diff --git a/packages/mediacenter/xbmc/patches/xbmc-990.24-PR2375.patch b/packages/mediacenter/xbmc/patches/xbmc-990.24-PR2375.patch
new file mode 100644
index 0000000000..ddf3dd7e81
--- /dev/null
+++ b/packages/mediacenter/xbmc/patches/xbmc-990.24-PR2375.patch
@@ -0,0 +1,31 @@
+From 8717c162b8c5092d77672351ce3bfa2d4e7e32d8 Mon Sep 17 00:00:00 2001
+From: fritsch <peter.fruehberger@gmail.com>
+Date: Tue, 5 Mar 2013 22:17:15 +0100
+Subject: [PATCH] AE: Fix resample of e.g. 192 khz to 48 khz audio by scaling
+ the usual suspects with the src sampleRate
+
+---
+ xbmc/cores/AudioEngine/Engines/SoftAE/SoftAEStream.cpp |    7 +++++++
+ 1 file changed, 7 insertions(+)
+
+diff --git a/xbmc/cores/AudioEngine/Engines/SoftAE/SoftAEStream.cpp b/xbmc/cores/AudioEngine/Engines/SoftAE/SoftAEStream.cpp
+index 258dcac..a64beb1 100644
+--- a/xbmc/cores/AudioEngine/Engines/SoftAE/SoftAEStream.cpp
++++ b/xbmc/cores/AudioEngine/Engines/SoftAE/SoftAEStream.cpp
+@@ -196,6 +196,13 @@ void CSoftAEStream::Initialize()
+     m_ssrcData.data_out      = (float*)_aligned_malloc(m_format.m_frameSamples * (int)std::ceil(m_ssrcData.src_ratio) * sizeof(float), 16);
+     m_ssrcData.output_frames = m_format.m_frames * (long)std::ceil(m_ssrcData.src_ratio);
+     m_ssrcData.end_of_input  = 0;
++    // we must buffer the same amount as before but taking the source sample rate into account
++    // there is no reason to decrease the buffer for upsampling
++    if (m_internalRatio < 1)
++    {
++      m_waterLevel *= (1.0 / m_internalRatio);
++      m_refillBuffer = m_waterLevel;
++    }
+   }
+ 
+   m_limiter.SetSamplerate(AE.GetSampleRate());
+-- 
+1.7.10
+