diff --git a/src/CMakeLists.txt b/src/CMakeLists.txt
index babd21c6..798da385 100644
--- a/src/CMakeLists.txt
+++ b/src/CMakeLists.txt
@@ -13,6 +13,7 @@ if(DUSK_TARGET_SYSTEM STREQUAL "linux" OR DUSK_TARGET_SYSTEM STREQUAL "knulli")
   add_subdirectory(dusklinux)
   add_subdirectory(dusksdl2)
   add_subdirectory(duskgl)
+  add_subdirectory(duskmad)
 
 elseif(DUSK_TARGET_SYSTEM STREQUAL "psp")
   add_subdirectory(duskpsp)
@@ -26,5 +27,6 @@ elseif(DUSK_TARGET_SYSTEM STREQUAL "vita")
 
 elseif(DUSK_TARGET_SYSTEM STREQUAL "wii" OR DUSK_TARGET_SYSTEM STREQUAL "gamecube")
   add_subdirectory(duskdolphin)
-  
+  add_subdirectory(duskmad)
+
 endif()
\ No newline at end of file
diff --git a/src/dusk/audio/stream/audiostream.c b/src/dusk/audio/stream/audiostream.c
index ed3d55f9..550faabb 100644
--- a/src/dusk/audio/stream/audiostream.c
+++ b/src/dusk/audio/stream/audiostream.c
@@ -7,6 +7,7 @@
 
 #include "audiostream.h"
 #include "assert/assert.h"
+#include "util/math.h"
 
 errorret_t audioStreamInit(audiostream_t *stream, assetentry_t *asset) {
   assertNotNull(stream, "Stream cannot be NULL.");
@@ -154,6 +155,24 @@ void audioStreamGetPanFactors(
   *outRight = pan < 0 ? (1.0f + pan) : 1.0f;
 }
 
+size_t audioStreamComputeEndFrame(
+  const audiostream_t *stream,
+  const size_t startFrame,
+  const size_t totalFrames
+) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+
+  size_t endFrame = totalFrames;
+  if((stream->state & AUDIO_STREAM_STATE_LOOPING) && stream->loopStart >= 0) {
+    endFrame = mathMin(
+      (size_t) (stream->loopStart * stream->sampleRate), totalFrames
+    );
+  }
+  if(startFrame >= endFrame) endFrame = totalFrames;
+
+  return endFrame;
+}
+
 void audioStreamSetVolume(audiostream_t *stream, const uint8_t volume) {
   assertNotNull(stream, "Stream cannot be NULL.");
   stream->volume = volume;
diff --git a/src/dusk/audio/stream/audiostream.h b/src/dusk/audio/stream/audiostream.h
index 1ef4c342..e7eb4371 100644
--- a/src/dusk/audio/stream/audiostream.h
+++ b/src/dusk/audio/stream/audiostream.h
@@ -261,6 +261,27 @@ void audioStreamGetPanFactors(
   float_t *outRight
 );
 
+/**
+ * Computes the stop frame for a playback pass starting at startFrame: the
+ * current loop segment's end (stream->loopStart, converted to frames and
+ * clamped to totalFrames) if looping is enabled and a loop start is set,
+ * or totalFrames itself otherwise - including when startFrame has already
+ * reached or passed the computed loop end (e.g. a seek landing in an
+ * outro past the loop point), in which case this pass plays out to the
+ * true end of the stream once instead of only the loop segment.
+ *
+ * @param stream The audio stream to compute this for.
+ * @param startFrame This pass's starting frame offset.
+ * @param totalFrames The stream's total decoded frame count
+ *                     (audioStreamGetTotalFrames()).
+ * @return This pass's stop frame offset.
+ */
+size_t audioStreamComputeEndFrame(
+  const audiostream_t *stream,
+  const size_t startFrame,
+  const size_t totalFrames
+);
+
 /**
  * Sets the playback volume of the given audio stream.
  *
diff --git a/src/dusk/thread/thread.c b/src/dusk/thread/thread.c
index 96e928a5..f0d09195 100644
--- a/src/dusk/thread/thread.c
+++ b/src/dusk/thread/thread.c
@@ -109,9 +109,9 @@ bool_t threadShouldStop(thread_t *thread) {
     
     threadMutexLock(&thread->stateMutex);
     thread->state = THREAD_STATE_STOPPED;
+    thread->threadId = 0;
     threadMutexSignal(&thread->stateMutex);
     threadMutexUnlock(&thread->stateMutex);
-    thread->threadId = 0;
     return NULL;
   }
 #endif
\ No newline at end of file
diff --git a/src/duskdolphin/audio/CMakeLists.txt b/src/duskdolphin/audio/CMakeLists.txt
index 2fc45395..a692cdb5 100644
--- a/src/duskdolphin/audio/CMakeLists.txt
+++ b/src/duskdolphin/audio/CMakeLists.txt
@@ -11,15 +11,7 @@ target_sources(${DUSK_LIBRARY_TARGET_NAME}
 )
 
 # No hardware MP3 decoder used on Dolphin (see audiostreammp3decoder.h's
-# own comment on why libogc's MP3Player wrapper isn't a fit) - use the
-# same shared libmad-based software backend as Linux, sourced directly
-# from src/dusk/audio rather than through its own unconditional
-# CMakeLists.txt so platforms with a hardware decoder (PSP) never pull
-# libmad in at all. Unlike Linux, libmad here comes bundled with the
-# devkitPPC/libogc toolchain itself (see cmake/targets/dolphin.cmake's
-# own "mad" link entry) rather than via find_package() - unnecessary
-# for, and not reliable when cross-compiling.
-target_sources(${DUSK_LIBRARY_TARGET_NAME}
-  PUBLIC
-    "${DUSK_SOURCES_DIR}/dusk/audio/stream/audiostreammp3decodersw.c"
-)
+# own comment on why libogc's MP3Player wrapper isn't a fit) - uses the
+# shared libmad-based software backend + decode-ahead ring in
+# src/duskmad instead (also used by Linux), see that directory's own
+# CMakeLists.txt for how it's sourced/linked.
diff --git a/src/duskdolphin/audio/audiostreamdolphin.c b/src/duskdolphin/audio/audiostreamdolphin.c
index 80d9fa76..a54ffbab 100644
--- a/src/duskdolphin/audio/audiostreamdolphin.c
+++ b/src/duskdolphin/audio/audiostreamdolphin.c
@@ -35,12 +35,21 @@ errorret_t audioStreamDolphinInit(audiostream_t *stream) {
   stream->platform.volumeLeft = 0;
   stream->platform.volumeRight = 0;
 
+  errorChain(audioStreamMp3RingInitIfNeeded(stream));
+
   errorOk();
 }
 
 errorret_t audioStreamDolphinDispose(audiostream_t *stream) {
   assertNotNull(stream, "Stream cannot be NULL.");
 
+  // Must happen first: audioStreamMp3Dispose() (called right after this
+  // function returns, from the shared audioStreamDispose()) tears down
+  // stream->mp3's decoder/file - the decode thread must be fully parked
+  // before that, not just before this function's own ASND/buffer cleanup
+  // below.
+  errorChain(audioStreamMp3RingDisposeIfNeeded(stream));
+
   ASND_StopVoice(stream->platform.voiceId);
   memoryFree(stream->platform.buffer[0]);
   memoryFree(stream->platform.buffer[1]);
@@ -53,25 +62,15 @@ errorret_t audioStreamDolphinBuffer(audiostream_t *stream) {
 
   const size_t totalFrames = audioStreamGetTotalFrames(stream);
 
-  // endFrame is this pass's stop point - the current loop segment's end,
-  // or the true end of the clip if not looping. Same math as PSP/Linux -
-  // see their own comments.
-  size_t endFrame = totalFrames;
-  if((stream->state & AUDIO_STREAM_STATE_LOOPING) && stream->loopStart >= 0) {
-    endFrame = mathMin(
-      (size_t) (stream->loopStart * stream->sampleRate), totalFrames
-    );
-  }
-
   const size_t startFrame = mathMin(stream->startFrame, totalFrames);
   stream->startFrame = 0;
   stream->seeking = false;
 
-  // A seek (or a loop restart landing exactly on the loop end) can put
-  // startFrame at or past endFrame - e.g. seeking into an outro after the
-  // loop point. Play out to the true end of the buffer once instead, same
-  // as PSP/Linux.
-  if(startFrame >= endFrame) endFrame = totalFrames;
+  // This pass's stop point - the current loop segment's end, or the true
+  // end of the clip if not looping (or if startFrame already landed past
+  // the loop end, e.g. a seek into an outro). See
+  // audioStreamComputeEndFrame()'s own comment.
+  const size_t endFrame = audioStreamComputeEndFrame(stream, startFrame, totalFrames);
 
   float_t leftFactor, rightFactor;
   audioStreamGetPanFactors(stream, &leftFactor, &rightFactor);
@@ -89,16 +88,32 @@ errorret_t audioStreamDolphinBuffer(audiostream_t *stream) {
   stream->platform.nextBufferIndex = 0;
   stream->platform.primed = false;
 
+  // For MP3 streams, the background decode thread reads via the same
+  // stream->mp3 decoder/file audioStreamSeek() below is about to mutate -
+  // must be fully stopped first, every time a new pass begins (initial
+  // play, explicit seek, or loop restart), not just once at stream
+  // teardown. Blocks until that's guaranteed and resets the ring to empty
+  // for this fresh pass.
+  errorChain(audioStreamMp3RingStopIfNeeded(stream));
+
   errorChain(audioStreamSeek(stream, startFrame));
   stream->platform.position = startFrame;
   stream->platform.endFrame = endFrame;
 
+  // Launch this pass's decode thread now that position/endFrame are
+  // finalized - it starts filling the ring immediately in the background,
+  // ahead of audioStreamDolphinFeed() below even asking for anything.
+  errorChain(audioStreamMp3RingStartIfNeeded(stream, startFrame, endFrame));
+
   // Prime only the first buffer here - the second window can't be queued
   // back-to-back with this one, since ASND_SetVoice() leaves VOICE_UPDATE
   // set until its own audio-DMA-interrupt tick picks it up, and
   // ASND_AddVoice() refuses to queue anything while that's still pending
   // (see audioStreamDolphinFeed()'s own comment). Later polls queue
-  // everything after this.
+  // everything after this. For MP3 streams this first call will likely
+  // find the ring still empty (the decode thread just started) and do
+  // nothing - the next few audioStreamDolphinIsFinished() polls pick it
+  // up once decode has caught up.
   errorChain(audioStreamDolphinFeed(stream));
 
   errorOk();
@@ -138,9 +153,20 @@ errorret_t audioStreamDolphinFeed(audiostream_t *stream) {
 
   uint8_t *buffer = stream->platform.buffer[stream->platform.nextBufferIndex];
   size_t framesRead = 0;
-  errorChain(audioStreamRead(
-    stream, (int16_t *) buffer, framesToRead, &framesRead
+  bool_t notReadyYet = false;
+  // For MP3 streams, drains already-decoded PCM out of the background ring
+  // instead of decoding directly - the expensive part (asset I/O + libmad
+  // decode) already happened on the decode thread, off the main thread.
+  // See audiostreammp3ring.h's own top-of-file comment for why. For
+  // everything else, this is just a plain audioStreamRead().
+  errorChain(audioStreamReadForPlayback(
+    stream, (int16_t *) buffer, framesToRead, &framesRead, &notReadyYet
   ));
+  if(notReadyYet) {
+    // Decode hasn't caught up to a full window yet - not an error, just
+    // try again next poll.
+    errorOk();
+  }
 
   const size_t bytesRead = framesRead * frameSize;
   const size_t alignedBytes = (bytesRead + 31) & ~((size_t) 31);
diff --git a/src/duskdolphin/audio/audiostreamdolphin.h b/src/duskdolphin/audio/audiostreamdolphin.h
index 363e3a23..f0afff80 100644
--- a/src/duskdolphin/audio/audiostreamdolphin.h
+++ b/src/duskdolphin/audio/audiostreamdolphin.h
@@ -7,6 +7,7 @@
 
 #pragma once
 #include "error/error.h"
+#include "audiostreammp3ring.h"
 
 typedef struct audiostream_s audiostream_t;
 
@@ -63,13 +64,20 @@ typedef struct {
   // call for that pass's first (ASND_SetVoice()) window.
   int32_t volumeLeft;
   int32_t volumeRight;
+
+  // Background decode-ahead ring, shared with Linux - only initialized/used
+  // for AUDIO_STREAM_TYPE_MP3 streams (PCM/WAV reads are cheap plain
+  // copies and don't need this). See audiostreammp3ring.h's own comment.
+  audiostreammp3ring_t mp3Ring;
 } audiostreamdolphin_t;
 
 /**
  * Initializes the GameCube/Wii-specific playback state of an audio stream:
  * allocates an ASND voice slot (ASND_GetFirstUnusedVoice()) and this
  * stream's two fixed-size double-buffer windows (sized from its channel
- * count, AUDIO_DOLPHIN_WINDOW_FRAMES frames each).
+ * count, AUDIO_DOLPHIN_WINDOW_FRAMES frames each). For MP3 streams, also
+ * initializes the background decode-ahead ring (audioStreamMp3RingInit()) -
+ * see audiostreammp3ring.h's own comment.
  *
  * @param stream The audio stream to initialize.
  * @return Error state if any.
@@ -78,8 +86,10 @@ errorret_t audioStreamDolphinInit(audiostream_t *stream);
 
 /**
  * Disposes the GameCube/Wii-specific playback state of an audio stream:
- * stops its voice (releasing the slot back to ASND_GetFirstUnusedVoice())
- * and frees its two buffers.
+ * for MP3 streams, first fully stops and frees the background decode ring
+ * (audioStreamMp3RingDispose()) - must happen before the shared MP3
+ * layer tears down its decoder/file - then stops the voice (releasing the
+ * slot back to ASND_GetFirstUnusedVoice()) and frees its two buffers.
  *
  * @param stream The audio stream to dispose.
  * @return Error state if any.
@@ -90,10 +100,14 @@ errorret_t audioStreamDolphinDispose(audiostream_t *stream);
  * Starts a new playback pass: seeks the stream's decode position to
  * stream->startFrame, determines this pass's loop-segment end, resolves
  * this pass's volume/pan, stops whatever the voice was doing before (so a
- * fresh pass never plays out stale queued windows from the last one), and
- * primes it via audioStreamDolphinFeed() - once for the first
- * (ASND_SetVoice()) window, once more to also queue the second
- * (ASND_AddVoice()) window ahead of time.
+ * fresh pass never plays out stale queued windows from the last one). For
+ * MP3 streams, also stops the previous pass's background decode thread
+ * (audioStreamMp3RingStop(), which must complete before the seek
+ * below - see its own comment) and starts a fresh one for this pass
+ * (audioStreamMp3RingStart()) once position/endFrame are finalized.
+ * Finally primes the first window via audioStreamDolphinFeed() - for MP3
+ * streams this may find the ring still empty and do nothing yet; the next
+ * few polls pick it up once decode has caught up.
  *
  * @param stream The audio stream to output.
  * @return Error state if any.
@@ -110,6 +124,10 @@ errorret_t audioStreamDolphinBuffer(audiostream_t *stream);
  * ASND_AddVoice()). A no-op once this pass's platform.position has
  * reached platform.endFrame.
  *
+ * For MP3 streams, the PCM data comes from draining the background decode
+ * ring (audioStreamMp3RingDrain()) instead of decoding directly -
+ * see audiostreammp3ring.h's own comment for why.
+ *
  * @param stream The audio stream to feed.
  * @return Error state if any.
  */
diff --git a/src/duskdolphin/audio/audiostreammp3decoder.h b/src/duskdolphin/audio/audiostreammp3decoder.h
index 7d77039c..f39288d6 100644
--- a/src/duskdolphin/audio/audiostreammp3decoder.h
+++ b/src/duskdolphin/audio/audiostreammp3decoder.h
@@ -16,4 +16,4 @@
 // see that header's own documentation for the actual struct/interface.
 // Conveniently, libmad ships as part of libogc itself (gc/mad.h) rather
 // than needing to be fetched separately, unlike on Linux.
-#include "audio/stream/audiostreammp3decodersw.h"
+#include "audiostreammp3decodersw.h"
diff --git a/src/dusklinux/audio/CMakeLists.txt b/src/dusklinux/audio/CMakeLists.txt
index c6c014d7..a2accbc6 100644
--- a/src/dusklinux/audio/CMakeLists.txt
+++ b/src/dusklinux/audio/CMakeLists.txt
@@ -10,17 +10,14 @@ target_sources(${DUSK_LIBRARY_TARGET_NAME}
     audiostreamlinux.c
 )
 
-# No hardware MP3 decoder on Linux - use the shared libmad-based software
-# backend (also used by Dolphin), sourced directly from src/dusk/audio
-# rather than through its own unconditional CMakeLists.txt so platforms
-# with a hardware decoder (PSP) never pull libmad in at all. See
-# audiostreammp3decodersw.h's own comment on why libmad (GPL-licensed)
-# replaced this project's original minimp3-based implementation.
+# No hardware MP3 decoder on Linux - uses the shared libmad-based software
+# backend + decode-ahead ring in src/duskmad instead (also used by
+# Dolphin). See audiostreammp3decodersw.h's own comment on why libmad
+# (GPL-licensed) replaced this project's original minimp3-based
+# implementation. Linking stays here (rather than in src/duskmad itself)
+# since Linux and Dolphin genuinely obtain libmad differently -
+# find_package() here, the bundled devkitPPC/libogc toolchain copy there.
 if(NOT libmad_FOUND)
   find_package(libmad REQUIRED)
 endif()
 target_link_libraries(${DUSK_LIBRARY_TARGET_NAME} PUBLIC libmad::mad)
-target_sources(${DUSK_LIBRARY_TARGET_NAME}
-  PUBLIC
-    "${DUSK_SOURCES_DIR}/dusk/audio/stream/audiostreammp3decodersw.c"
-)
diff --git a/src/dusklinux/audio/audiostreamlinux.c b/src/dusklinux/audio/audiostreamlinux.c
index 61960b51..ac963946 100644
--- a/src/dusklinux/audio/audiostreamlinux.c
+++ b/src/dusklinux/audio/audiostreamlinux.c
@@ -39,12 +39,20 @@ errorret_t audioStreamLinuxInit(audiostream_t *stream) {
     errorThrow("Failed to open SDL2 audio device: %s", SDL_GetError());
   }
 
+  errorChain(audioStreamMp3RingInitIfNeeded(stream));
+
   errorOk();
 }
 
 errorret_t audioStreamLinuxDispose(audiostream_t *stream) {
   assertNotNull(stream, "Stream cannot be NULL.");
 
+  // Must happen first: audioStreamMp3Dispose() (called right after this
+  // function returns, from the shared audioStreamDispose()) tears down
+  // stream->mp3's decoder/file - the decode thread must be fully parked
+  // before that, not just before this function's own SDL cleanup below.
+  errorChain(audioStreamMp3RingDisposeIfNeeded(stream));
+
   SDL_CloseAudioDevice(stream->platform.device);
 
   errorOk();
@@ -63,17 +71,11 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
   stream->startFrame = 0;
   stream->seeking = false;
 
-  size_t endFrame = totalFrames;
-  if((stream->state & AUDIO_STREAM_STATE_LOOPING) && stream->loopStart >= 0) {
-    endFrame = mathMin(
-      (size_t) (stream->loopStart * stream->sampleRate), totalFrames
-    );
-  }
-  // A seek (or a loop restart landing exactly on the loop end) can put
-  // startFrame at or past endFrame - e.g. seeking into an outro after the
-  // loop point. Play out to the true end of the buffer once instead, same
-  // as PSP.
-  if(startFrame >= endFrame) endFrame = totalFrames;
+  // This pass's stop point - the current loop segment's end, or the true
+  // end of the clip if not looping (or if startFrame already landed past
+  // the loop end, e.g. a seek into an outro). See
+  // audioStreamComputeEndFrame()'s own comment.
+  const size_t endFrame = audioStreamComputeEndFrame(stream, startFrame, totalFrames);
 
   // Only an explicit seek discards whatever's still queued and jumps -
   // a natural loop restart deliberately leaves the previous pass's tail
@@ -84,6 +86,17 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
   if(seeking) {
     SDL_ClearQueuedAudio(stream->platform.device);
   }
+
+  // For MP3 streams, the background decode thread reads via the same
+  // stream->mp3 decoder/file audioStreamSeek() below is about to mutate -
+  // must be fully stopped first, every time a new pass begins. Unlike the
+  // SDL_ClearQueuedAudio() call above, this is unconditional regardless of
+  // the seeking flag: that flag only controls whether the SDL queue's
+  // leftover tail is discarded (for gapless natural loop restarts), not
+  // whether decode needs to re-seek - MP3 has no cheap seek either way,
+  // see audioStreamSeek()'s own comment.
+  errorChain(audioStreamMp3RingStopIfNeeded(stream));
+
   errorChain(audioStreamSeek(stream, startFrame));
 
   stream->platform.position = startFrame;
@@ -91,6 +104,11 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
   stream->platform.passStartTicks = SDL_GetTicks();
   stream->platform.passStartPosition = startFrame;
 
+  // Launch this pass's decode thread now that position/endFrame are
+  // finalized - it starts filling the ring immediately in the background,
+  // ahead of audioStreamLinuxFeed() below even asking for anything.
+  errorChain(audioStreamMp3RingStartIfNeeded(stream, startFrame, endFrame));
+
   errorChain(audioStreamLinuxFeed(stream));
 
   SDL_PauseAudioDevice(stream->platform.device, 0);
@@ -150,11 +168,25 @@ errorret_t audioStreamLinuxFeed(audiostream_t *stream) {
 
   int16_t *chunk = memoryAllocate(framesToRead * frameSize);
   size_t framesRead = 0;
-  errorret_t ret = audioStreamRead(stream, chunk, framesToRead, &framesRead);
+  bool_t notReadyYet = false;
+  // For MP3 streams, drains already-decoded PCM out of the background ring
+  // instead of decoding directly - the expensive part (asset I/O + libmad
+  // decode) already happened on the decode thread, off the main thread.
+  // See audiostreammp3ring.h's own top-of-file comment for why. For
+  // everything else, this is just a plain audioStreamRead().
+  errorret_t ret = audioStreamReadForPlayback(
+    stream, chunk, framesToRead, &framesRead, &notReadyYet
+  );
   if(errorIsNotOk(ret)) {
     memoryFree(chunk);
     errorChain(ret);
   }
+  if(notReadyYet) {
+    // Decode hasn't caught up to a full window yet - not an error, just
+    // try again next poll.
+    memoryFree(chunk);
+    errorOk();
+  }
 
   const size_t bytesRead = framesRead * frameSize;
   int queued;
diff --git a/src/dusklinux/audio/audiostreamlinux.h b/src/dusklinux/audio/audiostreamlinux.h
index a10f4139..6d6dfb39 100644
--- a/src/dusklinux/audio/audiostreamlinux.h
+++ b/src/dusklinux/audio/audiostreamlinux.h
@@ -7,6 +7,7 @@
 
 #pragma once
 #include "error/error.h"
+#include "audiostreammp3ring.h"
 #include <SDL2/SDL.h>
 
 typedef struct audiostream_s audiostream_t;
@@ -31,11 +32,19 @@ typedef struct {
   // SDL_GetQueuedAudioSize() alone).
   Uint32 passStartTicks;
   size_t passStartPosition;
+
+  // Background decode-ahead ring, shared with Dolphin - only
+  // initialized/used for AUDIO_STREAM_TYPE_MP3 streams (PCM/WAV reads are
+  // cheap plain copies and don't need this). See audiostreammp3ring.h's
+  // own comment.
+  audiostreammp3ring_t mp3Ring;
 } audiostreamlinux_t;
 
 /**
  * Initializes the Linux-specific playback state of an audio stream by
- * opening an SDL2 audio device matching the stream's PCM format.
+ * opening an SDL2 audio device matching the stream's PCM format. For MP3
+ * streams, also initializes the background decode-ahead ring
+ * (audioStreamMp3RingInit()) - see audiostreammp3ring.h's own comment.
  *
  * @param stream The audio stream to initialize.
  * @return Error state if any.
@@ -43,8 +52,10 @@ typedef struct {
 errorret_t audioStreamLinuxInit(audiostream_t *stream);
 
 /**
- * Disposes the Linux-specific playback state of an audio stream, closing
- * its SDL2 audio device.
+ * Disposes the Linux-specific playback state of an audio stream: for MP3
+ * streams, first fully stops and frees the background decode ring
+ * (audioStreamMp3RingDispose()) - must happen before the shared MP3 layer
+ * tears down its decoder/file - then closes the SDL2 audio device.
  *
  * @param stream The audio stream to dispose.
  * @return Error state if any.
@@ -55,8 +66,14 @@ errorret_t audioStreamLinuxDispose(audiostream_t *stream);
  * Starts a new playback pass: seeks the stream's PCM read cursor to
  * stream->startFrame (clearing the SDL queue first if this is an explicit
  * seek rather than a natural loop restart - see stream->seeking's own
- * comment), determines this pass's loop-segment end, and queues the first
- * window via audioStreamLinuxFeed().
+ * comment), determines this pass's loop-segment end. For MP3 streams, also
+ * stops the previous pass's background decode thread
+ * (audioStreamMp3RingStop(), which must complete before the seek below -
+ * see its own comment) and starts a fresh one for this pass
+ * (audioStreamMp3RingStart()) once position/endFrame are finalized. Finally
+ * queues the first window via audioStreamLinuxFeed() - for MP3 streams
+ * this may find the ring still empty and queue nothing yet; the next few
+ * polls pick it up once decode has caught up.
  *
  * @param stream The audio stream to output.
  * @return Error state if any.
@@ -69,6 +86,10 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream);
  * advancing platform.position by however many frames were actually read.
  * A no-op once platform.position has reached platform.endFrame.
  *
+ * For MP3 streams, the PCM data comes from draining the background decode
+ * ring (audioStreamMp3RingDrain()) instead of decoding directly - see
+ * audiostreammp3ring.h's own comment for why.
+ *
  * Kept as a small, bounded read/queue operation (rather than the whole
  * pass at once) specifically so a long clip is never fully resident in
  * memory at once - called both by audioStreamLinuxBuffer() (the first
diff --git a/src/dusklinux/audio/audiostreammp3decoder.h b/src/dusklinux/audio/audiostreammp3decoder.h
index 8a268837..2cb931d8 100644
--- a/src/dusklinux/audio/audiostreammp3decoder.h
+++ b/src/dusklinux/audio/audiostreammp3decoder.h
@@ -12,4 +12,4 @@
 // header's own documentation for the actual struct/interface, including
 // why libmad (GPL-licensed) was chosen over the MIT-licensed minimp3 this
 // used to be.
-#include "audio/stream/audiostreammp3decodersw.h"
+#include "audiostreammp3decodersw.h"
diff --git a/src/duskmad/CMakeLists.txt b/src/duskmad/CMakeLists.txt
new file mode 100644
index 00000000..94fb2d7a
--- /dev/null
+++ b/src/duskmad/CMakeLists.txt
@@ -0,0 +1,33 @@
+# Copyright (c) 2026 Dominic Masters
+#
+# This software is released under the MIT License.
+# https://opensource.org/licenses/MIT
+
+# libmad-based MP3 decode, shared by every platform that doesn't have (or
+# use) a hardware MP3 decoder - currently Linux/knulli and Dolphin
+# (Wii/GameCube), both via the same software decoder and background
+# decode-ahead ring (audiostreammp3decodersw.c, audiostreammp3ring.c). See
+# src/CMakeLists.txt for which DUSK_TARGET_SYSTEM values pull this
+# directory in at all - PSP uses its own hardware sceMp3 decoder instead
+# (src/duskpsp/audio/audiostreammp3decoder.c) and never references
+# anything here.
+#
+# Note: libmad is GPL-licensed, unlike the rest of this project - see
+# audiostreammp3decodersw.h's own comment. Linking/finding libmad itself
+# stays with each platform's own audio CMakeLists.txt (src/dusklinux/audio,
+# cmake/targets/dolphin.cmake) rather than living here, since Linux and
+# Dolphin genuinely obtain it differently (find_package() vs the bundled
+# devkitPPC/libogc toolchain copy) - this file only owns the shared source.
+
+# Includes
+target_include_directories(${DUSK_LIBRARY_TARGET_NAME}
+  PUBLIC
+    ${CMAKE_CURRENT_LIST_DIR}
+)
+
+# Sources
+target_sources(${DUSK_LIBRARY_TARGET_NAME}
+  PUBLIC
+    audiostreammp3decodersw.c
+    audiostreammp3ring.c
+)
diff --git a/src/dusk/audio/stream/audiostreammp3decodersw.c b/src/duskmad/audiostreammp3decodersw.c
similarity index 98%
rename from src/dusk/audio/stream/audiostreammp3decodersw.c
rename to src/duskmad/audiostreammp3decodersw.c
index c99b922f..bc8a6857 100644
--- a/src/dusk/audio/stream/audiostreammp3decodersw.c
+++ b/src/duskmad/audiostreammp3decodersw.c
@@ -6,8 +6,8 @@
  */
 
 #include "audiostreammp3decodersw.h"
-#include "audiostream.h"
-#include "audiostreammp3.h"
+#include "audio/stream/audiostream.h"
+#include "audio/stream/audiostreammp3.h"
 #include "assert/assert.h"
 #include "util/memory.h"
 
diff --git a/src/dusk/audio/stream/audiostreammp3decodersw.h b/src/duskmad/audiostreammp3decodersw.h
similarity index 100%
rename from src/dusk/audio/stream/audiostreammp3decodersw.h
rename to src/duskmad/audiostreammp3decodersw.h
diff --git a/src/duskmad/audiostreammp3ring.c b/src/duskmad/audiostreammp3ring.c
new file mode 100644
index 00000000..342bf88a
--- /dev/null
+++ b/src/duskmad/audiostreammp3ring.c
@@ -0,0 +1,303 @@
+/**
+ * Copyright (c) 2026 Dominic Masters
+ *
+ * This software is released under the MIT License.
+ * https://opensource.org/licenses/MIT
+ */
+
+#include "audiostreammp3ring.h"
+#include "audio/stream/audiostream.h"
+#include "assert/assert.h"
+#include "util/memory.h"
+#include "util/math.h"
+#include <unistd.h>
+
+errorret_t audioStreamMp3RingInit(audiostream_t *stream) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+  assertTrue(stream->type == AUDIO_STREAM_TYPE_MP3, "Stream is not MP3.");
+
+  audiostreammp3ring_t *ring = &stream->platform.mp3Ring;
+  const size_t frameSize = stream->channels * sizeof(int16_t);
+
+  ring->data = (int16_t *) memoryAllocate(AUDIO_MP3_RING_FRAMES * frameSize);
+  ring->scratch = (int16_t *) memoryAllocate(AUDIO_MP3_RING_STEP_FRAMES * frameSize);
+  ring->readPos = 0;
+  ring->writePos = 0;
+  ring->filled = 0;
+  ring->reachedEnd = false;
+  ring->failed = false;
+  ring->startFrame = 0;
+  ring->endFrame = 0;
+  threadMutexInit(&ring->lock);
+  threadInit(&ring->thread, audioStreamMp3RingThreadFunc);
+
+  errorOk();
+}
+
+errorret_t audioStreamMp3RingDispose(audiostream_t *stream) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+  assertTrue(stream->type == AUDIO_STREAM_TYPE_MP3, "Stream is not MP3.");
+
+  audiostreammp3ring_t *ring = &stream->platform.mp3Ring;
+
+  // Must fully park the decode thread before freeing anything it might
+  // still be touching, or before the caller lets the shared MP3 layer
+  // tear down stream->mp3's decoder/file out from under it.
+  threadStop(&ring->thread);
+  threadMutexDispose(&ring->lock);
+  memoryFree(ring->data);
+  memoryFree(ring->scratch);
+  ring->data = NULL;
+  ring->scratch = NULL;
+
+  errorOk();
+}
+
+errorret_t audioStreamMp3RingStop(audiostream_t *stream) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+  assertTrue(stream->type == AUDIO_STREAM_TYPE_MP3, "Stream is not MP3.");
+
+  audiostreammp3ring_t *ring = &stream->platform.mp3Ring;
+
+  // Blocking - guarantees the decode thread is fully parked (not mid-read)
+  // before the caller runs audioStreamSeek(), which mutates the same
+  // stream->mp3 decoder/file this thread reads via audioStreamRead(). A
+  // safe no-op if the thread was never started or already finished this
+  // pass on its own (reached endFrame).
+  threadStop(&ring->thread);
+
+  threadMutexLock(&ring->lock);
+  ring->readPos = 0;
+  ring->writePos = 0;
+  ring->filled = 0;
+  ring->reachedEnd = false;
+  ring->failed = false;
+  threadMutexUnlock(&ring->lock);
+
+  errorOk();
+}
+
+errorret_t audioStreamMp3RingStart(
+  audiostream_t *stream, size_t startFrame, size_t endFrame
+) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+  assertTrue(stream->type == AUDIO_STREAM_TYPE_MP3, "Stream is not MP3.");
+
+  audiostreammp3ring_t *ring = &stream->platform.mp3Ring;
+
+  // Finalized here, before the thread is launched - thread creation itself
+  // is the happens-before edge that makes these writes visible to the new
+  // thread, so no lock is needed around this handoff.
+  ring->startFrame = startFrame;
+  ring->endFrame = endFrame;
+  ring->thread.data = stream;
+  threadStartRequest(&ring->thread);
+
+  errorOk();
+}
+
+size_t audioStreamMp3RingDrain(
+  audiostream_t *stream,
+  int16_t *out,
+  size_t maxFrames,
+  bool_t *outDecodeDone
+) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+  assertNotNull(out, "Out buffer cannot be NULL.");
+  assertNotNull(outDecodeDone, "outDecodeDone cannot be NULL.");
+  assertTrue(stream->type == AUDIO_STREAM_TYPE_MP3, "Stream is not MP3.");
+
+  audiostreammp3ring_t *ring = &stream->platform.mp3Ring;
+  const size_t channels = stream->channels;
+
+  threadMutexLock(&ring->lock);
+  const size_t available = ring->filled;
+  const bool_t decodeDone = ring->reachedEnd || ring->failed;
+
+  // Wait for a full window rather than handing back a partial one, unless
+  // decode has genuinely nothing more to give this pass - keeps a
+  // platform's own output handoff cadence matching its previous
+  // whole-window-per-call behavior instead of fragmenting into many tiny
+  // calls just because decode happened to be a little behind at this exact
+  // poll.
+  if(available < maxFrames && !decodeDone) {
+    threadMutexUnlock(&ring->lock);
+    *outDecodeDone = false;
+    return 0;
+  }
+
+  const size_t toCopy = mathMin(available, maxFrames);
+  const size_t firstPart = mathMin(toCopy, AUDIO_MP3_RING_FRAMES - ring->readPos);
+  memoryCopy(out, ring->data + ring->readPos * channels, firstPart * channels * sizeof(int16_t));
+  if(toCopy > firstPart) {
+    memoryCopy(
+      out + firstPart * channels, ring->data,
+      (toCopy - firstPart) * channels * sizeof(int16_t)
+    );
+  }
+  ring->readPos = (ring->readPos + toCopy) % AUDIO_MP3_RING_FRAMES;
+  ring->filled -= toCopy;
+  threadMutexUnlock(&ring->lock);
+
+  *outDecodeDone = decodeDone;
+  return toCopy;
+}
+
+bool_t audioStreamMp3RingFailed(const audiostream_t *stream) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+  assertTrue(stream->type == AUDIO_STREAM_TYPE_MP3, "Stream is not MP3.");
+
+  return stream->platform.mp3Ring.failed;
+}
+
+void audioStreamMp3RingThreadFunc(thread_t *thread) {
+  assertNotNull(thread, "Thread cannot be NULL.");
+
+  audiostream_t *stream = (audiostream_t *) thread->data;
+  audiostreammp3ring_t *ring = &stream->platform.mp3Ring;
+  const size_t channels = stream->channels;
+
+  // Captured once - only ever written by audioStreamMp3RingStart() before
+  // this thread was launched (a platform's Feed() advances its own
+  // position field as it drains the ring, never these). Tracks its own
+  // progress in a thread-local counter rather than re-reading endFrame
+  // repeatedly, though it wouldn't change mid-pass either way.
+  const size_t startFrame = ring->startFrame;
+  const size_t endFrame = ring->endFrame;
+  size_t decoded = 0;
+
+  while(!threadShouldStop(thread)) {
+    threadMutexLock(&ring->lock);
+    const size_t room = AUDIO_MP3_RING_FRAMES - ring->filled;
+    threadMutexUnlock(&ring->lock);
+
+    if(room == 0) {
+      // Never sleeps while holding the lock - re-checks threadShouldStop()
+      // immediately after waking, bounding a stop request's worst-case
+      // wait here to one poll interval, not a whole ring's worth of idle.
+      usleep(AUDIO_MP3_RING_POLL_MICROS);
+      continue;
+    }
+
+    if(startFrame + decoded >= endFrame) break;
+    const size_t framesRemaining = endFrame - (startFrame + decoded);
+    const size_t step = mathMin(
+      mathMin(room, framesRemaining), (size_t) AUDIO_MP3_RING_STEP_FRAMES
+    );
+
+    size_t framesRead = 0;
+    // The expensive call (asset I/O + libmad decode) - deliberately
+    // outside any lock, so it never blocks the main thread's own
+    // audioStreamMp3RingDrain() calls while it runs.
+    errorret_t ret = audioStreamRead(stream, ring->scratch, step, &framesRead);
+    if(errorIsNotOk(ret)) {
+      errorCatch(errorPrint(ret));
+      threadMutexLock(&ring->lock);
+      ring->failed = true;
+      threadMutexUnlock(&ring->lock);
+      return;
+    }
+    if(framesRead == 0) break; // Genuine early end of underlying data.
+
+    threadMutexLock(&ring->lock);
+    const size_t firstPart = mathMin(
+      framesRead, AUDIO_MP3_RING_FRAMES - ring->writePos
+    );
+    memoryCopy(
+      ring->data + ring->writePos * channels, ring->scratch,
+      firstPart * channels * sizeof(int16_t)
+    );
+    if(framesRead > firstPart) {
+      memoryCopy(
+        ring->data, ring->scratch + firstPart * channels,
+        (framesRead - firstPart) * channels * sizeof(int16_t)
+      );
+    }
+    ring->writePos = (ring->writePos + framesRead) % AUDIO_MP3_RING_FRAMES;
+    ring->filled += framesRead;
+    threadMutexUnlock(&ring->lock);
+
+    decoded += framesRead;
+    // Loops back to the top - threadShouldStop() is rechecked before the
+    // NEXT audioStreamRead(), not just once at the top of an outer loop,
+    // so a stop request doesn't have to wait through a whole pass.
+  }
+
+  threadMutexLock(&ring->lock);
+  ring->reachedEnd = true;
+  threadMutexUnlock(&ring->lock);
+}
+
+errorret_t audioStreamMp3RingInitIfNeeded(audiostream_t *stream) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+
+  if(stream->type == AUDIO_STREAM_TYPE_MP3) {
+    errorChain(audioStreamMp3RingInit(stream));
+  }
+
+  errorOk();
+}
+
+errorret_t audioStreamMp3RingDisposeIfNeeded(audiostream_t *stream) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+
+  if(stream->type == AUDIO_STREAM_TYPE_MP3) {
+    errorChain(audioStreamMp3RingDispose(stream));
+  }
+
+  errorOk();
+}
+
+errorret_t audioStreamMp3RingStopIfNeeded(audiostream_t *stream) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+
+  if(stream->type == AUDIO_STREAM_TYPE_MP3) {
+    errorChain(audioStreamMp3RingStop(stream));
+  }
+
+  errorOk();
+}
+
+errorret_t audioStreamMp3RingStartIfNeeded(
+  audiostream_t *stream, size_t startFrame, size_t endFrame
+) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+
+  if(stream->type == AUDIO_STREAM_TYPE_MP3) {
+    errorChain(audioStreamMp3RingStart(stream, startFrame, endFrame));
+  }
+
+  errorOk();
+}
+
+errorret_t audioStreamReadForPlayback(
+  audiostream_t *stream,
+  int16_t *out,
+  size_t maxFrames,
+  size_t *outFramesRead,
+  bool_t *outNotReadyYet
+) {
+  assertNotNull(stream, "Stream cannot be NULL.");
+  assertNotNull(out, "Out buffer cannot be NULL.");
+  assertNotNull(outFramesRead, "outFramesRead cannot be NULL.");
+  assertNotNull(outNotReadyYet, "outNotReadyYet cannot be NULL.");
+
+  *outNotReadyYet = false;
+
+  if(stream->type == AUDIO_STREAM_TYPE_MP3) {
+    bool_t decodeDone = false;
+    *outFramesRead = audioStreamMp3RingDrain(stream, out, maxFrames, &decodeDone);
+    if(*outFramesRead == 0 && !decodeDone) {
+      *outNotReadyYet = true;
+      errorOk();
+    }
+    if(*outFramesRead == 0 && audioStreamMp3RingFailed(stream)) {
+      errorThrow("Background MP3 decode failed.");
+    }
+    errorOk();
+  }
+
+  errorChain(audioStreamRead(stream, out, maxFrames, outFramesRead));
+
+  errorOk();
+}
diff --git a/src/duskmad/audiostreammp3ring.h b/src/duskmad/audiostreammp3ring.h
new file mode 100644
index 00000000..c1a5562b
--- /dev/null
+++ b/src/duskmad/audiostreammp3ring.h
@@ -0,0 +1,288 @@
+/**
+ * Copyright (c) 2026 Dominic Masters
+ *
+ * This software is released under the MIT License.
+ * https://opensource.org/licenses/MIT
+ */
+
+#pragma once
+#include "error/error.h"
+#include "thread/thread.h"
+#include "thread/threadmutex.h"
+
+typedef struct audiostream_s audiostream_t;
+
+// How many frames of decoded PCM the ring holds - deliberately generous
+// (~1.5s at 44100Hz stereo, 256KB) so the background decode thread can run
+// well ahead of what a platform's Feed() actually needs per poll, without
+// being a meaningful fraction of even GameCube's 24MB (~1.1%) total RAM.
+#define AUDIO_MP3_RING_FRAMES (64 * 1024)
+
+// How many frames the decode thread reads per audioStreamRead() call -
+// bounds both normal decode-thread iteration granularity and, more
+// importantly, audioStreamMp3RingStop()'s worst-case blocking wait (roughly
+// one of these reads' worth of asset I/O + libmad decode, since the thread
+// only rechecks threadShouldStop() between reads, not mid-read).
+#define AUDIO_MP3_RING_STEP_FRAMES 4096
+
+// How long the decode thread sleeps between polls when the ring is already
+// full - matches the shared asset-load thread's own idle-poll interval
+// (src/dusk/asset/asset.c) rather than inventing a new convention.
+#define AUDIO_MP3_RING_POLL_MICROS 1000
+
+/**
+ * Background MP3 decode-ahead ring, shared by every platform whose MP3
+ * decode is expensive enough to hitch the main thread if done synchronously
+ * (currently Linux and Dolphin, both via the libmad-based software decoder
+ * - audiostreammp3decodersw.c; PSP doesn't use this, see below). Moves
+ * audioStreamRead()'s real cost (asset I/O + libmad decode) off the main
+ * thread onto a background thread that decodes ahead into this ring - a
+ * platform's own Feed() then drains already-decoded PCM out of it (a
+ * cheap, lock-protected copy) instead of decoding directly. Sits upstream
+ * of whatever a platform does with the PCM afterward (SDL_QueueAudio on
+ * Linux, ASND_SetVoice()/ASND_AddVoice() on Dolphin) - this struct knows
+ * nothing about either.
+ *
+ * Only used for AUDIO_STREAM_TYPE_MP3 streams - PCM/WAV reads are already
+ * cheap plain memory copies and don't need this. Not used by PSP: its
+ * hardware sceMp3 decoder is a completely different API (see
+ * src/duskpsp/audio/audiostreammp3decoder.c), and PSP's own audio backend
+ * already has its own background thread + ring - for *output* (draining
+ * to hardware), not decode - layering this under it would just be a
+ * second, redundant ring.
+ *
+ * The background thread is one-shot per playback pass, not a persistent
+ * worker (unlike PSP's own output thread, which idles cheaply between
+ * passes) - a fresh thread is started for every new pass (initial play,
+ * explicit seek, or loop restart), since those are rare relative to
+ * per-frame polls and a brief synchronous stop-then-restart at pass
+ * boundaries is a fine trade for not needing a mid-pass "abandon and
+ * reset" handshake.
+ */
+typedef struct {
+  int16_t *data;    // AUDIO_MP3_RING_FRAMES * channels * sizeof(int16_t)
+  int16_t *scratch;  // AUDIO_MP3_RING_STEP_FRAMES * channels * sizeof(int16_t),
+                      // the decode thread's own read target - never touched
+                      // by the main thread, so it needs no lock protection.
+
+  // Guards exactly filled/readPos/writePos/reachedEnd/failed below,
+  // grouped together for simplicity rather than split per-field (same
+  // rationale as the PSP backend's own ring lock). Never held across
+  // audioStreamRead() or usleep() (decode thread) or a platform's own
+  // output call (main thread).
+  threadmutex_t lock;
+
+  // Next frame audioStreamMp3RingDrain() (main thread) reads.
+  size_t readPos;
+  // Next frame the decode thread writes.
+  size_t writePos;
+  // How many valid frames are currently buffered.
+  size_t filled;
+
+  // Set by the decode thread once it has produced every frame this pass
+  // will ever have (reached endFrame, or the underlying read returned 0
+  // frames early) - tells Drain() there's nothing more coming, so it
+  // should hand back whatever's left rather than waiting for a full
+  // window that will never arrive.
+  bool_t reachedEnd;
+  // Set by the decode thread if audioStreamRead() itself errored - Drain()
+  // treats this the same as reachedEnd for draining purposes, but a
+  // platform's Feed() should surface it as a real thrown error once the
+  // ring is empty.
+  bool_t failed;
+
+  // This pass's frame bounds, set by audioStreamMp3RingStart() before the
+  // thread is launched and read-only after (thread creation is the
+  // happens-before edge) - stored here rather than read from a platform's
+  // own platform_t fields, since this struct is shared across platforms
+  // whose own position/endFrame-equivalent fields aren't guaranteed to
+  // exist or be named the same way.
+  size_t startFrame;
+  size_t endFrame;
+
+  // One-shot per pass - see this struct's own top-of-file comment.
+  thread_t thread;
+} audiostreammp3ring_t;
+
+/**
+ * Allocates this stream's ring/scratch buffers (sized from its channel
+ * count) and initializes its mutex and thread_t - does not start decoding
+ * anything yet, that's audioStreamMp3RingStart()'s job. Call once, from a
+ * platform's own Init(), only for AUDIO_STREAM_TYPE_MP3 streams.
+ *
+ * @param stream The audio stream to initialize. Must be AUDIO_STREAM_TYPE_MP3.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingInit(audiostream_t *stream);
+
+/**
+ * Blocking-stops the decode thread (a no-op if it was never started or has
+ * already finished this pass) and frees the ring/scratch buffers. Call
+ * from a platform's own Dispose() to guarantee the thread is fully parked
+ * before the rest of this stream's (and the shared MP3 layer's) teardown
+ * runs.
+ *
+ * @param stream The audio stream to dispose. Must be AUDIO_STREAM_TYPE_MP3.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingDispose(audiostream_t *stream);
+
+/**
+ * Blocking-stops the decode thread (safe no-op if not running) and resets
+ * the ring to empty, ready for a fresh pass. Must be called (and must
+ * return) before audioStreamSeek() runs for a new pass, since the decode
+ * thread reads via the same stream->mp3 decoder/file audioStreamSeek()
+ * mutates - and before audioStreamMp3RingStart() launches the next one.
+ *
+ * @param stream The audio stream to stop. Must be AUDIO_STREAM_TYPE_MP3.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingStop(audiostream_t *stream);
+
+/**
+ * Starts a fresh one-shot decode thread for the pass a platform's own
+ * Buffer() just set up (stream->platform.position/endFrame - or
+ * equivalent, whatever the platform's own struct calls them - already
+ * finalized) - decodes forward via audioStreamRead() into the ring until
+ * it reaches endFrame, filling back up whenever audioStreamMp3RingDrain()
+ * makes room.
+ *
+ * @param stream The audio stream to start decoding. Must be AUDIO_STREAM_TYPE_MP3.
+ * @param startFrame This pass's starting frame offset.
+ * @param endFrame This pass's stop frame offset.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingStart(
+  audiostream_t *stream, size_t startFrame, size_t endFrame
+);
+
+/**
+ * Copies up to maxFrames already-decoded frames out of the ring into `out`,
+ * advancing the ring's read cursor. Only returns fewer than maxFrames (down
+ * to 0) once *outDecodeDone is also true (the decode thread has produced
+ * everything it ever will for this pass, or hit an error) - otherwise it
+ * waits for a full window rather than handing back a partial one, so a
+ * platform's own output handoff doesn't fragment into many tiny calls just
+ * because decode happened to be a little behind at this exact poll.
+ *
+ * @param stream The audio stream to drain from. Must be AUDIO_STREAM_TYPE_MP3.
+ * @param out Destination buffer, sized for at least maxFrames frames.
+ * @param maxFrames Maximum number of frames to copy out.
+ * @param outDecodeDone Set to true if the decode thread has nothing more to
+ *                       give this pass (cleanly finished or failed).
+ * @return The number of frames actually copied out.
+ */
+size_t audioStreamMp3RingDrain(
+  audiostream_t *stream,
+  int16_t *out,
+  size_t maxFrames,
+  bool_t *outDecodeDone
+);
+
+/**
+ * True if the decode thread hit a real read error this pass (see the
+ * `failed` field's own comment) - a platform's Feed() checks this once
+ * Drain() has returned 0 frames with decode reported done, to decide
+ * whether that's a genuine end of stream or something to surface as an
+ * error.
+ *
+ * @param stream The audio stream to check. Must be AUDIO_STREAM_TYPE_MP3.
+ * @return true if the decode thread failed this pass.
+ */
+bool_t audioStreamMp3RingFailed(const audiostream_t *stream);
+
+/**
+ * The decode thread's own entry point (passed to threadInit()). Reads
+ * forward via audioStreamRead() in AUDIO_MP3_RING_STEP_FRAMES chunks,
+ * writing into the ring whenever there's room, until it reaches this
+ * pass's endFrame, the underlying read returns 0 frames, or a real read
+ * error occurs - never touches stream->onLoop/onEnd or any other
+ * main-thread-owned stream state, only the ring and its own captured
+ * startFrame/endFrame.
+ *
+ * @param thread This ring's own thread_t (thread->data is the audiostream_t).
+ */
+void audioStreamMp3RingThreadFunc(thread_t *thread);
+
+// The functions below are thin, type-checking wrappers around the ones
+// above, meant to be called unconditionally from a platform's own
+// Init()/Dispose()/Buffer()/Feed() - each is a no-op (or, for
+// audioStreamReadForPlayback(), falls through to a plain audioStreamRead())
+// for anything other than an AUDIO_STREAM_TYPE_MP3 stream, so callers don't
+// need their own `if(stream->type == AUDIO_STREAM_TYPE_MP3)` check at every
+// call site. This is what actually keeps Linux's and Dolphin's own
+// Init/Dispose/Buffer/Feed implementations close to each other - see
+// audiostreamlinux.c/audiostreamdolphin.c for how they're used.
+
+/**
+ * Calls audioStreamMp3RingInit() if stream is an MP3 stream, otherwise a
+ * no-op.
+ *
+ * @param stream The audio stream to initialize.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingInitIfNeeded(audiostream_t *stream);
+
+/**
+ * Calls audioStreamMp3RingDispose() if stream is an MP3 stream, otherwise a
+ * no-op.
+ *
+ * @param stream The audio stream to dispose.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingDisposeIfNeeded(audiostream_t *stream);
+
+/**
+ * Calls audioStreamMp3RingStop() if stream is an MP3 stream, otherwise a
+ * no-op. Safe (and cheap) to call unconditionally at the start of every
+ * platform Buffer() pass, before audioStreamSeek().
+ *
+ * @param stream The audio stream to stop.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingStopIfNeeded(audiostream_t *stream);
+
+/**
+ * Calls audioStreamMp3RingStart() if stream is an MP3 stream, otherwise a
+ * no-op. Safe to call unconditionally once a platform Buffer() pass has
+ * finalized its startFrame/endFrame, after audioStreamSeek().
+ *
+ * @param stream The audio stream to start decoding.
+ * @param startFrame This pass's starting frame offset.
+ * @param endFrame This pass's stop frame offset.
+ * @return Error state if any.
+ */
+errorret_t audioStreamMp3RingStartIfNeeded(
+  audiostream_t *stream, size_t startFrame, size_t endFrame
+);
+
+/**
+ * Reads up to maxFrames frames of decoded PCM for playback into `out`: for
+ * MP3 streams, drains the background decode ring
+ * (audioStreamMp3RingDrain()); for anything else, reads directly via
+ * audioStreamRead(). The single call a platform's own Feed() needs to get
+ * its next window's worth of PCM, regardless of stream type.
+ *
+ * *outNotReadyYet distinguishes the two ways this can return 0 frames
+ * without erroring: true means an MP3 stream's decode thread simply hasn't
+ * produced a full window yet (retry next poll, don't treat as end of
+ * stream); false with 0 frames means genuinely nothing left to read this
+ * pass. A background MP3 decode failure surfaces as a real thrown error
+ * (via errorThrow) rather than through either of these.
+ *
+ * @param stream The audio stream to read from.
+ * @param out Destination buffer, sized for at least maxFrames frames.
+ * @param maxFrames Maximum number of frames to read.
+ * @param outFramesRead Set to the number of frames actually read.
+ * @param outNotReadyYet Set to true if this returned 0 frames only because
+ *                        decode hasn't caught up yet, not because playback
+ *                        has actually reached the end.
+ * @return Error state - a background MP3 decode failure surfaces here.
+ */
+errorret_t audioStreamReadForPlayback(
+  audiostream_t *stream,
+  int16_t *out,
+  size_t maxFrames,
+  size_t *outFramesRead,
+  bool_t *outNotReadyYet
+);