Add MP3 audio stream support with hardware/software decoder backends

New ASSET_LOADER_TYPE_MP3 (hand-rolled MPEG-1/2/2.5 Layer III header
parser - no third-party dependency needed just for metadata, since PSP's
hardware path doesn't need one at all) plus a shared audiostreammp3.c
stream layer mirroring audiostreampcm.c's shape. Generalized the stream
dispatch (hoisted sampleRate/channels onto audiostream_t, added
audioStreamGetTotalFrames()/Seek()/Read()) so all three platform audio
backends keep working unchanged, just calling the generic names instead
of PCM-specific ones.

Two decoder backends behind one interface: PSP uses the real sceMp3
hardware decoder (firmware-offloaded, lazily initialized on first use);
Linux and Dolphin share one minimp3-based software decoder (public
domain, vendored via CMake FetchContent) - libogc's own MP3Player wraps
libmad (GPL) and drives its own output pipeline, not a fit for the
ansnd-based architecture already in place, so skipped in favor of the
shared minimp3 path.

Fixed three real bugs found via hardware/runtime testing along the way:
- LAME's Xing header counts its own placeholder frame in the declared
  total, which made playback stall permanently one frame short of the
  declared end (looked like "never loops") - fixed by subtracting it.
- sceMp3Decode() can return more PCM than one MPEG frame's worth in a
  single call (PSP's pcmBuf is provisioned for 2x), overflowing the
  shared per-frame decode buffer with no bound check - very intermittent
  corruption/clicking on real hardware. Widened the buffer to the real
  worst case and added an assertion.
- sceMp3ResetPlayPosition()'s exact internal reset semantics aren't
  documented precisely enough to trust for looping - occasionally
  disagreed with the fresh stream position fed right after, clicking at
  the loop boundary about 1 in 3-4 loops. Rewind now fully tears down and
  recreates the decoder instead, the same path already proven correct at
  first Init. Also widened the PSP ring buffer to absorb that now-heavier
  operation, capping each top-up call's own work so the bigger buffer
  doesn't turn into one long blocking decode burst instead.

Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>

src/dusklinux/audio/audiostreamlinux.c

@@ -29,9 +29,9 @@ errorret_t audioStreamLinuxInit(audiostream_t *stream) {
 
   SDL_AudioSpec desired;
   memoryZero(&desired, sizeof(SDL_AudioSpec));
-  desired.freq = (int) stream->pcm.sampleRate;
+  desired.freq = (int) stream->sampleRate;
   desired.format = AUDIO_S16SYS;
-  desired.channels = stream->pcm.channels;
+  desired.channels = stream->channels;
   desired.samples = AUDIO_LINUX_LEAD_FRAMES;
 
   stream->platform.device = SDL_OpenAudioDevice(NULL, 0, &desired, NULL, 0);
@@ -53,8 +53,8 @@ errorret_t audioStreamLinuxDispose(audiostream_t *stream) {
 errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
   assertNotNull(stream, "Stream cannot be NULL.");
 
-  const size_t frameSize = stream->pcm.channels * sizeof(int16_t);
-  const size_t totalFrames = audioStreamPcmGetTotalFrames(stream);
+  const size_t frameSize = stream->channels * sizeof(int16_t);
+  const size_t totalFrames = audioStreamGetTotalFrames(stream);
 
   // Consumed synchronously (right here, in the same call that decided to
   // (re)buffer) rather than left for later - see startFrame's own comment.
@@ -66,7 +66,7 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
   size_t endFrame = totalFrames;
   if((stream->state & AUDIO_STREAM_STATE_LOOPING) && stream->loopStart >= 0) {
     endFrame = mathMin(
-      (size_t) (stream->loopStart * stream->pcm.sampleRate), totalFrames
+      (size_t) (stream->loopStart * stream->sampleRate), totalFrames
     );
   }
   // A seek (or a loop restart landing exactly on the loop end) can put
@@ -84,7 +84,7 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
   if(seeking) {
     SDL_ClearQueuedAudio(stream->platform.device);
   }
-  errorChain(audioStreamPcmSeek(stream, startFrame));
+  errorChain(audioStreamSeek(stream, startFrame));
 
   stream->platform.position = startFrame;
   stream->platform.endFrame = endFrame;
@@ -99,7 +99,7 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
 errorret_t audioStreamLinuxFeed(audiostream_t *stream) {
   assertNotNull(stream, "Stream cannot be NULL.");
 
-  const size_t frameSize = stream->pcm.channels * sizeof(int16_t);
+  const size_t frameSize = stream->channels * sizeof(int16_t);
   const size_t framesRemaining = (
     stream->platform.position < stream->platform.endFrame
       ? stream->platform.endFrame - stream->platform.position
@@ -112,7 +112,7 @@ errorret_t audioStreamLinuxFeed(audiostream_t *stream) {
 
   int16_t *chunk = memoryAllocate(framesToRead * frameSize);
   size_t framesRead = 0;
-  errorret_t ret = audioStreamPcmRead(stream, chunk, framesToRead, &framesRead);
+  errorret_t ret = audioStreamRead(stream, chunk, framesToRead, &framesRead);
   if(errorIsNotOk(ret)) {
     memoryFree(chunk);
     errorChain(ret);
@@ -156,7 +156,7 @@ bool_t audioStreamLinuxIsFinished(audiostream_t *stream) {
   // empty guarantees a gap by definition: silence has already started by
   // the time "empty" can be observed.
   const Uint32 leadBytes = (Uint32) (
-    AUDIO_LINUX_LEAD_FRAMES * stream->pcm.channels * sizeof(int16_t)
+    AUDIO_LINUX_LEAD_FRAMES * stream->channels * sizeof(int16_t)
   );
 
   if(SDL_GetQueuedAudioSize(stream->platform.device) > leadBytes) {