Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/codecs/ilbc/hp_input.c

@@ -0,0 +1,88 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.c
+
+******************************************************************/
+
+#include "defines.h"
+
+/*----------------------------------------------------------------*
+ *  high-pass filter of input with *0.5 and saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpInput(
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
+                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                                                                   is assumed to be 1.0 */
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                                                                   yhi[n-2] ylow[n-2] */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len)      /* (i)   Number of samples to filter */
+{
+  size_t i;
+  int32_t tmpW32;
+  int32_t tmpW32b;
+
+  for (i=0; i<len; i++) {
+
+    /*
+        y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+        + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+    */
+
+    tmpW32 = y[1] * ba[3];  /* (-a[1])*y[i-1] (low part) */
+    tmpW32 += y[3] * ba[4];  /* (-a[2])*y[i-2] (low part) */
+    tmpW32 = (tmpW32>>15);
+    tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
+    tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
+    tmpW32 = (tmpW32<<1);
+
+    tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
+    tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
+    tmpW32 += x[1] * ba[2];  /* b[2]*x[i-2] */
+
+    /* Update state (input part) */
+    x[1] = x[0];
+    x[0] = signal[i];
+
+    /* Rounding in Q(12+1), i.e. add 2^12 */
+    tmpW32b = tmpW32 + 4096;
+
+    /* Saturate (to 2^28) so that the HP filtered signal does not overflow */
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)268435455, tmpW32b, (int32_t)-268435456);
+
+    /* Convert back to Q0 and multiply with 0.5 */
+    signal[i] = (int16_t)(tmpW32b >> 13);
+
+    /* Update state (filtered part) */
+    y[2] = y[0];
+    y[3] = y[1];
+
+    /* upshift tmpW32 by 3 with saturation */
+    if (tmpW32>268435455) {
+      tmpW32 = WEBRTC_SPL_WORD32_MAX;
+    } else if (tmpW32<-268435456) {
+      tmpW32 = WEBRTC_SPL_WORD32_MIN;
+    } else {
+      tmpW32 <<= 3;
+    }
+
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1);
+  }
+
+  return;
+}