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Diffstat (limited to 'src/common_audio/signal_processing_library/main/source/get_scaling_square.c')
-rw-r--r--src/common_audio/signal_processing_library/main/source/get_scaling_square.c44
1 files changed, 44 insertions, 0 deletions
diff --git a/src/common_audio/signal_processing_library/main/source/get_scaling_square.c b/src/common_audio/signal_processing_library/main/source/get_scaling_square.c
new file mode 100644
index 0000000000..dccbf334fd
--- /dev/null
+++ b/src/common_audio/signal_processing_library/main/source/get_scaling_square.c
@@ -0,0 +1,44 @@
+/*
+ * 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_GetScalingSquare().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_GetScalingSquare(WebRtc_Word16 *in_vector, int in_vector_length, int times)
+{
+ int nbits = WebRtcSpl_GetSizeInBits(times);
+ int i;
+ WebRtc_Word16 smax = -1;
+ WebRtc_Word16 sabs;
+ WebRtc_Word16 *sptr = in_vector;
+ int t;
+ int looptimes = in_vector_length;
+
+ for (i = looptimes; i > 0; i--)
+ {
+ sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
+ smax = (sabs > smax ? sabs : smax);
+ }
+ t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
+
+ if (smax == 0)
+ {
+ return 0; // Since norm(0) returns 0
+ } else
+ {
+ return (t > nbits) ? 0 : nbits - t;
+ }
+}