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/include/aec_dump.h

@@ -0,0 +1,114 @@
+/*
+ *  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_INCLUDE_AEC_DUMP_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/deprecation.h"
+
+namespace webrtc {
+
+// Struct for passing current config from APM without having to
+// include protobuf headers.
+struct InternalAPMConfig {
+  InternalAPMConfig();
+  InternalAPMConfig(const InternalAPMConfig&);
+  InternalAPMConfig(InternalAPMConfig&&);
+
+  InternalAPMConfig& operator=(const InternalAPMConfig&);
+  InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
+
+  bool operator==(const InternalAPMConfig& other);
+
+  bool aec_enabled = false;
+  bool aec_delay_agnostic_enabled = false;
+  bool aec_drift_compensation_enabled = false;
+  bool aec_extended_filter_enabled = false;
+  int aec_suppression_level = 0;
+  bool aecm_enabled = false;
+  bool aecm_comfort_noise_enabled = false;
+  int aecm_routing_mode = 0;
+  bool agc_enabled = false;
+  int agc_mode = 0;
+  bool agc_limiter_enabled = false;
+  bool hpf_enabled = false;
+  bool ns_enabled = false;
+  int ns_level = 0;
+  bool transient_suppression_enabled = false;
+  bool noise_robust_agc_enabled = false;
+  bool pre_amplifier_enabled = false;
+  float pre_amplifier_fixed_gain_factor = 1.f;
+  std::string experiments_description = "";
+};
+
+// An interface for recording configuration and input/output streams
+// of the Audio Processing Module. The recordings are called
+// 'aec-dumps' and are stored in a protobuf format defined in
+// debug.proto.
+// The Write* methods are always safe to call concurrently or
+// otherwise for all implementing subclasses. The intended mode of
+// operation is to create a protobuf object from the input, and send
+// it away to be written to file asynchronously.
+class AecDump {
+ public:
+  struct AudioProcessingState {
+    int delay;
+    int drift;
+    int level;
+    bool keypress;
+  };
+
+  virtual ~AecDump() = default;
+
+  // Logs Event::Type INIT message.
+  virtual void WriteInitMessage(const ProcessingConfig& api_format,
+                                int64_t time_now_ms) = 0;
+  RTC_DEPRECATED void WriteInitMessage(const ProcessingConfig& api_format) {
+    WriteInitMessage(api_format, 0);
+  }
+
+  // Logs Event::Type STREAM message. To log an input/output pair,
+  // call the AddCapture* and AddAudioProcessingState methods followed
+  // by a WriteCaptureStreamMessage call.
+  virtual void AddCaptureStreamInput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamOutput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamInput(const int16_t* const data,
+                                     int num_channels,
+                                     int samples_per_channel) = 0;
+  virtual void AddCaptureStreamOutput(const int16_t* const data,
+                                      int num_channels,
+                                      int samples_per_channel) = 0;
+  virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
+  virtual void WriteCaptureStreamMessage() = 0;
+
+  // Logs Event::Type REVERSE_STREAM message.
+  virtual void WriteRenderStreamMessage(const int16_t* const data,
+                                        int num_channels,
+                                        int samples_per_channel) = 0;
+  virtual void WriteRenderStreamMessage(
+      const AudioFrameView<const float>& src) = 0;
+
+  virtual void WriteRuntimeSetting(
+      const AudioProcessing::RuntimeSetting& runtime_setting) = 0;
+
+  // Logs Event::Type CONFIG message.
+  virtual void WriteConfig(const InternalAPMConfig& config) = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_