Changed OLA window for neteq. Old code didnt work well with 48khz

webrtc/modules/audio_coding/neteq/normal.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
 
 #include <string.h>  // Access to size_t.
 
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Forward declarations.
 class BackgroundNoise;
 class DecoderDatabase;
 class Expand;
 
 // This class provides the "Normal" DSP operation, that is performed when
 // there is no data loss, no need to stretch the timing of the signal, and
 // no other "special circumstances" are at hand.
 class Normal {
  public:
-  Normal(int fs_hz, DecoderDatabase* decoder_database,
+  Normal(int fs_hz,
+         DecoderDatabase* decoder_database,
          const BackgroundNoise& background_noise,
          Expand* expand)
       : fs_hz_(fs_hz),
         decoder_database_(decoder_database),
         background_noise_(background_noise),
-        expand_(expand) {
-  }
+        expand_(expand),
+        samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
+        default_win_slope_Q14_((1 << 14) / samples_per_ms_) {}

[hlundin-webrtc, 2017/03/28 12:21:04]

You get compilation errors here for Win64. SEe
https://build.chromium.org/p/tryserver.webrtc/builders/win_compile_x64_dbg/bu....

You must add an explicit cast to int16_t. Suggest you use
rtc::dchecked_cast<int16_t>(...) 
and also 
#include "webrtc/base/safe_conversions.h"

 
   virtual ~Normal() {}
 
   // Performs the "Normal" operation. The decoder data is supplied in |input|,
   // having |length| samples in total for all channels (interleaved). The
   // result is written to |output|. The number of channels allocated in
   // |output| defines the number of channels that will be used when
   // de-interleaving |input|. |last_mode| contains the mode used in the previous
   // GetAudio call (i.e., not the current one), and |external_mute_factor| is
   // a pointer to the mute factor in the NetEqImpl class.
   int Process(const int16_t* input, size_t length,
               Modes last_mode,
               int16_t* external_mute_factor_array,
               AudioMultiVector* output);
 
  private:
   int fs_hz_;
   DecoderDatabase* decoder_database_;
   const BackgroundNoise& background_noise_;
   Expand* expand_;
+  const size_t samples_per_ms_;
+  const int16_t default_win_slope_Q14_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Normal);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_