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}

common_audio/signal_processing/downsample_fast.c

@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/sanitizer.h"
+
+// TODO(Bjornv): Change the function parameter order to WebRTC code style.
+// C version of WebRtcSpl_DownsampleFast() for generic platforms.
+int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
+                              size_t data_in_length,
+                              int16_t* data_out,
+                              size_t data_out_length,
+                              const int16_t* __restrict coefficients,
+                              size_t coefficients_length,
+                              int factor,
+                              size_t delay) {
+  int16_t* const original_data_out = data_out;
+  size_t i = 0;
+  size_t j = 0;
+  int32_t out_s32 = 0;
+  size_t endpos = delay + factor * (data_out_length - 1) + 1;
+
+  // Return error if any of the running conditions doesn't meet.
+  if (data_out_length == 0 || coefficients_length == 0
+                           || data_in_length < endpos) {
+    return -1;
+  }
+
+  rtc_MsanCheckInitialized(coefficients, sizeof(coefficients[0]),
+                           coefficients_length);
+
+  for (i = delay; i < endpos; i += factor) {
+    out_s32 = 2048;  // Round value, 0.5 in Q12.
+
+    for (j = 0; j < coefficients_length; j++) {
+      rtc_MsanCheckInitialized(&data_in[i - j], sizeof(data_in[0]), 1);
+      out_s32 += coefficients[j] * data_in[i - j];  // Q12.
+    }
+
+    out_s32 >>= 12;  // Q0.
+
+    // Saturate and store the output.
+    *data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
+  }
+
+  RTC_DCHECK_EQ(original_data_out + data_out_length, data_out);
+  rtc_MsanCheckInitialized(original_data_out, sizeof(original_data_out[0]),
+                           data_out_length);
+
+  return 0;
+}