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

webrtc/modules/audio_coding/neteq/normal.cc

 /*
  *  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 112 matching lines @@
         // Shift 14 with proper rounding.
         (*output)[channel_ix][i] =
             static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
             external_mute_factor_array[channel_ix] + increment, 16384));
       }
 
       // Interpolate the expanded data into the new vector.
       // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
-      RTC_DCHECK_LT(fs_shift, 3);  // Will always be 0, 1, or, 2.
-      increment = 4 >> fs_shift;
-      int fraction = increment;
-      // Don't interpolate over more samples than what is in output. When this
-      // cap strikes, the interpolation will likely sound worse, but this is an
-      // emergency operation in response to unexpected input.
-      const size_t interp_len_samples =
-          std::min(static_cast<size_t>(8 * fs_mult), output->Size());
-      for (size_t i = 0; i < interp_len_samples; ++i) {
-        // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
-        // now for legacy bit-exactness.
-        RTC_DCHECK_LT(channel_ix, output->Channels());
-        RTC_DCHECK_LT(i, output->Size());
+      size_t win_length = samples_per_ms_;
+      int16_t win_slope_Q14 = default_win_slope_Q14_;
+      RTC_DCHECK_LT(channel_ix, output->Channels());
+      if (win_length > output->Size()) {
+        win_length = output->Size();
+        win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+      }
+      int16_t win_up_Q14 = 0;
+      for (size_t i = 0; i < win_length; i++) {
+        win_up_Q14 += win_slope_Q14;
         (*output)[channel_ix][i] =
-            static_cast<int16_t>((fraction * (*output)[channel_ix][i] +
-                (32 - fraction) * expanded[channel_ix][i] + 8) >> 5);
-        fraction += increment;
+            (win_up_Q14 * (*output)[channel_ix][i] +
+             ((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
+            14;
+      }
+      if (fs_hz_ == 48000) {
+        RTC_DCHECK_EQ(win_up_Q14, (1 << 14) - 16);
+      } else {
+        RTC_DCHECK_EQ(win_up_Q14, 1 << 14);
       }
     }
   } else if (last_mode == kModeRfc3389Cng) {
     RTC_DCHECK_EQ(output->Channels(), 1);  // Not adapted for multi-channel yet.
     static const size_t kCngLength = 48;
     RTC_DCHECK_LE(8 * fs_mult, kCngLength);
     int16_t cng_output[kCngLength];
     // Reset mute factor and start up fresh.
     external_mute_factor_array[0] = 16384;
     ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
 
     if (cng_decoder) {
       // Generate long enough for 48kHz.
       if (!cng_decoder->Generate(cng_output, 0)) {
         // Error returned; set return vector to all zeros.
         memset(cng_output, 0, sizeof(cng_output));
       }
     } else {
       // If no CNG instance is defined, just copy from the decoded data.
       // (This will result in interpolating the decoded with itself.)
       (*output)[0].CopyTo(fs_mult * 8, 0, cng_output);
     }
     // Interpolate the CNG into the new vector.
     // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
-    RTC_DCHECK_LT(fs_shift, 3);  // Will always be 0, 1, or, 2.
-    int16_t increment = 4 >> fs_shift;
-    int16_t fraction = increment;
-    for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
-      // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
-      // for legacy bit-exactness.
-      (*output)[0][i] = (fraction * (*output)[0][i] +
-          (32 - fraction) * cng_output[i] + 8) >> 5;
-      fraction += increment;
+    size_t win_length = samples_per_ms_;
+    int16_t win_slope_Q14 = default_win_slope_Q14_;
+    if (win_length > kCngLength) {
+      win_length = kCngLength;
+      win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+    }
+    int16_t win_up_Q14 = 0;
+    for (size_t i = 0; i < win_length; i++) {
+      win_up_Q14 += win_slope_Q14;
+      (*output)[0][i] =
+          (win_up_Q14 * (*output)[0][i] +
+           ((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
+          14;
+    }
+    if (fs_hz_ == 48000) {
+      RTC_DCHECK_EQ(win_up_Q14, (1 << 14) - 16);
+    } else {
+      RTC_DCHECK_EQ(win_up_Q14, 1 << 14);
     }
   } else if (external_mute_factor_array[0] < 16384) {
     // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
     // still ramping up from previous muting.
     // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
     int increment = 64 / fs_mult;
     size_t length_per_channel = length / output->Channels();
     for (size_t i = 0; i < length_per_channel; i++) {
       for (size_t channel_ix = 0; channel_ix < output->Channels();
           ++channel_ix) {
         // Scale with mute factor.
         RTC_DCHECK_LT(channel_ix, output->Channels());
         RTC_DCHECK_LT(i, output->Size());
         int32_t scaled_signal = (*output)[channel_ix][i] *
             external_mute_factor_array[channel_ix];
         // Shift 14 with proper rounding.
         (*output)[channel_ix][i] =
             static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
             16384, external_mute_factor_array[channel_ix] + increment));
       }
     }
   }
 
   return static_cast<int>(length);
 }
 
 }  // namespace webrtc