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/render_delay_buffer.h

@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2017 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_RENDER_DELAY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+
+namespace webrtc {
+
+// Class for buffering the incoming render blocks such that these may be
+// extracted with a specified delay.
+class RenderDelayBuffer {
+ public:
+  enum class BufferingEvent {
+    kNone,
+    kRenderUnderrun,
+    kRenderOverrun,
+    kApiCallSkew
+  };
+
+  static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
+                                   int sample_rate_hz,
+                                   size_t num_render_channels);
+  virtual ~RenderDelayBuffer() = default;
+
+  // Resets the buffer alignment.
+  virtual void Reset() = 0;
+
+  // Inserts a block into the buffer.
+  virtual BufferingEvent Insert(
+      const std::vector<std::vector<std::vector<float>>>& block) = 0;
+
+  // Updates the buffers one step based on the specified buffer delay. Returns
+  // an enum indicating whether there was a special event that occurred.
+  virtual BufferingEvent PrepareCaptureProcessing() = 0;
+
+  // Called on capture blocks where PrepareCaptureProcessing is not called.
+  virtual void HandleSkippedCaptureProcessing() = 0;
+
+  // Sets the buffer delay and returns a bool indicating whether the delay
+  // changed.
+  virtual bool AlignFromDelay(size_t delay) = 0;
+
+  // Sets the buffer delay from the most recently reported external delay.
+  virtual void AlignFromExternalDelay() = 0;
+
+  // Gets the buffer delay.
+  virtual size_t Delay() const = 0;
+
+  // Gets the buffer delay.
+  virtual size_t MaxDelay() const = 0;
+
+  // Returns the render buffer for the echo remover.
+  virtual RenderBuffer* GetRenderBuffer() = 0;
+
+  // Returns the downsampled render buffer.
+  virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
+
+  // Returns the maximum non calusal offset that can occur in the delay buffer.
+  static int DelayEstimatorOffset(const EchoCanceller3Config& config);
+
+  // Provides an optional external estimate of the audio buffer delay.
+  virtual void SetAudioBufferDelay(int delay_ms) = 0;
+
+  // Returns whether an external delay estimate has been reported via
+  // SetAudioBufferDelay.
+  virtual bool HasReceivedBufferDelay() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_