Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 63 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter.h b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.h
new file mode 100644
index 0000000000..d4d556349c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2018 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_AGC2_LIMITER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+class Limiter {
+ public:
+  Limiter(int sample_rate_hz,
+          ApmDataDumper* apm_data_dumper,
+          absl::string_view histogram_name_prefix);
+  Limiter(const Limiter& limiter) = delete;
+  Limiter& operator=(const Limiter& limiter) = delete;
+  ~Limiter();
+
+  // Applies limiter and hard-clipping to `signal`.
+  void Process(AudioFrameView<float> signal);
+  InterpolatedGainCurve::Stats GetGainCurveStats() const;
+
+  // Supported rates must be
+  // * supported by FixedDigitalLevelEstimator
+  // * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
+  //   so that samples_per_channel fit in the
+  //   per_sample_scaling_factors_ array.
+  void SetSampleRate(int sample_rate_hz);
+
+  // Resets the internal state.
+  void Reset();
+
+  float LastAudioLevel() const;
+
+ private:
+  const InterpolatedGainCurve interp_gain_curve_;
+  FixedDigitalLevelEstimator level_estimator_;
+  ApmDataDumper* const apm_data_dumper_ = nullptr;
+
+  // Work array containing the sub-frame scaling factors to be interpolated.
+  std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
+  std::array<float, kMaximalNumberOfSamplesPerChannel>
+      per_sample_scaling_factors_ = {};
+  float last_scaling_factor_ = 1.f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_