({stage2.get(), upsampledLen}, output, chFactor);
 }
 
 }
@@ -310,7 +308,7 @@ static void mixdigi_convert_sound(const unsigned i)
 		int convertedSize = dlen * upFactor * out_channels * formatFactor;
 
 		auto cvtbuf = std::make_unique<Uint8[]>(convertedSize);
-		convert_audio(data, reinterpret_cast<int16_t*>(cvtbuf.get()), dlen, upFactor, out_channels);
+		convert_audio({data, dlen}, reinterpret_cast<int16_t*>(cvtbuf.get()), upFactor, out_channels);
 
 		SoundChunks[i].abuf = cvtbuf.release();
 		SoundChunks[i].alen = convertedSize;