summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h
diff options
context:
space:
mode:
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h')
-rw-r--r--third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h62
1 files changed, 62 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h
new file mode 100644
index 0000000000..14612508c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (c) 2021 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_CLIPPING_PREDICTOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+// Frame-wise clipping prediction and clipped level step estimation. Analyzes
+// 10 ms multi-channel frames and estimates an analog mic level decrease step
+// to possibly avoid clipping when predicted. `Analyze()` and
+// `EstimateClippedLevelStep()` can be called in any order.
+class ClippingPredictor {
+ public:
+ virtual ~ClippingPredictor() = default;
+
+ virtual void Reset() = 0;
+
+ // Analyzes a 10 ms multi-channel audio frame.
+ virtual void Analyze(const AudioFrameView<const float>& frame) = 0;
+
+ // Predicts if clipping is going to occur for the specified `channel` in the
+ // near-future and, if so, it returns a recommended analog mic level decrease
+ // step. Returns absl::nullopt if clipping is not predicted.
+ // `level` is the current analog mic level, `default_step` is the amount the
+ // mic level is lowered by the analog controller with every clipping event and
+ // `min_mic_level` and `max_mic_level` is the range of allowed analog mic
+ // levels.
+ virtual absl::optional<int> EstimateClippedLevelStep(
+ int channel,
+ int level,
+ int default_step,
+ int min_mic_level,
+ int max_mic_level) const = 0;
+};
+
+// Creates a ClippingPredictor based on the provided `config`. When enabled,
+// the following must hold for `config`:
+// `window_length < reference_window_length + reference_window_delay`.
+// Returns `nullptr` if `config.enabled` is false.
+std::unique_ptr<ClippingPredictor> CreateClippingPredictor(
+ int num_channels,
+ const AudioProcessing::Config::GainController1::AnalogGainController::
+ ClippingPredictor& config);
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_