diff --git a/src/decoder/Bridge.cxx b/src/decoder/Bridge.cxx
index 33f06adda..b4d9fd076 100644
--- a/src/decoder/Bridge.cxx
+++ b/src/decoder/Bridge.cxx
@@ -34,7 +34,6 @@
 #include "input/cache/Manager.hxx"
 #include "input/cache/Stream.hxx"
 #include "fs/Path.hxx"
-#include "util/ConstBuffer.hxx"
 #include "util/StringBuffer.hxx"
 
 #include <cassert>
diff --git a/src/decoder/plugins/FaadDecoderPlugin.cxx b/src/decoder/plugins/FaadDecoderPlugin.cxx
index 0a4eb7530..357043ca0 100644
--- a/src/decoder/plugins/FaadDecoderPlugin.cxx
+++ b/src/decoder/plugins/FaadDecoderPlugin.cxx
@@ -24,9 +24,9 @@
 #include "pcm/CheckAudioFormat.hxx"
 #include "tag/Handler.hxx"
 #include "util/ScopeExit.hxx"
-#include "util/ConstBuffer.hxx"
 #include "util/Domain.hxx"
 #include "util/Math.hxx"
+#include "util/SpanCast.hxx"
 #include "Log.hxx"
 
 #include <cassert>
@@ -65,27 +65,28 @@ static size_t
 adts_find_frame(DecoderBuffer &buffer)
 {
 	while (true) {
-		auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Need(8));
-		if (data.IsNull())
+		auto data = FromBytesStrict<const uint8_t>(buffer.Need(8));
+		if (data.data() == nullptr)
 			/* failed */
 			return 0;
 
 		/* find the 0xff marker */
-		auto p = (const uint8_t *)std::memchr(data.data, 0xff, data.size);
+		auto p = (const uint8_t *)std::memchr(data.data(), 0xff,
+						      data.size());
 		if (p == nullptr) {
 			/* no marker - discard the buffer */
 			buffer.Clear();
 			continue;
 		}
 
-		if (p > data.data) {
+		if (p > data.data()) {
 			/* discard data before 0xff */
-			buffer.Consume(p - data.data);
+			buffer.Consume(p - data.data());
 			continue;
 		}
 
 		/* is it a frame? */
-		const size_t frame_length = adts_check_frame(data.data);
+		const size_t frame_length = adts_check_frame(data.data());
 		if (frame_length == 0) {
 			/* it's just some random 0xff byte; discard it
 			   and continue searching */
@@ -123,11 +124,11 @@ adts_song_duration(DecoderBuffer &buffer)
 			break;
 
 		if (frames == 0) {
-			auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Read());
+			auto data = FromBytesStrict<const uint8_t>(buffer.Read());
 			assert(!data.empty());
-			assert(frame_length <= data.size);
+			assert(frame_length <= data.size());
 
-			sample_rate = adts_sample_rates[(data.data[2] & 0x3c) >> 2];
+			sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2];
 			if (sample_rate == 0)
 				break;
 		}
@@ -162,28 +163,28 @@ adts_song_duration(DecoderBuffer &buffer)
 static SignedSongTime
 faad_song_duration(DecoderBuffer &buffer, InputStream &is)
 {
-	auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Need(5));
-	if (data.IsNull())
+	auto data = FromBytesStrict<const uint8_t>(buffer.Need(5));
+	if (data.data() == nullptr)
 		return SignedSongTime::Negative();
 
 	size_t tagsize = 0;
-	if (data.size >= 10 && !memcmp(data.data, "ID3", 3)) {
+	if (data.size() >= 10 && !memcmp(data.data(), "ID3", 3)) {
 		/* skip the ID3 tag */
 
-		tagsize = (data.data[6] << 21) | (data.data[7] << 14) |
-		    (data.data[8] << 7) | (data.data[9] << 0);
+		tagsize = (data[6] << 21) | (data[7] << 14) |
+		    (data[8] << 7) | (data[9] << 0);
 
 		tagsize += 10;
 
 		if (!buffer.Skip(tagsize))
 			return SignedSongTime::Negative();
 
-		data = ConstBuffer<uint8_t>::FromVoid(buffer.Need(5));
-		if (data.IsNull())
+		data = FromBytesStrict<const uint8_t>(buffer.Need(5));
+		if (data.data() == nullptr)
 			return SignedSongTime::Negative();
 	}
 
-	if (data.size >= 8 && adts_check_frame(data.data) > 0) {
+	if (data.size() >= 8 && adts_check_frame(data.data()) > 0) {
 		/* obtain the duration from the ADTS header */
 
 		if (!is.IsSeekable())
@@ -199,23 +200,23 @@ faad_song_duration(DecoderBuffer &buffer, InputStream &is)
 		buffer.Clear();
 
 		return song_length;
-	} else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) {
+	} else if (data.size() >= 5 && memcmp(data.data(), "ADIF", 4) == 0) {
 		/* obtain the duration from the ADIF header */
 
 		if (!is.KnownSize())
 			return SignedSongTime::Negative();
 
-		size_t skip_size = (data.data[4] & 0x80) ? 9 : 0;
+		size_t skip_size = (data[4] & 0x80) ? 9 : 0;
 
-		if (8 + skip_size > data.size)
+		if (8 + skip_size > data.size())
 			/* not enough data yet; skip parsing this
 			   header */
 			return SignedSongTime::Negative();
 
-		unsigned bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) |
-			(data.data[5 + skip_size] << 11) |
-			(data.data[6 + skip_size] << 3) |
-			(data.data[7 + skip_size] & 0xE0);
+		unsigned bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
+			(data[5 + skip_size] << 11) |
+			(data[6 + skip_size] << 3) |
+			(data[7 + skip_size] & 0xE0);
 
 		const auto size = is.GetSize();
 		if (bit_rate == 0)
@@ -251,7 +252,7 @@ static void
 faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer &buffer,
 		  AudioFormat &audio_format)
 {
-	auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Read());
+	auto data = FromBytesStrict<const uint8_t>(buffer.Read());
 	if (data.empty())
 		throw std::runtime_error("Empty file");
 
@@ -259,8 +260,8 @@ faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer &buffer,
 	unsigned long sample_rate;
 	long nbytes = NeAACDecInit(decoder,
 				   /* deconst hack, libfaad requires this */
-				   const_cast<unsigned char *>(data.data),
-				   data.size,
+				   const_cast<unsigned char *>(data.data()),
+				   data.size(),
 				   &sample_rate, &channels);
 	if (nbytes < 0)
 		throw std::runtime_error("Not an AAC stream");
@@ -279,14 +280,14 @@ static const void *
 faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer &buffer,
 		    NeAACDecFrameInfo *frame_info)
 {
-	auto data = ConstBuffer<uint8_t>::FromVoid(buffer.Read());
+	auto data = FromBytesStrict<const uint8_t>(buffer.Read());
 	if (data.empty())
 		return nullptr;
 
 	return NeAACDecDecode(decoder, frame_info,
 			      /* deconst hack, libfaad requires this */
-			      const_cast<uint8_t *>(data.data),
-			      data.size);
+			      const_cast<uint8_t *>(data.data()),
+			      data.size());
 }
 
 /**
diff --git a/src/decoder/plugins/FfmpegDecoderPlugin.cxx b/src/decoder/plugins/FfmpegDecoderPlugin.cxx
index 87e80289e..384e31fcc 100644
--- a/src/decoder/plugins/FfmpegDecoderPlugin.cxx
+++ b/src/decoder/plugins/FfmpegDecoderPlugin.cxx
@@ -42,7 +42,6 @@
 #include "input/InputStream.hxx"
 #include "pcm/CheckAudioFormat.hxx"
 #include "util/ScopeExit.hxx"
-#include "util/ConstBuffer.hxx"
 #include "util/StringAPI.hxx"
 #include "Log.hxx"
 
@@ -216,23 +215,20 @@ FfmpegSendFrame(DecoderClient &client, InputStream *is,
 		size_t &skip_bytes,
 		FfmpegBuffer &buffer)
 {
-	ConstBuffer<void> output_buffer =
-		Ffmpeg::InterleaveFrame(frame, buffer);
+	auto output_buffer = Ffmpeg::InterleaveFrame(frame, buffer);
 
 	if (skip_bytes > 0) {
-		if (skip_bytes >= output_buffer.size) {
-			skip_bytes -= output_buffer.size;
+		if (skip_bytes >= output_buffer.size()) {
+			skip_bytes -= output_buffer.size();
 			return DecoderCommand::NONE;
 		}
 
-		output_buffer.data =
-			(const uint8_t *)output_buffer.data + skip_bytes;
-		output_buffer.size -= skip_bytes;
+		output_buffer = output_buffer.subspan(skip_bytes);
 		skip_bytes = 0;
 	}
 
 	return client.SubmitData(is,
-				 output_buffer.data, output_buffer.size,
+				 output_buffer.data(), output_buffer.size(),
 				 codec_context.bit_rate / 1000);
 }
 
diff --git a/src/decoder/plugins/FlacCommon.hxx b/src/decoder/plugins/FlacCommon.hxx
index 533c38307..f5c5a5809 100644
--- a/src/decoder/plugins/FlacCommon.hxx
+++ b/src/decoder/plugins/FlacCommon.hxx
@@ -27,10 +27,12 @@
 #include "FlacInput.hxx"
 #include "FlacPcm.hxx"
 #include "../DecoderAPI.hxx"
-#include "util/ConstBuffer.hxx"
 
 #include <FLAC/stream_decoder.h>
 
+#include <cstddef>
+#include <span>
+
 struct FlacDecoder : public FlacInput {
 	/**
 	 * Has DecoderClient::Ready() been called yet?
@@ -63,7 +65,7 @@ struct FlacDecoder : public FlacInput {
 	 * If this is non-empty, then DecoderBridge::SubmitData()
 	 * should be called.
 	 */
-	ConstBuffer<void> chunk = nullptr;
+	std::span<const std::byte> chunk = {};
 
 	FlacDecoder(DecoderClient &_client, InputStream &_input_stream)
 		:FlacInput(_input_stream, &_client) {}
diff --git a/src/decoder/plugins/FlacDecoderPlugin.cxx b/src/decoder/plugins/FlacDecoderPlugin.cxx
index 80d124127..d2e84b388 100644
--- a/src/decoder/plugins/FlacDecoderPlugin.cxx
+++ b/src/decoder/plugins/FlacDecoderPlugin.cxx
@@ -153,10 +153,10 @@ FlacSubmitToClient(DecoderClient &client, FlacDecoder &d) noexcept
 
 	if (!d.chunk.empty()) {
 		auto cmd = client.SubmitData(d.GetInputStream(),
-					     d.chunk.data,
-					     d.chunk.size,
+					     d.chunk.data(),
+					     d.chunk.size(),
 					     d.kbit_rate);
-		d.chunk = nullptr;
+		d.chunk = {};
 		if (cmd != DecoderCommand::NONE)
 			return cmd;
 	}
diff --git a/src/decoder/plugins/FlacPcm.cxx b/src/decoder/plugins/FlacPcm.cxx
index 94e4c2ac2..9d4d0435c 100644
--- a/src/decoder/plugins/FlacPcm.cxx
+++ b/src/decoder/plugins/FlacPcm.cxx
@@ -21,7 +21,6 @@
 #include "pcm/CheckAudioFormat.hxx"
 #include "lib/xiph/FlacAudioFormat.hxx"
 #include "util/RuntimeError.hxx"
-#include "util/ConstBuffer.hxx"
 
 #include <cassert>
 
@@ -69,7 +68,7 @@ FlacImport(T *dest, const FLAC__int32 *const src[], size_t n_frames,
 }
 
 template<typename T>
-static ConstBuffer<void>
+static std::span<const std::byte>
 FlacImport(PcmBuffer &buffer, const FLAC__int32 *const src[], size_t n_frames,
 	   unsigned n_channels)
 {
@@ -77,10 +76,10 @@ FlacImport(PcmBuffer &buffer, const FLAC__int32 *const src[], size_t n_frames,
 	size_t dest_size = n_samples * sizeof(T);
 	T *dest = (T *)buffer.Get(dest_size);
 	FlacImport(dest, src, n_frames, n_channels);
-	return {dest, dest_size};
+	return std::as_bytes(std::span{dest, n_samples});
 }
 
-ConstBuffer<void>
+std::span<const std::byte>
 FlacPcmImport::Import(const FLAC__int32 *const src[], size_t n_frames)
 {
 	switch (audio_format.format) {
diff --git a/src/decoder/plugins/FlacPcm.hxx b/src/decoder/plugins/FlacPcm.hxx
index ea0363d04..3f84a3b15 100644
--- a/src/decoder/plugins/FlacPcm.hxx
+++ b/src/decoder/plugins/FlacPcm.hxx
@@ -25,7 +25,8 @@
 
 #include <FLAC/ordinals.h>
 
-template<typename T> struct ConstBuffer;
+#include <cstddef>
+#include <span>
 
 /**
  * This class imports libFLAC PCM data into a PCM format supported by
@@ -47,8 +48,8 @@ public:
 		return audio_format;
 	}
 
-	ConstBuffer<void> Import(const FLAC__int32 *const src[],
-				 size_t n_frames);
+	std::span<const std::byte> Import(const FLAC__int32 *const src[],
+					  size_t n_frames);
 };
 
 #endif
diff --git a/src/lib/ffmpeg/Interleave.cxx b/src/lib/ffmpeg/Interleave.cxx
index bb83081ed..f391e3f39 100644
--- a/src/lib/ffmpeg/Interleave.cxx
+++ b/src/lib/ffmpeg/Interleave.cxx
@@ -32,7 +32,7 @@ extern "C" {
 
 namespace Ffmpeg {
 
-ConstBuffer<void>
+std::span<const std::byte>
 InterleaveFrame(const AVFrame &frame, FfmpegBuffer &buffer)
 {
 	assert(frame.nb_samples > 0);
@@ -49,9 +49,9 @@ InterleaveFrame(const AVFrame &frame, FfmpegBuffer &buffer)
 	if (data_size < 0)
 		throw MakeFfmpegError(data_size);
 
-	void *output_buffer;
+	std::byte *output_buffer;
 	if (av_sample_fmt_is_planar(format) && channels > 1) {
-		output_buffer = buffer.GetT<uint8_t>(data_size);
+		output_buffer = buffer.GetT<std::byte>(data_size);
 		if (output_buffer == nullptr)
 			/* Not enough memory - shouldn't happen */
 			throw std::bad_alloc();
@@ -61,7 +61,7 @@ InterleaveFrame(const AVFrame &frame, FfmpegBuffer &buffer)
 			      n_frames,
 			      av_get_bytes_per_sample(format));
 	} else {
-		output_buffer = frame.extended_data[0];
+		output_buffer = (std::byte *)frame.extended_data[0];
 	}
 
 	return { output_buffer, (size_t)data_size };
diff --git a/src/lib/ffmpeg/Interleave.hxx b/src/lib/ffmpeg/Interleave.hxx
index 33d0c0312..b2bf59e30 100644
--- a/src/lib/ffmpeg/Interleave.hxx
+++ b/src/lib/ffmpeg/Interleave.hxx
@@ -20,8 +20,9 @@
 #ifndef MPD_FFMPEG_INTERLEAVE_HXX
 #define MPD_FFMPEG_INTERLEAVE_HXX
 
+#include <span>
+
 struct AVFrame;
-template<typename T> struct ConstBuffer;
 class FfmpegBuffer;
 
 namespace Ffmpeg {
@@ -32,7 +33,7 @@ namespace Ffmpeg {
  *
  * Throws on error.
  */
-ConstBuffer<void>
+std::span<const std::byte>
 InterleaveFrame(const AVFrame &frame, FfmpegBuffer &buffer);
 
 } // namespace Ffmpeg