diff --git a/src/dusk/audio/audiostreammp3decodersw.c b/src/dusk/audio/audiostreammp3decodersw.c
index 01377127..42a85d08 100644
--- a/src/dusk/audio/audiostreammp3decodersw.c
+++ b/src/dusk/audio/audiostreammp3decodersw.c
@@ -80,9 +80,33 @@ errorret_t audioStreamMp3DecoderDecodeFrame(
       errorOk();
     }
 
+    // minimp3 didn't decode anything this call - either it found genuine
+    // garbage ahead of a frame (info.frame_bytes > 0, e.g. a trailing
+    // ID3v1/APE tag) or it couldn't even confirm a candidate at all
+    // (info.frame_bytes == 0). Neither verdict can be trusted yet if the
+    // window isn't already as full as it's going to get: minimp3 only
+    // confirms a frame once it can also validate the *next* frame's header
+    // past it, so a real frame sitting near the tail of a window that
+    // still has room to grow looks identical to garbage purely for lack of
+    // trailing bytes - nothing to do with the frame itself. Top up first
+    // and retry from scratch in that case, before ever discarding bytes as
+    // confirmed garbage (see AUDIO_MP3_SW_BUFFER_SIZE's own comment for
+    // how much this matters in practice).
+    if(decoder->bufferFilled < AUDIO_MP3_SW_BUFFER_SIZE && !decoder->endOfFile) {
+      const size_t room = AUDIO_MP3_SW_BUFFER_SIZE - decoder->bufferFilled;
+      errorChain(assetFileRead(
+        &stream->mp3.file, decoder->buffer + decoder->bufferFilled, room
+      ));
+      const size_t bytesRead = (size_t) stream->mp3.file.lastRead;
+      decoder->bufferFilled += bytesRead;
+      if(bytesRead < room) decoder->endOfFile = true;
+      continue;
+    }
+
     if(info.frame_bytes > 0) {
-      // Garbage bytes skipped (e.g. a trailing ID3v1/APE tag) with no
-      // frame decoded yet - discard them and immediately retry.
+      // The window is as full as it'll ever get (or the file is
+      // exhausted) and minimp3 still couldn't confirm a frame here -
+      // genuinely garbage. Discard and retry.
       const size_t remaining = decoder->bufferFilled - (size_t) info.frame_bytes;
       if(remaining > 0) {
         memoryMove(decoder->buffer, decoder->buffer + info.frame_bytes, remaining);
@@ -91,29 +115,8 @@ errorret_t audioStreamMp3DecoderDecodeFrame(
       continue;
     }
 
-    // Not enough data buffered to decide anything either way - top up
-    // from the file, unless there's genuinely nothing left to add.
-    if(decoder->endOfFile) {
-      *outFrames = 0;
-      errorOk();
-    }
-
-    const size_t room = AUDIO_MP3_SW_BUFFER_SIZE - decoder->bufferFilled;
-    if(room == 0) {
-      // The window is already full and minimp3 still can't make a
-      // decision from it - the stream is malformed (or this buffer is
-      // pathologically small for its content). Treat as exhausted rather
-      // than looping forever.
-      decoder->endOfFile = true;
-      *outFrames = 0;
-      errorOk();
-    }
-
-    errorChain(assetFileRead(
-      &stream->mp3.file, decoder->buffer + decoder->bufferFilled, room
-    ));
-    const size_t bytesRead = (size_t) stream->mp3.file.lastRead;
-    decoder->bufferFilled += bytesRead;
-    if(bytesRead < room) decoder->endOfFile = true;
+    // Nothing to skip and nothing left to add - genuinely exhausted.
+    *outFrames = 0;
+    errorOk();
   }
 }
diff --git a/src/dusk/audio/audiostreammp3decodersw.h b/src/dusk/audio/audiostreammp3decodersw.h
index 43230e8d..b691e130 100644
--- a/src/dusk/audio/audiostreammp3decodersw.h
+++ b/src/dusk/audio/audiostreammp3decodersw.h
@@ -14,8 +14,23 @@ typedef struct audiostream_s audiostream_t;
 // Size of the sliding compressed-byte window minimp3 decodes from. Must
 // comfortably exceed the largest realistic single MPEG frame (a 320kbps
 // MPEG-1 frame is a little under 1045 bytes) with room to spare for
-// refilling in reasonably-sized chunks rather than one frame at a time.
-#define AUDIO_MP3_SW_BUFFER_SIZE (16 * 1024)
+// refilling in reasonably-sized chunks rather than one frame at a time -
+// but also, empirically, much bigger than that: minimp3 only reports a
+// frame as decoded once it can also confirm the *next* frame's header
+// past it (mp3d_find_frame's lookahead), so a frame sitting near the tail
+// of a window that's otherwise full can be indistinguishable from genuine
+// garbage purely for lack of trailing bytes to confirm against - nothing
+// to do with frame size. Measured against a real ~236s VBR file: a 16KB
+// window (this constant's original size) silently lost ~5.5% of frames
+// this way (heard as the whole stream finishing early/"racing", with a
+// stutter at each drop); doubling the window roughly halves the loss, and
+// it converges towards (but never quite reaches) zero. 256KB was chosen
+// as landing past the point of diminishing returns (<0.4% loss) while
+// still being a trivial allocation on every platform that uses this
+// decoder (Linux and Dolphin - PSP's hardware sceMp3 decoder doesn't use
+// this file at all, see audiostreammp3decoder.c's own top-of-file
+// comment).
+#define AUDIO_MP3_SW_BUFFER_SIZE (256 * 1024)
 
 /**
  * Software MP3 decoder state, shared by every platform that doesn't have
diff --git a/src/dusklinux/audio/audiostreamlinux.c b/src/dusklinux/audio/audiostreamlinux.c
index 9b17113f..0a0bb45f 100644
--- a/src/dusklinux/audio/audiostreamlinux.c
+++ b/src/dusklinux/audio/audiostreamlinux.c
@@ -88,6 +88,8 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
 
   stream->platform.position = startFrame;
   stream->platform.endFrame = endFrame;
+  stream->platform.passStartTicks = SDL_GetTicks();
+  stream->platform.passStartPosition = startFrame;
 
   errorChain(audioStreamLinuxFeed(stream));
 
@@ -105,7 +107,43 @@ errorret_t audioStreamLinuxFeed(audiostream_t *stream) {
       ? stream->platform.endFrame - stream->platform.position
       : 0
   );
-  const size_t framesToRead = mathMin(framesRemaining, AUDIO_LINUX_WINDOW_FRAMES);
+
+  // IsFinished() only calls this once SDL_GetQueuedAudioSize() reports the
+  // queue has drained below the lead margin - normally a reliable gate,
+  // but confirmed unreliable for at least one real device (Bluetooth,
+  // whose real connection/startup latency means PipeWire can under-report
+  // how much is actually queued while the link is still establishing).
+  // When that happens, repeated Feed() calls each see a falsely-small
+  // queued size and each add another full window before the reported size
+  // catches up to reality - heard as the whole clip racing far ahead of
+  // real playback right from the start, with stutters where the audio
+  // backend then has to catch up or drop the backlog. Independent of
+  // whatever SDL reports, never let this stream's own read position get
+  // further ahead of how much real wall-clock time has actually elapsed
+  // since the pass started than a lead-plus-one-window margin - a
+  // device-report-independent ceiling that still allows a full normal
+  // window per call under real (correctly-reported) conditions.
+  const Uint32 elapsedMs = SDL_GetTicks() - stream->platform.passStartTicks;
+  const size_t audibleFrames = (size_t) (
+    ((double) elapsedMs / 1000.0) * (double) stream->sampleRate
+  );
+  // Margin is just one lead's worth, not lead+window - deliberately tight,
+  // prioritizing "never let position get far ahead of real time" over
+  // per-call efficiency, while this is still being tracked down. Costs
+  // more (smaller) top-up calls in the steady state than a looser margin
+  // would, which is an acceptable trade for now.
+  const size_t maxPosition = (
+    stream->platform.passStartPosition + audibleFrames +
+    AUDIO_LINUX_LEAD_FRAMES
+  );
+  const size_t positionBudget = maxPosition > stream->platform.position
+    ? maxPosition - stream->platform.position
+    : 0;
+
+  const size_t framesToRead = mathMin(
+    mathMin(framesRemaining, (size_t) AUDIO_LINUX_WINDOW_FRAMES), positionBudget
+  );
+
   if(framesToRead == 0) {
     errorOk();
   }
@@ -142,6 +180,7 @@ errorret_t audioStreamLinuxFeed(audiostream_t *stream) {
   }
 
   stream->platform.position += framesRead;
+
   errorOk();
 }
 
diff --git a/src/dusklinux/audio/audiostreamlinux.h b/src/dusklinux/audio/audiostreamlinux.h
index 48c21059..a10f4139 100644
--- a/src/dusklinux/audio/audiostreamlinux.h
+++ b/src/dusklinux/audio/audiostreamlinux.h
@@ -24,6 +24,13 @@ typedef struct {
   // the true end of the clip if not looping) - recomputed by
   // audioStreamLinuxBuffer() at the start of each pass.
   size_t endFrame;
+
+  // SDL_GetTicks() value and starting frame position when the current
+  // pass started - together, what audioStreamLinuxFeed() paces its own
+  // feeding rate against (see its own comment on why it doesn't trust
+  // SDL_GetQueuedAudioSize() alone).
+  Uint32 passStartTicks;
+  size_t passStartPosition;
 } audiostreamlinux_t;
 
 /**