SDL: audio: Fix resampler overflowing input buffer.

SDL Commits
1 
From 91cd5478be5aaf684636d64662c4a63acb09afdc Mon Sep 17 00:00:00 2001
From: "Ryan C. Gordon" <[EMAIL REDACTED]>
Date: Sat, 12 Aug 2023 00:52:11 -0400
Subject: [PATCH] audio: Fix resampler overflowing input buffer.

This usually manifests as a clicking sound, because it often produces
a value outside the range -1.0f to 1.0f based on whatever random data
is past the buffer, which later stages of audio conversion will clamp
to a maximum value for the audio format. Since this tends to be a single
bad sample generated at the end of the resampled buffer, it sounds like
a repeating click in streamed data.

I'd like a more efficient means to clamp this value to not overflow the
buffer, but this puts out the immediate fire.
---
 src/audio/SDL_audiocvt.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

diff --git a/src/audio/SDL_audiocvt.c b/src/audio/SDL_audiocvt.c
index f1517ae6c188..d16c1c337cf9 100644
--- a/src/audio/SDL_audiocvt.c
+++ b/src/audio/SDL_audiocvt.c
@@ -81,7 +81,11 @@ static void ResampleAudio(const int chans, const int inrate, const int outrate,
     int i, j, chan;
 
     for (i = 0; i < outframes; i++) {
-        const int srcindex = (int)((Sint64)i * inrate / outrate);
+        int srcindex = (int)((Sint64)i * inrate / outrate);
+        if (srcindex >= inframes) {  // !!! FIXME: can we clamp this without an if statement on each iteration?
+            srcindex = inframes - 1;
+        }
+
         /* Calculating the following way avoids subtraction or modulo of large
          * floats which have low result precision.
          *   interpolation1