Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/modules/audio_processing/aec3/aec3_common.h

@@ -0,0 +1,114 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+#ifdef _MSC_VER /* visual c++ */
+#define ALIGN16_BEG __declspec(align(16))
+#define ALIGN16_END
+#else /* gcc or icc */
+#define ALIGN16_BEG
+#define ALIGN16_END __attribute__((aligned(16)))
+#endif
+
+enum class Aec3Optimization { kNone, kSse2, kAvx2, kNeon };
+
+constexpr int kNumBlocksPerSecond = 250;
+
+constexpr int kMetricsReportingIntervalBlocks = 10 * kNumBlocksPerSecond;
+constexpr int kMetricsComputationBlocks = 7;
+constexpr int kMetricsCollectionBlocks =
+    kMetricsReportingIntervalBlocks - kMetricsComputationBlocks;
+
+constexpr size_t kFftLengthBy2 = 64;
+constexpr size_t kFftLengthBy2Plus1 = kFftLengthBy2 + 1;
+constexpr size_t kFftLengthBy2Minus1 = kFftLengthBy2 - 1;
+constexpr size_t kFftLength = 2 * kFftLengthBy2;
+constexpr size_t kFftLengthBy2Log2 = 6;
+
+constexpr int kRenderTransferQueueSizeFrames = 100;
+
+constexpr size_t kMaxNumBands = 3;
+constexpr size_t kFrameSize = 160;
+constexpr size_t kSubFrameLength = kFrameSize / 2;
+
+constexpr size_t kBlockSize = kFftLengthBy2;
+constexpr size_t kBlockSizeLog2 = kFftLengthBy2Log2;
+
+constexpr size_t kExtendedBlockSize = 2 * kFftLengthBy2;
+constexpr size_t kMatchedFilterWindowSizeSubBlocks = 32;
+constexpr size_t kMatchedFilterAlignmentShiftSizeSubBlocks =
+    kMatchedFilterWindowSizeSubBlocks * 3 / 4;
+
+// TODO(peah): Integrate this with how it is done inside audio_processing_impl.
+constexpr size_t NumBandsForRate(int sample_rate_hz) {
+  return static_cast<size_t>(sample_rate_hz / 16000);
+}
+
+constexpr bool ValidFullBandRate(int sample_rate_hz) {
+  return sample_rate_hz == 16000 || sample_rate_hz == 32000 ||
+         sample_rate_hz == 48000;
+}
+
+constexpr int GetTimeDomainLength(int filter_length_blocks) {
+  return filter_length_blocks * kFftLengthBy2;
+}
+
+constexpr size_t GetDownSampledBufferSize(size_t down_sampling_factor,
+                                          size_t num_matched_filters) {
+  return kBlockSize / down_sampling_factor *
+         (kMatchedFilterAlignmentShiftSizeSubBlocks * num_matched_filters +
+          kMatchedFilterWindowSizeSubBlocks + 1);
+}
+
+constexpr size_t GetRenderDelayBufferSize(size_t down_sampling_factor,
+                                          size_t num_matched_filters,
+                                          size_t filter_length_blocks) {
+  return GetDownSampledBufferSize(down_sampling_factor, num_matched_filters) /
+             (kBlockSize / down_sampling_factor) +
+         filter_length_blocks + 1;
+}
+
+// Detects what kind of optimizations to use for the code.
+Aec3Optimization DetectOptimization();
+
+// Computes the log2 of the input in a fast an approximate manner.
+float FastApproxLog2f(const float in);
+
+// Returns dB from a power quantity expressed in log2.
+float Log2TodB(const float in_log2);
+
+static_assert(1 << kBlockSizeLog2 == kBlockSize,
+              "Proper number of shifts for blocksize");
+
+static_assert(1 << kFftLengthBy2Log2 == kFftLengthBy2,
+              "Proper number of shifts for the fft length");
+
+static_assert(1 == NumBandsForRate(16000), "Number of bands for 16 kHz");
+static_assert(2 == NumBandsForRate(32000), "Number of bands for 32 kHz");
+static_assert(3 == NumBandsForRate(48000), "Number of bands for 48 kHz");
+
+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  // MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_