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.
  */
 
@@ skipping 21 matching lines @@
 // no other "special circumstances" are at hand.
 class Normal {
  public:
   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) {
+          // Setup Overlapp Add to 1ms
+          ola_win_length_ = fs_hz / 1000;

[hlundin-webrtc, 2017/03/23 11:41:51]

Make these two member variables const, and set them in the initializer list of
the ctor instead.

[hlundin-webrtc, 2017/03/23 11:41:51]

rtc::CheckedDivExact(fs_hz_, 1000)

+          ola_win_slope_Q14_ = (1 << 14)/ola_win_length_;

[hlundin-webrtc, 2017/03/23 11:41:51]

Add spaces on both sides of /.
I suggest you run "git cl format" to fix all formatting nits.

   }
 
   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_;

[hlundin-webrtc, 2017/03/23 11:41:51]

Please, make this const while you are at it.

   DecoderDatabase* decoder_database_;
   const BackgroundNoise& background_noise_;
   Expand* expand_;
 

[hlundin-webrtc, 2017/03/23 11:41:52]

No blank line, please.

+  int ola_win_length_;

[hlundin-webrtc, 2017/03/23 11:41:51]

const size_t samples_per_ms_;

+  int16_t ola_win_slope_Q14_;

[hlundin-webrtc, 2017/03/23 11:41:51]

const int16_t default_win_slope_Q14_;

+
   RTC_DISALLOW_COPY_AND_ASSIGN(Normal);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_