Index: webrtc/modules/audio_processing/aec3/aec_state.h |
diff --git a/webrtc/modules/audio_processing/aec3/aec_state.h b/webrtc/modules/audio_processing/aec3/aec_state.h |
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index 0000000000000000000000000000000000000000..e3502b4a4f5b5e42ae0446e0bb305bc98747c285 |
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+/* |
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ |
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ |
+ |
+#include <algorithm> |
+#include <memory> |
+#include <vector> |
+ |
+#include "webrtc/base/array_view.h" |
+#include "webrtc/base/constructormagic.h" |
+#include "webrtc/base/optional.h" |
+#include "webrtc/modules/audio_processing/aec3/aec3_common.h" |
+#include "webrtc/modules/audio_processing/aec3/echo_path_variability.h" |
+#include "webrtc/modules/audio_processing/aec3/erle_estimator.h" |
+#include "webrtc/modules/audio_processing/aec3/erl_estimator.h" |
+#include "webrtc/modules/audio_processing/aec3/fft_buffer.h" |
+ |
+namespace webrtc { |
+ |
+class ApmDataDumper; |
+ |
+// Handles the state and the conditions for the echo removal functionality. |
+class AecState { |
+ public: |
+ AecState(); |
+ ~AecState(); |
+ |
+ // Returns whether the linear filter estimate is usable. |
+ bool UsableLinearEstimate() const { return usable_linear_estimate_; } |
+ |
+ // Returns whether there has been echo leakage detected. |
+ bool EchoLeakageDetected() const { return echo_leakage_detected_; } |
+ |
+ // Returns whether it is possible at all to use the model based echo removal |
+ // functionalities. |
+ bool ModelBasedAecFeasible() const { return model_based_aec_feasible_; } |
+ |
+ // Returns whether the render signal is currently active. |
+ bool ActiveRender() const { return active_render_counter_ > 0; } |
+ |
+ // Returns the ERLE. |
+ const std::array<float, kFftLengthBy2Plus1>& Erle() const { |
+ return erle_estimator_.Erle(); |
+ } |
+ |
+ // Returns the ERL. |
+ const std::array<float, kFftLengthBy2Plus1>& Erl() const { |
+ return erl_estimator_.Erl(); |
+ } |
+ |
+ // Returns the delay estimate based on the linear filter. |
+ rtc::Optional<size_t> FilterDelay() const { return filter_delay_; } |
+ |
+ // Returns the externally provided delay. |
+ rtc::Optional<size_t> ExternalDelay() const { return external_delay_; } |
+ |
+ // Returns the bands where the linear filter is reliable. |
+ const std::array<bool, kFftLengthBy2Plus1>& BandsWithReliableFilter() const { |
+ return bands_with_reliable_filter_; |
+ } |
+ |
+ // Reports whether the filter is poorly aligned. |
+ bool PoorlyAlignedFilter() const { |
+ return FilterDelay() ? *FilterDelay() > 0.75f * filter_length_ : false; |
+ } |
+ |
+ // Returns the strength of the filter. |
+ const std::array<float, kFftLengthBy2Plus1>& FilterEstimateStrength() const { |
+ return filter_estimate_strength_; |
+ } |
+ |
+ // Returns whether the capture signal is saturated. |
+ bool SaturatedCapture() const { return capture_signal_saturation_; } |
+ |
+ // Updates the capture signal saturation. |
+ void UpdateCaptureSaturation(bool capture_signal_saturation) { |
+ capture_signal_saturation_ = capture_signal_saturation; |
+ } |
+ |
+ // Returns whether a probable headset setup has been detected. |
+ bool HeadsetDetected() const { return headset_detected_; } |
+ |
+ // Updates the aec state. |
+ void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>& |
+ filter_frequency_response, |
+ const rtc::Optional<size_t>& external_delay_samples, |
+ const FftBuffer& X_buffer, |
+ const std::array<float, kFftLengthBy2Plus1>& E2_main, |
+ const std::array<float, kFftLengthBy2Plus1>& E2_shadow, |
+ const std::array<float, kFftLengthBy2Plus1>& Y2, |
+ rtc::ArrayView<const float> x, |
+ const EchoPathVariability& echo_path_variability, |
+ bool echo_leakage_detected); |
+ |
+ private: |
+ static int instance_count_; |
+ std::unique_ptr<ApmDataDumper> data_dumper_; |
+ ErlEstimator erl_estimator_; |
+ ErleEstimator erle_estimator_; |
+ int echo_path_change_counter_; |
+ int active_render_counter_; |
+ bool usable_linear_estimate_ = false; |
+ bool echo_leakage_detected_ = false; |
+ bool model_based_aec_feasible_ = false; |
+ bool capture_signal_saturation_ = false; |
+ bool headset_detected_ = false; |
+ rtc::Optional<size_t> filter_delay_; |
+ rtc::Optional<size_t> external_delay_; |
+ std::array<bool, kFftLengthBy2Plus1> bands_with_reliable_filter_; |
+ std::array<float, kFftLengthBy2Plus1> filter_estimate_strength_; |
+ size_t filter_length_; |
+ |
+ RTC_DISALLOW_COPY_AND_ASSIGN(AecState); |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ |