Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 66 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h
new file mode 100644
index 0000000000..d26b55950c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h
@@ -0,0 +1,66 @@
+/*
+ *  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_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
+
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+// Produces a smooth signal level estimate from an input audio
+// stream. The estimate smoothing is done through exponential
+// filtering.
+class FixedDigitalLevelEstimator {
+ public:
+  // Sample rates are allowed if the number of samples in a frame
+  // (sample_rate_hz * kFrameDurationMs / 1000) is divisible by
+  // kSubFramesInSample. For kFrameDurationMs=10 and
+  // kSubFramesInSample=20, this means that sample_rate_hz has to be
+  // divisible by 2000.
+  FixedDigitalLevelEstimator(int sample_rate_hz,
+                             ApmDataDumper* apm_data_dumper);
+
+  FixedDigitalLevelEstimator(const FixedDigitalLevelEstimator&) = delete;
+  FixedDigitalLevelEstimator& operator=(const FixedDigitalLevelEstimator&) =
+      delete;
+
+  // The input is assumed to be in FloatS16 format. Scaled input will
+  // produce similarly scaled output. A frame of with kFrameDurationMs
+  // ms of audio produces a level estimates in the same scale. The
+  // level estimate contains kSubFramesInFrame values.
+  std::array<float, kSubFramesInFrame> ComputeLevel(
+      const AudioFrameView<const float>& float_frame);
+
+  // Rate may be changed at any time (but not concurrently) from the
+  // value passed to the constructor. The class is not thread safe.
+  void SetSampleRate(int sample_rate_hz);
+
+  // Resets the level estimator internal state.
+  void Reset();
+
+  float LastAudioLevel() const { return filter_state_level_; }
+
+ private:
+  void CheckParameterCombination();
+
+  ApmDataDumper* const apm_data_dumper_ = nullptr;
+  float filter_state_level_;
+  int samples_in_frame_;
+  int samples_in_sub_frame_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_