Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 62 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.h b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.h
new file mode 100644
index 0000000000..2e4aaa4ba7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.h
@@ -0,0 +1,62 @@
+/*
+ *  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_SIGNAL_ANALYZER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
+
+#include <algorithm>
+#include <array>
+#include <cstddef>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Provides functionality for analyzing the properties of the render signal.
+class RenderSignalAnalyzer {
+ public:
+  explicit RenderSignalAnalyzer(const EchoCanceller3Config& config);
+  ~RenderSignalAnalyzer();
+
+  RenderSignalAnalyzer(const RenderSignalAnalyzer&) = delete;
+  RenderSignalAnalyzer& operator=(const RenderSignalAnalyzer&) = delete;
+
+  // Updates the render signal analysis with the most recent render signal.
+  void Update(const RenderBuffer& render_buffer,
+              const absl::optional<size_t>& delay_partitions);
+
+  // Returns true if the render signal is poorly exciting.
+  bool PoorSignalExcitation() const {
+    RTC_DCHECK_LT(2, narrow_band_counters_.size());
+    return std::any_of(narrow_band_counters_.begin(),
+                       narrow_band_counters_.end(),
+                       [](size_t a) { return a > 10; });
+  }
+
+  // Zeros the array around regions with narrow bands signal characteristics.
+  void MaskRegionsAroundNarrowBands(
+      std::array<float, kFftLengthBy2Plus1>* v) const;
+
+  absl::optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
+
+ private:
+  const int strong_peak_freeze_duration_;
+  std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
+  absl::optional<int> narrow_peak_band_;
+  size_t narrow_peak_counter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_