Use 20 ms as the default audio buffer size on Android

It turns out that Android will not pick a low latency buffer size by default.

src/audio/aaudio/SDL_aaudio.c

@@ -321,11 +321,15 @@ static bool BuildAAudioStream(SDL_AudioDevice *device)
     ctx.AAudioStreamBuilder_setSampleRate(builder, device->spec.freq);
     ctx.AAudioStreamBuilder_setChannelCount(builder, device->spec.channels);
 
-    // If no specific buffer size has been requested, the device will pick the optimal
-	if(SDL_GetHint(SDL_HINT_AUDIO_DEVICE_SAMPLE_FRAMES)) {
-	    ctx.AAudioStreamBuilder_setBufferCapacityInFrames(builder, 2 * device->sample_frames); // AAudio requires that the buffer capacity is at least
-	    ctx.AAudioStreamBuilder_setFramesPerDataCallback(builder, device->sample_frames);      // twice the size of the data callback buffer size
-	}
+    int32_t sample_frames;
+    if (SDL_GetHint(SDL_HINT_AUDIO_DEVICE_SAMPLE_FRAMES)) {
+        sample_frames = device->sample_frames;
+    } else {
+        // Use 20 ms for the default audio buffer size
+        sample_frames = (device->spec.freq / 50);
+    }
+    ctx.AAudioStreamBuilder_setBufferCapacityInFrames(builder, 2 * sample_frames); // AAudio requires that the buffer capacity is at least
+    ctx.AAudioStreamBuilder_setFramesPerDataCallback(builder, sample_frames);      // twice the size of the data callback buffer size
 
     const aaudio_direction_t direction = (recording ? AAUDIO_DIRECTION_INPUT : AAUDIO_DIRECTION_OUTPUT);
     ctx.AAudioStreamBuilder_setDirection(builder, direction);