Fixing sample-rate dependent band-split filter issues in AEC3

webrtc/modules/audio_processing/aec3/aec3_common.h

 /*
  *  Copyright (c) 2016 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 50 matching lines @@
 constexpr float kFixedEchoPathGain = 100;
 
 constexpr size_t kRenderDelayBufferSize =
     (3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize);
 
 constexpr size_t kMaxApiCallsJitterBlocks = 20;
 constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2;
 static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks,
               "Requirement to ensure buffer overflow detection");
 
+// TODO(peah): Integrate this with how it is done inside audio_processing_impl.
 constexpr size_t NumBandsForRate(int sample_rate_hz) {
+#ifdef WEBRTC_ARCH_ARM_FAMILY
+  return std::min(2, static_cast<size_t>(

[hlundin-webrtc, 2017/04/07 09:55:06]

I think this would be more readable as 
return static_cast<size_t>(sample_rate_hz == 8000 ? 1 : 2);

[hlundin-webrtc, 2017/04/07 09:56:35]

My bad. How about
return static_cast<size_t>(sample_rate_hz <= 16000 ? 1 : 2);

[peah-webrtc, 2017/04/07 10:00:16]

Awesome! Much better!
Done.

+                         sample_rate_hz == 8000 ? 1 : sample_rate_hz / 16000));
+#else
   return static_cast<size_t>(sample_rate_hz == 8000 ? 1
                                                     : sample_rate_hz / 16000);
+#endif
 }
 constexpr int LowestBandRate(int sample_rate_hz) {
   return sample_rate_hz == 8000 ? sample_rate_hz : 16000;
 }
 
 constexpr bool ValidFullBandRate(int sample_rate_hz) {
   return sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
          sample_rate_hz == 32000 || sample_rate_hz == 48000;
 }
 
 // Detects what kind of optimizations to use for the code.
 Aec3Optimization DetectOptimization();
 
 static_assert(1 == NumBandsForRate(8000), "Number of bands for 8 kHz");
 static_assert(1 == NumBandsForRate(16000), "Number of bands for 16 kHz");
 static_assert(2 == NumBandsForRate(32000), "Number of bands for 32 kHz");
+#ifdef WEBRTC_ARCH_ARM_FAMILY
+static_assert(2 == NumBandsForRate(48000), "Number of bands for 48 kHz");
+#else
 static_assert(3 == NumBandsForRate(48000), "Number of bands for 48 kHz");
+#endif
 
 static_assert(8000 == LowestBandRate(8000), "Sample rate of band 0 for 8 kHz");
 static_assert(16000 == LowestBandRate(16000),
               "Sample rate of band 0 for 16 kHz");
 static_assert(16000 == LowestBandRate(32000),
               "Sample rate of band 0 for 32 kHz");
 static_assert(16000 == LowestBandRate(48000),
               "Sample rate of band 0 for 48 kHz");
 
 static_assert(ValidFullBandRate(8000),
               "Test that 8 kHz is a valid sample rate");
 static_assert(ValidFullBandRate(16000),
               "Test that 16 kHz is a valid sample rate");
 static_assert(ValidFullBandRate(32000),
               "Test that 32 kHz is a valid sample rate");
 static_assert(ValidFullBandRate(48000),
               "Test that 48 kHz is a valid sample rate");
 static_assert(!ValidFullBandRate(8001),
               "Test that 8001 Hz is not a valid sample rate");
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_