1 files changed, 59 insertions, 0 deletions
diff --git a/src/common_audio/signal_processing/downsample_fast.c b/src/common_audio/signal_processing/downsample_fast.c
new file mode 100644
index 0000000000..cce463c5d3
--- /dev/null
+++ b/src/common_audio/signal_processing/downsample_fast.c
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_DownsampleFast().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_DownsampleFast(WebRtc_Word16 *in_ptr, WebRtc_Word16 in_length,
+                             WebRtc_Word16 *out_ptr, WebRtc_Word16 out_length,
+                             WebRtc_Word16 *B, WebRtc_Word16 B_length, WebRtc_Word16 factor,
+                             WebRtc_Word16 delay)
+{
+    WebRtc_Word32 o;
+    int i, j;
+
+    WebRtc_Word16 *downsampled_ptr = out_ptr;
+    WebRtc_Word16 *b_ptr;
+    WebRtc_Word16 *x_ptr;
+    WebRtc_Word16 endpos = delay
+            + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(factor, (out_length - 1)) + 1;
+
+    if (in_length < endpos)
+    {
+        return -1;
+    }
+
+    for (i = delay; i < endpos; i += factor)
+    {
+        b_ptr = &B[0];
+        x_ptr = &in_ptr[i];
+
+        o = (WebRtc_Word32)2048; // Round val
+
+        for (j = 0; j < B_length; j++)
+        {
+            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
+        }
+
+        o = WEBRTC_SPL_RSHIFT_W32(o, 12);
+
+        // If output is higher than 32768, saturate it. Same with negative side
+
+        *downsampled_ptr++ = WebRtcSpl_SatW32ToW16(o);
+    }
+
+    return 0;
+}