Fixed PSP trying to load entirely into memory

src/dusklinux/audio/audiostreamlinux.c

@@ -10,12 +10,20 @@
 #include "assert/assert.h"
 #include "util/memory.h"
 #include "util/math.h"
+#include "error/error.h"
 
 // How many frames of lead time to keep queued ahead of playback. Matches
 // SDL_AudioSpec.samples below - the device's own internal buffer size - so
 // this is "never let the queue run drier than one SDL-internal buffer."
 #define AUDIO_LINUX_LEAD_FRAMES 4096
 
+// How many frames audioStreamLinuxFeed() reads and queues per call - a
+// few multiples of the lead margin, so one top-up comfortably outlasts
+// the time between Update() calls without ever needing to hold more than
+// this much decoded audio in memory at once, regardless of how long the
+// underlying clip is.
+#define AUDIO_LINUX_WINDOW_FRAMES (AUDIO_LINUX_LEAD_FRAMES * 4)
+
 errorret_t audioStreamLinuxInit(audiostream_t *stream) {
   assertNotNull(stream, "Stream cannot be NULL.");
 
@@ -46,7 +54,7 @@ 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 = stream->dataSize / frameSize;
+  const size_t totalFrames = audioStreamPcmGetTotalFrames(stream);
 
   // Consumed synchronously (right here, in the same call that decided to
   // (re)buffer) rather than left for later - see startFrame's own comment.
@@ -67,57 +75,106 @@ errorret_t audioStreamLinuxBuffer(audiostream_t *stream) {
   // as PSP.
   if(startFrame >= endFrame) endFrame = totalFrames;
 
-  const size_t bytes = (endFrame - startFrame) * frameSize;
-  const uint8_t *segment = stream->data + (startFrame * frameSize);
-
   // Only an explicit seek discards whatever's still queued and jumps -
   // a natural loop restart deliberately leaves the previous pass's tail
   // (AUDIO_LINUX_LEAD_FRAMES worth) queued and appends the new pass after
   // it, which is what makes looping gapless (see IsFinished()'s comment).
-  // Clearing on every Buffer() call would destroy that overlap and
+  // Clearing here on every pass would destroy that overlap and
   // reintroduce the exact gap this was built to avoid.
   if(seeking) {
     SDL_ClearQueuedAudio(stream->platform.device);
   }
+  errorChain(audioStreamPcmSeek(stream, startFrame));
 
-  int queued;
-  if(stream->volume == 0xFF) {
-    // Nothing to mix at full volume - queue the segment directly instead of
-    // allocating/zeroing a same-size scratch buffer just to copy it in.
-    queued = SDL_QueueAudio(stream->platform.device, segment, (Uint32) bytes);
-  } else {
-    uint8_t *mixed = memoryAllocate(bytes);
-    memoryZero(mixed, bytes);
-    SDL_MixAudioFormat(
-      mixed, segment, AUDIO_S16SYS, (Uint32) bytes,
-      (stream->volume * SDL_MIX_MAXVOLUME) / 0xFF
-    );
-    queued = SDL_QueueAudio(stream->platform.device, mixed, (Uint32) bytes);
-    memoryFree(mixed);
-  }
+  stream->platform.position = startFrame;
+  stream->platform.endFrame = endFrame;
 
-  if(queued != 0) {
-    errorThrow("Failed to queue SDL2 audio data: %s", SDL_GetError());
-  }
+  errorChain(audioStreamLinuxFeed(stream));
 
   SDL_PauseAudioDevice(stream->platform.device, 0);
 
   errorOk();
 }
 
+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 framesRemaining = (
+    stream->platform.position < stream->platform.endFrame
+      ? stream->platform.endFrame - stream->platform.position
+      : 0
+  );
+  const size_t framesToRead = mathMin(framesRemaining, AUDIO_LINUX_WINDOW_FRAMES);
+  if(framesToRead == 0) {
+    errorOk();
+  }
+
+  int16_t *chunk = memoryAllocate(framesToRead * frameSize);
+  size_t framesRead = 0;
+  errorret_t ret = audioStreamPcmRead(stream, chunk, framesToRead, &framesRead);
+  if(errorIsNotOk(ret)) {
+    memoryFree(chunk);
+    errorChain(ret);
+  }
+
+  const size_t bytesRead = framesRead * frameSize;
+  int queued;
+
+  if(stream->volume == 0xFF) {
+    // Nothing to mix at full volume - queue the window directly instead of
+    // allocating/zeroing a same-size scratch buffer just to copy it in.
+    queued = SDL_QueueAudio(stream->platform.device, chunk, (Uint32) bytesRead);
+  } else {
+    uint8_t *mixed = memoryAllocate(bytesRead);
+    memoryZero(mixed, bytesRead);
+    SDL_MixAudioFormat(
+      mixed, (uint8_t *) chunk, AUDIO_S16SYS, (Uint32) bytesRead,
+      (stream->volume * SDL_MIX_MAXVOLUME) / 0xFF
+    );
+    queued = SDL_QueueAudio(stream->platform.device, mixed, (Uint32) bytesRead);
+    memoryFree(mixed);
+  }
+  memoryFree(chunk);
+
+  if(queued != 0) {
+    errorThrow("Failed to queue SDL2 audio data: %s", SDL_GetError());
+  }
+
+  stream->platform.position += framesRead;
+  errorOk();
+}
+
 bool_t audioStreamLinuxIsFinished(audiostream_t *stream) {
   assertNotNull(stream, "Stream cannot be NULL.");
 
-  // Reports "finished" (ready to be re-buffered) with AUDIO_LINUX_LEAD_FRAMES
-  // of margin still queued, rather than waiting for the queue to actually
-  // run dry. SDL_QueueAudio only ever appends to a FIFO, so queuing the next
-  // loop this early never causes overlap - it just avoids ever going silent
-  // while our once-per-frame Update() notices and catches up. Waiting for
-  // truly empty (as this used to) guarantees a gap by definition: silence
-  // has already started by the time "empty" can be observed.
+  // Reports "finished" (ready to loop/end) with AUDIO_LINUX_LEAD_FRAMES of
+  // margin still queued, rather than waiting for the queue to actually run
+  // dry - queuing the next pass this early never causes overlap (SDL's
+  // queue is a plain FIFO), it just avoids ever going silent while our
+  // once-per-frame Update() notices and catches up. Waiting for truly
+  // 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)
   );
 
-  return SDL_GetQueuedAudioSize(stream->platform.device) <= leadBytes;
+  if(SDL_GetQueuedAudioSize(stream->platform.device) > leadBytes) {
+    return false;
+  }
+
+  // Below the lead margin - if more of the current pass is left to read,
+  // top up now rather than reporting finished, which would otherwise
+  // trigger a full loop-restart/end while still mid-pass, just because
+  // the queue happened to run low.
+  if(stream->platform.position < stream->platform.endFrame) {
+    errorret_t ret = audioStreamLinuxFeed(stream);
+    if(errorIsNotOk(ret)) {
+      errorCatch(errorPrint(ret));
+      return true; // Can't recover - let the shared layer end/loop it.
+    }
+    return false;
+  }
+
+  return true;
 }