From c7629704b470e75326cb6d181940de3f90958aef Mon Sep 17 00:00:00 2001
From: "Ryan C. Gordon" <[EMAIL REDACTED]>
Date: Tue, 9 May 2023 10:56:34 -0400
Subject: [PATCH] audio: SDL_ConvertAudioSamples shouldn't calculate its output
buffer size.
Just use what the AudioStream calculates instead.
---
src/audio/SDL_audio.c | 81 +++++++++++++++++--------------------------
1 file changed, 32 insertions(+), 49 deletions(-)
diff --git a/src/audio/SDL_audio.c b/src/audio/SDL_audio.c
index 65729a1f617f..487c7a5593f0 100644
--- a/src/audio/SDL_audio.c
+++ b/src/audio/SDL_audio.c
@@ -1555,73 +1555,56 @@ void SDL_CalculateAudioSpec(SDL_AudioSpec *spec)
spec->size *= spec->samples;
}
-int SDL_ConvertAudioSamples(
- SDL_AudioFormat src_format, Uint8 src_channels, int src_rate, const Uint8 *src_data, int src_len,
- SDL_AudioFormat dst_format, Uint8 dst_channels, int dst_rate, Uint8 **dst_data, int *dst_len)
+/* !!! FIXME: move this to SDL_audiocvt.c */
+int SDL_ConvertAudioSamples(SDL_AudioFormat src_format, Uint8 src_channels, int src_rate, const Uint8 *src_data, int src_len,
+ SDL_AudioFormat dst_format, Uint8 dst_channels, int dst_rate, Uint8 **dst_data, int *dst_len)
{
int ret = -1;
SDL_AudioStream *stream = NULL;
+ Uint8 *dst = NULL;
+ int dstlen = 0;
- int src_samplesize, dst_samplesize;
- int real_dst_len;
+ if (dst_data) {
+ *dst_data = NULL;
+ }
+ if (dst_len) {
+ *dst_len = 0;
+ }
if (src_data == NULL) {
return SDL_InvalidParamError("src_data");
- }
- if (src_len < 0) {
+ } else if (src_len < 0) {
return SDL_InvalidParamError("src_len");
- }
- if (dst_data == NULL) {
+ } else if (dst_data == NULL) {
return SDL_InvalidParamError("dst_data");
- }
- if (dst_len == NULL) {
+ } else if (dst_len == NULL) {
return SDL_InvalidParamError("dst_len");
}
- *dst_data = NULL;
- *dst_len = 0;
-
stream = SDL_CreateAudioStream(src_format, src_channels, src_rate, dst_format, dst_channels, dst_rate);
- if (stream == NULL) {
- goto end;
- }
-
- src_samplesize = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels;
- dst_samplesize = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels;
-
- src_len &= ~(src_samplesize - 1);
- *dst_len = dst_samplesize * (src_len / src_samplesize);
- if (src_rate < dst_rate) {
- const double mult = ((double)dst_rate) / ((double)src_rate);
- *dst_len *= (int) SDL_ceil(mult);
- }
-
- *dst_len &= ~(dst_samplesize - 1);
- *dst_data = (Uint8 *)SDL_malloc(*dst_len);
- if (*dst_data == NULL) {
- goto end;
- }
-
- if (SDL_PutAudioStreamData(stream, src_data, src_len) < 0 ||
- SDL_FlushAudioStream(stream) < 0) {
- goto end;
+ if (stream != NULL) {
+ if ((SDL_PutAudioStreamData(stream, src_data, src_len) == 0) && (SDL_FlushAudioStream(stream) == 0)) {
+ dstlen = SDL_GetAudioStreamAvailable(stream);
+ if (dstlen >= 0) {
+ dst = (Uint8 *)SDL_malloc(dstlen);
+ if (!dst) {
+ SDL_OutOfMemory();
+ } else {
+ ret = (SDL_GetAudioStreamData(stream, dst, dstlen) >= 0) ? 0 : -1;
+ }
+ }
+ }
}
- real_dst_len = SDL_GetAudioStreamData(stream, *dst_data, *dst_len);
- if (real_dst_len < 0) {
- goto end;
+ if (ret == -1) {
+ SDL_free(dst);
+ } else {
+ *dst_data = dst;
+ *dst_len = dstlen;
}
- *dst_len = real_dst_len;
- ret = 0;
-
-end:
- if (ret != 0) {
- SDL_free(*dst_data);
- *dst_data = NULL;
- *dst_len = 0;
- }
SDL_DestroyAudioStream(stream);
return ret;
}
+