From 40a6a445ce2c9777aade8517b294115616586d07 Mon Sep 17 00:00:00 2001
From: Brick <[EMAIL REDACTED]>
Date: Sun, 20 Aug 2023 16:15:03 +0100
Subject: [PATCH] Update resample_offset inside ResampleAudio
---
src/audio/SDL_audiocvt.c | 15 ++++++---------
1 file changed, 6 insertions(+), 9 deletions(-)
diff --git a/src/audio/SDL_audiocvt.c b/src/audio/SDL_audiocvt.c
index 68a0976adfbb..af1ce64513b5 100644
--- a/src/audio/SDL_audiocvt.c
+++ b/src/audio/SDL_audiocvt.c
@@ -67,14 +67,14 @@ static int GetHistoryBufferSampleFrames(const Sint32 required_resampler_frames)
static void ResampleAudio(const int chans, const int inrate, const int outrate,
const float *lpadding, const float *rpadding,
const float *inbuf, const int inframes,
- float *outbuf, const int outframes, const Sint64 offset)
+ float *outbuf, const int outframes, Sint64* resample_offset)
{
const int paddinglen = GetResamplerPaddingFrames(inrate, outrate);
float *dst = outbuf;
int i, j, chan;
const Sint64 srcstep = GetResampleRate(inrate, outrate);
- Sint64 srcpos = offset;
+ Sint64 srcpos = *resample_offset;
for (i = 0; i < outframes; i++) {
int srcindex = (int)(Sint32)(srcpos >> 32);
@@ -116,6 +116,8 @@ static void ResampleAudio(const int chans, const int inrate, const int outrate,
*(dst++) = outsample;
}
}
+
+ *resample_offset = srcpos - ((Sint64)inframes << 32);
}
/*
@@ -825,7 +827,6 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int le
int future_buffer_filled_frames = stream->future_buffer_filled_frames;
Uint8 *future_buffer = stream->future_buffer;
Uint8 *history_buffer = stream->history_buffer;
- Sint64 resample_offset = stream->resample_offset;
float *resample_outbuf;
int input_frames;
int output_frames;
@@ -854,12 +855,8 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int le
if (dst_rate != src_rate) {
// Make sure this matches the logic used in ResampleAudio
const Sint64 srcstep = GetResampleRate(src_rate, dst_rate);
-
- Sint64 nextpos = (output_frames * srcstep) + resample_offset;
- Sint64 lastpos = nextpos - srcstep;
-
+ const Sint64 lastpos = ((output_frames - 1) * srcstep) + stream->resample_offset;
input_frames = (int)(Sint32)(lastpos >> 32) + 1;
- stream->resample_offset = nextpos - ((Sint64)input_frames << 32);
if (input_frames == 0) { // uhoh, not enough input frames!
// if they are upsampling and we end up needing less than a frame of input, we reject it because it would cause artifacts on future reads to eat a full input frame.
@@ -989,7 +986,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int le
ResampleAudio(pre_resample_channels, src_rate, dst_rate,
stream->left_padding, stream->right_padding,
(const float *) workbuf, input_frames,
- resample_outbuf, output_frames, resample_offset);
+ resample_outbuf, output_frames, &stream->resample_offset);
// Get us to the final format!
// see if we can do the conversion in-place (will fit in `buf` while in-progress), or if we need to do it in the workbuf and copy it over